From vdubois at b2b2c.com  Tue Oct  1 09:45:17 2024
From: vdubois at b2b2c.com (Vincent Dubois)
Date: Tue, 1 Oct 2024 13:45:17 +0000
Subject: [Spce-user] How to setup subscriber as SIP Trunk
In-Reply-To: <YQBPR0101MB4650334A5FAE46DD11BFB087A6772@YQBPR0101MB4650.CANPRD01.PROD.OUTLOOK.COM>
References: <CAFkRdGtpiVMK67fxOOO4vX=bmtzVqY_Q2iE-Us9k3fhaa3R-Rg@mail.gmail.com>
 <YQBPR0101MB4650334A5FAE46DD11BFB087A6772@YQBPR0101MB4650.CANPRD01.PROD.OUTLOOK.COM>
Message-ID: <YQBPR0101MB4650C71FBEDF95DE5A75E0A6A6772@YQBPR0101MB4650.CANPRD01.PROD.OUTLOOK.COM>

Hi

If you do not want authentication, and just want to validate based on IP address this is what we do:
(PS: in our case everything goes thru an SBC first, so it is just different internal IPs of our SBCs)

In the subscriber:

  *   add a Permanent Registration like
contact: sip:sip at 1.2.3.4:5060  with the IP of the destination
expires: 2157-01-01

In the subscriber preferences

  *   under Trusted Sources you want to add their IP again something like
Source IP: 1.2.3.4
Protocol: ANY
From-Pattern: .*

  *   under Number Manipulation
e164_to_ruri: true     (unless that customer only has one number, in that case you could have put the number in the permanent registration, and this wouldn?t be needed)

  *   under Access Restrictions
allowed_clis: *   (again if they have use multiple numbers, if not then you could put the number there)

I would not recommend this with public IP, because source IP can be faked and any Invite from the IP would be accepted.
But in case of private IP space, or like us private SBC Ips, it works well.

PS: in our case we actually use a different interface, so in the permanent registration we set the socket to that other interface, and in sub pref (under NAT and Media Flow Control) we also set the rtp_interface to that other interface.


Vincent Dubois | Analyste programmeur
B2B2C
514-908-5420
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From pa.ka12 at outlook.com  Fri Oct  4 05:01:08 2024
From: pa.ka12 at outlook.com (Pa Ka)
Date: Fri, 4 Oct 2024 09:01:08 +0000
Subject: [Spce-user] incoming Session fails using ngcp CE mr12.4.1 and SIP.js
Message-ID: <IA2P220MB1912D25CDEB706382873774FEA722@IA2P220MB1912.NAMP220.PROD.OUTLOOK.COM>

Hey Please help me to understand this. incoming session on my SIP.js application are failing.

But, outgoingcall are functioning, I can initiate call to the PSTN.

Second, Incoming session works perfectly when using Zoiper softphone application installed on Windows ( Zoiper and SIP.js are using the same configuration from ngcp c5 switch which make me think it is not ngcp c5 issue, but prefer to ask here!!!)

No 404 fallback is enabled on the portal to the specific address used to log into both phone.

This IP 172.110.70.15 , is for a local pc


I'd would like to know if it is SIP.js codes or Sipwise (ngcp CE mr12.4.1 ) configuration that cause the error.  Below are the sip proxy from Kamailio-proxy.log .


When call from PSTN arrives, the script rewrite called party to remove the leading 1

NOTICE: ROUTE_APPLY_REWRITE_RULE <script>: Rewriting called party '<199999101>' to '<99999101>' - R=<sip:19999101 at 172.110.70.155:57432;transport=tcp;rinstance=6492442> ID=<1B6D3B76 at 208.72.120.66> UA='<null>'

Oct  4 04:31:39.478180 sips proxy[1641]: NOTICE: BRANCH_ROUTE_SBC <script>: Setting P-Called-Party-ID 'sip:<99999101>@<sip.sipdomain.com;user=phone><sip:>' - R=<sip:99999101 at 172.110.70.155:57432;transport=tcp;rinstance=6492442;user=phone> ID=<1B6D3B76 at pstnIP.66> UA='<null>'


This is the first request to the B2B

New request on proxy for the B LEG of the call - M=INVITE R=<sip:99999101 at 172.110.70.155:57432;transport=tcp;rinstance=6492442;user=phone> F=<sip:calllerNumber64 at sip.sipdomain.com;user=phone> T=<sip:99999101 at sip.sipdomain..com;user=phone> IP=<<null>>:<<null>> (<127.0.0.1>:<5080>) ID=<1B6D3B76 at pstnIP_b2b-1> UA='<null>' DESTIP=<127.0.0.1>:<6052>

Oct  4 04:31:39.539462 sips proxy[1650]: NOTICE: REPLY_ROUTE_NAT_CONTINUE <script>: Sending reply S=404 M=INVITE fs='<127.0.0.1>:<6052>' du='<127.0.0.1>:<6050>' - ID=<1B6D3B76 at pstnIP> UA='Vogat/1.0'

After B2B, it fails with this error

Oct  4 04:31:39.539496 sips proxy[1650]: NOTICE: FAILURE_ROUTE_LOCAL <script>: Failure route for local call, code='404' - R=<sip:usernamesubcriber999 at 172.110.70.155:57432;transport=tcp;rinstance=6492442> ID=<1B6D3B76 at pstnIP> UA='<null>'
Oct  4 04:31:39.540023 sips proxy[1650]: NOTICE: FAILURE_ROUTE_LOCAL <script>: Fallback from E164 to user DISABLED due to 'no_404_fallback' set - R=<sip:usernamesubscriber999 at 172.110.70.155:57432;transport=tcp;rinstance=6492442> ID=<1B6D3B76 at pstnIP> UA='<null>'

thank you
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From pa.ka12 at outlook.com  Fri Oct  4 20:50:51 2024
From: pa.ka12 at outlook.com (Pa Ka)
Date: Sat, 5 Oct 2024 00:50:51 +0000
Subject: [Spce-user] incoming Session fails using ngcp CE mr12.4.1 and
 SIP.js
In-Reply-To: <IA2P220MB1912D25CDEB706382873774FEA722@IA2P220MB1912.NAMP220.PROD.OUTLOOK.COM>
References: <IA2P220MB1912D25CDEB706382873774FEA722@IA2P220MB1912.NAMP220.PROD.OUTLOOK.COM>
Message-ID: <IA2P220MB1912D6FB6826CF76EECAF51AEA732@IA2P220MB1912.NAMP220.PROD.OUTLOOK.COM>

Please ignore the previous email. I managed to resolve the issue it was a SIP.js error. Thanks
________________________________
From: Pa Ka <pa.ka12 at outlook.com>
Sent: Friday, October 4, 2024 5:01:08 AM
To: spce-user at lists.sipwise.com <spce-user at lists.sipwise.com>
Subject: incoming Session fails using ngcp CE mr12.4.1 and SIP.js


Hey Please help me to understand this. incoming session on my SIP.js application are failing.



But, outgoingcall are functioning, I can initiate call to the PSTN.



Second, Incoming session works perfectly when using Zoiper softphone application installed on Windows ( Zoiper and SIP.js are using the same configuration from ngcp c5 switch which make me think it is not ngcp c5 issue, but prefer to ask here!!!)



No 404 fallback is enabled on the portal to the specific address used to log into both phone.



This IP 172.110.70.15 , is for a local pc





I?d would like to know if it is SIP.js codes or Sipwise (ngcp CE mr12.4.1 ) configuration that cause the error.  Below are the sip proxy from Kamailio-proxy.log .





When call from PSTN arrives, the script rewrite called party to remove the leading 1



NOTICE: ROUTE_APPLY_REWRITE_RULE <script>: Rewriting called party '?199999101?' to '?99999101?' - R=?sip:19999101 at 172.110.70.155:57432;transport=tcp;rinstance=6492442? ID=?1B6D3B76 at 208.72.120.66? UA='<null>'



Oct  4 04:31:39.478180 sips proxy[1641]: NOTICE: BRANCH_ROUTE_SBC <script>: Setting P-Called-Party-ID 'sip:?99999101?@?sip.sipdomain.com;user=phone?<sip:>' - R=?sip:99999101 at 172.110.70.155:57432;transport=tcp;rinstance=6492442;user=phone? ID=?1B6D3B76 at pstnIP.66? UA='<null>'





This is the first request to the B2B


New request on proxy for the B LEG of the call - M=INVITE R=?sip:99999101 at 172.110.70.155:57432;transport=tcp;rinstance=6492442;user=phone? F=?sip:calllerNumber64 at sip.sipdomain.com;user=phone? T=?sip:99999101 at sip.sipdomain..com;user=phone? IP=?<null>?:?<null>? (?127.0.0.1?:?5080?) ID=?1B6D3B76 at pstnIP_b2b-1? UA='<null>' DESTIP=?127.0.0.1?:?6052?



Oct  4 04:31:39.539462 sips proxy[1650]: NOTICE: REPLY_ROUTE_NAT_CONTINUE <script>: Sending reply S=404 M=INVITE fs='?127.0.0.1?:?6052?' du='?127.0.0.1?:?6050?' - ID=?1B6D3B76 at pstnIP? UA='Vogat/1.0'



After B2B, it fails with this error



Oct  4 04:31:39.539496 sips proxy[1650]: NOTICE: FAILURE_ROUTE_LOCAL <script>: Failure route for local call, code='404' - R=?sip:usernamesubcriber999 at 172.110.70.155:57432;transport=tcp;rinstance=6492442? ID=?1B6D3B76 at pstnIP? UA='<null>'

Oct  4 04:31:39.540023 sips proxy[1650]: NOTICE: FAILURE_ROUTE_LOCAL <script>: Fallback from E164 to user DISABLED due to 'no_404_fallback' set - R=?sip:usernamesubscriber999 at 172.110.70.155:57432;transport=tcp;rinstance=6492442? ID=?1B6D3B76 at pstnIP? UA='<null>'

thank you
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From edo.ekunwe at gmail.com  Mon Oct  7 00:48:51 2024
From: edo.ekunwe at gmail.com (Edo)
Date: Sun, 6 Oct 2024 23:48:51 -0500
Subject: [Spce-user] Troubleshooting request
Message-ID: <CAFkRdGtKH2MtMwdPoQES6JdU+yjSbYG8wp2dzhpfhkFMUeDH3w@mail.gmail.com>

Hi,

I tried to make a call to a subscriber (SIP Trunk setup) and I got the
following error message.
*****

Oct  7 06:43:50.891692 spce proxy[837565]: ERROR: ROUTE_PSTN_SELECT_PEER pv
[pv_core.c:336]: pv_get_ruri_attr(): failed to parse the R-URI

Oct  7 06:43:50.891806 spce proxy[837565]: ERROR: ROUTE_LOAD_PEER_OUT_PREF
pv [pv_core.c:261]: pv_get_ruri(): failed to parse the R-URI

Oct  7 06:43:50.891827 spce proxy[837565]: NOTICE: ROUTE_LOAD_PEER_OUT_PREF
<script>: Load peer preferences for peer host '1' - R=?<null>?
ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk PBX 18.24.3'

Oct  7 06:43:50.893690 spce proxy[837565]: ERROR: ROUTE_PSTN pv
[pv_core.c:336]: pv_get_ruri_attr(): failed to parse the R-URI

Oct  7 06:43:50.895431 spce proxy[837565]: ERROR: ROUTE_PSTN pv
[pv_core.c:336]: pv_get_ruri_attr(): failed to parse the R-URI

Oct  7 06:43:50.895538 spce proxy[837565]: ERROR: <null> pv
[pv_core.c:336]: pv_get_ruri_attr(): failed to parse the R-URI

Oct  7 06:43:50.895558 spce proxy[837565]: ERROR: <null> pv
[pv_core.c:336]: pv_get_ruri_attr(): failed to parse the R-URI

Oct  7 06:43:50.901133 spce proxy[837565]: ERROR: ROUTE_OUTBOUND pv
[pv_core.c:336]: pv_get_ruri_attr(): failed to parse the R-URI

Oct  7 06:43:50.901156 spce proxy[837565]: ERROR: ROUTE_OUTBOUND pv
[pv_core.c:336]: pv_get_ruri_attr(): failed to parse the R-URI

Oct  7 06:43:50.901829 spce proxy[837565]: ERROR: ROUTE_APPLY_REWRITE_RULE
pv [pv_core.c:261]: pv_get_ruri(): failed to parse the R-URI
*****
 I am looking for where I can set the system to a specific R-URI or check
the system to use network provided R-RURI.

Please help.

Thanks


-- 
-----
Ekunwe
Tel: 601.497.3932
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From dbusso at sipwise.com  Mon Oct  7 02:11:04 2024
From: dbusso at sipwise.com (Dario Busso)
Date: Mon, 7 Oct 2024 08:11:04 +0200
Subject: [Spce-user] [EXTERNAL]  Troubleshooting request
In-Reply-To: <CAFkRdGtKH2MtMwdPoQES6JdU+yjSbYG8wp2dzhpfhkFMUeDH3w@mail.gmail.com>
References: <CAFkRdGtKH2MtMwdPoQES6JdU+yjSbYG8wp2dzhpfhkFMUeDH3w@mail.gmail.com>
Message-ID: <CAAyzhJTi6W28bqohSAqJtP5FgyyAeeuqbKK_BTxoE+f+Yzs1yA@mail.gmail.com>

Looking at the error log, it seems that the INVITE received by SPCE is
malformed.
You need to check the peer system setup, not the SPCE.


Dario B.


Il lun 7 ott 2024, 06:50 Edo <edo.ekunwe at gmail.com> ha scritto:

> Hi,
>
> I tried to make a call to a subscriber (SIP Trunk setup) and I got the
> following error message.
> *****
>
> Oct  7 06:43:50.891692 spce proxy[837565]: ERROR: ROUTE_PSTN_SELECT_PEER
> pv [pv_core.c:336]: pv_get_ruri_attr(): failed to parse the R-URI
>
> Oct  7 06:43:50.891806 spce proxy[837565]: ERROR:
> ROUTE_LOAD_PEER_OUT_PREF pv [pv_core.c:261]: pv_get_ruri(): failed to parse
> the R-URI
>
> Oct  7 06:43:50.891827 spce proxy[837565]: NOTICE:
> ROUTE_LOAD_PEER_OUT_PREF <script>: Load peer preferences for peer host '1'
> - R=?<null>? ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk PBX
> 18.24.3'
>
> Oct  7 06:43:50.893690 spce proxy[837565]: ERROR: ROUTE_PSTN pv
> [pv_core.c:336]: pv_get_ruri_attr(): failed to parse the R-URI
>
> Oct  7 06:43:50.895431 spce proxy[837565]: ERROR: ROUTE_PSTN pv
> [pv_core.c:336]: pv_get_ruri_attr(): failed to parse the R-URI
>
> Oct  7 06:43:50.895538 spce proxy[837565]: ERROR: <null> pv
> [pv_core.c:336]: pv_get_ruri_attr(): failed to parse the R-URI
>
> Oct  7 06:43:50.895558 spce proxy[837565]: ERROR: <null> pv
> [pv_core.c:336]: pv_get_ruri_attr(): failed to parse the R-URI
>
> Oct  7 06:43:50.901133 spce proxy[837565]: ERROR: ROUTE_OUTBOUND pv
> [pv_core.c:336]: pv_get_ruri_attr(): failed to parse the R-URI
>
> Oct  7 06:43:50.901156 spce proxy[837565]: ERROR: ROUTE_OUTBOUND pv
> [pv_core.c:336]: pv_get_ruri_attr(): failed to parse the R-URI
>
> Oct  7 06:43:50.901829 spce proxy[837565]: ERROR:
> ROUTE_APPLY_REWRITE_RULE pv [pv_core.c:261]: pv_get_ruri(): failed to parse
> the R-URI
> *****
>  I am looking for where I can set the system to a specific R-URI or check
> the system to use network provided R-RURI.
>
> Please help.
>
> Thanks
>
>
> --
> -----
> Ekunwe
> Tel: 601.497.3932
> --
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/mailman/listinfo/spce-user_lists.sipwise.com
>
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From dbusso at sipwise.com  Mon Oct  7 03:16:10 2024
From: dbusso at sipwise.com (Dario Busso)
Date: Mon, 7 Oct 2024 09:16:10 +0200
Subject: [Spce-user] [EXTERNAL] incoming Session fails using ngcp CE
 mr12.4.1 and SIP.js
In-Reply-To: <IA2P220MB1912D25CDEB706382873774FEA722@IA2P220MB1912.NAMP220.PROD.OUTLOOK.COM>
References: <IA2P220MB1912D25CDEB706382873774FEA722@IA2P220MB1912.NAMP220.PROD.OUTLOOK.COM>
Message-ID: <CAAyzhJQRB6D5e+8K7QmEeefdOQddGM5B5M8YRmr1n-NSX_zL2A@mail.gmail.com>

Hi Pa Ka.
I suggest always updating the system to the most recent LTS; in your case,
Mr12.5.1 was released at the end of September.
About this error: Could you check the RW rules applied, and the CLI
received from the peer? You can see this information in the logs when
verbose is activated for the proxy.
KR
Dario

On Fri, Oct 4, 2024 at 11:02?AM Pa Ka <pa.ka12 at outlook.com> wrote:

> Hey Please help me to understand this. incoming session on my SIP.js
> application are failing.
>
>
>
> *But, outgoingcall are functioning, I can initiate call to the PSTN.*
>
>
>
> *Second, Incoming session works perfectly when using Zoiper softphone
> application installed on Windows ( Zoiper and SIP.js are using the same
> configuration from ngcp c5 switch which make me think it is not ngcp c5
> issue, but prefer to ask here!!!)*
>
>
>
> *No 404 fallback is enabled on the portal to the specific address used to
> log into both phone.*
>
>
>
> *This IP 172.110.70.15 , is for a local pc*
>
>
>
>
>
> I?d would like to know if it is SIP.js codes or Sipwise (ngcp CE mr12.4.1
> ) configuration that cause the error.  Below are the sip proxy from
> Kamailio-proxy.log .
>
>
>
>
>
> When call from PSTN arrives, the script rewrite called party to remove the
> leading 1
>
>
>
> NOTICE: ROUTE_APPLY_REWRITE_RULE <script>: Rewriting called party
> '?199999101?' to '?99999101?' - R=?sip:19999101 at 172.110.70.155:57432;transport=tcp;rinstance=6492442?
> ID=?1B6D3B76 at 208.72.120.66? UA='<null>'
>
>
>
> Oct  4 04:31:39.478180 sips proxy[1641]: NOTICE: BRANCH_ROUTE_SBC
> <script>: Setting P-Called-Party-ID '
> sip:?99999101?@?sip.sipdomain.com;user=phone?' -
> R=?sip:99999101 at 172.110.70.155:57432;transport=tcp;rinstance=6492442;user=phone?
> ID=?1B6D3B76 at pstnIP.66? UA='<null>'
>
>
>
>
>
> This is the first request to the B2B
>
> New request on proxy for the B LEG of the call - M=INVITE
> R=?sip:99999101 at 172.110.70.155:57432;transport=tcp;rinstance=6492442;user=phone?
> F=?sip:calllerNumber64 at sip.sipdomain.com;user=phone?
> T=?sip:99999101 at sip.sipdomain..com;user=phone? IP=?<null>?:?<null>?
> (?127.0.0.1?:?5080?) ID=?1B6D3B76 at pstnIP_b2b-1? UA='<null>'
> DESTIP=?127.0.0.1?:?6052?
>
>
>
> Oct  4 04:31:39.539462 sips proxy[1650]: NOTICE: REPLY_ROUTE_NAT_CONTINUE
> <script>: Sending reply S=404 M=INVITE fs='?127.0.0.1?:?6052?'
> du='?127.0.0.1?:?6050?' - ID=?1B6D3B76 at pstnIP? UA='Vogat/1.0'
>
>
>
> After B2B, it fails with this error
>
>
>
> Oct  4 04:31:39.539496 sips proxy[1650]: NOTICE: FAILURE_ROUTE_LOCAL
> <script>: Failure route for local call, code='404' -
> R=?sip:usernamesubcriber999 at 172.110.70.155:57432;transport=tcp;rinstance=6492442?
> ID=?1B6D3B76 at pstnIP? UA='<null>'
>
> Oct  4 04:31:39.540023 sips proxy[1650]: NOTICE: FAILURE_ROUTE_LOCAL
> <script>: Fallback from E164 to user DISABLED due to 'no_404_fallback' set
> - R=?sip:usernamesubscriber999 at 172.110.70.155:57432;transport=tcp;rinstance=6492442?
> ID=?1B6D3B76 at pstnIP? UA='<null>'
>
> thank you
> --
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/mailman/listinfo/spce-user_lists.sipwise.com
>
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From edo.ekunwe at gmail.com  Mon Oct  7 09:35:14 2024
From: edo.ekunwe at gmail.com (Edo)
Date: Mon, 7 Oct 2024 08:35:14 -0500
Subject: [Spce-user] [EXTERNAL] Troubleshooting request (Dario Busso)
In-Reply-To: <mailman.171.1728285423.2942163.spce-user_lists.sipwise.com@lists.sipwise.com>
References: <mailman.171.1728285423.2942163.spce-user_lists.sipwise.com@lists.sipwise.com>
Message-ID: <CAFkRdGuAqoK0z5X3Gq1NPXVexjX+gfcM+Lp9jXb-AJ5Z5ghgrQ@mail.gmail.com>

Hi Dario,

I am not sure if your earlier response of malformed invite was directed at
me so I decided to paste the sip logs from a sample call, maybe we can get
more information that may help with troubleshooting.

****

Oct  7 06:43:50.807051 spce proxy[837572]: NOTICE: DEFAULT_ROUTE <script>:
New request on proxy - M=INVITE R=?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX:5060?
F=?sip:XXCallerIDXX at XXX.XXX.86.104?
T=?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX? IP=?XXX.XXX.188.51?:?5060?
(?127.0.0.1?:?5060?) ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk
PBX 18.24.3' DESTIP=?127.0.0.1?:?5062?

Oct  7 06:43:50.807084 spce proxy[837572]: NOTICE: DEFAULT_ROUTE <script>:
Sending reply S=100 Trying M=INVITE fs='?127.0.0.1?:?5062?'
du='?127.0.0.1?:?5060?' - R=?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX:5060?
ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk PBX 18.24.3'

Oct  7 06:43:50.807375 spce proxy[837572]: NOTICE:
ROUTE_LOAD_CALLEE_DOMAIN_PREF <script>: Load domain preferences for callee
'?XXX.XXX.XXX.XXX?' - R=?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX:5060?
ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk PBX 18.24.3'

Oct  7 06:43:50.811499 spce proxy[837572]: NOTICE: ROUTE_AUTH <script>:
Authentication failed, no credentials -
R=?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX:5060?
ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk PBX 18.24.3'
Auth=?<null>?

Oct  7 06:43:50.811581 spce proxy[837572]: NOTICE: ROUTE_AUTH <script>:
Sending reply S=407 fs='?127.0.0.1?:?5062?' du='?127.0.0.1?:?5060?' -
ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk PBX 18.24.3'

Oct  7 06:43:50.812257 spce proxy[837568]: NOTICE: sl:filtered-ack
<script>: New request on proxy - M=ACK
R=?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX:5060?
F=?sip:XXCallerIDXX at XXX.XXX.86.104?
T=?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX? IP=?<null>?:?<null>?
(?127.0.0.1?:?5060?) ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='<null>'
DESTIP=?127.0.0.1?:?5062?

Oct  7 06:43:50.872198 spce proxy[837565]: NOTICE: DEFAULT_ROUTE <script>:
New request on proxy - M=INVITE R=?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX:5060?
F=?sip:XXCallerIDXX at XXX.XXX.86.104?
T=?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX? IP=?XXX.XXX.188.51?:?5060?
(?127.0.0.1?:?5060?) ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk
PBX 18.24.3' DESTIP=?127.0.0.1?:?5062?

Oct  7 06:43:50.872229 spce proxy[837565]: NOTICE: DEFAULT_ROUTE <script>:
Sending reply S=100 Trying M=INVITE fs='?127.0.0.1?:?5062?'
du='?127.0.0.1?:?5060?' - R=?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX:5060?
ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk PBX 18.24.3'

Oct  7 06:43:50.872504 spce proxy[837565]: NOTICE:
ROUTE_LOAD_CALLEE_DOMAIN_PREF <script>: Load domain preferences for callee
'?XXX.XXX.XXX.XXX?' - R=?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX:5060?
ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk PBX 18.24.3'

Oct  7 06:43:50.877414 spce proxy[837565]: NOTICE: ROUTE_FIND_CALLER
<script>: Sending reply S=101 Connecting fs='?127.0.0.1?:?5062?'
du='?127.0.0.1?:?5060?' - R=?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX:5060?
ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk PBX 18.24.3'

Oct  7 06:43:50.877751 spce proxy[837565]: NOTICE: ROUTE_LOAD_CALLER_PREF
<script>: Load caller preferences for uuid
'?86ba28d9-153f-4cee-800c-f34c1efea74a?' and domain '?XXX.XXX.86.104?' -
R=?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX:5060?
ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk PBX 18.24.3'

Oct  7 06:43:50.883279 spce proxy[837565]: NOTICE:
ROUTE_LOAD_CALLER_CONTRACT_PREF <script>: Load caller contract preferences
for contract id '?2?' and ip '?XXX.XXX.188.51?' -
R=?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX:5060?
ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk PBX 18.24.3'

Oct  7 06:43:50.883583 spce proxy[837565]: NOTICE:
ROUTE_LOAD_CALLER_RESELLER_PREF <script>: Load caller reseller preferences
for reseller id '?1?' - R=?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX:5060?
ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk PBX 18.24.3'

Oct  7 06:43:50.884273 spce proxy[837565]: NOTICE: ROUTE_DLG_MANAGE
<script>: Dialog managed,
lua_dlg_callid:[?3e2e30bf-6132-478a-bdd0-292e139c39c5?] -
R=?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX:5060?
ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk PBX 18.24.3'

Oct  7 06:43:50.884348 spce proxy[837565]: NOTICE: ROUTE_GET_CALLER_CLI
<script>: User-Provided CLI '?XXCallerIDXX?' taken from From-User -
R=?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX:5060?
ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk PBX 18.24.3'

Oct  7 06:43:50.888566 spce proxy[837565]: NOTICE: ROUTE_APPLY_REWRITE_RULE
<script>: Applying caller-in usr/domain rewrite rules on user-provided CLI
using dpid '7' - R=?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX:5060?
ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk PBX 18.24.3'

Oct  7 06:43:50.888839 spce proxy[837565]: NOTICE: ROUTE_CHECK_CLI_ALLOWED
<script>: User-provided CLI '?XXCallerIDXX?' accepted (matched user
preference) - R=?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX:5060?
ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk PBX 18.24.3'

Oct  7 06:43:50.888915 spce proxy[837565]: NOTICE: ROUTE_INVITE <script>:
Setting '?XXCallerIDXX?@?XXX.XXX.86.104?' as initiating user-provided CLI -
R=?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX:5060?
ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk PBX 18.24.3'

Oct  7 06:43:50.888963 spce proxy[837565]: NOTICE: ROUTE_INVITE <script>:
Setting '?XXCallerIDXX?@?XXX.XXX.86.104?' as initiating network-provided
CLI - R=?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX:5060?
ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk PBX 18.24.3'

Oct  7 06:43:50.889037 spce proxy[837565]: NOTICE: ROUTE_APPLY_REWRITE_RULE
<script>: Applying callee-in usr/domain rewrite rules on called party using
dpid '8' - R=?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX:5060?
ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk PBX 18.24.3'

Oct  7 06:43:50.890596 spce proxy[837565]: NOTICE: ROUTE_FIND_CALLEE
<script>: Callee is not local - R=?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX:5060?
ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk PBX 18.24.3'

Oct  7 06:43:50.890869 spce proxy[837565]: NOTICE: ROUTE_PSTN <script>:
Call to SIP Peering - R=?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX:5060?
ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk PBX 18.24.3'

Oct  7 06:43:50.890924 spce proxy[837565]: NOTICE: ROUTE_PSTN <script>:
Load gws matching calling part '?sip:XXCallerIDXX at XXX.XXX.86.104?' and
called user '?xxxxxxxxxxx2607244?' and called part
'?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX:5060?' -
R=?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX:5060?
ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk PBX 18.24.3'

Oct  7 06:43:50.891692 spce proxy[837565]: ERROR: ROUTE_PSTN_SELECT_PEER pv
[pv_core.c:336]: pv_get_ruri_attr(): failed to parse the R-URI

Oct  7 06:43:50.891806 spce proxy[837565]: ERROR: ROUTE_LOAD_PEER_OUT_PREF
pv [pv_core.c:261]: pv_get_ruri(): failed to parse the R-URI

Oct  7 06:43:50.891827 spce proxy[837565]: NOTICE: ROUTE_LOAD_PEER_OUT_PREF
<script>: Load peer preferences for peer host '1' - R=?<null>?
ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk PBX 18.24.3'

Oct  7 06:43:50.893690 spce proxy[837565]: ERROR: ROUTE_PSTN pv
[pv_core.c:336]: pv_get_ruri_attr(): failed to parse the R-URI

Oct  7 06:43:50.895431 spce proxy[837565]: ERROR: ROUTE_PSTN pv
[pv_core.c:336]: pv_get_ruri_attr(): failed to parse the R-URI

Oct  7 06:43:50.895538 spce proxy[837565]: ERROR: <null> pv
[pv_core.c:336]: pv_get_ruri_attr(): failed to parse the R-URI

Oct  7 06:43:50.895558 spce proxy[837565]: ERROR: <null> pv
[pv_core.c:336]: pv_get_ruri_attr(): failed to parse the R-URI

Oct  7 06:43:50.901133 spce proxy[837565]: ERROR: ROUTE_OUTBOUND pv
[pv_core.c:336]: pv_get_ruri_attr(): failed to parse the R-URI

Oct  7 06:43:50.901156 spce proxy[837565]: ERROR: ROUTE_OUTBOUND pv
[pv_core.c:336]: pv_get_ruri_attr(): failed to parse the R-URI

Oct  7 06:43:50.901829 spce proxy[837565]: ERROR: ROUTE_APPLY_REWRITE_RULE
pv [pv_core.c:261]: pv_get_ruri(): failed to parse the R-URI

Oct  7 06:43:50.901850 spce proxy[837565]: NOTICE: ROUTE_APPLY_REWRITE_RULE
<script>: Applying acc caller-out peer rewrite rules on acc caller party
using dpid '3' - R=?<null>? ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5?
UA='Asterisk PBX 18.24.3'

Oct  7 06:43:50.902770 spce proxy[837565]: ERROR: ROUTE_ACC_CALLER pv
[pv_core.c:261]: pv_get_ruri(): failed to parse the R-URI

Oct  7 06:43:50.902792 spce proxy[837565]: NOTICE: ROUTE_ACC_CALLER
<script>: Setting acc source-leg for uuid
'?86ba28d9-153f-4cee-800c-f34c1efea74a?':
'?{"gpp":["70770",null,null,null,null,null,null,null,null,null],"clir":0,"uuid":"86ba28d9-153f-4cee-800c-f34c1efea74a","u":"XXCallerIDXX","d":"XXX.XXX.86.104","cli":"XXCallerIDXX","s_id":"70770","c_id":"70770","a_id":2,"pau":"","par":"","s":"call","ip":"XXX.XXX.188.51","t":1728276230.902101,"lnp_p":"","cli_out":"XXCallerIDXX","lnp_t":"","pai":"","ppi":"","div":"","last_hih":"","cid":"","u2u":"","lcr":0,"cc_quota":"","cc_sub":"1","cc_cust":"1"}?'
- R=?<null>? ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk PBX
18.24.3'

Oct  7 06:43:50.903156 spce proxy[837565]: ERROR: ROUTE_APPLY_REWRITE_RULE
pv [pv_core.c:261]: pv_get_ruri(): failed to parse the R-URI

Oct  7 06:43:50.903176 spce proxy[837565]: NOTICE: ROUTE_APPLY_REWRITE_RULE
<script>: Applying acc callee-out peer rewrite rules on acc called party
using dpid '4' - R=?<null>? ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5?
UA='Asterisk PBX 18.24.3'

Oct  7 06:43:50.903656 spce proxy[837565]: ERROR: ROUTE_ACC_CALLEE pv
[pv_core.c:261]: pv_get_ruri(): failed to parse the R-URI

Oct  7 06:43:50.903679 spce proxy[837565]: NOTICE: ROUTE_ACC_CALLEE
<script>: Setting acc destination-leg for uuid '?0?':
'?{"gpp":[null,null,null,null,null,null,null,null,null,null],"plu":0,"s_id":"","c_id":"","a_id":0,"dialed":"xxxxxxxxxxx2607244","uuid":"0","u":"<null>","d":"<null>","u_in":"xxxxxxxxxxx2607244","d_in":"XXX.XXX.XXX.XXX","lcr":1,"lnp_p":"","u_out":"0","lnp_t":"","fci":"","cc_quota":"","cc_sub":"1","cc_cust":""}?'
- R=?<null>? ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk PBX
18.24.3'

Oct  7 06:43:50.903837 spce proxy[837565]: ERROR: ROUTE_OUTBOUND tm
[t_lookup.c:1827]: new_t(): uri invalid

Oct  7 06:43:50.903858 spce proxy[837565]: ERROR: ROUTE_OUTBOUND tm
[t_lookup.c:1979]: t_newtran(): new_t failed

Oct  7 06:43:50.903913 spce proxy[837565]: ERROR: ROUTE_OUTBOUND sl
[sl_funcs.c:428]: sl_reply_error(): stateless error reply used: I'm
terribly sorry, server error occurred (1/SL)

Oct  7 06:43:50.904591 spce proxy[837570]: NOTICE: sl:filtered-ack
<script>: New request on proxy - M=ACK
R=?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX:5060?
F=?sip:XXCallerIDXX at XXX.XXX.86.104?
T=?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX? IP=?<null>?:?<null>?
(?127.0.0.1?:?5060?) ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='<null>'
DESTIP=?127.0.0.1?:?5062?


****

Thanks



On Mon, Oct 7, 2024 at 2:17?AM <spce-user-request at lists.sipwise.com> wrote:

> Send Spce-user mailing list submissions to
>         spce-user at lists.sipwise.com
>
> To subscribe or unsubscribe via the World Wide Web, visit
>
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> or, via email, send a message with subject or body 'help' to
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>
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of Spce-user digest..."
>
>
> Today's Topics:
>
>    1. Troubleshooting request (Edo)
>    2. Re: [EXTERNAL]  Troubleshooting request (Dario Busso)
>    3. Re: [EXTERNAL] incoming Session fails using ngcp CE mr12.4.1
>       and SIP.js (Dario Busso)
>
>
> ----------------------------------------------------------------------
>
> Message: 1
> Date: Sun, 6 Oct 2024 23:48:51 -0500
> From: Edo <edo.ekunwe at gmail.com>
> To: spce-user at lists.sipwise.com
> Subject: [Spce-user] Troubleshooting request
> Message-ID:
>         <
> CAFkRdGtKH2MtMwdPoQES6JdU+yjSbYG8wp2dzhpfhkFMUeDH3w at mail.gmail.com>
> Content-Type: text/plain; charset="utf-8"
>
> Hi,
>
> I tried to make a call to a subscriber (SIP Trunk setup) and I got the
> following error message.
> *****
>
> Oct  7 06:43:50.891692 spce proxy[837565]: ERROR: ROUTE_PSTN_SELECT_PEER pv
> [pv_core.c:336]: pv_get_ruri_attr(): failed to parse the R-URI
>
> Oct  7 06:43:50.891806 spce proxy[837565]: ERROR: ROUTE_LOAD_PEER_OUT_PREF
> pv [pv_core.c:261]: pv_get_ruri(): failed to parse the R-URI
>
> Oct  7 06:43:50.891827 spce proxy[837565]: NOTICE: ROUTE_LOAD_PEER_OUT_PREF
> <script>: Load peer preferences for peer host '1' - R=?<null>?
> ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk PBX 18.24.3'
>
> Oct  7 06:43:50.893690 spce proxy[837565]: ERROR: ROUTE_PSTN pv
> [pv_core.c:336]: pv_get_ruri_attr(): failed to parse the R-URI
>
> Oct  7 06:43:50.895431 spce proxy[837565]: ERROR: ROUTE_PSTN pv
> [pv_core.c:336]: pv_get_ruri_attr(): failed to parse the R-URI
>
> Oct  7 06:43:50.895538 spce proxy[837565]: ERROR: <null> pv
> [pv_core.c:336]: pv_get_ruri_attr(): failed to parse the R-URI
>
> Oct  7 06:43:50.895558 spce proxy[837565]: ERROR: <null> pv
> [pv_core.c:336]: pv_get_ruri_attr(): failed to parse the R-URI
>
> Oct  7 06:43:50.901133 spce proxy[837565]: ERROR: ROUTE_OUTBOUND pv
> [pv_core.c:336]: pv_get_ruri_attr(): failed to parse the R-URI
>
> Oct  7 06:43:50.901156 spce proxy[837565]: ERROR: ROUTE_OUTBOUND pv
> [pv_core.c:336]: pv_get_ruri_attr(): failed to parse the R-URI
>
> Oct  7 06:43:50.901829 spce proxy[837565]: ERROR: ROUTE_APPLY_REWRITE_RULE
> pv [pv_core.c:261]: pv_get_ruri(): failed to parse the R-URI
> *****
>  I am looking for where I can set the system to a specific R-URI or check
> the system to use network provided R-RURI.
>
> Please help.
>
> Thanks
>
>
> --
> -----
> Ekunwe
> Tel: 601.497.3932
> -------------- next part --------------
> An HTML attachment was scrubbed...
> URL: <
> http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20241006/d65e45db/attachment-0001.htm
> >
>
> ------------------------------
>
> Message: 2
> Date: Mon, 7 Oct 2024 08:11:04 +0200
> From: Dario Busso <dbusso at sipwise.com>
> To: Edo <edo.ekunwe at gmail.com>
> Cc: spce-user <spce-user at lists.sipwise.com>
> Subject: Re: [Spce-user] [EXTERNAL]  Troubleshooting request
> Message-ID:
>         <
> CAAyzhJTi6W28bqohSAqJtP5FgyyAeeuqbKK_BTxoE+f+Yzs1yA at mail.gmail.com>
> Content-Type: text/plain; charset="utf-8"
>
> Looking at the error log, it seems that the INVITE received by SPCE is
> malformed.
> You need to check the peer system setup, not the SPCE.
>
>
> Dario B.
>
>
> Il lun 7 ott 2024, 06:50 Edo <edo.ekunwe at gmail.com> ha scritto:
>
> > Hi,
> >
> > I tried to make a call to a subscriber (SIP Trunk setup) and I got the
> > following error message.
> > *****
> >
> > Oct  7 06:43:50.891692 spce proxy[837565]: ERROR: ROUTE_PSTN_SELECT_PEER
> > pv [pv_core.c:336]: pv_get_ruri_attr(): failed to parse the R-URI
> >
> > Oct  7 06:43:50.891806 spce proxy[837565]: ERROR:
> > ROUTE_LOAD_PEER_OUT_PREF pv [pv_core.c:261]: pv_get_ruri(): failed to
> parse
> > the R-URI
> >
> > Oct  7 06:43:50.891827 spce proxy[837565]: NOTICE:
> > ROUTE_LOAD_PEER_OUT_PREF <script>: Load peer preferences for peer host
> '1'
> > - R=?<null>? ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk PBX
> > 18.24.3'
> >
> > Oct  7 06:43:50.893690 spce proxy[837565]: ERROR: ROUTE_PSTN pv
> > [pv_core.c:336]: pv_get_ruri_attr(): failed to parse the R-URI
> >
> > Oct  7 06:43:50.895431 spce proxy[837565]: ERROR: ROUTE_PSTN pv
> > [pv_core.c:336]: pv_get_ruri_attr(): failed to parse the R-URI
> >
> > Oct  7 06:43:50.895538 spce proxy[837565]: ERROR: <null> pv
> > [pv_core.c:336]: pv_get_ruri_attr(): failed to parse the R-URI
> >
> > Oct  7 06:43:50.895558 spce proxy[837565]: ERROR: <null> pv
> > [pv_core.c:336]: pv_get_ruri_attr(): failed to parse the R-URI
> >
> > Oct  7 06:43:50.901133 spce proxy[837565]: ERROR: ROUTE_OUTBOUND pv
> > [pv_core.c:336]: pv_get_ruri_attr(): failed to parse the R-URI
> >
> > Oct  7 06:43:50.901156 spce proxy[837565]: ERROR: ROUTE_OUTBOUND pv
> > [pv_core.c:336]: pv_get_ruri_attr(): failed to parse the R-URI
> >
> > Oct  7 06:43:50.901829 spce proxy[837565]: ERROR:
> > ROUTE_APPLY_REWRITE_RULE pv [pv_core.c:261]: pv_get_ruri(): failed to
> parse
> > the R-URI
> > *****
> >  I am looking for where I can set the system to a specific R-URI or check
> > the system to use network provided R-RURI.
> >
> > Please help.
> >
> > Thanks
> >
> >
> > --
> > -----
> > Ekunwe
> > Tel: 601.497.3932
> > --
> > Spce-user mailing list
> > Spce-user at lists.sipwise.com
> > http://lists.sipwise.com/mailman/listinfo/spce-user_lists.sipwise.com
> >
> -------------- next part --------------
> An HTML attachment was scrubbed...
> URL: <
> http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20241007/4df0c094/attachment-0001.htm
> >
>
> ------------------------------
>
> Message: 3
> Date: Mon, 7 Oct 2024 09:16:10 +0200
> From: Dario Busso <dbusso at sipwise.com>
> To: Pa Ka <pa.ka12 at outlook.com>
> Cc: "spce-user at lists.sipwise.com" <spce-user at lists.sipwise.com>
> Subject: Re: [Spce-user] [EXTERNAL] incoming Session fails using ngcp
>         CE mr12.4.1 and SIP.js
> Message-ID:
>         <
> CAAyzhJQRB6D5e+8K7QmEeefdOQddGM5B5M8YRmr1n-NSX_zL2A at mail.gmail.com>
> Content-Type: text/plain; charset="utf-8"
>
> Hi Pa Ka.
> I suggest always updating the system to the most recent LTS; in your case,
> Mr12.5.1 was released at the end of September.
> About this error: Could you check the RW rules applied, and the CLI
> received from the peer? You can see this information in the logs when
> verbose is activated for the proxy.
> KR
> Dario
>
> On Fri, Oct 4, 2024 at 11:02?AM Pa Ka <pa.ka12 at outlook.com> wrote:
>
> > Hey Please help me to understand this. incoming session on my SIP.js
> > application are failing.
> >
> >
> >
> > *But, outgoingcall are functioning, I can initiate call to the PSTN.*
> >
> >
> >
> > *Second, Incoming session works perfectly when using Zoiper softphone
> > application installed on Windows ( Zoiper and SIP.js are using the same
> > configuration from ngcp c5 switch which make me think it is not ngcp c5
> > issue, but prefer to ask here!!!)*
> >
> >
> >
> > *No 404 fallback is enabled on the portal to the specific address used to
> > log into both phone.*
> >
> >
> >
> > *This IP 172.110.70.15 , is for a local pc*
> >
> >
> >
> >
> >
> > I?d would like to know if it is SIP.js codes or Sipwise (ngcp CE mr12.4.1
> > ) configuration that cause the error.  Below are the sip proxy from
> > Kamailio-proxy.log .
> >
> >
> >
> >
> >
> > When call from PSTN arrives, the script rewrite called party to remove
> the
> > leading 1
> >
> >
> >
> > NOTICE: ROUTE_APPLY_REWRITE_RULE <script>: Rewriting called party
> > '?199999101?' to '?99999101?' - R=?sip:19999101 at 172.110.70.
> 155:57432;transport=tcp;rinstance=6492442?
> > ID=?1B6D3B76 at 208.72.120.66? UA='<null>'
> >
> >
> >
> > Oct  4 04:31:39.478180 sips proxy[1641]: NOTICE: BRANCH_ROUTE_SBC
> > <script>: Setting P-Called-Party-ID '
> > sip:?99999101?@?sip.sipdomain.com;user=phone?' -
> > R=?sip:99999101 at 172.110.70.
> 155:57432;transport=tcp;rinstance=6492442;user=phone?
> > ID=?1B6D3B76 at pstnIP.66? UA='<null>'
> >
> >
> >
> >
> >
> > This is the first request to the B2B
> >
> > New request on proxy for the B LEG of the call - M=INVITE
> > R=?sip:99999101 at 172.110.70.
> 155:57432;transport=tcp;rinstance=6492442;user=phone?
> > F=?sip:calllerNumber64 at sip.sipdomain.com;user=phone?
> > T=?sip:99999101 at sip.sipdomain..com;user=phone? IP=?<null>?:?<null>?
> > (?127.0.0.1?:?5080?) ID=?1B6D3B76 at pstnIP_b2b-1? UA='<null>'
> > DESTIP=?127.0.0.1?:?6052?
> >
> >
> >
> > Oct  4 04:31:39.539462 sips proxy[1650]: NOTICE: REPLY_ROUTE_NAT_CONTINUE
> > <script>: Sending reply S=404 M=INVITE fs='?127.0.0.1?:?6052?'
> > du='?127.0.0.1?:?6050?' - ID=?1B6D3B76 at pstnIP? UA='Vogat/1.0'
> >
> >
> >
> > After B2B, it fails with this error
> >
> >
> >
> > Oct  4 04:31:39.539496 sips proxy[1650]: NOTICE: FAILURE_ROUTE_LOCAL
> > <script>: Failure route for local call, code='404' -
> > R=?sip:usernamesubcriber999 at 172.110.70.155:57432
> ;transport=tcp;rinstance=6492442?
> > ID=?1B6D3B76 at pstnIP? UA='<null>'
> >
> > Oct  4 04:31:39.540023 sips proxy[1650]: NOTICE: FAILURE_ROUTE_LOCAL
> > <script>: Fallback from E164 to user DISABLED due to 'no_404_fallback'
> set
> > - R=?sip:usernamesubscriber999 at 172.110.70.155:57432
> ;transport=tcp;rinstance=6492442?
> > ID=?1B6D3B76 at pstnIP? UA='<null>'
> >
> > thank you
> > --
> > Spce-user mailing list
> > Spce-user at lists.sipwise.com
> > http://lists.sipwise.com/mailman/listinfo/spce-user_lists.sipwise.com
> >
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>
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-- 
-----
Ekunwe
Tel: 601.497.3932
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From edo.ekunwe at gmail.com  Mon Oct  7 09:36:40 2024
From: edo.ekunwe at gmail.com (Edo)
Date: Mon, 7 Oct 2024 08:36:40 -0500
Subject: [Spce-user] [EXTERNAL]  Troubleshooting request
In-Reply-To: <CAAyzhJTi6W28bqohSAqJtP5FgyyAeeuqbKK_BTxoE+f+Yzs1yA@mail.gmail.com>
References: <CAFkRdGtKH2MtMwdPoQES6JdU+yjSbYG8wp2dzhpfhkFMUeDH3w@mail.gmail.com>
 <CAAyzhJTi6W28bqohSAqJtP5FgyyAeeuqbKK_BTxoE+f+Yzs1yA@mail.gmail.com>
Message-ID: <CAFkRdGuEw-FfHym8Of+fUf3aYvcC7+==nB_dyi5qMJJg91A=nQ@mail.gmail.com>

Hi Dario,

I am not sure if your earlier response of malformed invite was directed at
me so I decided to paste the sip logs from a sample call, maybe we can get
more information that may help with troubleshooting.

****

Oct  7 06:43:50.807051 spce proxy[837572]: NOTICE: DEFAULT_ROUTE <script>:
New request on proxy - M=INVITE R=?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX:5060?
F=?sip:XXCallerIDXX at XXX.XXX.86.104?
T=?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX? IP=?XXX.XXX.188.51?:?5060?
(?127.0.0.1?:?5060?) ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk
PBX 18.24.3' DESTIP=?127.0.0.1?:?5062?

Oct  7 06:43:50.807084 spce proxy[837572]: NOTICE: DEFAULT_ROUTE <script>:
Sending reply S=100 Trying M=INVITE fs='?127.0.0.1?:?5062?'
du='?127.0.0.1?:?5060?' - R=?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX:5060?
ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk PBX 18.24.3'

Oct  7 06:43:50.807375 spce proxy[837572]: NOTICE:
ROUTE_LOAD_CALLEE_DOMAIN_PREF <script>: Load domain preferences for callee
'?XXX.XXX.XXX.XXX?' - R=?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX:5060?
ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk PBX 18.24.3'

Oct  7 06:43:50.811499 spce proxy[837572]: NOTICE: ROUTE_AUTH <script>:
Authentication failed, no credentials -
R=?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX:5060?
ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk PBX 18.24.3'
Auth=?<null>?

Oct  7 06:43:50.811581 spce proxy[837572]: NOTICE: ROUTE_AUTH <script>:
Sending reply S=407 fs='?127.0.0.1?:?5062?' du='?127.0.0.1?:?5060?' -
ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk PBX 18.24.3'

Oct  7 06:43:50.812257 spce proxy[837568]: NOTICE: sl:filtered-ack
<script>: New request on proxy - M=ACK
R=?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX:5060?
F=?sip:XXCallerIDXX at XXX.XXX.86.104?
T=?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX? IP=?<null>?:?<null>?
(?127.0.0.1?:?5060?) ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='<null>'
DESTIP=?127.0.0.1?:?5062?

Oct  7 06:43:50.872198 spce proxy[837565]: NOTICE: DEFAULT_ROUTE <script>:
New request on proxy - M=INVITE R=?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX:5060?
F=?sip:XXCallerIDXX at XXX.XXX.86.104?
T=?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX? IP=?XXX.XXX.188.51?:?5060?
(?127.0.0.1?:?5060?) ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk
PBX 18.24.3' DESTIP=?127.0.0.1?:?5062?

Oct  7 06:43:50.872229 spce proxy[837565]: NOTICE: DEFAULT_ROUTE <script>:
Sending reply S=100 Trying M=INVITE fs='?127.0.0.1?:?5062?'
du='?127.0.0.1?:?5060?' - R=?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX:5060?
ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk PBX 18.24.3'

Oct  7 06:43:50.872504 spce proxy[837565]: NOTICE:
ROUTE_LOAD_CALLEE_DOMAIN_PREF <script>: Load domain preferences for callee
'?XXX.XXX.XXX.XXX?' - R=?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX:5060?
ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk PBX 18.24.3'

Oct  7 06:43:50.877414 spce proxy[837565]: NOTICE: ROUTE_FIND_CALLER
<script>: Sending reply S=101 Connecting fs='?127.0.0.1?:?5062?'
du='?127.0.0.1?:?5060?' - R=?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX:5060?
ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk PBX 18.24.3'

Oct  7 06:43:50.877751 spce proxy[837565]: NOTICE: ROUTE_LOAD_CALLER_PREF
<script>: Load caller preferences for uuid
'?86ba28d9-153f-4cee-800c-f34c1efea74a?' and domain '?XXX.XXX.86.104?' -
R=?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX:5060?
ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk PBX 18.24.3'

Oct  7 06:43:50.883279 spce proxy[837565]: NOTICE:
ROUTE_LOAD_CALLER_CONTRACT_PREF <script>: Load caller contract preferences
for contract id '?2?' and ip '?XXX.XXX.188.51?' -
R=?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX:5060?
ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk PBX 18.24.3'

Oct  7 06:43:50.883583 spce proxy[837565]: NOTICE:
ROUTE_LOAD_CALLER_RESELLER_PREF <script>: Load caller reseller preferences
for reseller id '?1?' - R=?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX:5060?
ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk PBX 18.24.3'

Oct  7 06:43:50.884273 spce proxy[837565]: NOTICE: ROUTE_DLG_MANAGE
<script>: Dialog managed,
lua_dlg_callid:[?3e2e30bf-6132-478a-bdd0-292e139c39c5?] -
R=?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX:5060?
ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk PBX 18.24.3'

Oct  7 06:43:50.884348 spce proxy[837565]: NOTICE: ROUTE_GET_CALLER_CLI
<script>: User-Provided CLI '?XXCallerIDXX?' taken from From-User -
R=?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX:5060?
ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk PBX 18.24.3'

Oct  7 06:43:50.888566 spce proxy[837565]: NOTICE: ROUTE_APPLY_REWRITE_RULE
<script>: Applying caller-in usr/domain rewrite rules on user-provided CLI
using dpid '7' - R=?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX:5060?
ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk PBX 18.24.3'

Oct  7 06:43:50.888839 spce proxy[837565]: NOTICE: ROUTE_CHECK_CLI_ALLOWED
<script>: User-provided CLI '?XXCallerIDXX?' accepted (matched user
preference) - R=?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX:5060?
ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk PBX 18.24.3'

Oct  7 06:43:50.888915 spce proxy[837565]: NOTICE: ROUTE_INVITE <script>:
Setting '?XXCallerIDXX?@?XXX.XXX.86.104?' as initiating user-provided CLI -
R=?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX:5060?
ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk PBX 18.24.3'

Oct  7 06:43:50.888963 spce proxy[837565]: NOTICE: ROUTE_INVITE <script>:
Setting '?XXCallerIDXX?@?XXX.XXX.86.104?' as initiating network-provided
CLI - R=?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX:5060?
ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk PBX 18.24.3'

Oct  7 06:43:50.889037 spce proxy[837565]: NOTICE: ROUTE_APPLY_REWRITE_RULE
<script>: Applying callee-in usr/domain rewrite rules on called party using
dpid '8' - R=?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX:5060?
ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk PBX 18.24.3'

Oct  7 06:43:50.890596 spce proxy[837565]: NOTICE: ROUTE_FIND_CALLEE
<script>: Callee is not local - R=?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX:5060?
ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk PBX 18.24.3'

Oct  7 06:43:50.890869 spce proxy[837565]: NOTICE: ROUTE_PSTN <script>:
Call to SIP Peering - R=?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX:5060?
ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk PBX 18.24.3'

Oct  7 06:43:50.890924 spce proxy[837565]: NOTICE: ROUTE_PSTN <script>:
Load gws matching calling part '?sip:XXCallerIDXX at XXX.XXX.86.104?' and
called user '?xxxxxxxxxxx2607244?' and called part
'?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX:5060?' -
R=?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX:5060?
ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk PBX 18.24.3'

Oct  7 06:43:50.891692 spce proxy[837565]: ERROR: ROUTE_PSTN_SELECT_PEER pv
[pv_core.c:336]: pv_get_ruri_attr(): failed to parse the R-URI

Oct  7 06:43:50.891806 spce proxy[837565]: ERROR: ROUTE_LOAD_PEER_OUT_PREF
pv [pv_core.c:261]: pv_get_ruri(): failed to parse the R-URI

Oct  7 06:43:50.891827 spce proxy[837565]: NOTICE: ROUTE_LOAD_PEER_OUT_PREF
<script>: Load peer preferences for peer host '1' - R=?<null>?
ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk PBX 18.24.3'

Oct  7 06:43:50.893690 spce proxy[837565]: ERROR: ROUTE_PSTN pv
[pv_core.c:336]: pv_get_ruri_attr(): failed to parse the R-URI

Oct  7 06:43:50.895431 spce proxy[837565]: ERROR: ROUTE_PSTN pv
[pv_core.c:336]: pv_get_ruri_attr(): failed to parse the R-URI

Oct  7 06:43:50.895538 spce proxy[837565]: ERROR: <null> pv
[pv_core.c:336]: pv_get_ruri_attr(): failed to parse the R-URI

Oct  7 06:43:50.895558 spce proxy[837565]: ERROR: <null> pv
[pv_core.c:336]: pv_get_ruri_attr(): failed to parse the R-URI

Oct  7 06:43:50.901133 spce proxy[837565]: ERROR: ROUTE_OUTBOUND pv
[pv_core.c:336]: pv_get_ruri_attr(): failed to parse the R-URI

Oct  7 06:43:50.901156 spce proxy[837565]: ERROR: ROUTE_OUTBOUND pv
[pv_core.c:336]: pv_get_ruri_attr(): failed to parse the R-URI

Oct  7 06:43:50.901829 spce proxy[837565]: ERROR: ROUTE_APPLY_REWRITE_RULE
pv [pv_core.c:261]: pv_get_ruri(): failed to parse the R-URI

Oct  7 06:43:50.901850 spce proxy[837565]: NOTICE: ROUTE_APPLY_REWRITE_RULE
<script>: Applying acc caller-out peer rewrite rules on acc caller party
using dpid '3' - R=?<null>? ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5?
UA='Asterisk PBX 18.24.3'

Oct  7 06:43:50.902770 spce proxy[837565]: ERROR: ROUTE_ACC_CALLER pv
[pv_core.c:261]: pv_get_ruri(): failed to parse the R-URI

Oct  7 06:43:50.902792 spce proxy[837565]: NOTICE: ROUTE_ACC_CALLER
<script>: Setting acc source-leg for uuid
'?86ba28d9-153f-4cee-800c-f34c1efea74a?':
'?{"gpp":["70770",null,null,null,null,null,null,null,null,null],"clir":0,"uuid":"86ba28d9-153f-4cee-800c-f34c1efea74a","u":"XXCallerIDXX","d":"XXX.XXX.86.104","cli":"XXCallerIDXX","s_id":"70770","c_id":"70770","a_id":2,"pau":"","par":"","s":"call","ip":"XXX.XXX.188.51","t":1728276230.902101,"lnp_p":"","cli_out":"XXCallerIDXX","lnp_t":"","pai":"","ppi":"","div":"","last_hih":"","cid":"","u2u":"","lcr":0,"cc_quota":"","cc_sub":"1","cc_cust":"1"}?'
- R=?<null>? ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk PBX
18.24.3'

Oct  7 06:43:50.903156 spce proxy[837565]: ERROR: ROUTE_APPLY_REWRITE_RULE
pv [pv_core.c:261]: pv_get_ruri(): failed to parse the R-URI

Oct  7 06:43:50.903176 spce proxy[837565]: NOTICE: ROUTE_APPLY_REWRITE_RULE
<script>: Applying acc callee-out peer rewrite rules on acc called party
using dpid '4' - R=?<null>? ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5?
UA='Asterisk PBX 18.24.3'

Oct  7 06:43:50.903656 spce proxy[837565]: ERROR: ROUTE_ACC_CALLEE pv
[pv_core.c:261]: pv_get_ruri(): failed to parse the R-URI

Oct  7 06:43:50.903679 spce proxy[837565]: NOTICE: ROUTE_ACC_CALLEE
<script>: Setting acc destination-leg for uuid '?0?':
'?{"gpp":[null,null,null,null,null,null,null,null,null,null],"plu":0,"s_id":"","c_id":"","a_id":0,"dialed":"xxxxxxxxxxx2607244","uuid":"0","u":"<null>","d":"<null>","u_in":"xxxxxxxxxxx2607244","d_in":"XXX.XXX.XXX.XXX","lcr":1,"lnp_p":"","u_out":"0","lnp_t":"","fci":"","cc_quota":"","cc_sub":"1","cc_cust":""}?'
- R=?<null>? ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk PBX
18.24.3'

Oct  7 06:43:50.903837 spce proxy[837565]: ERROR: ROUTE_OUTBOUND tm
[t_lookup.c:1827]: new_t(): uri invalid

Oct  7 06:43:50.903858 spce proxy[837565]: ERROR: ROUTE_OUTBOUND tm
[t_lookup.c:1979]: t_newtran(): new_t failed

Oct  7 06:43:50.903913 spce proxy[837565]: ERROR: ROUTE_OUTBOUND sl
[sl_funcs.c:428]: sl_reply_error(): stateless error reply used: I'm
terribly sorry, server error occurred (1/SL)

Oct  7 06:43:50.904591 spce proxy[837570]: NOTICE: sl:filtered-ack
<script>: New request on proxy - M=ACK
R=?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX:5060?
F=?sip:XXCallerIDXX at XXX.XXX.86.104?
T=?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX? IP=?<null>?:?<null>?
(?127.0.0.1?:?5060?) ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='<null>'
DESTIP=?127.0.0.1?:?5062?


****

Thanks


On Mon, Oct 7, 2024 at 1:11?AM Dario Busso <dbusso at sipwise.com> wrote:

> Looking at the error log, it seems that the INVITE received by SPCE is
> malformed.
> You need to check the peer system setup, not the SPCE.
>
>
> Dario B.
>
>
> Il lun 7 ott 2024, 06:50 Edo <edo.ekunwe at gmail.com> ha scritto:
>
>> Hi,
>>
>> I tried to make a call to a subscriber (SIP Trunk setup) and I got the
>> following error message.
>> *****
>>
>> Oct  7 06:43:50.891692 spce proxy[837565]: ERROR: ROUTE_PSTN_SELECT_PEER
>> pv [pv_core.c:336]: pv_get_ruri_attr(): failed to parse the R-URI
>>
>> Oct  7 06:43:50.891806 spce proxy[837565]: ERROR:
>> ROUTE_LOAD_PEER_OUT_PREF pv [pv_core.c:261]: pv_get_ruri(): failed to parse
>> the R-URI
>>
>> Oct  7 06:43:50.891827 spce proxy[837565]: NOTICE:
>> ROUTE_LOAD_PEER_OUT_PREF <script>: Load peer preferences for peer host '1'
>> - R=?<null>? ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk PBX
>> 18.24.3'
>>
>> Oct  7 06:43:50.893690 spce proxy[837565]: ERROR: ROUTE_PSTN pv
>> [pv_core.c:336]: pv_get_ruri_attr(): failed to parse the R-URI
>>
>> Oct  7 06:43:50.895431 spce proxy[837565]: ERROR: ROUTE_PSTN pv
>> [pv_core.c:336]: pv_get_ruri_attr(): failed to parse the R-URI
>>
>> Oct  7 06:43:50.895538 spce proxy[837565]: ERROR: <null> pv
>> [pv_core.c:336]: pv_get_ruri_attr(): failed to parse the R-URI
>>
>> Oct  7 06:43:50.895558 spce proxy[837565]: ERROR: <null> pv
>> [pv_core.c:336]: pv_get_ruri_attr(): failed to parse the R-URI
>>
>> Oct  7 06:43:50.901133 spce proxy[837565]: ERROR: ROUTE_OUTBOUND pv
>> [pv_core.c:336]: pv_get_ruri_attr(): failed to parse the R-URI
>>
>> Oct  7 06:43:50.901156 spce proxy[837565]: ERROR: ROUTE_OUTBOUND pv
>> [pv_core.c:336]: pv_get_ruri_attr(): failed to parse the R-URI
>>
>> Oct  7 06:43:50.901829 spce proxy[837565]: ERROR:
>> ROUTE_APPLY_REWRITE_RULE pv [pv_core.c:261]: pv_get_ruri(): failed to parse
>> the R-URI
>> *****
>>  I am looking for where I can set the system to a specific R-URI or check
>> the system to use network provided R-RURI.
>>
>> Please help.
>>
>> Thanks
>>
>>
>> --
>> -----
>> Ekunwe
>> Tel: 601.497.3932
>> --
>> Spce-user mailing list
>> Spce-user at lists.sipwise.com
>> http://lists.sipwise.com/mailman/listinfo/spce-user_lists.sipwise.com
>>
>

-- 
-----
Ekunwe
Tel: 601.497.3932
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From pa.ka12 at outlook.com  Mon Oct  7 16:51:48 2024
From: pa.ka12 at outlook.com (Pa Ka)
Date: Mon, 7 Oct 2024 20:51:48 +0000
Subject: [Spce-user] incoming Session fails using ngcp CE mr12.4.1 and
 SIP.js
In-Reply-To: <IA2P220MB1912D6FB6826CF76EECAF51AEA732@IA2P220MB1912.NAMP220.PROD.OUTLOOK.COM>
References: <IA2P220MB1912D25CDEB706382873774FEA722@IA2P220MB1912.NAMP220.PROD.OUTLOOK.COM>
 <IA2P220MB1912D6FB6826CF76EECAF51AEA732@IA2P220MB1912.NAMP220.PROD.OUTLOOK.COM>
Message-ID: <IA2P220MB1912006646C449DFF0BA69BAEA7D2@IA2P220MB1912.NAMP220.PROD.OUTLOOK.COM>

I was trying to upgrade from mr12.4.1 to mr 12.5.1 from azure vm, got this error when reach to this step ngcp-upgrade-pre-checks mr12.5.1 . kindly guide ! thank you


=> Running /tmp/ngcp-upgrade-pre-checks_5uE1SiWV/upgrade-mr12.4-to-mr12.5.1-checks-only
================================================================================
Sipwise C5 CE detected...
2024-10-07 16:47:27: [01/23] Running: /usr/share/ngcp-upgrade/steps/mr4.2/check_mandatory_variables
Time taken to run step mr4.2/check_mandatory_variables: 0.01s

2024-10-07 16:47:27: [02/23] Running: /usr/share/ngcp-upgrade/steps/mr5.5/report_check_start
Checking for upgrade from 'mr12.4.1' to 'mr12.5.1' started on Mon Oct  7 16:47:27 EDT 2024
Time taken to run step mr5.5/report_check_start: 0.02s

2024-10-07 16:47:27: [03/23] Running: /usr/share/ngcp-upgrade/steps/mr9.5/check_dedicated_fallback
ERROR: Fallback partition '/dev/mapper/ngcp-root' is not a block device
Time taken to run step mr9.5/check_dedicated_fallback: 0.01s

ERROR: the step 'mr9.5/check_dedicated_fallback' failed, upgrade check aborted!

From: Pa Ka <pa.ka12 at outlook.com>
Sent: Friday, October 4, 2024 8:51 PM
To: spce-user at lists.sipwise.com
Subject: Re: incoming Session fails using ngcp CE mr12.4.1 and SIP.js

Please ignore the previous email. I managed to resolve the issue it was a SIP.js error. Thanks
________________________________
From: Pa Ka <pa.ka12 at outlook.com<mailto:pa.ka12 at outlook.com>>
Sent: Friday, October 4, 2024 5:01:08 AM
To: spce-user at lists.sipwise.com<mailto:spce-user at lists.sipwise.com> <spce-user at lists.sipwise.com<mailto:spce-user at lists.sipwise.com>>
Subject: incoming Session fails using ngcp CE mr12.4.1 and SIP.js


Hey Please help me to understand this. incoming session on my SIP.js application are failing.



But, outgoingcall are functioning, I can initiate call to the PSTN.



Second, Incoming session works perfectly when using Zoiper softphone application installed on Windows ( Zoiper and SIP.js are using the same configuration from ngcp c5 switch which make me think it is not ngcp c5 issue, but prefer to ask here!!!)



No 404 fallback is enabled on the portal to the specific address used to log into both phone.



This IP 172.110.70.15 , is for a local pc





I'd would like to know if it is SIP.js codes or Sipwise (ngcp CE mr12.4.1 ) configuration that cause the error.  Below are the sip proxy from Kamailio-proxy.log .





When call from PSTN arrives, the script rewrite called party to remove the leading 1



NOTICE: ROUTE_APPLY_REWRITE_RULE <script>: Rewriting called party '<199999101>' to '<99999101>' - R=<sip:19999101 at 172.110.70.155:57432;transport=tcp;rinstance=6492442> ID=<1B6D3B76 at 208.72.120.66> UA='<null>'



Oct  4 04:31:39.478180 sips proxy[1641]: NOTICE: BRANCH_ROUTE_SBC <script>: Setting P-Called-Party-ID 'sip:<99999101>@<sip.sipdomain.com;user=phone><sip:>' - R=<sip:99999101 at 172.110.70.155:57432;transport=tcp;rinstance=6492442;user=phone> ID=<1B6D3B76 at pstnIP.66> UA='<null>'





This is the first request to the B2B

New request on proxy for the B LEG of the call - M=INVITE R=<sip:99999101 at 172.110.70.155:57432;transport=tcp;rinstance=6492442;user=phone> F=<sip:calllerNumber64 at sip.sipdomain.com;user=phone> T=<sip:99999101 at sip.sipdomain..com;user=phone> IP=<<null>>:<<null>> (<127.0.0.1>:<5080>) ID=<1B6D3B76 at pstnIP_b2b-1> UA='<null>' DESTIP=<127.0.0.1>:<6052>



Oct  4 04:31:39.539462 sips proxy[1650]: NOTICE: REPLY_ROUTE_NAT_CONTINUE <script>: Sending reply S=404 M=INVITE fs='<127.0.0.1>:<6052>' du='<127.0.0.1>:<6050>' - ID=<1B6D3B76 at pstnIP> UA='Vogat/1.0'



After B2B, it fails with this error



Oct  4 04:31:39.539496 sips proxy[1650]: NOTICE: FAILURE_ROUTE_LOCAL <script>: Failure route for local call, code='404' - R=<sip:usernamesubcriber999 at 172.110.70.155:57432;transport=tcp;rinstance=6492442> ID=<1B6D3B76 at pstnIP> UA='<null>'

Oct  4 04:31:39.540023 sips proxy[1650]: NOTICE: FAILURE_ROUTE_LOCAL <script>: Fallback from E164 to user DISABLED due to 'no_404_fallback' set - R=<sip:usernamesubscriber999 at 172.110.70.155:57432;transport=tcp;rinstance=6492442> ID=<1B6D3B76 at pstnIP> UA='<null>'

thank you
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From vdubois at b2b2c.com  Tue Oct  1 09:42:02 2024
From: vdubois at b2b2c.com (Vincent Dubois)
Date: Tue, 1 Oct 2024 13:42:02 +0000
Subject: [Spce-user] How to setup subscriber as SIP Trunk
In-Reply-To: <CAFkRdGtpiVMK67fxOOO4vX=bmtzVqY_Q2iE-Us9k3fhaa3R-Rg@mail.gmail.com>
References: <CAFkRdGtpiVMK67fxOOO4vX=bmtzVqY_Q2iE-Us9k3fhaa3R-Rg@mail.gmail.com>
Message-ID: <YQBPR0101MB4650334A5FAE46DD11BFB087A6772@YQBPR0101MB4650.CANPRD01.PROD.OUTLOOK.COM>

Hi

If you do not want authentication, and just want to validate based on IP address this is what we do:
(PS: in our case everything goes thru an SBC first, so it is just different internal IPs of our SBCs)

In the subscriber:

  *   add a Permanent Registration like
contact: sip:sip at 1.2.3.4:5060  with the IP of the destination
expires: 2157-01-01

In the subscriber preferences

  *   under Trusted Sources you want to add their IP again something like
Source IP: 1.2.3.4
Protocol: ANY
From-Pattern: .*

  *   under Number Manipulation
e164_to_ruri: true     (unless that customer only has one number, in that case you could have put the number in the permanent registration, and this wouldn?t be needed)

  *   under Access Restrictions
allowed_clis: *   (again if they have use multiple numbers, if not then you could put the number there)

I would not recommend this with public IP, because source IP can be faked and any Invite from the IP would be accepted.
But in case of private IP space, or like us private SBC Ips, it works well.

PS: in our case we actually use a different interface, so in the permanent registration we set the socket to that other interface, and in sub pref (under NAT and Media Flow Control) we also set the rtp_interface to that other interface.


Vincent Dubois | Analyste programmeur
B2B2C
514-908-5420

From: Edo <edo.ekunwe at gmail.com>
Sent: Sunday, September 29, 2024 6:20 AM
To: spce-user at lists.sipwise.com
Subject: [Spce-user] How to setup subscriber as SIP Trunk

CAUTION : External Mail. Do not click on links or open attachments you do not trust.
ATTENTION : Courriel externe. Ne cliquez pas sur des liens et n'ouvrez pas de pi?ces jointes auxquels vous ne faites pas confiance.

All I am looking for example configuration(s) of how to setup a subscriber as a SIP trunk as mentioned below. I am really trying to handle multiple users behind another IP PBX system with IP authentication
[cid:image001.png at 01DB13E4.15AC0D70]
I looked at subscriber preferences but could not find any idea leading to success.

Please assist.

--
-----
Ekunwe
Tel: 601.497.3932
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From dbusso at sipwise.com  Tue Oct  8 02:59:12 2024
From: dbusso at sipwise.com (Dario Busso)
Date: Tue, 8 Oct 2024 08:59:12 +0200
Subject: [Spce-user] [EXTERNAL] RE: incoming Session fails using ngcp CE
 mr12.4.1 and SIP.js
In-Reply-To: <IA2P220MB1912006646C449DFF0BA69BAEA7D2@IA2P220MB1912.NAMP220.PROD.OUTLOOK.COM>
References: <IA2P220MB1912D25CDEB706382873774FEA722@IA2P220MB1912.NAMP220.PROD.OUTLOOK.COM>
 <IA2P220MB1912D6FB6826CF76EECAF51AEA732@IA2P220MB1912.NAMP220.PROD.OUTLOOK.COM>
 <IA2P220MB1912006646C449DFF0BA69BAEA7D2@IA2P220MB1912.NAMP220.PROD.OUTLOOK.COM>
Message-ID: <CAAyzhJRQXiQ1bRirzK5e=hezcRV9iqYc5ZTJU_Ghg2tRg9ZPFA@mail.gmail.com>

Execute this commands and paste the output, please.
mount
df -h

Dario

On Mon, Oct 7, 2024 at 10:51?PM Pa Ka <pa.ka12 at outlook.com> wrote:

> I was trying to upgrade from mr12.4.1 to mr 12.5.1 from azure vm, got this
> error when reach to this step ngcp-upgrade-pre-checks mr12.5.1 . kindly
> guide ! thank you
>
>
> => Running
> /tmp/ngcp-upgrade-pre-checks_5uE1SiWV/upgrade-mr12.4-to-mr12.5.1-checks-only
>
>
> ================================================================================
>
> Sipwise C5 CE detected...
>
> 2024-10-07 16:47:27: [01/23] Running:
> /usr/share/ngcp-upgrade/steps/mr4.2/check_mandatory_variables
>
> Time taken to run step mr4.2/check_mandatory_variables: 0.01s
>
>
>
> 2024-10-07 16:47:27: [02/23] Running:
> /usr/share/ngcp-upgrade/steps/mr5.5/report_check_start
>
> Checking for upgrade from 'mr12.4.1' to 'mr12.5.1' started on Mon Oct  7
> 16:47:27 EDT 2024
>
> Time taken to run step mr5.5/report_check_start: 0.02s
>
>
>
> 2024-10-07 16:47:27: [03/23] Running:
> /usr/share/ngcp-upgrade/steps/mr9.5/check_dedicated_fallback
>
> ERROR: Fallback partition '/dev/mapper/ngcp-root' is not a block device
>
> Time taken to run step mr9.5/check_dedicated_fallback: 0.01s
>
>
>
> ERROR: the step 'mr9.5/check_dedicated_fallback' failed, upgrade check
> aborted!
>
>
>
> *From:* Pa Ka <pa.ka12 at outlook.com>
> *Sent:* Friday, October 4, 2024 8:51 PM
> *To:* spce-user at lists.sipwise.com
> *Subject:* Re: incoming Session fails using ngcp CE mr12.4.1 and SIP.js
>
>
>
> Please ignore the previous email. I managed to resolve the issue it was a
> SIP.js error. Thanks
> ------------------------------
>
> *From:* Pa Ka <pa.ka12 at outlook.com>
> *Sent:* Friday, October 4, 2024 5:01:08 AM
> *To:* spce-user at lists.sipwise.com <spce-user at lists.sipwise.com>
> *Subject:* incoming Session fails using ngcp CE mr12.4.1 and SIP.js
>
>
>
> Hey Please help me to understand this. incoming session on my SIP.js
> application are failing.
>
>
>
> *But, outgoingcall are functioning, I can initiate call to the PSTN.*
>
>
>
> *Second, Incoming session works perfectly when using Zoiper softphone
> application installed on Windows ( Zoiper and SIP.js are using the same
> configuration from ngcp c5 switch which make me think it is not ngcp c5
> issue, but prefer to ask here!!!)*
>
>
>
> *No 404 fallback is enabled on the portal to the specific address used to
> log into both phone.*
>
>
>
> *This IP 172.110.70.15 , is for a local pc*
>
>
>
>
>
> I?d would like to know if it is SIP.js codes or Sipwise (ngcp CE mr12.4.1
> ) configuration that cause the error.  Below are the sip proxy from
> Kamailio-proxy.log .
>
>
>
>
>
> When call from PSTN arrives, the script rewrite called party to remove the
> leading 1
>
>
>
> NOTICE: ROUTE_APPLY_REWRITE_RULE <script>: Rewriting called party
> '?199999101?' to '?99999101?' - R=?sip:19999101 at 172.110.70.155:57432;transport=tcp;rinstance=6492442?
> ID=?1B6D3B76 at 208.72.120.66? UA='<null>'
>
>
>
> Oct  4 04:31:39.478180 sips proxy[1641]: NOTICE: BRANCH_ROUTE_SBC
> <script>: Setting P-Called-Party-ID '
> sip:?99999101?@?sip.sipdomain.com;user=phone?' -
> R=?sip:99999101 at 172.110.70.155:57432;transport=tcp;rinstance=6492442;user=phone?
> ID=?1B6D3B76 at pstnIP.66? UA='<null>'
>
>
>
>
>
> This is the first request to the B2B
>
> New request on proxy for the B LEG of the call - M=INVITE
> R=?sip:99999101 at 172.110.70.155:57432;transport=tcp;rinstance=6492442;user=phone?
> F=?sip:calllerNumber64 at sip.sipdomain.com;user=phone?
> T=?sip:99999101 at sip.sipdomain..com;user=phone? IP=?<null>?:?<null>?
> (?127.0.0.1?:?5080?) ID=?1B6D3B76 at pstnIP_b2b-1? UA='<null>'
> DESTIP=?127.0.0.1?:?6052?
>
>
>
> Oct  4 04:31:39.539462 sips proxy[1650]: NOTICE: REPLY_ROUTE_NAT_CONTINUE
> <script>: Sending reply S=404 M=INVITE fs='?127.0.0.1?:?6052?'
> du='?127.0.0.1?:?6050?' - ID=?1B6D3B76 at pstnIP? UA='Vogat/1.0'
>
>
>
> After B2B, it fails with this error
>
>
>
> Oct  4 04:31:39.539496 sips proxy[1650]: NOTICE: FAILURE_ROUTE_LOCAL
> <script>: Failure route for local call, code='404' -
> R=?sip:usernamesubcriber999 at 172.110.70.155:57432;transport=tcp;rinstance=6492442?
> ID=?1B6D3B76 at pstnIP? UA='<null>'
>
> Oct  4 04:31:39.540023 sips proxy[1650]: NOTICE: FAILURE_ROUTE_LOCAL
> <script>: Fallback from E164 to user DISABLED due to 'no_404_fallback' set
> - R=?sip:usernamesubscriber999 at 172.110.70.155:57432;transport=tcp;rinstance=6492442?
> ID=?1B6D3B76 at pstnIP? UA='<null>'
>
> thank you
>
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From pa.ka12 at outlook.com  Tue Oct  8 08:11:01 2024
From: pa.ka12 at outlook.com (Pa Ka)
Date: Tue, 8 Oct 2024 12:11:01 +0000
Subject: [Spce-user] [EXTERNAL] RE: incoming Session fails using ngcp CE
 mr12.4.1 and SIP.js
In-Reply-To: <CAAyzhJRQXiQ1bRirzK5e=hezcRV9iqYc5ZTJU_Ghg2tRg9ZPFA@mail.gmail.com>
References: <IA2P220MB1912D25CDEB706382873774FEA722@IA2P220MB1912.NAMP220.PROD.OUTLOOK.COM>
 <IA2P220MB1912D6FB6826CF76EECAF51AEA732@IA2P220MB1912.NAMP220.PROD.OUTLOOK.COM>
 <IA2P220MB1912006646C449DFF0BA69BAEA7D2@IA2P220MB1912.NAMP220.PROD.OUTLOOK.COM>
 <CAAyzhJRQXiQ1bRirzK5e=hezcRV9iqYc5ZTJU_Ghg2tRg9ZPFA@mail.gmail.com>
Message-ID: <IA2P220MB1912C96FE7EA690D3C0EC3E5EA7E2@IA2P220MB1912.NAMP220.PROD.OUTLOOK.COM>

This is the result


root at sips:/home/sysadmin# df -h

Filesystem      Size  Used Avail Use% Mounted on
udev            1.9G     0  1.9G   0% /dev
tmpfs           388M  912K  387M   1% /run
/dev/sdb1        30G  5.9G   22G  21% /
tmpfs           1.9G  192K  1.9G   1% /dev/shm
tmpfs           5.0M     0  5.0M   0% /run/lock
/dev/sdb15      124M   12M  113M  10% /boot/efi
/dev/sda1       7.8G   24K  7.4G   1% /mnt
tmpfs           388M  4.0K  388M   1% /run/user/1000

From: Dario Busso <dbusso at sipwise.com>
Sent: Tuesday, October 8, 2024 2:59 AM
To: Pa Ka <pa.ka12 at outlook.com>
Cc: spce-user at lists.sipwise.com
Subject: Re: [EXTERNAL] RE: incoming Session fails using ngcp CE mr12.4.1 and SIP.js

Execute this commands and paste the output, please.
mount
df -h

Dario

On Mon, Oct 7, 2024 at 10:51?PM Pa Ka <pa.ka12 at outlook.com<mailto:pa.ka12 at outlook.com>> wrote:
I was trying to upgrade from mr12.4.1 to mr 12.5.1 from azure vm, got this error when reach to this step ngcp-upgrade-pre-checks mr12.5.1 . kindly guide ! thank you


=> Running /tmp/ngcp-upgrade-pre-checks_5uE1SiWV/upgrade-mr12.4-to-mr12.5.1-checks-only
================================================================================
Sipwise C5 CE detected...
2024-10-07 16:47:27: [01/23] Running: /usr/share/ngcp-upgrade/steps/mr4.2/check_mandatory_variables
Time taken to run step mr4.2/check_mandatory_variables: 0.01s

2024-10-07 16:47:27: [02/23] Running: /usr/share/ngcp-upgrade/steps/mr5.5/report_check_start
Checking for upgrade from 'mr12.4.1' to 'mr12.5.1' started on Mon Oct  7 16:47:27 EDT 2024
Time taken to run step mr5.5/report_check_start: 0.02s

2024-10-07 16:47:27: [03/23] Running: /usr/share/ngcp-upgrade/steps/mr9.5/check_dedicated_fallback
ERROR: Fallback partition '/dev/mapper/ngcp-root' is not a block device
Time taken to run step mr9.5/check_dedicated_fallback: 0.01s

ERROR: the step 'mr9.5/check_dedicated_fallback' failed, upgrade check aborted!

From: Pa Ka <pa.ka12 at outlook.com<mailto:pa.ka12 at outlook.com>>
Sent: Friday, October 4, 2024 8:51 PM
To: spce-user at lists.sipwise.com<mailto:spce-user at lists.sipwise.com>
Subject: Re: incoming Session fails using ngcp CE mr12.4.1 and SIP.js

Please ignore the previous email. I managed to resolve the issue it was a SIP.js error. Thanks
________________________________
From: Pa Ka <pa.ka12 at outlook.com<mailto:pa.ka12 at outlook.com>>
Sent: Friday, October 4, 2024 5:01:08 AM
To: spce-user at lists.sipwise.com<mailto:spce-user at lists.sipwise.com> <spce-user at lists.sipwise.com<mailto:spce-user at lists.sipwise.com>>
Subject: incoming Session fails using ngcp CE mr12.4.1 and SIP.js


Hey Please help me to understand this. incoming session on my SIP.js application are failing.



But, outgoingcall are functioning, I can initiate call to the PSTN.



Second, Incoming session works perfectly when using Zoiper softphone application installed on Windows ( Zoiper and SIP.js are using the same configuration from ngcp c5 switch which make me think it is not ngcp c5 issue, but prefer to ask here!!!)



No 404 fallback is enabled on the portal to the specific address used to log into both phone.



This IP 172.110.70.15 , is for a local pc





I?d would like to know if it is SIP.js codes or Sipwise (ngcp CE mr12.4.1 ) configuration that cause the error.  Below are the sip proxy from Kamailio-proxy.log .





When call from PSTN arrives, the script rewrite called party to remove the leading 1



NOTICE: ROUTE_APPLY_REWRITE_RULE <script>: Rewriting called party '?199999101?' to '?99999101?' - R=?sip:19999101 at 172.110.70.155:57432;transport=tcp;rinstance=6492442? ID=?1B6D3B76 at 208.72.120.66<mailto:1B6D3B76 at 208.72.120.66>? UA='<null>'



Oct  4 04:31:39.478180 sips proxy[1641]: NOTICE: BRANCH_ROUTE_SBC <script>: Setting P-Called-Party-ID 'sip:?99999101?@?sip.sipdomain.com;user=phone?' - R=?sip:99999101 at 172.110.70.155:57432;transport=tcp;rinstance=6492442;user=phone? ID=?1B6D3B76 at pstnIP.66? UA='<null>'





This is the first request to the B2B

New request on proxy for the B LEG of the call - M=INVITE R=?sip:99999101 at 172.110.70.155:57432;transport=tcp;rinstance=6492442;user=phone? F=?sip:calllerNumber64 at sip.sipdomain.com<mailto:sip%3AcalllerNumber64 at sip.sipdomain.com>;user=phone? T=?sip:99999101 at sip.sipdomain..com;user=phone? IP=?<null>?:?<null>? (?127.0.0.1?:?5080?) ID=?1B6D3B76 at pstnIP_b2b-1? UA='<null>' DESTIP=?127.0.0.1?:?6052?



Oct  4 04:31:39.539462 sips proxy[1650]: NOTICE: REPLY_ROUTE_NAT_CONTINUE <script>: Sending reply S=404 M=INVITE fs='?127.0.0.1?:?6052?' du='?127.0.0.1?:?6050?' - ID=?1B6D3B76 at pstnIP? UA='Vogat/1.0'



After B2B, it fails with this error



Oct  4 04:31:39.539496 sips proxy[1650]: NOTICE: FAILURE_ROUTE_LOCAL <script>: Failure route for local call, code='404' - R=?sip:usernamesubcriber999 at 172.110.70.155:57432;transport=tcp;rinstance=6492442? ID=?1B6D3B76 at pstnIP? UA='<null>'

Oct  4 04:31:39.540023 sips proxy[1650]: NOTICE: FAILURE_ROUTE_LOCAL <script>: Fallback from E164 to user DISABLED due to 'no_404_fallback' set - R=?sip:usernamesubscriber999 at 172.110.70.155:57432;transport=tcp;rinstance=6492442? ID=?1B6D3B76 at pstnIP? UA='<null>'

thank you
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From dbusso at sipwise.com  Tue Oct  8 12:19:13 2024
From: dbusso at sipwise.com (Dario Busso)
Date: Tue, 8 Oct 2024 18:19:13 +0200
Subject: [Spce-user] [EXTERNAL] RE: incoming Session fails using ngcp CE
 mr12.4.1 and SIP.js
In-Reply-To: <IA2P220MB1912C96FE7EA690D3C0EC3E5EA7E2@IA2P220MB1912.NAMP220.PROD.OUTLOOK.COM>
References: <IA2P220MB1912D25CDEB706382873774FEA722@IA2P220MB1912.NAMP220.PROD.OUTLOOK.COM>
 <IA2P220MB1912D6FB6826CF76EECAF51AEA732@IA2P220MB1912.NAMP220.PROD.OUTLOOK.COM>
 <IA2P220MB1912006646C449DFF0BA69BAEA7D2@IA2P220MB1912.NAMP220.PROD.OUTLOOK.COM>
 <CAAyzhJRQXiQ1bRirzK5e=hezcRV9iqYc5ZTJU_Ghg2tRg9ZPFA@mail.gmail.com>
 <IA2P220MB1912C96FE7EA690D3C0EC3E5EA7E2@IA2P220MB1912.NAMP220.PROD.OUTLOOK.COM>
Message-ID: <CAAyzhJT0HwcJZRN4kjot3foErUjs=s+-CPB__qcHNbzhQJUafQ@mail.gmail.com>

As you can see the expected partition mapping for the upgrade is not
present in you deployment.
The upgrade scripts expects something like this.
~# df -h
Filesystem                 Size  Used Avail Use% Mounted on
udev                        32G     0   32G   0% /dev
tmpfs                      6.3G  1.4M  6.3G   1% /run
/dev/mapper/ngcp-fallback  9.8G  5.3G  4.0G  58% /
tmpfs                       32G   33M   32G   1% /dev/shm
tmpfs                      5.0M     0  5.0M   0% /run/lock
/dev/mapper/ngcp-root      9.8G  4.7G  4.6G  51% /ngcp-fallback
/dev/sdb2                  486M  176K  485M   1% /boot/efi
/dev/mapper/ngcp-data      377G   28G  330G   8% /ngcp-data
tmpfs                      6.3G     0  6.3G   0% /run/user/0
With partitions
/dev/mapper/ngcp-root
/dev/mapper/ngcp-fallback
I guess the only options is to re-install.
Dario


On Tue, Oct 8, 2024 at 2:11?PM Pa Ka <pa.ka12 at outlook.com> wrote:

> This is the result
>
>
> root at sips:/home/sysadmin# df -h
>
>
>
> Filesystem      Size  Used Avail Use% Mounted on
>
> udev            1.9G     0  1.9G   0% /dev
>
> tmpfs           388M  912K  387M   1% /run
>
> /dev/sdb1        30G  5.9G   22G  21% /
>
> tmpfs           1.9G  192K  1.9G   1% /dev/shm
>
> tmpfs           5.0M     0  5.0M   0% /run/lock
>
> /dev/sdb15      124M   12M  113M  10% /boot/efi
>
> /dev/sda1       7.8G   24K  7.4G   1% /mnt
>
> tmpfs           388M  4.0K  388M   1% /run/user/1000
>
>
>
> *From:* Dario Busso <dbusso at sipwise.com>
> *Sent:* Tuesday, October 8, 2024 2:59 AM
> *To:* Pa Ka <pa.ka12 at outlook.com>
> *Cc:* spce-user at lists.sipwise.com
> *Subject:* Re: [EXTERNAL] RE: incoming Session fails using ngcp CE
> mr12.4.1 and SIP.js
>
>
>
> Execute this commands and paste the output, please.
>
> mount
>
> df -h
>
>
>
> Dario
>
>
>
> On Mon, Oct 7, 2024 at 10:51?PM Pa Ka <pa.ka12 at outlook.com> wrote:
>
> I was trying to upgrade from mr12.4.1 to mr 12.5.1 from azure vm, got this
> error when reach to this step ngcp-upgrade-pre-checks mr12.5.1 . kindly
> guide ! thank you
>
>
> => Running
> /tmp/ngcp-upgrade-pre-checks_5uE1SiWV/upgrade-mr12.4-to-mr12.5.1-checks-only
>
>
> ================================================================================
>
> Sipwise C5 CE detected...
>
> 2024-10-07 16:47:27: [01/23] Running:
> /usr/share/ngcp-upgrade/steps/mr4.2/check_mandatory_variables
>
> Time taken to run step mr4.2/check_mandatory_variables: 0.01s
>
>
>
> 2024-10-07 16:47:27: [02/23] Running:
> /usr/share/ngcp-upgrade/steps/mr5.5/report_check_start
>
> Checking for upgrade from 'mr12.4.1' to 'mr12.5.1' started on Mon Oct  7
> 16:47:27 EDT 2024
>
> Time taken to run step mr5.5/report_check_start: 0.02s
>
>
>
> 2024-10-07 16:47:27: [03/23] Running:
> /usr/share/ngcp-upgrade/steps/mr9.5/check_dedicated_fallback
>
> ERROR: Fallback partition '/dev/mapper/ngcp-root' is not a block device
>
> Time taken to run step mr9.5/check_dedicated_fallback: 0.01s
>
>
>
> ERROR: the step 'mr9.5/check_dedicated_fallback' failed, upgrade check
> aborted!
>
>
>
> *From:* Pa Ka <pa.ka12 at outlook.com>
> *Sent:* Friday, October 4, 2024 8:51 PM
> *To:* spce-user at lists.sipwise.com
> *Subject:* Re: incoming Session fails using ngcp CE mr12.4.1 and SIP.js
>
>
>
> Please ignore the previous email. I managed to resolve the issue it was a
> SIP.js error. Thanks
> ------------------------------
>
> *From:* Pa Ka <pa.ka12 at outlook.com>
> *Sent:* Friday, October 4, 2024 5:01:08 AM
> *To:* spce-user at lists.sipwise.com <spce-user at lists.sipwise.com>
> *Subject:* incoming Session fails using ngcp CE mr12.4.1 and SIP.js
>
>
>
> Hey Please help me to understand this. incoming session on my SIP.js
> application are failing.
>
>
>
> *But, outgoingcall are functioning, I can initiate call to the PSTN.*
>
>
>
> *Second, Incoming session works perfectly when using Zoiper softphone
> application installed on Windows ( Zoiper and SIP.js are using the same
> configuration from ngcp c5 switch which make me think it is not ngcp c5
> issue, but prefer to ask here!!!)*
>
>
>
> *No 404 fallback is enabled on the portal to the specific address used to
> log into both phone.*
>
>
>
> *This IP 172.110.70.15 , is for a local pc*
>
>
>
>
>
> I?d would like to know if it is SIP.js codes or Sipwise (ngcp CE mr12.4.1
> ) configuration that cause the error.  Below are the sip proxy from
> Kamailio-proxy.log .
>
>
>
>
>
> When call from PSTN arrives, the script rewrite called party to remove the
> leading 1
>
>
>
> NOTICE: ROUTE_APPLY_REWRITE_RULE <script>: Rewriting called party
> '?199999101?' to '?99999101?' - R=?sip:19999101 at 172.110.70.155:57432;transport=tcp;rinstance=6492442?
> ID=?1B6D3B76 at 208.72.120.66? UA='<null>'
>
>
>
> Oct  4 04:31:39.478180 sips proxy[1641]: NOTICE: BRANCH_ROUTE_SBC
> <script>: Setting P-Called-Party-ID 'sip:?99999101?@?sip.sipdomain.com;user=phone?'
> - R=?sip:99999101 at 172.110.70.155:57432;transport=tcp;rinstance=6492442;user=phone?
> ID=?1B6D3B76 at pstnIP.66? UA='<null>'
>
>
>
>
>
> This is the first request to the B2B
>
> New request on proxy for the B LEG of the call - M=INVITE
> R=?sip:99999101 at 172.110.70.155:57432;transport=tcp;rinstance=6492442;user=phone?
> F=?sip:calllerNumber64 at sip.sipdomain.com;user=phone?
> T=?sip:99999101 at sip.sipdomain..com;user=phone? IP=?<null>?:?<null>?
> (?127.0.0.1?:?5080?) ID=?1B6D3B76 at pstnIP_b2b-1? UA='<null>'
> DESTIP=?127.0.0.1?:?6052?
>
>
>
> Oct  4 04:31:39.539462 sips proxy[1650]: NOTICE: REPLY_ROUTE_NAT_CONTINUE
> <script>: Sending reply S=404 M=INVITE fs='?127.0.0.1?:?6052?'
> du='?127.0.0.1?:?6050?' - ID=?1B6D3B76 at pstnIP? UA='Vogat/1.0'
>
>
>
> After B2B, it fails with this error
>
>
>
> Oct  4 04:31:39.539496 sips proxy[1650]: NOTICE: FAILURE_ROUTE_LOCAL
> <script>: Failure route for local call, code='404' -
> R=?sip:usernamesubcriber999 at 172.110.70.155:57432;transport=tcp;rinstance=6492442?
> ID=?1B6D3B76 at pstnIP? UA='<null>'
>
> Oct  4 04:31:39.540023 sips proxy[1650]: NOTICE: FAILURE_ROUTE_LOCAL
> <script>: Fallback from E164 to user DISABLED due to 'no_404_fallback' set
> - R=?sip:usernamesubscriber999 at 172.110.70.155:57432;transport=tcp;rinstance=6492442?
> ID=?1B6D3B76 at pstnIP? UA='<null>'
>
> thank you
>
>
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From dbusso at sipwise.com  Wed Oct  9 03:47:14 2024
From: dbusso at sipwise.com (Dario Busso)
Date: Wed, 9 Oct 2024 09:47:14 +0200
Subject: [Spce-user] [EXTERNAL]  Troubleshooting request
In-Reply-To: <CAFkRdGuEw-FfHym8Of+fUf3aYvcC7+==nB_dyi5qMJJg91A=nQ@mail.gmail.com>
References: <CAFkRdGtKH2MtMwdPoQES6JdU+yjSbYG8wp2dzhpfhkFMUeDH3w@mail.gmail.com>
 <CAAyzhJTi6W28bqohSAqJtP5FgyyAeeuqbKK_BTxoE+f+Yzs1yA@mail.gmail.com>
 <CAFkRdGuEw-FfHym8Of+fUf3aYvcC7+==nB_dyi5qMJJg91A=nQ@mail.gmail.com>
Message-ID: <CAAyzhJTwqX6UvLBHeo6YmSEkw-0ibfiPCmSf_1ffJzfYTLfbjQ@mail.gmail.com>

Hi Edo.
Can you share the full SIP INVITE message the kamailio proxy receive?
Dario

On Mon, Oct 7, 2024 at 3:36?PM Edo <edo.ekunwe at gmail.com> wrote:

> Hi Dario,
>
> I am not sure if your earlier response of malformed invite was directed at
> me so I decided to paste the sip logs from a sample call, maybe we can get
> more information that may help with troubleshooting.
>
> ****
>
> Oct  7 06:43:50.807051 spce proxy[837572]: NOTICE: DEFAULT_ROUTE
> <script>: New request on proxy - M=INVITE
> R=?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX:5060?
> F=?sip:XXCallerIDXX at XXX.XXX.86.104?
> T=?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX? IP=?XXX.XXX.188.51?:?5060?
> (?127.0.0.1?:?5060?) ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk
> PBX 18.24.3' DESTIP=?127.0.0.1?:?5062?
>
> Oct  7 06:43:50.807084 spce proxy[837572]: NOTICE: DEFAULT_ROUTE
> <script>: Sending reply S=100 Trying M=INVITE fs='?127.0.0.1?:?5062?'
> du='?127.0.0.1?:?5060?' - R=?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX:5060?
> ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk PBX 18.24.3'
>
> Oct  7 06:43:50.807375 spce proxy[837572]: NOTICE:
> ROUTE_LOAD_CALLEE_DOMAIN_PREF <script>: Load domain preferences for callee
> '?XXX.XXX.XXX.XXX?' - R=?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX:5060?
> ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk PBX 18.24.3'
>
> Oct  7 06:43:50.811499 spce proxy[837572]: NOTICE: ROUTE_AUTH <script>:
> Authentication failed, no credentials -
> R=?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX:5060?
> ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk PBX 18.24.3'
> Auth=?<null>?
>
> Oct  7 06:43:50.811581 spce proxy[837572]: NOTICE: ROUTE_AUTH <script>:
> Sending reply S=407 fs='?127.0.0.1?:?5062?' du='?127.0.0.1?:?5060?' -
> ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk PBX 18.24.3'
>
> Oct  7 06:43:50.812257 spce proxy[837568]: NOTICE: sl:filtered-ack
> <script>: New request on proxy - M=ACK
> R=?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX:5060?
> F=?sip:XXCallerIDXX at XXX.XXX.86.104?
> T=?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX? IP=?<null>?:?<null>?
> (?127.0.0.1?:?5060?) ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='<null>'
> DESTIP=?127.0.0.1?:?5062?
>
> Oct  7 06:43:50.872198 spce proxy[837565]: NOTICE: DEFAULT_ROUTE
> <script>: New request on proxy - M=INVITE
> R=?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX:5060?
> F=?sip:XXCallerIDXX at XXX.XXX.86.104?
> T=?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX? IP=?XXX.XXX.188.51?:?5060?
> (?127.0.0.1?:?5060?) ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk
> PBX 18.24.3' DESTIP=?127.0.0.1?:?5062?
>
> Oct  7 06:43:50.872229 spce proxy[837565]: NOTICE: DEFAULT_ROUTE
> <script>: Sending reply S=100 Trying M=INVITE fs='?127.0.0.1?:?5062?'
> du='?127.0.0.1?:?5060?' - R=?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX:5060?
> ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk PBX 18.24.3'
>
> Oct  7 06:43:50.872504 spce proxy[837565]: NOTICE:
> ROUTE_LOAD_CALLEE_DOMAIN_PREF <script>: Load domain preferences for callee
> '?XXX.XXX.XXX.XXX?' - R=?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX:5060?
> ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk PBX 18.24.3'
>
> Oct  7 06:43:50.877414 spce proxy[837565]: NOTICE: ROUTE_FIND_CALLER
> <script>: Sending reply S=101 Connecting fs='?127.0.0.1?:?5062?'
> du='?127.0.0.1?:?5060?' - R=?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX:5060?
> ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk PBX 18.24.3'
>
> Oct  7 06:43:50.877751 spce proxy[837565]: NOTICE: ROUTE_LOAD_CALLER_PREF
> <script>: Load caller preferences for uuid
> '?86ba28d9-153f-4cee-800c-f34c1efea74a?' and domain '?XXX.XXX.86.104?' -
> R=?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX:5060?
> ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk PBX 18.24.3'
>
> Oct  7 06:43:50.883279 spce proxy[837565]: NOTICE:
> ROUTE_LOAD_CALLER_CONTRACT_PREF <script>: Load caller contract preferences
> for contract id '?2?' and ip '?XXX.XXX.188.51?' -
> R=?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX:5060?
> ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk PBX 18.24.3'
>
> Oct  7 06:43:50.883583 spce proxy[837565]: NOTICE:
> ROUTE_LOAD_CALLER_RESELLER_PREF <script>: Load caller reseller preferences
> for reseller id '?1?' - R=?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX:5060?
> ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk PBX 18.24.3'
>
> Oct  7 06:43:50.884273 spce proxy[837565]: NOTICE: ROUTE_DLG_MANAGE
> <script>: Dialog managed,
> lua_dlg_callid:[?3e2e30bf-6132-478a-bdd0-292e139c39c5?] -
> R=?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX:5060?
> ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk PBX 18.24.3'
>
> Oct  7 06:43:50.884348 spce proxy[837565]: NOTICE: ROUTE_GET_CALLER_CLI
> <script>: User-Provided CLI '?XXCallerIDXX?' taken from From-User -
> R=?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX:5060?
> ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk PBX 18.24.3'
>
> Oct  7 06:43:50.888566 spce proxy[837565]: NOTICE:
> ROUTE_APPLY_REWRITE_RULE <script>: Applying caller-in usr/domain rewrite
> rules on user-provided CLI using dpid '7' -
> R=?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX:5060?
> ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk PBX 18.24.3'
>
> Oct  7 06:43:50.888839 spce proxy[837565]: NOTICE:
> ROUTE_CHECK_CLI_ALLOWED <script>: User-provided CLI '?XXCallerIDXX?'
> accepted (matched user preference) -
> R=?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX:5060?
> ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk PBX 18.24.3'
>
> Oct  7 06:43:50.888915 spce proxy[837565]: NOTICE: ROUTE_INVITE <script>:
> Setting '?XXCallerIDXX?@?XXX.XXX.86.104?' as initiating user-provided CLI -
> R=?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX:5060?
> ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk PBX 18.24.3'
>
> Oct  7 06:43:50.888963 spce proxy[837565]: NOTICE: ROUTE_INVITE <script>:
> Setting '?XXCallerIDXX?@?XXX.XXX.86.104?' as initiating network-provided
> CLI - R=?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX:5060?
> ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk PBX 18.24.3'
>
> Oct  7 06:43:50.889037 spce proxy[837565]: NOTICE:
> ROUTE_APPLY_REWRITE_RULE <script>: Applying callee-in usr/domain rewrite
> rules on called party using dpid '8' -
> R=?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX:5060?
> ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk PBX 18.24.3'
>
> Oct  7 06:43:50.890596 spce proxy[837565]: NOTICE: ROUTE_FIND_CALLEE
> <script>: Callee is not local - R=?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX:5060?
> ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk PBX 18.24.3'
>
> Oct  7 06:43:50.890869 spce proxy[837565]: NOTICE: ROUTE_PSTN <script>:
> Call to SIP Peering - R=?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX:5060?
> ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk PBX 18.24.3'
>
> Oct  7 06:43:50.890924 spce proxy[837565]: NOTICE: ROUTE_PSTN <script>:
> Load gws matching calling part '?sip:XXCallerIDXX at XXX.XXX.86.104?' and
> called user '?xxxxxxxxxxx2607244?' and called part
> '?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX:5060?' -
> R=?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX:5060?
> ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk PBX 18.24.3'
>
> Oct  7 06:43:50.891692 spce proxy[837565]: ERROR: ROUTE_PSTN_SELECT_PEER
> pv [pv_core.c:336]: pv_get_ruri_attr(): failed to parse the R-URI
>
> Oct  7 06:43:50.891806 spce proxy[837565]: ERROR:
> ROUTE_LOAD_PEER_OUT_PREF pv [pv_core.c:261]: pv_get_ruri(): failed to parse
> the R-URI
>
> Oct  7 06:43:50.891827 spce proxy[837565]: NOTICE:
> ROUTE_LOAD_PEER_OUT_PREF <script>: Load peer preferences for peer host '1'
> - R=?<null>? ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk PBX
> 18.24.3'
>
> Oct  7 06:43:50.893690 spce proxy[837565]: ERROR: ROUTE_PSTN pv
> [pv_core.c:336]: pv_get_ruri_attr(): failed to parse the R-URI
>
> Oct  7 06:43:50.895431 spce proxy[837565]: ERROR: ROUTE_PSTN pv
> [pv_core.c:336]: pv_get_ruri_attr(): failed to parse the R-URI
>
> Oct  7 06:43:50.895538 spce proxy[837565]: ERROR: <null> pv
> [pv_core.c:336]: pv_get_ruri_attr(): failed to parse the R-URI
>
> Oct  7 06:43:50.895558 spce proxy[837565]: ERROR: <null> pv
> [pv_core.c:336]: pv_get_ruri_attr(): failed to parse the R-URI
>
> Oct  7 06:43:50.901133 spce proxy[837565]: ERROR: ROUTE_OUTBOUND pv
> [pv_core.c:336]: pv_get_ruri_attr(): failed to parse the R-URI
>
> Oct  7 06:43:50.901156 spce proxy[837565]: ERROR: ROUTE_OUTBOUND pv
> [pv_core.c:336]: pv_get_ruri_attr(): failed to parse the R-URI
>
> Oct  7 06:43:50.901829 spce proxy[837565]: ERROR:
> ROUTE_APPLY_REWRITE_RULE pv [pv_core.c:261]: pv_get_ruri(): failed to parse
> the R-URI
>
> Oct  7 06:43:50.901850 spce proxy[837565]: NOTICE:
> ROUTE_APPLY_REWRITE_RULE <script>: Applying acc caller-out peer rewrite
> rules on acc caller party using dpid '3' - R=?<null>?
> ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk PBX 18.24.3'
>
> Oct  7 06:43:50.902770 spce proxy[837565]: ERROR: ROUTE_ACC_CALLER pv
> [pv_core.c:261]: pv_get_ruri(): failed to parse the R-URI
>
> Oct  7 06:43:50.902792 spce proxy[837565]: NOTICE: ROUTE_ACC_CALLER
> <script>: Setting acc source-leg for uuid
> '?86ba28d9-153f-4cee-800c-f34c1efea74a?':
> '?{"gpp":["70770",null,null,null,null,null,null,null,null,null],"clir":0,"uuid":"86ba28d9-153f-4cee-800c-f34c1efea74a","u":"XXCallerIDXX","d":"XXX.XXX.86.104","cli":"XXCallerIDXX","s_id":"70770","c_id":"70770","a_id":2,"pau":"","par":"","s":"call","ip":"XXX.XXX.188.51","t":1728276230.902101,"lnp_p":"","cli_out":"XXCallerIDXX","lnp_t":"","pai":"","ppi":"","div":"","last_hih":"","cid":"","u2u":"","lcr":0,"cc_quota":"","cc_sub":"1","cc_cust":"1"}?'
> - R=?<null>? ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk PBX
> 18.24.3'
>
> Oct  7 06:43:50.903156 spce proxy[837565]: ERROR:
> ROUTE_APPLY_REWRITE_RULE pv [pv_core.c:261]: pv_get_ruri(): failed to parse
> the R-URI
>
> Oct  7 06:43:50.903176 spce proxy[837565]: NOTICE:
> ROUTE_APPLY_REWRITE_RULE <script>: Applying acc callee-out peer rewrite
> rules on acc called party using dpid '4' - R=?<null>?
> ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk PBX 18.24.3'
>
> Oct  7 06:43:50.903656 spce proxy[837565]: ERROR: ROUTE_ACC_CALLEE pv
> [pv_core.c:261]: pv_get_ruri(): failed to parse the R-URI
>
> Oct  7 06:43:50.903679 spce proxy[837565]: NOTICE: ROUTE_ACC_CALLEE
> <script>: Setting acc destination-leg for uuid '?0?':
> '?{"gpp":[null,null,null,null,null,null,null,null,null,null],"plu":0,"s_id":"","c_id":"","a_id":0,"dialed":"xxxxxxxxxxx2607244","uuid":"0","u":"<null>","d":"<null>","u_in":"xxxxxxxxxxx2607244","d_in":"XXX.XXX.XXX.XXX","lcr":1,"lnp_p":"","u_out":"0","lnp_t":"","fci":"","cc_quota":"","cc_sub":"1","cc_cust":""}?'
> - R=?<null>? ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk PBX
> 18.24.3'
>
> Oct  7 06:43:50.903837 spce proxy[837565]: ERROR: ROUTE_OUTBOUND tm
> [t_lookup.c:1827]: new_t(): uri invalid
>
> Oct  7 06:43:50.903858 spce proxy[837565]: ERROR: ROUTE_OUTBOUND tm
> [t_lookup.c:1979]: t_newtran(): new_t failed
>
> Oct  7 06:43:50.903913 spce proxy[837565]: ERROR: ROUTE_OUTBOUND sl
> [sl_funcs.c:428]: sl_reply_error(): stateless error reply used: I'm
> terribly sorry, server error occurred (1/SL)
>
> Oct  7 06:43:50.904591 spce proxy[837570]: NOTICE: sl:filtered-ack
> <script>: New request on proxy - M=ACK
> R=?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX:5060?
> F=?sip:XXCallerIDXX at XXX.XXX.86.104?
> T=?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX? IP=?<null>?:?<null>?
> (?127.0.0.1?:?5060?) ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='<null>'
> DESTIP=?127.0.0.1?:?5062?
>
>
> ****
>
> Thanks
>
>
> On Mon, Oct 7, 2024 at 1:11?AM Dario Busso <dbusso at sipwise.com> wrote:
>
>> Looking at the error log, it seems that the INVITE received by SPCE is
>> malformed.
>> You need to check the peer system setup, not the SPCE.
>>
>>
>> Dario B.
>>
>>
>> Il lun 7 ott 2024, 06:50 Edo <edo.ekunwe at gmail.com> ha scritto:
>>
>>> Hi,
>>>
>>> I tried to make a call to a subscriber (SIP Trunk setup) and I got the
>>> following error message.
>>> *****
>>>
>>> Oct  7 06:43:50.891692 spce proxy[837565]: ERROR:
>>> ROUTE_PSTN_SELECT_PEER pv [pv_core.c:336]: pv_get_ruri_attr(): failed to
>>> parse the R-URI
>>>
>>> Oct  7 06:43:50.891806 spce proxy[837565]: ERROR:
>>> ROUTE_LOAD_PEER_OUT_PREF pv [pv_core.c:261]: pv_get_ruri(): failed to parse
>>> the R-URI
>>>
>>> Oct  7 06:43:50.891827 spce proxy[837565]: NOTICE:
>>> ROUTE_LOAD_PEER_OUT_PREF <script>: Load peer preferences for peer host '1'
>>> - R=?<null>? ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk PBX
>>> 18.24.3'
>>>
>>> Oct  7 06:43:50.893690 spce proxy[837565]: ERROR: ROUTE_PSTN pv
>>> [pv_core.c:336]: pv_get_ruri_attr(): failed to parse the R-URI
>>>
>>> Oct  7 06:43:50.895431 spce proxy[837565]: ERROR: ROUTE_PSTN pv
>>> [pv_core.c:336]: pv_get_ruri_attr(): failed to parse the R-URI
>>>
>>> Oct  7 06:43:50.895538 spce proxy[837565]: ERROR: <null> pv
>>> [pv_core.c:336]: pv_get_ruri_attr(): failed to parse the R-URI
>>>
>>> Oct  7 06:43:50.895558 spce proxy[837565]: ERROR: <null> pv
>>> [pv_core.c:336]: pv_get_ruri_attr(): failed to parse the R-URI
>>>
>>> Oct  7 06:43:50.901133 spce proxy[837565]: ERROR: ROUTE_OUTBOUND pv
>>> [pv_core.c:336]: pv_get_ruri_attr(): failed to parse the R-URI
>>>
>>> Oct  7 06:43:50.901156 spce proxy[837565]: ERROR: ROUTE_OUTBOUND pv
>>> [pv_core.c:336]: pv_get_ruri_attr(): failed to parse the R-URI
>>>
>>> Oct  7 06:43:50.901829 spce proxy[837565]: ERROR:
>>> ROUTE_APPLY_REWRITE_RULE pv [pv_core.c:261]: pv_get_ruri(): failed to parse
>>> the R-URI
>>> *****
>>>  I am looking for where I can set the system to a specific R-URI or
>>> check the system to use network provided R-RURI.
>>>
>>> Please help.
>>>
>>> Thanks
>>>
>>>
>>> --
>>> -----
>>> Ekunwe
>>> Tel: 601.497.3932
>>> --
>>> Spce-user mailing list
>>> Spce-user at lists.sipwise.com
>>> http://lists.sipwise.com/mailman/listinfo/spce-user_lists.sipwise.com
>>>
>>
>
> --
> -----
> Ekunwe
> Tel: 601.497.3932
>
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From mmalkov at sipwise.com  Wed Oct  9 05:11:57 2024
From: mmalkov at sipwise.com (Mykola Malkov)
Date: Wed, 9 Oct 2024 12:11:57 +0300
Subject: [Spce-user] [EXTERNAL] RE: incoming Session fails using ngcp CE
 mr12.4.1 and SIP.js
In-Reply-To: <IA2P220MB1912C96FE7EA690D3C0EC3E5EA7E2@IA2P220MB1912.NAMP220.PROD.OUTLOOK.COM>
References: <IA2P220MB1912D25CDEB706382873774FEA722@IA2P220MB1912.NAMP220.PROD.OUTLOOK.COM>
 <IA2P220MB1912D6FB6826CF76EECAF51AEA732@IA2P220MB1912.NAMP220.PROD.OUTLOOK.COM>
 <IA2P220MB1912006646C449DFF0BA69BAEA7D2@IA2P220MB1912.NAMP220.PROD.OUTLOOK.COM>
 <CAAyzhJRQXiQ1bRirzK5e=hezcRV9iqYc5ZTJU_Ghg2tRg9ZPFA@mail.gmail.com>
 <IA2P220MB1912C96FE7EA690D3C0EC3E5EA7E2@IA2P220MB1912.NAMP220.PROD.OUTLOOK.COM>
Message-ID: <0bed1610-7135-4558-bfca-39ba47f33ab8@sipwise.com>


> This is the result
>
>
> root at sips:/home/sysadmin# df -h
>
> Filesystem Size? Used Avail Use% Mounted on
>
> udev 1.9G???? 0? 1.9G?? 0% /dev
>
> tmpfs 388M? 912K? 387M?? 1% /run
>
> /dev/sdb1 30G? 5.9G?? 22G? 21% /
>
> tmpfs 1.9G? 192K? 1.9G?? 1% /dev/shm
>
> tmpfs 5.0M???? 0? 5.0M?? 0% /run/lock
>
> /dev/sdb15 124M?? 12M? 113M? 10% /boot/efi
>
> /dev/sda1 7.8G?? 24K? 7.4G?? 1% /mnt
>
> tmpfs 388M? 4.0K? 388M?? 1% /run/user/1000
>
The upgrade procedure relies on specific partition schema 
https://www.sipwise.com/doc/mr12.5.1/spce/ce/mr12.5.1/upgrade/upgrade.html
The majority of cloud providers (like microsoft or amazon) doesn't 
provide tools to modify it. So in this case the only way to upgrade will 
be reinstalling/migration/etc.

-- 
WBR
---
Mykola Malkov
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From spce at lard.at  Mon Oct 14 13:29:36 2024
From: spce at lard.at (spce at lard.at)
Date: Mon, 14 Oct 2024 19:29:36 +0200
Subject: [Spce-user] Header Manipulation in CE?
Message-ID: <2d0c92c7-ec4d-4117-a004-7d035c321907@lard.at>

Hello,

i would like to add a Reason-header to a CANCEL-message. Is such a kind 
of header manipulation possible in sipwise C5 CE?

Version: sipwise 12.5.1 CE


Thx

Malte


From edo.ekunwe at gmail.com  Wed Oct  9 21:25:42 2024
From: edo.ekunwe at gmail.com (Edo)
Date: Wed, 9 Oct 2024 20:25:42 -0500
Subject: [Spce-user] [EXTERNAL]  Troubleshooting request
In-Reply-To: <CAAyzhJTwqX6UvLBHeo6YmSEkw-0ibfiPCmSf_1ffJzfYTLfbjQ@mail.gmail.com>
References: <CAFkRdGtKH2MtMwdPoQES6JdU+yjSbYG8wp2dzhpfhkFMUeDH3w@mail.gmail.com>
 <CAAyzhJTi6W28bqohSAqJtP5FgyyAeeuqbKK_BTxoE+f+Yzs1yA@mail.gmail.com>
 <CAFkRdGuEw-FfHym8Of+fUf3aYvcC7+==nB_dyi5qMJJg91A=nQ@mail.gmail.com>
 <CAAyzhJTwqX6UvLBHeo6YmSEkw-0ibfiPCmSf_1ffJzfYTLfbjQ@mail.gmail.com>
Message-ID: <CAFkRdGu-PjJMY8R9WjAReE7hKa08vaSR5j3gHVgtaH9zzZ9=Og@mail.gmail.com>

Sure, here you go! I saw some weird characters showing up in front of the
INVITE and OPTIONS packets. I just cut a few before and after lines

*****


]'g?R?E???,@?????OPTIONS sip:nagios at voip.sipwise.local:5060 SIP/2.0

Record-Route: <sip:127.0.0.1;lr=on;ftag=14e81d56;ngcplb=yes;socket=udp:
127.0.0.1:5060>

Via: SIP/2.0/UDP
127.0.0.1;branch=z9hG4bKaad3.b18546410bf87bb4131a6f674164370a.0

Via: SIP/2.0/UDP 127.0.0.2:8806
;received=127.0.0.1;branch=z9hG4bK.7dc0cb48;rport=8806;alias

From: sip:sipsak at 127.0.0.2:8806;tag=14e81d56

To: sip:nagios at voip.sipwise.local:5060

Call-ID: 350756182 at 127.0.0.2

CSeq: 1 OPTIONS

Contact: sip:sipsak at 127.0.0.2:8806

Content-Length: 0

Max-Forwards: 69

User-Agent: sipsak 0.9.8.1

Accept: text/plain

P-NGCP-Src-Ip: 127.0.0.1

P-NGCP-Src-Port: 8806

P-NGCP-Src-Proto: udp

P-NGCP-Src-Af: 4

P-Sock-Info: udp:127.0.0.1:5060

P-LB-Uptime: 1233637


]'gT


   E??-@???????SIP/2.0 100 Trying

Via: SIP/2.0/UDP
127.0.0.1;branch=z9hG4bKaad3.b18546410bf87bb4131a6f674164370a.0

Via: SIP/2.0/UDP 127.0.0.2:8806
;received=127.0.0.1;branch=z9hG4bK.7dc0cb48;rport=8806;alias

From: sip:sipsak at 127.0.0.2:8806;tag=14e81d56

To: sip:nagios at voip.sipwise.local:5060

Call-ID: 350756182 at 127.0.0.2

CSeq: 1 OPTIONS

P-Out-Socket: udp:127.0.0.1:5060

P-NGCP-Auth-IP: 127.0.0.1

P-NGCP-Auth-UA: sipsak 0.9.8.1

Server: Sipwise NGCP Proxy mr12.5.1

Content-Length: 0


]'g?T?E??.@?,??l?SIP/2.0 200 Alive

Record-Route: <sip:127.0.0.1;lr=on;ftag=14e81d56;ngcplb=yes;socket=udp:
127.0.0.1:5060>

Via: SIP/2.0/UDP
127.0.0.1;branch=z9hG4bKaad3.b18546410bf87bb4131a6f674164370a.0

Via: SIP/2.0/UDP 127.0.0.2:8806
;received=127.0.0.1;branch=z9hG4bK.7dc0cb48;rport=8806;alias

From: sip:sipsak at 127.0.0.2:8806;tag=14e81d56

To: sip:nagios at voip.sipwise.local
:5060;tag=95c37a12bff1a2c36d72bf8333176544.78550000

Call-ID: 350756182 at 127.0.0.2

CSeq: 1 OPTIONS

P-Out-Socket: udp:127.0.0.1:5060

P-NGCP-Auth-IP: 127.0.0.1

P-NGCP-Auth-UA: sipsak 0.9.8.1

Server: Sipwise NGCP Proxy mr12.5.1

Content-Length: 0


]'g?U?E???/@?2?"f???SIP/2.0 200 Alive

Record-Route: <sip:127.0.0.1;lr=on;ftag=14e81d56;ngcplb=yes;socket=udp:
127.0.0.1:5060>

Via: SIP/2.0/UDP 127.0.0.2:8806
;received=127.0.0.1;branch=z9hG4bK.7dc0cb48;rport=8806;alias

From: sip:sipsak at 127.0.0.2:8806;tag=14e81d56

To: sip:nagios at voip.sipwise.local
:5060;tag=95c37a12bff1a2c36d72bf8333176544.78550000

Call-ID: 350756182 at 127.0.0.2

CSeq: 1 OPTIONS

Server: Sipwise NGCP Proxy mr12.5.1

Content-Length: 0


]'g?

    ??5??E?@4?7W?32L?????CINVITE
sip:TECHPREFIX-CC-AC-2607244 at SPCE-IP-ADDRESS:5060 SIP/2.0

Via: SIP/2.0/UDP
SENDER-IP-ADDRESS:5060;rport;branch=z9hG4bKPj0caddbe0-6e81-4d41-a123-74e2037e9eb6

From: <sip:16014973932 at 172.31.86.104
>;tag=453eb646-d5b9-4f6b-b685-9c95c12c88a8

To: <sip:TECHPREFIX-CC-AC-2607244 at SPCE-IP-ADDRESS>

Contact: <sip:asterisk at SENDER-IP-ADDRESS:5060>

Call-ID: 52b26324-383b-4016-9e18-f231521a67a3

CSeq: 7613 INVITE

Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE,
CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER

Supported: 100rel, timer, replaces, norefersub, histinfo

Session-Expires: 1800

Min-SE: 90

Max-Forwards: 70

User-Agent: Asterisk PBX 18.24.3

Content-Type: application/sdp

Content-Length:   280


v=0

o=- 433188941 433188941 IN IP4 SENDER-IP-ADDRESS

s=Asterisk

c=IN IP4 SENDER-IP-ADDRESS

t=0 0

m=audio 12136 RTP/AVP 8 18 101

a=rtpmap:8 PCMA/8000

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=maxptime:140

a=sendrecv

]'gw?

     ??$n?[E??Cg at or2L??W?3????YSIP/2.0 100 Trying

Via: SIP/2.0/UDP
SENDER-IP-ADDRESS:5060;rport=5060;branch=z9hG4bKPj0caddbe0-6e81-4d41-a123-74e2037e9eb6;received=SENDER-IP-ADDRESS

From: <sip:16014973932 at 172.31.86.104
>;tag=453eb646-d5b9-4f6b-b685-9c95c12c88a8

To: <sip:TECHPREFIX-CC-AC-2607244 at SPCE-IP-ADDRESS>

Call-ID: 52b26324-383b-4016-9e18-f231521a67a3

CSeq: 7613 INVITE

Server: Sipwise NGCP LB mr12.5.1

Content-Length: 0


]'g?

    ?E???n@??????INVITE sip:TECHPREFIX-CC-AC-2607244 at SPCE-IP-ADDRESS:5060
SIP/2.0

Record-Route:
<sip:127.0.0.1;r2=on;lr=on;ftag=453eb646-d5b9-4f6b-b685-9c95c12c88a8;ngcplb=yes;socket=udp:SPCE-IP-ADDRESS:5060>

Record-Route:
<sip:SPCE-IP-ADDRESS;r2=on;lr=on;ftag=453eb646-d5b9-4f6b-b685-9c95c12c88a8;ngcplb=yes;socket=udp:SPCE-IP-ADDRESS:5060>

Via: SIP/2.0/UDP
127.0.0.1;branch=z9hG4bK3d98.3101547a9123f5d14c7928ae47d19fdc.0

Via: SIP/2.0/UDP
SENDER-IP-ADDRESS:5060;received=SENDER-IP-ADDRESS;rport=5060;branch=z9hG4bKPj0caddbe0-6e81-4d41-a123-74e2037e9eb6

From: <sip:16014973932 at 172.31.86.104
>;tag=453eb646-d5b9-4f6b-b685-9c95c12c88a8

To: <sip:TECHPREFIX-CC-AC-2607244 at SPCE-IP-ADDRESS>

Contact: <sip:asterisk at SENDER-IP-ADDRESS:5060>

Call-ID: 52b26324-383b-4016-9e18-f231521a67a3

CSeq: 7613 INVITE

Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE,
CANCEL, UPDATE, PRACK, MESSAGE

Supported: 100rel, timer, replaces, norefersub, histinfo

Session-Expires: 1800

Min-SE: 90

Max-Forwards: 69

User-Agent: Asterisk PBX 18.24.3

Content-Type: application/sdp

Content-Length:   280

P-NGCP-Src-Ip: SENDER-IP-ADDRESS

P-NGCP-Src-Port: 5060

P-NGCP-Src-Proto: udp

P-NGCP-Src-Af: 4

P-Sock-Info: udp:SPCE-IP-ADDRESS:5060

P-LB-Uptime: 1233637


v=0

o=- 433188941 433188941 IN IP4 SENDER-IP-ADDRESS

s=Asterisk

c=IN IP4 SENDER-IP-ADDRESS

t=0 0

m=audio 12136 RTP/AVP 8 18 101

a=rtpmap:8 PCMA/8000

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=maxptime:140

a=sendrecv

]'gO?

     kk?E[?o@???G[SIP/2.0 100 Trying

Via: SIP/2.0/UDP
127.0.0.1;branch=z9hG4bK3d98.3101547a9123f5d14c7928ae47d19fdc.0

Via: SIP/2.0/UDP
SENDER-IP-ADDRESS:5060;received=SENDER-IP-ADDRESS;rport=5060;branch=z9hG4bKPj0caddbe0-6e81-4d41-a123-74e2037e9eb6

From: <sip:16014973932 at 172.31.86.104
>;tag=453eb646-d5b9-4f6b-b685-9c95c12c88a8

To: <sip:TECHPREFIX-CC-AC-2607244 at SPCE-IP-ADDRESS>

Call-ID: 52b26324-383b-4016-9e18-f231521a67a3

CSeq: 7613 INVITE

P-Out-Socket: udp:SPCE-IP-ADDRESS:5060

P-NGCP-Auth-IP: SENDER-IP-ADDRESS

P-NGCP-Auth-UA: Asterisk PBX 18.24.3

Server: Sipwise NGCP Proxy mr12.5.1

Content-Length: 0


]'g]?



    E??p@?n????SIP/2.0 407 Proxy Authentication Required

Via: SIP/2.0/UDP
127.0.0.1;branch=z9hG4bK3d98.3101547a9123f5d14c7928ae47d19fdc.0

Via: SIP/2.0/UDP
SENDER-IP-ADDRESS:5060;received=SENDER-IP-ADDRESS;rport=5060;branch=z9hG4bKPj0caddbe0-6e81-4d41-a123-74e2037e9eb6

From: <sip:16014973932 at 172.31.86.104
>;tag=453eb646-d5b9-4f6b-b685-9c95c12c88a8

To: <sip:TECHPREFIX-CC-AC-2607244 at SPCE-IP-ADDRESS
>;tag=95c37a12bff1a2c36d72bf8333176544.78550000

Call-ID: 52b26324-383b-4016-9e18-f231521a67a3

CSeq: 7613 INVITE

P-Out-Socket: udp:SPCE-IP-ADDRESS:5060

P-NGCP-Auth-IP: SENDER-IP-ADDRESS

P-NGCP-Auth-UA: Asterisk PBX 18.24.3

Proxy-Authenticate: Digest realm="172.31.86.104",
nonce="ZwcoiWcHJ11xDog/Bk4LxGa3PDdkpgik"

Server: Sipwise NGCP Proxy mr12.5.1

Content-Length: 0


]'g

   ?

    ?E???q@?

            ?????ACK sip:TECHPREFIX-CC-AC-2607244 at SPCE-IP-ADDRESS:5060
SIP/2.0

Via: SIP/2.0/UDP
127.0.0.1;branch=z9hG4bK3d98.3101547a9123f5d14c7928ae47d19fdc.0

From: <sip:16014973932 at 172.31.86.104
>;tag=453eb646-d5b9-4f6b-b685-9c95c12c88a8

To: <sip:TECHPREFIX-CC-AC-2607244 at SPCE-IP-ADDRESS
>;tag=95c37a12bff1a2c36d72bf8333176544.78550000

Call-ID: 52b26324-383b-4016-9e18-f231521a67a3

CSeq: 7613 ACK

Max-Forwards: 69

Content-Length: 0


]'g?

    QQ$n?[E?ACh at n?2L??W?3??-??SIP/2.0 407 Proxy Authentication Required

Via: SIP/2.0/UDP
SENDER-IP-ADDRESS:5060;received=SENDER-IP-ADDRESS;rport=5060;branch=z9hG4bKPj0caddbe0-6e81-4d41-a123-74e2037e9eb6

From: <sip:16014973932 at 172.31.86.104
>;tag=453eb646-d5b9-4f6b-b685-9c95c12c88a8

To: <sip:TECHPREFIX-CC-AC-2607244 at SPCE-IP-ADDRESS
>;tag=95c37a12bff1a2c36d72bf8333176544.78550000

Call-ID: 52b26324-383b-4016-9e18-f231521a67a3

CSeq: 7613 INVITE

Proxy-Authenticate: Digest realm="172.31.86.104",
nonce="ZwcoiWcHJ11xDog/Bk4LxGa3PDdkpgik"

Server: Sipwise NGCP Proxy mr12.5.1

Content-Length: 0


]'g?w

     ????5???xE?1 at 4?W?32L??????9ACK
sip:TECHPREFIX-CC-AC-2607244 at SPCE-IP-ADDRESS:5060 SIP/2.0

Via: SIP/2.0/UDP
SENDER-IP-ADDRESS:5060;rport;branch=z9hG4bKPj0caddbe0-6e81-4d41-a123-74e2037e9eb6

From: <sip:16014973932 at 172.31.86.104
>;tag=453eb646-d5b9-4f6b-b685-9c95c12c88a8

To: <sip:TECHPREFIX-CC-AC-2607244 at SPCE-IP-ADDRESS
>;tag=95c37a12bff1a2c36d72bf8333176544.78550000

Call-ID: 52b26324-383b-4016-9e18-f231521a67a3

CSeq: 7613 ACK

Max-Forwards: 70

User-Agent: Asterisk PBX 18.24.3

Content-Length:  0


]'g|

    ????5??E?2 at 4?0W?32L??????INVITE
sip:TECHPREFIX-CC-AC-2607244 at SPCE-IP-ADDRESS:5060 SIP/2.0

Via: SIP/2.0/UDP
SENDER-IP-ADDRESS:5060;rport;branch=z9hG4bKPjb41a3038-ac6c-489d-af75-a3366c6e1c49

From: <sip:16014973932 at 172.31.86.104
>;tag=453eb646-d5b9-4f6b-b685-9c95c12c88a8

To: <sip:TECHPREFIX-CC-AC-2607244 at SPCE-IP-ADDRESS>

Contact: <sip:asterisk at SENDER-IP-ADDRESS:5060>

Call-ID: 52b26324-383b-4016-9e18-f231521a67a3

CSeq: 7614 INVITE

Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE,
CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER

Supported: 100rel, timer, replaces, norefersub, histinfo

Session-Expires: 1800

Min-SE: 90

Max-Forwards: 70

User-Agent: Asterisk PBX 18.24.3

Proxy-Authorization: Digest username="USERNAME-FOR-SIPTRUNK",
realm="172.31.86.104", nonce="ZwcoiWcHJ11xDog/Bk4LxGa3PDdkpgik",
uri="sip:TECHPREFIX-CC-AC-2607244 at SPCE-IP-ADDRESS:5060",
response="37046e3b72e6758aa4c1e7b35ac3a215"

Content-Type: application/sdp

Content-Length:   280


v=0

o=- 433188941 433188941 IN IP4 SENDER-IP-ADDRESS

s=Asterisk

c=IN IP4 SENDER-IP-ADDRESS

t=0 0

m=audio 12136 RTP/AVP 8 18 101

a=rtpmap:8 PCMA/8000

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=maxptime:140

a=sendrecv

]'g?}

     ??$n?[E??Cv at oc2L??W?3????YSIP/2.0 100 Trying

Via: SIP/2.0/UDP
SENDER-IP-ADDRESS:5060;rport=5060;branch=z9hG4bKPjb41a3038-ac6c-489d-af75-a3366c6e1c49;received=SENDER-IP-ADDRESS

From: <sip:16014973932 at 172.31.86.104
>;tag=453eb646-d5b9-4f6b-b685-9c95c12c88a8

To: <sip:TECHPREFIX-CC-AC-2607244 at SPCE-IP-ADDRESS>

Call-ID: 52b26324-383b-4016-9e18-f231521a67a3

CSeq: 7614 INVITE

Server: Sipwise NGCP LB mr12.5.1

Content-Length: 0


]'ge~

     ???E???z@??????INVITE sip:TECHPREFIX-CC-AC-2607244 at SPCE-IP-ADDRESS:5060
SIP/2.0

Record-Route:
<sip:127.0.0.1;r2=on;lr=on;ftag=453eb646-d5b9-4f6b-b685-9c95c12c88a8;ngcplb=yes;socket=udp:SPCE-IP-ADDRESS:5060>

Record-Route:
<sip:SPCE-IP-ADDRESS;r2=on;lr=on;ftag=453eb646-d5b9-4f6b-b685-9c95c12c88a8;ngcplb=yes;socket=udp:SPCE-IP-ADDRESS:5060>

Via: SIP/2.0/UDP
127.0.0.1;branch=z9hG4bK0d98.db77909851726a22821601a9afe9a764.0

Via: SIP/2.0/UDP
SENDER-IP-ADDRESS:5060;received=SENDER-IP-ADDRESS;rport=5060;branch=z9hG4bKPjb41a3038-ac6c-489d-af75-a3366c6e1c49

From: <sip:16014973932 at 172.31.86.104
>;tag=453eb646-d5b9-4f6b-b685-9c95c12c88a8

To: <sip:TECHPREFIX-CC-AC-2607244 at SPCE-IP-ADDRESS>

Contact: <sip:asterisk at SENDER-IP-ADDRESS:5060>

Call-ID: 52b26324-383b-4016-9e18-f231521a67a3

CSeq: 7614 INVITE

Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE,
CANCEL, UPDATE, PRACK, MESSAGE

Supported: 100rel, timer, replaces, norefersub, histinfo

Session-Expires: 1800

Min-SE: 90

Max-Forwards: 69

User-Agent: Asterisk PBX 18.24.3

Proxy-Authorization: Digest username="USERNAME-FOR-SIPTRUNK",
realm="172.31.86.104", nonce="ZwcoiWcHJ11xDog/Bk4LxGa3PDdkpgik",
uri="sip:TECHPREFIX-CC-AC-2607244 at SPCE-IP-ADDRESS:5060",
response="37046e3b72e6758aa4c1e7b35ac3a215"

Content-Type: application/sdp

Content-Length:   280

P-NGCP-Src-Ip: SENDER-IP-ADDRESS

P-NGCP-Src-Port: 5060

P-NGCP-Src-Proto: udp

P-NGCP-Src-Af: 4

P-Sock-Info: udp:SPCE-IP-ADDRESS:5060

P-LB-Uptime: 1233637


v=0

o=- 433188941 433188941 IN IP4 SENDER-IP-ADDRESS

s=Asterisk

c=IN IP4 SENDER-IP-ADDRESS

t=0 0

m=audio 12136 RTP/AVP 8 18 101

a=rtpmap:8 PCMA/8000

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=maxptime:140

a=sendrecv

]'g?

    kk?E[?{@???G[SIP/2.0 100 Trying

Via: SIP/2.0/UDP
127.0.0.1;branch=z9hG4bK0d98.db77909851726a22821601a9afe9a764.0

Via: SIP/2.0/UDP
SENDER-IP-ADDRESS:5060;received=SENDER-IP-ADDRESS;rport=5060;branch=z9hG4bKPjb41a3038-ac6c-489d-af75-a3366c6e1c49

From: <sip:16014973932 at 172.31.86.104
>;tag=453eb646-d5b9-4f6b-b685-9c95c12c88a8

To: <sip:TECHPREFIX-CC-AC-2607244 at SPCE-IP-ADDRESS>

Call-ID: 52b26324-383b-4016-9e18-f231521a67a3

CSeq: 7614 INVITE

P-Out-Socket: udp:SPCE-IP-ADDRESS:5060

P-NGCP-Auth-IP: SENDER-IP-ADDRESS

P-NGCP-Auth-UA: Asterisk PBX 18.24.3

Server: Sipwise NGCP Proxy mr12.5.1

Content-Length: 0


]'g??

     EE7E5?|@?)??!5SIP/2.0 101 Connecting

Via: SIP/2.0/UDP
127.0.0.1;branch=z9hG4bK0d98.db77909851726a22821601a9afe9a764.0

Via: SIP/2.0/UDP
SENDER-IP-ADDRESS:5060;received=SENDER-IP-ADDRESS;rport=5060;branch=z9hG4bKPjb41a3038-ac6c-489d-af75-a3366c6e1c49

From: <sip:16014973932 at 172.31.86.104
>;tag=453eb646-d5b9-4f6b-b685-9c95c12c88a8

To: <sip:TECHPREFIX-CC-AC-2607244 at SPCE-IP-ADDRESS>

Call-ID: 52b26324-383b-4016-9e18-f231521a67a3

CSeq: 7614 INVITE

P-Out-Socket: udp:SPCE-IP-ADDRESS:5060

P-NGCP-Auth-IP: SENDER-IP-ADDRESS

P-NGCP-Auth-UA: Asterisk PBX 18.24.3

P-NGCP-Authorization: USERNAME-FOR-SIPTRUNK at 172.31.86.104

P-NGCP-Authorized: 1

P-NGCP-Caller-Info: <sip:USERNAME-FOR-SIPTRUNK at 172.31.86.104
>;ip=SENDER-IP-ADDRESS;port=5060

P-Caller-UUID: 86ba28d9-153f-4cee-800c-f34c1efea74a

Server: Sipwise NGCP Proxy mr12.5.1

Content-Length: 0


?E??@????u?SIP/2.0 500 I'm terribly sorry, server error occurred (1/SL)

Via: SIP/2.0/UDP
127.0.0.1;branch=z9hG4bK0d98.db77909851726a22821601a9afe9a764.0

Via: SIP/2.0/UDP
SENDER-IP-ADDRESS:5060;received=SENDER-IP-ADDRESS;rport=5060;branch=z9hG4bKPjb41a3038-ac6c-489d-af75-a3366c6e1c49

From: <sip:16014973932 at 172.31.86.104
>;tag=453eb646-d5b9-4f6b-b685-9c95c12c88a8

To: <sip:TECHPREFIX-CC-AC-2607244 at SPCE-IP-ADDRESS
>;tag=95c37a12bff1a2c36d72bf8333176544.78550000

Call-ID: 52b26324-383b-4016-9e18-f231521a67a3

CSeq: 7614 INVITE

P-Out-Socket: udp:SPCE-IP-ADDRESS:5060

P-NGCP-Auth-IP: SENDER-IP-ADDRESS

P-NGCP-Auth-UA: Asterisk PBX 18.24.3

P-NGCP-Authorization: USERNAME-FOR-SIPTRUNK at 172.31.86.104

P-NGCP-Authorized: 1

P-NGCP-Caller-Info: <sip:USERNAME-FOR-SIPTRUNK at 172.31.86.104
>;ip=SENDER-IP-ADDRESS;port=5060

P-Caller-UUID: 86ba28d9-153f-4cee-800c-f34c1efea74a

Server: Sipwise NGCP Proxy mr12.5.1

Content-Length: 0


?E???@???????ACK sip:TECHPREFIX-CC-AC-2607244 at SPCE-IP-ADDRESS:5060 SIP/2.0

Via: SIP/2.0/UDP
127.0.0.1;branch=z9hG4bK0d98.db77909851726a22821601a9afe9a764.0

From: <sip:16014973932 at 172.31.86.104
>;tag=453eb646-d5b9-4f6b-b685-9c95c12c88a8

To: <sip:TECHPREFIX-CC-AC-2607244 at SPCE-IP-ADDRESS
>;tag=95c37a12bff1a2c36d72bf8333176544.78550000

Call-ID: 52b26324-383b-4016-9e18-f231521a67a3

CSeq: 7614 ACK

Max-Forwards: 69

Content-Length: 0


$n?[E??C}@o2L??W?3????SIP/2.0 500 I'm terribly sorry, server error occurred
(1/SL)

Via: SIP/2.0/UDP
SENDER-IP-ADDRESS:5060;received=SENDER-IP-ADDRESS;rport=5060;branch=z9hG4bKPjb41a3038-ac6c-489d-af75-a3366c6e1c49

From: <sip:16014973932 at 172.31.86.104
>;tag=453eb646-d5b9-4f6b-b685-9c95c12c88a8

To: <sip:TECHPREFIX-CC-AC-2607244 at SPCE-IP-ADDRESS
>;tag=95c37a12bff1a2c36d72bf8333176544.78550000

Call-ID: 52b26324-383b-4016-9e18-f231521a67a3

CSeq: 7614 INVITE

Server: Sipwise NGCP Proxy mr12.5.1

Content-Length: 0


????5??E?x at 4??W?32L?????2fACK sip:TECHPREFIX-CC-AC-2607244 at SPCE-IP-ADDRESS:5060
SIP/2.0

Via: SIP/2.0/UDP
SENDER-IP-ADDRESS:5060;rport;branch=z9hG4bKPjb41a3038-ac6c-489d-af75-a3366c6e1c49

From: <sip:16014973932 at 172.31.86.104
>;tag=453eb646-d5b9-4f6b-b685-9c95c12c88a8

To: <sip:TECHPREFIX-CC-AC-2607244 at SPCE-IP-ADDRESS
>;tag=95c37a12bff1a2c36d72bf8333176544.78550000

Call-ID: 52b26324-383b-4016-9e18-f231521a67a3

CSeq: 7614 ACK

Max-Forwards: 70

User-Agent: Asterisk PBX 18.24.3

Content-Length:  0


g'gtB?E?g@@8??????OPTIONS sip:nagios at voip.sipwise.local:5060 SIP/2.0

Via: SIP/2.0/UDP 127.0.0.2:6357;branch=z9hG4bK.19821781;rport;alias

From: sip:sipsak at 127.0.0.2:6357;tag=70402e3a

To: sip:nagios at voip.sipwise.local:5060

Call-ID: 1883254330 at 127.0.0.2

CSeq: 1 OPTIONS

Contact: sip:sipsak at 127.0.0.2:6357

Content-Length: 0

Max-Forwards: 70

User-Agent: sipsak 0.9.8.1

Accept: text/plain


g'g?D?E???@?H????OPTIONS sip:nagios at voip.sipwise.local:5060 SIP/2.0

Record-Route: <sip:127.0.0.1;lr=on;ftag=70402e3a;ngcplb=yes;socket=udp:
127.0.0.1:5060>

Via: SIP/2.0/UDP
127.0.0.1;branch=z9hG4bKee64.5b46b33203888567f0eff8e159314096.0

Via: SIP/2.0/UDP 127.0.0.2:6357
;received=127.0.0.1;branch=z9hG4bK.19821781;rport=6357;alias

From: sip:sipsak at 127.0.0.2:6357;tag=70402e3a

To: sip:nagios at voip.sipwise.local:5060

Call-ID: 1883254330 at 127.0.0.2

CSeq: 1 OPTIONS

Contact: sip:sipsak at 127.0.0.2:6357

Content-Length: 0

Max-Forwards: 69

User-Agent: sipsak 0.9.8.1

Accept: text/plain

P-NGCP-Src-Ip: 127.0.0.1

P-NGCP-Src-Port: 6357

P-NGCP-Src-Proto: udp

P-NGCP-Src-Af: 4

P-Sock-Info: udp:127.0.0.1:5060

P-LB-Uptime: 1233647


g'g?E


    E??@???????SIP/2.0 100 Trying

Via: SIP/2.0/UDP
127.0.0.1;branch=z9hG4bKee64.5b46b33203888567f0eff8e159314096.0

Via: SIP/2.0/UDP 127.0.0.2:6357
;received=127.0.0.1;branch=z9hG4bK.19821781;rport=6357;alias

From: sip:sipsak at 127.0.0.2:6357;tag=70402e3a

To: sip:nagios at voip.sipwise.local:5060

Call-ID: 1883254330 at 127.0.0.2

CSeq: 1 OPTIONS

P-Out-Socket: udp:127.0.0.1:5060

P-NGCP-Auth-IP: 127.0.0.1

P-NGCP-Auth-UA: sipsak 0.9.8.1

Server: Sipwise NGCP Proxy mr12.5.1

Content-Length: 0


g'g?F?E??@?V??m?SIP/2.0 200 Alive

Record-Route: <sip:127.0.0.1;lr=on;ftag=70402e3a;ngcplb=yes;socket=udp:
127.0.0.1:5060>

Via: SIP/2.0/UDP
127.0.0.1;branch=z9hG4bKee64.5b46b33203888567f0eff8e159314096.0

Via: SIP/2.0/UDP 127.0.0.2:6357
;received=127.0.0.1;branch=z9hG4bK.19821781;rport=6357;alias

From: sip:sipsak at 127.0.0.2:6357;tag=70402e3a

To: sip:nagios at voip.sipwise.local
:5060;tag=95c37a12bff1a2c36d72bf8333176544.78550000

Call-ID: 1883254330 at 127.0.0.2

CSeq: 1 OPTIONS

P-Out-Socket: udp:127.0.0.1:5060

P-NGCP-Auth-IP: 127.0.0.1

P-NGCP-Auth-UA: sipsak 0.9.8.1

Server: Sipwise NGCP Proxy mr12.5.1

Content-Length: 0


g'g?G?E???@?\?????SIP/2.0 200 Alive

Record-Route: <sip:127.0.0.1;lr=on;ftag=70402e3a;ngcplb=yes;socket=udp:
127.0.0.1:5060>

Via: SIP/2.0/UDP 127.0.0.2:6357
;received=127.0.0.1;branch=z9hG4bK.19821781;rport=6357;alias

From: sip:sipsak at 127.0.0.2:6357;tag=70402e3a

To: sip:nagios at voip.sipwise.local
:5060;tag=95c37a12bff1a2c36d72bf8333176544.78550000

Call-ID: 1883254330 at 127.0.0.2

CSeq: 1 OPTIONS

Server: Sipwise NGCP Proxy mr12.5.1

Content-Length: 0


q'g?>?E???@@??"W????OPTIONS sip:nagios at voip.sipwise.local:5060 SIP/2.0

Via: SIP/2.0/UDP 127.0.0.2:8791;branch=z9hG4bK.55e71b1d;rport;alias

From: sip:sipsak at 127.0.0.2:8791;tag=796cec4e

To: sip:nagios at voip.sipwise.local:5060

Call-ID: 2037181518 at 127.0.0.2

CSeq: 1 OPTIONS

Contact: sip:sipsak at 127.0.0.2:8791

Content-Length: 0

Max-Forwards: 70

User-Agent: sipsak 0.9.8.1

Accept: text/plain


q'gA?E????.????OPTIONS sip:nagios at voip.sipwise.local:5060 SIP/2.0

Record-Route: <sip:127.0.0.1;lr=on;ftag=796cec4e;ngcplb=yes;socket=udp:
127.0.0.1:5060>

Via: SIP/2.0/UDP
127.0.0.1;branch=z9hG4bKc392.24aedd8ec91b388019f6bf5ffd507f46.0

Via: SIP/2.0/UDP 127.0.0.2:8791
;received=127.0.0.1;branch=z9hG4bK.55e71b1d;rport=8791;alias

From: sip:sipsak at 127.0.0.2:8791;tag=796cec4e

To: sip:nagios at voip.sipwise.local:5060

Call-ID: 2037181518 at 127.0.0.2

CSeq: 1 OPTIONS

Contact: sip:sipsak at 127.0.0.2:8791

Content-Length: 0

Max-Forwards: 69

User-Agent: sipsak 0.9.8.1

Accept: text/plain

P-NGCP-Src-Ip: 127.0.0.1

P-NGCP-Src-Port: 8791

P-NGCP-Src-Proto: udp

P-NGCP-Src-Af: 4

P-Sock-Info: udp:127.0.0.1:5060

P-LB-Uptime: 1233657


q'gvB


    E??@???????SIP/2.0 100 Trying

Via: SIP/2.0/UDP
127.0.0.1;branch=z9hG4bKc392.24aedd8ec91b388019f6bf5ffd507f46.0

Via: SIP/2.0/UDP 127.0.0.2:8791
;received=127.0.0.1;branch=z9hG4bK.55e71b1d;rport=8791;alias

From: sip:sipsak at 127.0.0.2:8791;tag=796cec4e

To: sip:nagios at voip.sipwise.local:5060

Call-ID: 2037181518 at 127.0.0.2

CSeq: 1 OPTIONS

P-Out-Socket: udp:127.0.0.1:5060

P-NGCP-Auth-IP: 127.0.0.1

P-NGCP-Auth-UA: sipsak 0.9.8.1

Server: Sipwise NGCP Proxy mr12.5.1

Content-Length: 0


q'g,C?E??@?<??m?SIP/2.0 200 Alive

Record-Route: <sip:127.0.0.1;lr=on;ftag=796cec4e;ngcplb=yes;socket=udp:
127.0.0.1:5060>

Via: SIP/2.0/UDP
127.0.0.1;branch=z9hG4bKc392.24aedd8ec91b388019f6bf5ffd507f46.0

Via: SIP/2.0/UDP 127.0.0.2:8791
;received=127.0.0.1;branch=z9hG4bK.55e71b1d;rport=8791;alias

From: sip:sipsak at 127.0.0.2:8791;tag=796cec4e

To: sip:nagios at voip.sipwise.local
:5060;tag=95c37a12bff1a2c36d72bf8333176544.78550000

Call-ID: 2037181518 at 127.0.0.2

CSeq: 1 OPTIONS

P-Out-Socket: udp:127.0.0.1:5060

P-NGCP-Auth-IP: 127.0.0.1

P-NGCP-Auth-UA: sipsak 0.9.8.1

Server: Sipwise NGCP Proxy mr12.5.1

Content-Length: 0




*****

On Wed, Oct 9, 2024 at 2:47?AM Dario Busso <dbusso at sipwise.com> wrote:

> Hi Edo.
> Can you share the full SIP INVITE message the kamailio proxy receive?
> Dario
>
> On Mon, Oct 7, 2024 at 3:36?PM Edo <edo.ekunwe at gmail.com> wrote:
>
>> Hi Dario,
>>
>> I am not sure if your earlier response of malformed invite was directed
>> at me so I decided to paste the sip logs from a sample call, maybe we can
>> get more information that may help with troubleshooting.
>>
>> ****
>>
>> Oct  7 06:43:50.807051 spce proxy[837572]: NOTICE: DEFAULT_ROUTE
>> <script>: New request on proxy - M=INVITE
>> R=?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX:5060?
>> F=?sip:XXCallerIDXX at XXX.XXX.86.104?
>> T=?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX? IP=?XXX.XXX.188.51?:?5060?
>> (?127.0.0.1?:?5060?) ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk
>> PBX 18.24.3' DESTIP=?127.0.0.1?:?5062?
>>
>> Oct  7 06:43:50.807084 spce proxy[837572]: NOTICE: DEFAULT_ROUTE
>> <script>: Sending reply S=100 Trying M=INVITE fs='?127.0.0.1?:?5062?'
>> du='?127.0.0.1?:?5060?' - R=?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX:5060?
>> ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk PBX 18.24.3'
>>
>> Oct  7 06:43:50.807375 spce proxy[837572]: NOTICE:
>> ROUTE_LOAD_CALLEE_DOMAIN_PREF <script>: Load domain preferences for callee
>> '?XXX.XXX.XXX.XXX?' - R=?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX:5060?
>> ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk PBX 18.24.3'
>>
>> Oct  7 06:43:50.811499 spce proxy[837572]: NOTICE: ROUTE_AUTH <script>:
>> Authentication failed, no credentials -
>> R=?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX:5060?
>> ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk PBX 18.24.3'
>> Auth=?<null>?
>>
>> Oct  7 06:43:50.811581 spce proxy[837572]: NOTICE: ROUTE_AUTH <script>:
>> Sending reply S=407 fs='?127.0.0.1?:?5062?' du='?127.0.0.1?:?5060?' -
>> ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk PBX 18.24.3'
>>
>> Oct  7 06:43:50.812257 spce proxy[837568]: NOTICE: sl:filtered-ack
>> <script>: New request on proxy - M=ACK
>> R=?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX:5060?
>> F=?sip:XXCallerIDXX at XXX.XXX.86.104?
>> T=?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX? IP=?<null>?:?<null>?
>> (?127.0.0.1?:?5060?) ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='<null>'
>> DESTIP=?127.0.0.1?:?5062?
>>
>> Oct  7 06:43:50.872198 spce proxy[837565]: NOTICE: DEFAULT_ROUTE
>> <script>: New request on proxy - M=INVITE
>> R=?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX:5060?
>> F=?sip:XXCallerIDXX at XXX.XXX.86.104?
>> T=?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX? IP=?XXX.XXX.188.51?:?5060?
>> (?127.0.0.1?:?5060?) ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk
>> PBX 18.24.3' DESTIP=?127.0.0.1?:?5062?
>>
>> Oct  7 06:43:50.872229 spce proxy[837565]: NOTICE: DEFAULT_ROUTE
>> <script>: Sending reply S=100 Trying M=INVITE fs='?127.0.0.1?:?5062?'
>> du='?127.0.0.1?:?5060?' - R=?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX:5060?
>> ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk PBX 18.24.3'
>>
>> Oct  7 06:43:50.872504 spce proxy[837565]: NOTICE:
>> ROUTE_LOAD_CALLEE_DOMAIN_PREF <script>: Load domain preferences for callee
>> '?XXX.XXX.XXX.XXX?' - R=?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX:5060?
>> ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk PBX 18.24.3'
>>
>> Oct  7 06:43:50.877414 spce proxy[837565]: NOTICE: ROUTE_FIND_CALLER
>> <script>: Sending reply S=101 Connecting fs='?127.0.0.1?:?5062?'
>> du='?127.0.0.1?:?5060?' - R=?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX:5060?
>> ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk PBX 18.24.3'
>>
>> Oct  7 06:43:50.877751 spce proxy[837565]: NOTICE:
>> ROUTE_LOAD_CALLER_PREF <script>: Load caller preferences for uuid
>> '?86ba28d9-153f-4cee-800c-f34c1efea74a?' and domain '?XXX.XXX.86.104?' -
>> R=?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX:5060?
>> ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk PBX 18.24.3'
>>
>> Oct  7 06:43:50.883279 spce proxy[837565]: NOTICE:
>> ROUTE_LOAD_CALLER_CONTRACT_PREF <script>: Load caller contract preferences
>> for contract id '?2?' and ip '?XXX.XXX.188.51?' -
>> R=?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX:5060?
>> ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk PBX 18.24.3'
>>
>> Oct  7 06:43:50.883583 spce proxy[837565]: NOTICE:
>> ROUTE_LOAD_CALLER_RESELLER_PREF <script>: Load caller reseller preferences
>> for reseller id '?1?' - R=?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX:5060?
>> ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk PBX 18.24.3'
>>
>> Oct  7 06:43:50.884273 spce proxy[837565]: NOTICE: ROUTE_DLG_MANAGE
>> <script>: Dialog managed,
>> lua_dlg_callid:[?3e2e30bf-6132-478a-bdd0-292e139c39c5?] -
>> R=?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX:5060?
>> ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk PBX 18.24.3'
>>
>> Oct  7 06:43:50.884348 spce proxy[837565]: NOTICE: ROUTE_GET_CALLER_CLI
>> <script>: User-Provided CLI '?XXCallerIDXX?' taken from From-User -
>> R=?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX:5060?
>> ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk PBX 18.24.3'
>>
>> Oct  7 06:43:50.888566 spce proxy[837565]: NOTICE:
>> ROUTE_APPLY_REWRITE_RULE <script>: Applying caller-in usr/domain rewrite
>> rules on user-provided CLI using dpid '7' -
>> R=?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX:5060?
>> ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk PBX 18.24.3'
>>
>> Oct  7 06:43:50.888839 spce proxy[837565]: NOTICE:
>> ROUTE_CHECK_CLI_ALLOWED <script>: User-provided CLI '?XXCallerIDXX?'
>> accepted (matched user preference) -
>> R=?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX:5060?
>> ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk PBX 18.24.3'
>>
>> Oct  7 06:43:50.888915 spce proxy[837565]: NOTICE: ROUTE_INVITE
>> <script>: Setting '?XXCallerIDXX?@?XXX.XXX.86.104?' as initiating
>> user-provided CLI - R=?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX:5060?
>> ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk PBX 18.24.3'
>>
>> Oct  7 06:43:50.888963 spce proxy[837565]: NOTICE: ROUTE_INVITE
>> <script>: Setting '?XXCallerIDXX?@?XXX.XXX.86.104?' as initiating
>> network-provided CLI - R=?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX:5060?
>> ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk PBX 18.24.3'
>>
>> Oct  7 06:43:50.889037 spce proxy[837565]: NOTICE:
>> ROUTE_APPLY_REWRITE_RULE <script>: Applying callee-in usr/domain rewrite
>> rules on called party using dpid '8' -
>> R=?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX:5060?
>> ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk PBX 18.24.3'
>>
>> Oct  7 06:43:50.890596 spce proxy[837565]: NOTICE: ROUTE_FIND_CALLEE
>> <script>: Callee is not local - R=?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX:5060?
>> ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk PBX 18.24.3'
>>
>> Oct  7 06:43:50.890869 spce proxy[837565]: NOTICE: ROUTE_PSTN <script>:
>> Call to SIP Peering - R=?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX:5060?
>> ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk PBX 18.24.3'
>>
>> Oct  7 06:43:50.890924 spce proxy[837565]: NOTICE: ROUTE_PSTN <script>:
>> Load gws matching calling part '?sip:XXCallerIDXX at XXX.XXX.86.104?' and
>> called user '?xxxxxxxxxxx2607244?' and called part
>> '?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX:5060?' -
>> R=?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX:5060?
>> ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk PBX 18.24.3'
>>
>> Oct  7 06:43:50.891692 spce proxy[837565]: ERROR: ROUTE_PSTN_SELECT_PEER
>> pv [pv_core.c:336]: pv_get_ruri_attr(): failed to parse the R-URI
>>
>> Oct  7 06:43:50.891806 spce proxy[837565]: ERROR:
>> ROUTE_LOAD_PEER_OUT_PREF pv [pv_core.c:261]: pv_get_ruri(): failed to parse
>> the R-URI
>>
>> Oct  7 06:43:50.891827 spce proxy[837565]: NOTICE:
>> ROUTE_LOAD_PEER_OUT_PREF <script>: Load peer preferences for peer host '1'
>> - R=?<null>? ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk PBX
>> 18.24.3'
>>
>> Oct  7 06:43:50.893690 spce proxy[837565]: ERROR: ROUTE_PSTN pv
>> [pv_core.c:336]: pv_get_ruri_attr(): failed to parse the R-URI
>>
>> Oct  7 06:43:50.895431 spce proxy[837565]: ERROR: ROUTE_PSTN pv
>> [pv_core.c:336]: pv_get_ruri_attr(): failed to parse the R-URI
>>
>> Oct  7 06:43:50.895538 spce proxy[837565]: ERROR: <null> pv
>> [pv_core.c:336]: pv_get_ruri_attr(): failed to parse the R-URI
>>
>> Oct  7 06:43:50.895558 spce proxy[837565]: ERROR: <null> pv
>> [pv_core.c:336]: pv_get_ruri_attr(): failed to parse the R-URI
>>
>> Oct  7 06:43:50.901133 spce proxy[837565]: ERROR: ROUTE_OUTBOUND pv
>> [pv_core.c:336]: pv_get_ruri_attr(): failed to parse the R-URI
>>
>> Oct  7 06:43:50.901156 spce proxy[837565]: ERROR: ROUTE_OUTBOUND pv
>> [pv_core.c:336]: pv_get_ruri_attr(): failed to parse the R-URI
>>
>> Oct  7 06:43:50.901829 spce proxy[837565]: ERROR:
>> ROUTE_APPLY_REWRITE_RULE pv [pv_core.c:261]: pv_get_ruri(): failed to parse
>> the R-URI
>>
>> Oct  7 06:43:50.901850 spce proxy[837565]: NOTICE:
>> ROUTE_APPLY_REWRITE_RULE <script>: Applying acc caller-out peer rewrite
>> rules on acc caller party using dpid '3' - R=?<null>?
>> ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk PBX 18.24.3'
>>
>> Oct  7 06:43:50.902770 spce proxy[837565]: ERROR: ROUTE_ACC_CALLER pv
>> [pv_core.c:261]: pv_get_ruri(): failed to parse the R-URI
>>
>> Oct  7 06:43:50.902792 spce proxy[837565]: NOTICE: ROUTE_ACC_CALLER
>> <script>: Setting acc source-leg for uuid
>> '?86ba28d9-153f-4cee-800c-f34c1efea74a?':
>> '?{"gpp":["70770",null,null,null,null,null,null,null,null,null],"clir":0,"uuid":"86ba28d9-153f-4cee-800c-f34c1efea74a","u":"XXCallerIDXX","d":"XXX.XXX.86.104","cli":"XXCallerIDXX","s_id":"70770","c_id":"70770","a_id":2,"pau":"","par":"","s":"call","ip":"XXX.XXX.188.51","t":1728276230.902101,"lnp_p":"","cli_out":"XXCallerIDXX","lnp_t":"","pai":"","ppi":"","div":"","last_hih":"","cid":"","u2u":"","lcr":0,"cc_quota":"","cc_sub":"1","cc_cust":"1"}?'
>> - R=?<null>? ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk PBX
>> 18.24.3'
>>
>> Oct  7 06:43:50.903156 spce proxy[837565]: ERROR:
>> ROUTE_APPLY_REWRITE_RULE pv [pv_core.c:261]: pv_get_ruri(): failed to parse
>> the R-URI
>>
>> Oct  7 06:43:50.903176 spce proxy[837565]: NOTICE:
>> ROUTE_APPLY_REWRITE_RULE <script>: Applying acc callee-out peer rewrite
>> rules on acc called party using dpid '4' - R=?<null>?
>> ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk PBX 18.24.3'
>>
>> Oct  7 06:43:50.903656 spce proxy[837565]: ERROR: ROUTE_ACC_CALLEE pv
>> [pv_core.c:261]: pv_get_ruri(): failed to parse the R-URI
>>
>> Oct  7 06:43:50.903679 spce proxy[837565]: NOTICE: ROUTE_ACC_CALLEE
>> <script>: Setting acc destination-leg for uuid '?0?':
>> '?{"gpp":[null,null,null,null,null,null,null,null,null,null],"plu":0,"s_id":"","c_id":"","a_id":0,"dialed":"xxxxxxxxxxx2607244","uuid":"0","u":"<null>","d":"<null>","u_in":"xxxxxxxxxxx2607244","d_in":"XXX.XXX.XXX.XXX","lcr":1,"lnp_p":"","u_out":"0","lnp_t":"","fci":"","cc_quota":"","cc_sub":"1","cc_cust":""}?'
>> - R=?<null>? ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk PBX
>> 18.24.3'
>>
>> Oct  7 06:43:50.903837 spce proxy[837565]: ERROR: ROUTE_OUTBOUND tm
>> [t_lookup.c:1827]: new_t(): uri invalid
>>
>> Oct  7 06:43:50.903858 spce proxy[837565]: ERROR: ROUTE_OUTBOUND tm
>> [t_lookup.c:1979]: t_newtran(): new_t failed
>>
>> Oct  7 06:43:50.903913 spce proxy[837565]: ERROR: ROUTE_OUTBOUND sl
>> [sl_funcs.c:428]: sl_reply_error(): stateless error reply used: I'm
>> terribly sorry, server error occurred (1/SL)
>>
>> Oct  7 06:43:50.904591 spce proxy[837570]: NOTICE: sl:filtered-ack
>> <script>: New request on proxy - M=ACK
>> R=?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX:5060?
>> F=?sip:XXCallerIDXX at XXX.XXX.86.104?
>> T=?sip:xxxxxxxxxxx2607244 at XXX.XXX.XXX.XXX? IP=?<null>?:?<null>?
>> (?127.0.0.1?:?5060?) ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='<null>'
>> DESTIP=?127.0.0.1?:?5062?
>>
>>
>> ****
>>
>> Thanks
>>
>>
>> On Mon, Oct 7, 2024 at 1:11?AM Dario Busso <dbusso at sipwise.com> wrote:
>>
>>> Looking at the error log, it seems that the INVITE received by SPCE is
>>> malformed.
>>> You need to check the peer system setup, not the SPCE.
>>>
>>>
>>> Dario B.
>>>
>>>
>>> Il lun 7 ott 2024, 06:50 Edo <edo.ekunwe at gmail.com> ha scritto:
>>>
>>>> Hi,
>>>>
>>>> I tried to make a call to a subscriber (SIP Trunk setup) and I got the
>>>> following error message.
>>>> *****
>>>>
>>>> Oct  7 06:43:50.891692 spce proxy[837565]: ERROR:
>>>> ROUTE_PSTN_SELECT_PEER pv [pv_core.c:336]: pv_get_ruri_attr(): failed to
>>>> parse the R-URI
>>>>
>>>> Oct  7 06:43:50.891806 spce proxy[837565]: ERROR:
>>>> ROUTE_LOAD_PEER_OUT_PREF pv [pv_core.c:261]: pv_get_ruri(): failed to parse
>>>> the R-URI
>>>>
>>>> Oct  7 06:43:50.891827 spce proxy[837565]: NOTICE:
>>>> ROUTE_LOAD_PEER_OUT_PREF <script>: Load peer preferences for peer host '1'
>>>> - R=?<null>? ID=?3e2e30bf-6132-478a-bdd0-292e139c39c5? UA='Asterisk PBX
>>>> 18.24.3'
>>>>
>>>> Oct  7 06:43:50.893690 spce proxy[837565]: ERROR: ROUTE_PSTN pv
>>>> [pv_core.c:336]: pv_get_ruri_attr(): failed to parse the R-URI
>>>>
>>>> Oct  7 06:43:50.895431 spce proxy[837565]: ERROR: ROUTE_PSTN pv
>>>> [pv_core.c:336]: pv_get_ruri_attr(): failed to parse the R-URI
>>>>
>>>> Oct  7 06:43:50.895538 spce proxy[837565]: ERROR: <null> pv
>>>> [pv_core.c:336]: pv_get_ruri_attr(): failed to parse the R-URI
>>>>
>>>> Oct  7 06:43:50.895558 spce proxy[837565]: ERROR: <null> pv
>>>> [pv_core.c:336]: pv_get_ruri_attr(): failed to parse the R-URI
>>>>
>>>> Oct  7 06:43:50.901133 spce proxy[837565]: ERROR: ROUTE_OUTBOUND pv
>>>> [pv_core.c:336]: pv_get_ruri_attr(): failed to parse the R-URI
>>>>
>>>> Oct  7 06:43:50.901156 spce proxy[837565]: ERROR: ROUTE_OUTBOUND pv
>>>> [pv_core.c:336]: pv_get_ruri_attr(): failed to parse the R-URI
>>>>
>>>> Oct  7 06:43:50.901829 spce proxy[837565]: ERROR:
>>>> ROUTE_APPLY_REWRITE_RULE pv [pv_core.c:261]: pv_get_ruri(): failed to parse
>>>> the R-URI
>>>> *****
>>>>  I am looking for where I can set the system to a specific R-URI or
>>>> check the system to use network provided R-RURI.
>>>>
>>>> Please help.
>>>>
>>>> Thanks
>>>>
>>>>
>>>> --
>>>> -----
>>>> Ekunwe
>>>> Tel: 601.497.3932
>>>> --
>>>> Spce-user mailing list
>>>> Spce-user at lists.sipwise.com
>>>> http://lists.sipwise.com/mailman/listinfo/spce-user_lists.sipwise.com
>>>>
>>>
>>
>> --
>> -----
>> Ekunwe
>> Tel: 601.497.3932
>>
>

-- 
-----
Ekunwe
Tel: 601.497.3932
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From mcapetta at sipwise.com  Tue Oct 15 06:13:45 2024
From: mcapetta at sipwise.com (Marco Capetta)
Date: Tue, 15 Oct 2024 12:13:45 +0200
Subject: [Spce-user] [EXTERNAL]  Header Manipulation in CE?
In-Reply-To: <2d0c92c7-ec4d-4117-a004-7d035c321907@lard.at>
References: <2d0c92c7-ec4d-4117-a004-7d035c321907@lard.at>
Message-ID: <CAEZOBv7qO6WpeVgROYqFu6zzBF9aF4__P+maWJZCeuHOWJteUQ@mail.gmail.com>

Hi,

Header Manipulation feature is available only in PRO and CARRIER systems.

If you are interested in our commercial product, feel free to contact our
sales team at sales at sipwise.com

Thank you
Marco

On Mon, Oct 14, 2024 at 7:30?PM spce at lard.at <spce at lard.at> wrote:

> Hello,
>
> i would like to add a Reason-header to a CANCEL-message. Is such a kind
> of header manipulation possible in sipwise C5 CE?
>
> Version: sipwise 12.5.1 CE
>
>
> Thx
>
> Malte
>
> --
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/mailman/listinfo/spce-user_lists.sipwise.com
>
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From manwe at sipdoc.net  Wed Oct 16 16:42:10 2024
From: manwe at sipdoc.net (Jon Bonilla (Manwe))
Date: Wed, 16 Oct 2024 22:42:10 +0200
Subject: [Spce-user] RATEOMAT_SHUFFLE_BATCH
Message-ID: <20241016224210.3c88e6d0@multivac.sipdoc.net>

Hi all

I'm checking rateomat to see if it can be speed up a little bit. I found that
there's a $shuffle_batch option which is supposed to shuffle grabbed cdrs to
avoid collisions in case more than one rateomat is running at the same time. 

I'm not sure if running 2 or more rateomat instances will speed up the rating
process and if it's documented somewhere.

Any guide or tip would be appreciated.


cheers,

Jon 

-- 
PekePBX, the multitenant PBX solution
https://pekepbx.com
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From rkrenn at sipwise.com  Wed Oct 16 21:21:02 2024
From: rkrenn at sipwise.com (rkrenn at sipwise.com)
Date: Thu, 17 Oct 2024 03:21:02 +0200
Subject: [Spce-user] [EXTERNAL]  RATEOMAT_SHUFFLE_BATCH
In-Reply-To: <20241016224210.3c88e6d0@multivac.sipdoc.net>
References: <20241016224210.3c88e6d0@multivac.sipdoc.net>
Message-ID: <001701db2032$d6e50af0$84af20d0$@sipwise.com>

Hi,

> I'm not sure if running 2 or more rateomat instances will speed up the
rating process and if it's documented somewhere.
Yes, check out with sales if you want to enable rate-o-mat multimaster mode.

br

-----Urspr?ngliche Nachricht-----
Von: Jon Bonilla (Manwe) <manwe at sipdoc.net> 
Gesendet: Mittwoch, 16. Oktober 2024 22:42
An: spce-user at lists.sipwise.com
Betreff: [EXTERNAL] [Spce-user] RATEOMAT_SHUFFLE_BATCH

Hi all

I'm checking rateomat to see if it can be speed up a little bit. I found
that there's a $shuffle_batch option which is supposed to shuffle grabbed
cdrs to avoid collisions in case more than one rateomat is running at the
same time. 

I'm not sure if running 2 or more rateomat instances will speed up the
rating process and if it's documented somewhere.

Any guide or tip would be appreciated.


cheers,

Jon 

--
PekePBX, the multitenant PBX solution
https://pekepbx.com



From manwe at sipdoc.net  Thu Oct 17 02:15:43 2024
From: manwe at sipdoc.net (Jon Bonilla (Manwe))
Date: Thu, 17 Oct 2024 08:15:43 +0200
Subject: [Spce-user] [EXTERNAL]  RATEOMAT_SHUFFLE_BATCH
In-Reply-To: <001701db2032$d6e50af0$84af20d0$@sipwise.com>
References: <20241016224210.3c88e6d0@multivac.sipdoc.net>
 <001701db2032$d6e50af0$84af20d0$@sipwise.com>
Message-ID: <20241017081543.19cf8604@multivac.sipdoc.net>

El Thu, 17 Oct 2024 03:21:02 +0200
<rkrenn at sipwise.com> escribi?:

> Hi,
> 
> > I'm not sure if running 2 or more rateomat instances will speed up the  
> rating process and if it's documented somewhere.
> Yes, check out with sales if you want to enable rate-o-mat multimaster mode.
> 


Hi Renne


I'd like to know how it works. I don't think sales will know that.

If both rateomat grab the same 100 cdrs, they shuffle them and then they go one
by one and as it's shuffled there are less race conditions as it seems. But not
sure about that.
And if it's shuffled the rating is not secuential any more, right?


Not sure about how they calculate the balances and the fraud limits and deal
with collisions. That's why I wanted to know if it's documented somewhere. 


 


-- 
PekePBX, the multitenant PBX solution
https://pekepbx.com
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From walter at myrepublic.net  Thu Oct 17 04:34:14 2024
From: walter at myrepublic.net (Walter Klomp)
Date: Thu, 17 Oct 2024 16:34:14 +0800
Subject: [Spce-user] mr12.5.1 - default administrator can't edit peering
 servers
Message-ID: <CAH9G5wapN2UvBoJHaQPJK=J6W2ntUV1ztYYas44s4uuCOWuQbg@mail.gmail.com>

Hi,

The default administrator from an install from ISO is only allowed to do
reseller stuff, so no domain and peering server.  I have to go into the
database to change the role to "is system" to be able to manipulate the
peering servers.

Perhaps something to look at ?

-- 
[image: Signature header]
[image: Signature image]
Walter Klomp
Head of Voice Network
[image: Email] walter at myrepublic.net
Beyond Work :
11 Lorong 3 Toa Payoh, #04-11/15, Jackson Square, Block B, Singapore 319579
www.myrepublic.net
* [image: Facebook-f] * <https://scribe-mail.myrepublic.net/s2/21iu7kgk> *
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The information in this message is privileged and confidential. If you are
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by return email. Please advise immediately if you or your employer does not
consent to email for messages of this kind. Any content that is not
concerned with MyRepublic business reflects the views of the sender only
and not those of MyRepublic.

-- 
The contents of this email and any attachments are confidential and may 
also be privileged. You must not disseminate the contents of this email and 
any attachments without permission of the sender. If you have received this 
email by mistake, please delete all copies and inform the sender 
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From aclark at structuredcommunications.co.uk  Thu Oct 17 04:40:57 2024
From: aclark at structuredcommunications.co.uk (Adam Clark)
Date: Thu, 17 Oct 2024 08:40:57 +0000
Subject: [Spce-user] mr12.5.1 - default administrator can't edit peering
 servers
In-Reply-To: <CAH9G5wapN2UvBoJHaQPJK=J6W2ntUV1ztYYas44s4uuCOWuQbg@mail.gmail.com>
References: <CAH9G5wapN2UvBoJHaQPJK=J6W2ntUV1ztYYas44s4uuCOWuQbg@mail.gmail.com>
Message-ID: <CWLP265MB10921D64445116FA1C09AEF7FC472@CWLP265MB1092.GBRP265.PROD.OUTLOOK.COM>

Myself and another member have seen thing. It only happens in set conditions.

But you are right they are missing from the GUI

I did offer the VM image we had of this very thing (broken) but it was never taken up or responded to.

Regards,

Adam

Sent from Outlook for Android<https://aka.ms/AAb9ysg>
________________________________
From: Walter Klomp via Spce-user <spce-user at lists.sipwise.com>
Sent: Thursday, October 17, 2024 9:34:14 AM
To: spce-user <spce-user at lists.sipwise.com>
Subject: [Spce-user] mr12.5.1 - default administrator can't edit peering servers

Hi,

The default administrator from an install from ISO is only allowed to do reseller stuff, so no domain and peering server.  I have to go into the database to change the role to "is system" to be able to manipulate the peering servers.

Perhaps something to look at ?

--

[Signature image]
Walter Klomp
Head of Voice Network
[Email]         walter at myrepublic.net<mailto:walter at myrepublic.net>
Beyond Work :
11 Lorong 3 Toa Payoh, #04-11/15, Jackson Square, Block B, Singapore 319579
www.myrepublic.net<http://www.myrepublic.net/>
[Facebook-f] <https://scribe-mail.myrepublic.net/s2/21iu7kgk>   [Instagram]  <https://scribe-mail.myrepublic.net/s2/kzqypegm>   [Twitter]  <https://scribe-mail.myrepublic.net/s2/geefi681>     [Linkedin-in]  <https://scribe-mail.myrepublic.net/s2/3bzu1nu9>
The information in this message is privileged and confidential. If you are not the intended recipient of this email, please don't read, copy, use, distribute or tell anyone about it; kindly destroy it and notify the sender by return email. Please advise immediately if you or your employer does not consent to email for messages of this kind. Any content that is not concerned with MyRepublic business reflects the views of the sender only and not those of MyRepublic.

The contents of this email and any attachments are confidential and may also be privileged. You must not disseminate the contents of this email and any attachments without permission of the sender. If you have received this email by mistake, please delete all copies and inform the sender immediately. You may refer to our company's Privacy Policy here<https://myrepublic.net/sg/legal/terms-of-use-policies/privacy-policy/>.
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From manwe at sipdoc.net  Thu Oct 17 07:37:51 2024
From: manwe at sipdoc.net (Jon Bonilla (Manwe))
Date: Thu, 17 Oct 2024 13:37:51 +0200
Subject: [Spce-user] mr12.5.1 - default administrator can't edit peering
 servers
In-Reply-To: <CAH9G5wapN2UvBoJHaQPJK=J6W2ntUV1ztYYas44s4uuCOWuQbg@mail.gmail.com>
References: <CAH9G5wapN2UvBoJHaQPJK=J6W2ntUV1ztYYas44s4uuCOWuQbg@mail.gmail.com>
Message-ID: <20241017133751.47a3eed2@multivac.sipdoc.net>

El Thu, 17 Oct 2024 16:34:14 +0800
Walter Klomp via Spce-user <spce-user at lists.sipwise.com> escribi?:

> Hi,
> 
> The default administrator from an install from ISO is only allowed to do
> reseller stuff, so no domain and peering server.  I have to go into the
> database to change the role to "is system" to be able to manipulate the
> peering servers.
> 

I did a 10.5 install yesterday and same happened to me.



-- 
PekePBX, the multitenant PBX solution
https://pekepbx.com
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From mcapetta at sipwise.com  Thu Oct 17 08:30:21 2024
From: mcapetta at sipwise.com (Marco Capetta)
Date: Thu, 17 Oct 2024 14:30:21 +0200
Subject: [Spce-user] [EXTERNAL] Re: mr12.5.1 - default administrator
 can't edit peering servers
In-Reply-To: <20241017133751.47a3eed2@multivac.sipdoc.net>
References: <CAH9G5wapN2UvBoJHaQPJK=J6W2ntUV1ztYYas44s4uuCOWuQbg@mail.gmail.com>
 <20241017133751.47a3eed2@multivac.sipdoc.net>
Message-ID: <CAEZOBv5EMAXT0KTAYQArjgjKqH=mnCOjSFt0zFP_MekGFVA4aw@mail.gmail.com>

Hi,

thank you for the report.

We opened an internal ticket with number MT#61306 to check the issue.

Thank you
Marco

On Thu, Oct 17, 2024 at 1:38?PM Jon Bonilla (Manwe) <manwe at sipdoc.net>
wrote:

> El Thu, 17 Oct 2024 16:34:14 +0800
> Walter Klomp via Spce-user <spce-user at lists.sipwise.com> escribi?:
>
> > Hi,
> >
> > The default administrator from an install from ISO is only allowed to do
> > reseller stuff, so no domain and peering server.  I have to go into the
> > database to change the role to "is system" to be able to manipulate the
> > peering servers.
> >
>
> I did a 10.5 install yesterday and same happened to me.
>
>
>
> --
> PekePBX, the multitenant PBX solution
> https://pekepbx.com
> --
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/mailman/listinfo/spce-user_lists.sipwise.com
>
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From mcapetta at sipwise.com  Thu Oct 17 12:31:54 2024
From: mcapetta at sipwise.com (Marco Capetta)
Date: Thu, 17 Oct 2024 18:31:54 +0200
Subject: [Spce-user] [EXTERNAL] Re: mr12.5.1 - default administrator
 can't edit peering servers
In-Reply-To: <CAEZOBv5EMAXT0KTAYQArjgjKqH=mnCOjSFt0zFP_MekGFVA4aw@mail.gmail.com>
References: <CAH9G5wapN2UvBoJHaQPJK=J6W2ntUV1ztYYas44s4uuCOWuQbg@mail.gmail.com>
 <20241017133751.47a3eed2@multivac.sipdoc.net>
 <CAEZOBv5EMAXT0KTAYQArjgjKqH=mnCOjSFt0zFP_MekGFVA4aw@mail.gmail.com>
Message-ID: <CAEZOBv7nxX0o8mGCxwwg-6mLoHWkPi1zHYY=uqrfPQgWGi6Cjw@mail.gmail.com>

Hi again,

can you please specify on which exact NGCP version this issue is happening?
And also if you use API, the NEW Admin UI or the OLD Admin UI?

I have just installed mr12.5.1 from the ISO and I'm able to create domains
and peering using the default "administrator".

Thank you
Marco

On Thu, Oct 17, 2024 at 2:30?PM Marco Capetta <mcapetta at sipwise.com> wrote:

> Hi,
>
> thank you for the report.
>
> We opened an internal ticket with number MT#61306 to check the issue.
>
> Thank you
> Marco
>
> On Thu, Oct 17, 2024 at 1:38?PM Jon Bonilla (Manwe) <manwe at sipdoc.net>
> wrote:
>
>> El Thu, 17 Oct 2024 16:34:14 +0800
>> Walter Klomp via Spce-user <spce-user at lists.sipwise.com> escribi?:
>>
>> > Hi,
>> >
>> > The default administrator from an install from ISO is only allowed to do
>> > reseller stuff, so no domain and peering server.  I have to go into the
>> > database to change the role to "is system" to be able to manipulate the
>> > peering servers.
>> >
>>
>> I did a 10.5 install yesterday and same happened to me.
>>
>>
>>
>> --
>> PekePBX, the multitenant PBX solution
>> https://pekepbx.com
>> --
>> Spce-user mailing list
>> Spce-user at lists.sipwise.com
>> http://lists.sipwise.com/mailman/listinfo/spce-user_lists.sipwise.com
>>
>
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From manwe at sipdoc.net  Thu Oct 17 16:01:11 2024
From: manwe at sipdoc.net (Jon Bonilla (Manwe))
Date: Thu, 17 Oct 2024 22:01:11 +0200
Subject: [Spce-user] [EXTERNAL] Re: mr12.5.1 - default administrator
 can't edit peering servers
In-Reply-To: <CAEZOBv7nxX0o8mGCxwwg-6mLoHWkPi1zHYY=uqrfPQgWGi6Cjw@mail.gmail.com>
References: <CAH9G5wapN2UvBoJHaQPJK=J6W2ntUV1ztYYas44s4uuCOWuQbg@mail.gmail.com>
 <20241017133751.47a3eed2@multivac.sipdoc.net>
 <CAEZOBv5EMAXT0KTAYQArjgjKqH=mnCOjSFt0zFP_MekGFVA4aw@mail.gmail.com>
 <CAEZOBv7nxX0o8mGCxwwg-6mLoHWkPi1zHYY=uqrfPQgWGi6Cjw@mail.gmail.com>
Message-ID: <20241017220111.048ed892@multivac.sipdoc.net>

El Thu, 17 Oct 2024 18:31:54 +0200
Marco Capetta via Spce-user <spce-user at lists.sipwise.com> escribi?:

> Hi again,
> 
> can you please specify on which exact NGCP version this issue is happening?
> And also if you use API, the NEW Admin UI or the OLD Admin UI?
> 
> I have just installed mr12.5.1 from the ISO and I'm able to create domains
> and peering using the default "administrator".
> 

Hi marco

What I did was install latest 10.5.X from the iso. In the admin web interface
(switched to old panel but I don't think that's relevant) I couldn't see
peering or reseller because the "administrator" user had a "reseller" role.

I edited the role directly in the billing database to fix that.


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From walter at myrepublic.net  Thu Oct 17 22:52:33 2024
From: walter at myrepublic.net (Walter Klomp)
Date: Fri, 18 Oct 2024 10:52:33 +0800
Subject: [Spce-user] [EXTERNAL] Re: mr12.5.1 - default administrator
 can't edit peering servers
In-Reply-To: <CAEZOBv7nxX0o8mGCxwwg-6mLoHWkPi1zHYY=uqrfPQgWGi6Cjw@mail.gmail.com>
References: <CAH9G5wapN2UvBoJHaQPJK=J6W2ntUV1ztYYas44s4uuCOWuQbg@mail.gmail.com>
 <20241017133751.47a3eed2@multivac.sipdoc.net>
 <CAEZOBv5EMAXT0KTAYQArjgjKqH=mnCOjSFt0zFP_MekGFVA4aw@mail.gmail.com>
 <CAEZOBv7nxX0o8mGCxwwg-6mLoHWkPi1zHYY=uqrfPQgWGi6Cjw@mail.gmail.com>
Message-ID: <CAH9G5wZ+2hfiKgSOuikPvExgGN9wbgnxgYBrRQRgdfYhMUrEQA@mail.gmail.com>

I am referring to mr12.5.1, but it also happens on mr10.5.1

I did not try the API directly, but when using the new panel, the
peering-server options do not show up.

Also using the API I could not create the peering groups and servers until
I changed the administrator role to add "is_system"

On Fri, Oct 18, 2024 at 12:32?AM Marco Capetta via Spce-user <
spce-user at lists.sipwise.com> wrote:

> Hi again,
>
> can you please specify on which exact NGCP version this issue is happening?
> And also if you use API, the NEW Admin UI or the OLD Admin UI?
>
> I have just installed mr12.5.1 from the ISO and I'm able to create domains
> and peering using the default "administrator".
>
> Thank you
> Marco
>
> On Thu, Oct 17, 2024 at 2:30?PM Marco Capetta <mcapetta at sipwise.com>
> wrote:
>
>> Hi,
>>
>> thank you for the report.
>>
>> We opened an internal ticket with number MT#61306 to check the issue.
>>
>> Thank you
>> Marco
>>
>> On Thu, Oct 17, 2024 at 1:38?PM Jon Bonilla (Manwe) <manwe at sipdoc.net>
>> wrote:
>>
>>> El Thu, 17 Oct 2024 16:34:14 +0800
>>> Walter Klomp via Spce-user <spce-user at lists.sipwise.com> escribi?:
>>>
>>> > Hi,
>>> >
>>> > The default administrator from an install from ISO is only allowed to
>>> do
>>> > reseller stuff, so no domain and peering server.  I have to go into the
>>> > database to change the role to "is system" to be able to manipulate the
>>> > peering servers.
>>> >
>>>
>>> I did a 10.5 install yesterday and same happened to me.
>>>
>>>
>>>
>>> --
>>> PekePBX, the multitenant PBX solution
>>> https://pekepbx.com
>>> --
>>> Spce-user mailing list
>>> Spce-user at lists.sipwise.com
>>> http://lists.sipwise.com/mailman/listinfo/spce-user_lists.sipwise.com
>>>
>> --
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/mailman/listinfo/spce-user_lists.sipwise.com
>


-- 
[image: Signature header]
[image: Signature image]
Walter Klomp
Head of Voice Network
[image: Email] walter at myrepublic.net
Beyond Work :
11 Lorong 3 Toa Payoh, #04-11/15, Jackson Square, Block B, Singapore 319579
www.myrepublic.net
* [image: Facebook-f] * <https://scribe-mail.myrepublic.net/s2/21iu7kgk> *
[image: Instagram] * <https://scribe-mail.myrepublic.net/s2/kzqypegm> *
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The information in this message is privileged and confidential. If you are
not the intended recipient of this email, please don't read, copy, use,
distribute or tell anyone about it; kindly destroy it and notify the sender
by return email. Please advise immediately if you or your employer does not
consent to email for messages of this kind. Any content that is not
concerned with MyRepublic business reflects the views of the sender only
and not those of MyRepublic.

-- 
The contents of this email and any attachments are confidential and may 
also be privileged. You must not disseminate the contents of this email and 
any attachments without permission of the sender. If you have received this 
email by mistake, please delete all copies and inform the sender 
immediately. You may refer to our company's Privacy Policy here 
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From rkrenn at sipwise.com  Fri Oct 18 02:29:38 2024
From: rkrenn at sipwise.com (rkrenn at sipwise.com)
Date: Fri, 18 Oct 2024 08:29:38 +0200
Subject: [Spce-user] [EXTERNAL]  RATEOMAT_SHUFFLE_BATCH
In-Reply-To: <20241017081543.19cf8604@multivac.sipdoc.net>
References: <20241016224210.3c88e6d0@multivac.sipdoc.net>	<001701db2032$d6e50af0$84af20d0$@sipwise.com>
 <20241017081543.19cf8604@multivac.sipdoc.net>
Message-ID: <007001db2127$1c3e7ce0$54bb76a0$@sipwise.com>

Hi,

> I'd like to know how it works. I don't think sales will know that.
These settings are related to the multi-master mode. If you require particular support on that enhanced functionality, you can nowadays request particular support for that:

*support and certain Pro features can nowadays be provided even to CE users*.

I thought it seems that would match your situation, hence feel free to get into contact with sales at sipwise.com.

br

-----Urspr?ngliche Nachricht-----
Von: Jon Bonilla (Manwe) <manwe at sipdoc.net> 
Gesendet: Donnerstag, 17. Oktober 2024 08:16
An: rkrenn at sipwise.com
Cc: spce-user at lists.sipwise.com
Betreff: Re: [EXTERNAL] [Spce-user] RATEOMAT_SHUFFLE_BATCH

El Thu, 17 Oct 2024 03:21:02 +0200
<rkrenn at sipwise.com> escribi?:

> Hi,
> 
> > I'm not sure if running 2 or more rateomat instances will speed up 
> > the
> rating process and if it's documented somewhere.
> Yes, check out with sales if you want to enable rate-o-mat multimaster mode.
> 


Hi Renne


I'd like to know how it works. I don't think sales will know that.

If both rateomat grab the same 100 cdrs, they shuffle them and then they go one by one and as it's shuffled there are less race conditions as it seems. But not sure about that.
And if it's shuffled the rating is not secuential any more, right?


Not sure about how they calculate the balances and the fraud limits and deal with collisions. That's why I wanted to know if it's documented somewhere. 


 


--
PekePBX, the multitenant PBX solution
https://pekepbx.com



From mcapetta at sipwise.com  Tue Oct 22 03:59:29 2024
From: mcapetta at sipwise.com (Marco Capetta)
Date: Tue, 22 Oct 2024 09:59:29 +0200
Subject: [Spce-user] [EXTERNAL] Re: mr12.5.1 - default administrator
 can't edit peering servers
In-Reply-To: <CAH9G5wZ+2hfiKgSOuikPvExgGN9wbgnxgYBrRQRgdfYhMUrEQA@mail.gmail.com>
References: <CAH9G5wapN2UvBoJHaQPJK=J6W2ntUV1ztYYas44s4uuCOWuQbg@mail.gmail.com>
 <20241017133751.47a3eed2@multivac.sipdoc.net>
 <CAEZOBv5EMAXT0KTAYQArjgjKqH=mnCOjSFt0zFP_MekGFVA4aw@mail.gmail.com>
 <CAEZOBv7nxX0o8mGCxwwg-6mLoHWkPi1zHYY=uqrfPQgWGi6Cjw@mail.gmail.com>
 <CAH9G5wZ+2hfiKgSOuikPvExgGN9wbgnxgYBrRQRgdfYhMUrEQA@mail.gmail.com>
Message-ID: <CAEZOBv7+cqBwfL7ON3e+V+_YpP9uBqTL_962G+ThVwFn+rEM+Q@mail.gmail.com>

Thank you all for the details.

It looks in fact a bug of the change password process.
If you don't change the password of the default administrator user, then
you can see and edit all the menu as expected.

I'll keep you updated when the issue will be resolved.

Thank you

On Fri, Oct 18, 2024 at 4:53?AM Walter Klomp <walter at myrepublic.net> wrote:

> I am referring to mr12.5.1, but it also happens on mr10.5.1
>
> I did not try the API directly, but when using the new panel, the
> peering-server options do not show up.
>
> Also using the API I could not create the peering groups and servers until
> I changed the administrator role to add "is_system"
>
> On Fri, Oct 18, 2024 at 12:32?AM Marco Capetta via Spce-user <
> spce-user at lists.sipwise.com> wrote:
>
>> Hi again,
>>
>> can you please specify on which exact NGCP version this issue is
>> happening?
>> And also if you use API, the NEW Admin UI or the OLD Admin UI?
>>
>> I have just installed mr12.5.1 from the ISO and I'm able to create
>> domains and peering using the default "administrator".
>>
>> Thank you
>> Marco
>>
>> On Thu, Oct 17, 2024 at 2:30?PM Marco Capetta <mcapetta at sipwise.com>
>> wrote:
>>
>>> Hi,
>>>
>>> thank you for the report.
>>>
>>> We opened an internal ticket with number MT#61306 to check the issue.
>>>
>>> Thank you
>>> Marco
>>>
>>> On Thu, Oct 17, 2024 at 1:38?PM Jon Bonilla (Manwe) <manwe at sipdoc.net>
>>> wrote:
>>>
>>>> El Thu, 17 Oct 2024 16:34:14 +0800
>>>> Walter Klomp via Spce-user <spce-user at lists.sipwise.com> escribi?:
>>>>
>>>> > Hi,
>>>> >
>>>> > The default administrator from an install from ISO is only allowed to
>>>> do
>>>> > reseller stuff, so no domain and peering server.  I have to go into
>>>> the
>>>> > database to change the role to "is system" to be able to manipulate
>>>> the
>>>> > peering servers.
>>>> >
>>>>
>>>> I did a 10.5 install yesterday and same happened to me.
>>>>
>>>>
>>>>
>>>> --
>>>> PekePBX, the multitenant PBX solution
>>>> https://pekepbx.com
>>>> --
>>>> Spce-user mailing list
>>>> Spce-user at lists.sipwise.com
>>>> http://lists.sipwise.com/mailman/listinfo/spce-user_lists.sipwise.com
>>>>
>>> --
>> Spce-user mailing list
>> Spce-user at lists.sipwise.com
>> http://lists.sipwise.com/mailman/listinfo/spce-user_lists.sipwise.com
>>
>
>
> --
> [image: Signature header]
> [image: Signature image]
> Walter Klomp
> Head of Voice Network
> [image: Email] walter at myrepublic.net
> Beyond Work :
> 11 Lorong 3 Toa Payoh, #04-11/15, Jackson Square, Block B, Singapore 319579
> www.myrepublic.net
> * [image: Facebook-f] * <https://scribe-mail.myrepublic.net/s2/21iu7kgk> *
> [image: Instagram] * <https://scribe-mail.myrepublic.net/s2/kzqypegm> *
> [image: Twitter] * <https://scribe-mail.myrepublic.net/s2/geefi681> *
> [image: Linkedin-in] * <https://scribe-mail.myrepublic.net/s2/3bzu1nu9>
> The information in this message is privileged and confidential. If you are
> not the intended recipient of this email, please don't read, copy, use,
> distribute or tell anyone about it; kindly destroy it and notify the sender
> by return email. Please advise immediately if you or your employer does not
> consent to email for messages of this kind. Any content that is not
> concerned with MyRepublic business reflects the views of the sender only
> and not those of MyRepublic.
>
> The contents of this email and any attachments are confidential and may
> also be privileged. You must not disseminate the contents of this email and
> any attachments without permission of the sender. If you have received this
> email by mistake, please delete all copies and inform the sender
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From aclark at structuredcommunications.co.uk  Thu Oct 17 12:41:49 2024
From: aclark at structuredcommunications.co.uk (Adam Clark)
Date: Thu, 17 Oct 2024 16:41:49 +0000
Subject: [Spce-user] [EXTERNAL] Re: mr12.5.1 - default administrator
 can't edit peering servers
In-Reply-To: <CAEZOBv7nxX0o8mGCxwwg-6mLoHWkPi1zHYY=uqrfPQgWGi6Cjw@mail.gmail.com>
References: <CAH9G5wapN2UvBoJHaQPJK=J6W2ntUV1ztYYas44s4uuCOWuQbg@mail.gmail.com>
 <20241017133751.47a3eed2@multivac.sipdoc.net>
 <CAEZOBv5EMAXT0KTAYQArjgjKqH=mnCOjSFt0zFP_MekGFVA4aw@mail.gmail.com>
 <CAEZOBv7nxX0o8mGCxwwg-6mLoHWkPi1zHYY=uqrfPQgWGi6Cjw@mail.gmail.com>
Message-ID: <CWLP265MB10927F5A143F57014A0AC4ABFC472@CWLP265MB1092.GBRP265.PROD.OUTLOOK.COM>

Hi Marco,

When I also experienced this a while back, I did explain in that post and also offered the VM of the broken setup but no one from Sipwise acknowledged it.

For us it was on mr11.5.1

Initial install showed the GUI options but after doing 1 of the following it vanished:


  1.  Changing the administrator password via the old interface
  2.  Using the old interface adding subscribers .ect

We literally had to take a backup every time we changed something encase it vanished.

There was someone else at the time who had the same thing that I spoke with directly via email.

Seems to occur very much during the initial setup of Sipwise after its installed.

I am glad to see others are also reporting it as its clearly a bug under set conditions and has pulled across to the new builds as well.

I will see if we still have the broken VPS image.



Kind Regards,

[Structured Communications Ltd]
Adam Clark
Chief Network Engineer | Structured Communications Ltd
Phone: 0203 301 4000<tel:0203%20301%204000> DDI: 0203 301 4004<tel:0203%20301%204004>
www.structuredcommunications.co.uk<http://www.structuredcommunications.co.uk/>
[facebook]<https://www.facebook.com/StructuredCommunications>
[twitter]<https://twitter.com/scommsltd>
[twitter]<https://www.linkedin.com/company/structured-communications-limited>

[cid:image005.jpg at 01DB20BB.D60C5FF0]

[cid:image006.png at 01DB20BB.D60C5FF0]
This e-mail and any attachments transmitted with it, including replies and forwarded copies subsequently transmitted, contains information which may be confidential and which may also be legally privileged. The content of this e-mail is for the exclusive use of the individual(s) or entity named above, if you are not the intended recipient, please note that any form of distribution, copying or use of this e-mail or the information in it is strictly prohibited and may be unlawful. If you have received this e-mail in error please reply to the sender at Structured Communications Ltd and then delete the e-mail immediately. Any views or opinions presented are solely those of the author, and may not necessarily represent those of Structured Communications Ltd. Structured Communications Ltd is registered in England and Wales, registered no 07667451, and its registered office is Unit 8 Foundry Court, Foundry Lane, Horsham, West Sussex RH13 5PY. Registered VAT number GB116507336.


From: Marco Capetta via Spce-user <spce-user at lists.sipwise.com>
Sent: 17 October 2024 17:32
To: spce-user at lists.sipwise.com
Subject: Re: [Spce-user] [EXTERNAL] Re: mr12.5.1 - default administrator can't edit peering servers

Hi again,

can you please specify on which exact NGCP version this issue is happening?
And also if you use API, the NEW Admin UI or the OLD Admin UI?

I have just installed mr12.5.1 from the ISO and I'm able to create domains and peering using the default "administrator".

Thank you
Marco

On Thu, Oct 17, 2024 at 2:30?PM Marco Capetta <mcapetta at sipwise.com<mailto:mcapetta at sipwise.com>> wrote:
Hi,

thank you for the report.

We opened an internal ticket with number MT#61306 to check the issue.

Thank you
Marco

On Thu, Oct 17, 2024 at 1:38?PM Jon Bonilla (Manwe) <manwe at sipdoc.net<mailto:manwe at sipdoc.net>> wrote:
El Thu, 17 Oct 2024 16:34:14 +0800
Walter Klomp via Spce-user <spce-user at lists.sipwise.com<mailto:spce-user at lists.sipwise.com>> escribi?:

> Hi,
>
> The default administrator from an install from ISO is only allowed to do
> reseller stuff, so no domain and peering server.  I have to go into the
> database to change the role to "is system" to be able to manipulate the
> peering servers.
>

I did a 10.5 install yesterday and same happened to me.



--
PekePBX, the multitenant PBX solution
https://pekepbx.com<https://pekepbx.com/>
--
Spce-user mailing list
Spce-user at lists.sipwise.com<mailto:Spce-user at lists.sipwise.com>
http://lists.sipwise.com/mailman/listinfo/spce-user_lists.sipwise.com
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From aalevi at protonmail.com  Thu Oct 31 07:43:12 2024
From: aalevi at protonmail.com (Aaron Levi)
Date: Thu, 31 Oct 2024 11:43:12 +0000
Subject: [Spce-user] Configured demo server
Message-ID: <v9xTfYOnBpOvUceYWoleqO63jwsgHfbNCb-K1fHubji-9wBQzYM4nysfGQYngxOzKQOchrEbaTCiU_hurzKv07kTktfy8RNTiaXL9ZJEntY=@protonmail.com>

Hello!

I'm very new to Sipwise CE software and would like to evaluate it as an internal voip router. I have one C5 voip server with retail clients and few old hardware PBXed connected via E1 interface, so I'd like to use the Sipwice CE to route calls between all my PBXes and route the external outgoing calls based on LCR.
Every voip / billing server I know use it's own logic and its difficult to understand at the start. Is there any correctly configured demo server available on the network?

Thanks,
Aalevi

Sent with [Proton Mail](https://proton.me/mail/home) secure email.
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From walter at myrepublic.net  Sat Oct 26 06:26:08 2024
From: walter at myrepublic.net (Walter Klomp)
Date: Sat, 26 Oct 2024 18:26:08 +0800
Subject: [Spce-user] [EXTERNAL] Re: mr12.5.1 - default administrator
 can't edit peering servers
In-Reply-To: <CWLP265MB10927F5A143F57014A0AC4ABFC472@CWLP265MB1092.GBRP265.PROD.OUTLOOK.COM>
References: <CAH9G5wapN2UvBoJHaQPJK=J6W2ntUV1ztYYas44s4uuCOWuQbg@mail.gmail.com>
 <20241017133751.47a3eed2@multivac.sipdoc.net>
 <CAEZOBv5EMAXT0KTAYQArjgjKqH=mnCOjSFt0zFP_MekGFVA4aw@mail.gmail.com>
 <CAEZOBv7nxX0o8mGCxwwg-6mLoHWkPi1zHYY=uqrfPQgWGi6Cjw@mail.gmail.com>
 <CWLP265MB10927F5A143F57014A0AC4ABFC472@CWLP265MB1092.GBRP265.PROD.OUTLOOK.COM>
Message-ID: <CAH9G5wZw8H0y_1V+R+5V7vc+Vr-0bKACeR3B0R1S5e=Qe=vEKA@mail.gmail.com>

I got feedback from Sipwise that they managed to reproduce it and they?ll
fix it. It?s a bug in the change password function.

[image: Signature header]
[image: Signature image]
Walter Klomp
Head of Voice Network
[image: Email] walter at myrepublic.net
Beyond Work :
11 Lorong 3 Toa Payoh, #04-11/15, Jackson Square, Block B, Singapore 319579
www.myrepublic.net
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The information in this message is privileged and confidential. If you are
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concerned with MyRepublic business reflects the views of the sender only
and not those of MyRepublic.


On Sat, 26 Oct 2024 at 02:14, Adam Clark <
aclark at structuredcommunications.co.uk> wrote:

> Hi Marco,
>
>
>
> When I also experienced this a while back, I did explain in that post and
> also offered the VM of the broken setup but no one from Sipwise
> acknowledged it.
>
>
>
> For us it was on mr11.5.1
>
>
>
> Initial install showed the GUI options but after doing 1 of the following
> it vanished:
>
>
>
>    1. Changing the administrator password via the old interface
>    2. Using the old interface adding subscribers .ect
>
>
>
> We literally had to take a backup every time we changed something encase
> it vanished.
>
>
>
> There was someone else at the time who had the same thing that I spoke
> with directly via email.
>
>
>
> Seems to occur very much during the initial setup of Sipwise after its
> installed.
>
>
>
> I am glad to see others are also reporting it as its clearly a bug under
> set conditions and has pulled across to the new builds as well.
>
>
>
> I will see if we still have the broken VPS image.
>
>
>
>
>
>
>
> Kind Regards,
>
>
>
> [image: Structured Communications Ltd]
>
> *Adam Clark*
>
> *Chief Network Engineer* | Structured Communications Ltd
>
> Phone: 0203 301 4000 <0203%20301%204000> DDI: 0203 301 4004
> <0203%20301%204004>
>
> www.structuredcommunications.co.uk
>
> [image: facebook] <https://www.facebook.com/StructuredCommunications>
>
> [image: twitter] <https://twitter.com/scommsltd>
>
> [image: twitter]
> <https://www.linkedin.com/company/structured-communications-limited>
>
>
>
>
> This e-mail and any attachments transmitted with it, including replies and
> forwarded copies subsequently transmitted, contains information which may
> be confidential and which may also be legally privileged. The content of
> this e-mail is for the exclusive use of the individual(s) or entity named
> above, if you are not the intended recipient, please note that any form of
> distribution, copying or use of this e-mail or the information in it is
> strictly prohibited and may be unlawful. If you have received this e-mail
> in error please reply to the sender at Structured Communications Ltd and
> then delete the e-mail immediately. Any views or opinions presented are
> solely those of the author, and may not necessarily represent those of
> Structured Communications Ltd. Structured Communications Ltd is registered
> in England and Wales, registered no 07667451, and its registered office is Unit
> 8 Foundry Court, Foundry Lane, Horsham, West Sussex RH13 5PY
> <https://www.google.com/maps/search/Unit+8+Foundry+Court,+Foundry+Lane,+Horsham,+West+Sussex%0D%0A+RH13+5PY?entry=gmail&source=g>.
> Registered VAT number GB116507336.
>
>
>
>
>
> *From:* Marco Capetta via Spce-user <spce-user at lists.sipwise.com>
> *Sent:* 17 October 2024 17:32
> *To:* spce-user at lists.sipwise.com
> *Subject:* Re: [Spce-user] [EXTERNAL] Re: mr12.5.1 - default
> administrator can't edit peering servers
>
>
>
> Hi again,
>
>
>
> can you please specify on which exact NGCP version this issue is happening?
>
> And also if you use API, the NEW Admin UI or the OLD Admin UI?
>
>
> I have just installed mr12.5.1 from the ISO and I'm able to create domains
> and peering using the default "administrator".
>
>
>
> Thank you
>
> Marco
>
>
>
> On Thu, Oct 17, 2024 at 2:30?PM Marco Capetta <mcapetta at sipwise.com>
> wrote:
>
> Hi,
>
>
>
> thank you for the report.
>
>
>
> We opened an internal ticket with number MT#61306 to check the issue.
>
>
>
> Thank you
>
> Marco
>
>
>
> On Thu, Oct 17, 2024 at 1:38?PM Jon Bonilla (Manwe) <manwe at sipdoc.net>
> wrote:
>
> El Thu, 17 Oct 2024 16:34:14 +0800
> Walter Klomp via Spce-user <spce-user at lists.sipwise.com> escribi?:
>
> > Hi,
> >
> > The default administrator from an install from ISO is only allowed to do
> > reseller stuff, so no domain and peering server.  I have to go into the
> > database to change the role to "is system" to be able to manipulate the
> > peering servers.
> >
>
> I did a 10.5 install yesterday and same happened to me.
>
>
>
> --
> PekePBX, the multitenant PBX solution
> https://pekepbx.com
>
>
> --
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/mailman/listinfo/spce-user_lists.sipwise.com
>
> --
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/mailman/listinfo/spce-user_lists.sipwise.com
>

-- 
The contents of this email and any attachments are confidential and may 
also be privileged. You must not disseminate the contents of this email and 
any attachments without permission of the sender. If you have received this 
email by mistake, please delete all copies and inform the sender 
immediately. You may refer to our company's Privacy Policy here 
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