[Spce-user] PSTN GW can register SIPWISE server but GW peerstatus always busy.

Tue Mar 26 21:15:08 EDT 2013

I have already a PSTN Gateway it is not a SIP trunk but it can convert SIP to PSTN.
And I the PSTN Gateway IP set private network. i.e. 192.168.102.254
In "SIP Peering" section of admin panel, I made "test group" and there have "test-gw-1" server it configured same like following;
-------------------
Name : test-gw-1
IP Address : 192.168.102.254 ->it is same IP address with PSTN Gateway because I want to use this PSTN.
Hostname : " "(empty)
Port : UDP
Weight : 1
-------------------
Also Peering Rules values it configured all empty. Callee Prefix, Callee Pattern, Caller Pattern, Description.
And as I opened the log file Kamailio-proxy.log, there are so many logs made at just one call.
So I attached two log files one is a call from SIP phone to SIP phone as locally and the other one is a call from SIP Phone to PSTN Gateway.
I don't know exactly what is point of problem in the log files why PSTN status is always busy.
But as I had inference, it is point why didn't call from SIP phone to PSTN Gateway and it status is always busy.
------------------
Mar 27 01:21:25 spce /usr/sbin/kamailio[1994]: INFO: <script>: No matching rewrite rules for '0112796171' found - R=sip:0112796171 at 192.168.102.254:5060;transport=udp ID=d8bb68a825c52d88dc2813b4e77ace43 at 192.168.102.60
Mar 27 01:21:25 spce /usr/sbin/kamailio[1994]: INFO: <script>: Setting P-Called-Party-ID '<sip:0112796171 at 192.168.102.254>' - R=sip:0112796171 at 192.168.102.254:5060;transport=udp ID=d8bb68a825c52d88dc2813b4e77ace43 at 192.168.102.60
Mar 27 01:21:25 spce /usr/sbin/kamailio[1994]: INFO: <script>: Writing sbc parameters  ;aleg_sst_enable=no;sst_enable=no - R=sip:0112796171 at 192.168.102.254:5060;transport=udp ID=d8bb68a825c52d88dc2813b4e77ace43 at 192.168.102.60
Mar 27 01:21:25 spce /usr/sbin/kamailio[1994]: INFO: <script>: Setting 'sip:192.168.102.254:5060' taken from R-URI as next hop after lb for PSTN call - R=sip:0112796171 at 192.168.102.254:5060;transport=udp ID=d8bb68a825c52d88dc2813b4e77ace43 at 192.168.102.60
Mar 27 01:21:25 spce /usr/sbin/kamailio[1994]: INFO: <script>: Appending P-D-URI 'sip:127.0.0.1:5060;received='sip:192.168.102.254:5060;transport=udp'' - R=sip:0112796171 at 192.168.102.254:5060;transport=udp ID=d8bb68a825c52d88dc2813b4e77ace43 at 192.168.102.60
Mar 27 01:21:25 spce /usr/sbin/kamailio[1994]: INFO: <script>: Forcing request via B2BUA 'sip:127.0.0.1:5080' - R=sip:0112796171 at 192.168.102.254:5060;transport=udp ID=d8bb68a825c52d88dc2813b4e77ace43 at 192.168.102.60
Mar 27 01:21:25 spce /usr/sbin/kamailio[1994]: INFO: <script>: Setting P-Caller-UUID to '0cbfd478-f7eb-445d-9f3c-6061b0f96eb2' - R=sip:0112796171 at 192.168.102.254:5060;transport=udp ID=d8bb68a825c52d88dc2813b4e77ace43 at 192.168.102.60
Mar 27 01:21:25 spce /usr/sbin/kamailio[1994]: INFO: <script>: Setting P-Callee-UUID to '0' - R=sip:0112796171 at 192.168.102.254:5060;transport=udp ID=d8bb68a825c52d88dc2813b4e77ace43 at 192.168.102.60
Mar 27 01:21:25 spce /usr/sbin/kamailio[1994]: INFO: <script>: Request leaving server, D-URI='sip:127.0.0.1:5080' - R=sip:0112796171 at 192.168.102.254:5060;transport=udp ID=d8bb68a825c52d88dc2813b4e77ace43 at 192.168.102.60
Mar 27 01:21:25 spce /usr/sbin/kamailio[1995]: INFO: <script>: NAT-Reply - S=100 - Connecting M=INVITE IP=192.168.102.60:5060 (127.0.0.1:5080) ID=d8bb68a825c52d88dc2813b4e77ace43 at 192.168.102.60
Mar 27 01:21:25 spce /usr/sbin/kamailio[1992]: INFO: <script>: NAT-Reply - S=486 - Busy Here M=INVITE IP=192.168.102.60:5060 (127.0.0.1:5080) ID=d8bb68a825c52d88dc2813b4e77ace43 at 192.168.102.60
Mar 27 01:21:25 spce /usr/sbin/kamailio[1992]: INFO: <script>: Failure route for PSTN call - R=sip:0112796171 at sip.baekdoo.local:5060;transport=udp ID=d8bb68a825c52d88dc2813b4e77ace43 at 192.168.102.60
------------------------

So I don't know exactly but the SIP Server tried to call the PSTN Gateway as the SIPWISE server IP 127.0.0.1:5080 as local SIP it referenced following a log in upper logs.
-------------------------------
Forcing request via B2BUA 'sip:127.0.0.1:5080' - R=sip:0112796171 at 192.168.102.254:5060;transport=udp
-------------------------------

But SIPWISE server couldn't reached to PSTN gateway because IP networks are different each others.
So SIPWISE can't reached the network 192.168.102.254:5060 that is only my opinion.

Please let me know if you find what was problem from connect with from SIP phone to PSTN network.

David Ho

----------------------원본메일-----------------------
보낸사람 : Andreas Granig <agranig at sipwise.com>
받는사람: spce-user at lists.sipwise.com
보낸날짜: 2013-03-26 22:09:50 GMT +0900 (Asia/Seoul)
제목: Re: [Spce-user] PSTN GW can register SIPWISE server but GW peerstatus always busy.

Hi,

On 03/26/2013 11:52 AM, 허윤 wrote:
> So now I trying to solve it could communicate with PSTN.
>
> If you were me, how to find the problem that couldn't communicate with PSTN?
>
> If you were me, where to configure in SIPWISE?

Well, do you already have a SIP trunk to some peering provider? If yes, 
did you configure it in the "SIP Peering" section of the admin panel 
(check the handbook for the how-to)? If yes, what did you put there and 
where, and what does the kamailio-proxy.log say for a call towards this 
peer?

Andreas

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