[Spce-user] Force CLI

Fri May 10 04:45:12 EDT 2013

Hi all,

here're some sip traces.

1.1.1.1 -> Asterisk Box
2.2.2.2 -> SPCE
3.3.3.3 -> SIP PEER

So I've registered 9030 extension on the Asterisk box. Made a call to an
external number (98769876), goes to 2.2.2.2 which is the SPCE.
2.2.2.2 SPCE goes to 3.3.3.3 which is the sip peer.

problem is ... even though user_cli or enum is set in the subscriber
details, that setting is not getting pulled through when the INVITE goes to
3.3.3.3 (sip peer). The 9030 still goes through and replaces whatever is
setup in the subscriber.

Can someone advise what can be done to rectify this or troubleshoot?

-----

U 2013/05/10 18:35:32.796564 1.1.1.1:50082 -> 2.2.2.2:5060
INVITE sip:98769876 at 2.2.2.2;cpd=on SIP/2.0.
Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK6459469e;rport.
From: "9030" <sip:9030 at 2.2.2.2>;tag=as4d29eab1.
To: <sip:98769876 at 2.2.2.2;cpd=on>.
Contact: <sip:9030 at 1.1.1.1>.
Call-ID: 10e674be2049959f12e51f5d1888106f at 2.2.2.2.
CSeq: 103 INVITE.
User-Agent: Asterisk PBX.
Max-Forwards: 70.
Remote-Party-ID: "9030" <sip:9030 at 2.2.2.2>;privacy=off;screen=no.
Proxy-Authorization: Digest username="61390207910", realm="2.2.2.2",
algorithm=MD5, uri="sip:98769876 at 2.2.2.2;cpd=on",
nonce="UYyygFGMsVSyU+TXii017pt2ZOlCur69",
response="d7b6d2581b0aa25ea902750b3325d6fa".
Date: Fri, 10 May 2013 08:35:34 GMT.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO.
Supported: replaces.
Content-Type: application/sdp.
Content-Length: 234.
.
v=0.
o=root 6108 6109 IN IP4 1.1.1.1.
s=session.
c=IN IP4 1.1.1.1.
t=0 0.
m=audio 38840 RTP/AVP 18 101.
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=sendrecv.

U 2013/05/10 18:35:32.814676 2.2.2.2:5060 -> 3.3.3.3:5060
INVITE sip:6198769876 at 3.3.3.3:5060;transport=udp SIP/2.0.
Max-Forwards: 10.
Record-Route:
<sip:2.2.2.2;r2=on;lr=on;ftag=244CC301-518CB154000C6AC8-062EA700;ngcplb=yes>.
Record-Route:
<sip:127.0.0.1;r2=on;lr=on;ftag=244CC301-518CB154000C6AC8-062EA700;ngcplb=yes>.
Via: SIP/2.0/UDP
2.2.2.2;branch=z9hG4bK3fe2.7771d74fcbc179eb5ca7b9068156b692.0.
Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bKNMqJPaAN;rport=5080.
From: <sip:9030 at 2.2.2.2>;tag=244CC301-518CB154000C6AC8-062EA700.
To: <sip:6198769876 at 3.3.3.3>.
CSeq: 10 INVITE.
Call-ID: 10e674be2049959f12e51f5d1888106f at 2.2.2.2_b2b-1.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO.
Supported: replaces.
P-Asserted-Identity: <sip:9030 at 2.2.2.2>.
Content-Type: application/sdp.
Content-Length: 250.
Contact: <sip:ngcp-lb at 2.2.2.2:5060;ngcpct='sip:127.0.0.1:5080'>.
.
v=0.
o=root 6108 6109 IN IP4 2.2.2.2.
s=session.
c=IN IP4 2.2.2.2.
t=0 0.
m=audio 32874 RTP/AVP 18 101.
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=sendrecv.
a=rtcp:32875.

On Fri, May 10, 2013 at 1:09 AM, Martin Wong <
martin.wong at binaryelements.com.au> wrote:

> Hi Kevin,
>
> unless I've stuffed something up... it's not working..
>
> Have set user_cli as well as the E164 as well. From the sip trace, I see
> that from the SPCE to the provider, the INVITE is as follows...
>
> 123.123.123.123 is the SPCE, 213.213.213.213 the provider.
>
> As you can see, somehow the From: header has 9030 which is the extension
> number of an Asterisk system. Therefore 9030 is not a valid number and thus
> the provider puts it as anonymous. I can't seem to force the CLI.
>
> Any ideas what's happening?
>
> U 2013/05/10 01:05:03.335037 123.123.123.123:5060 -> 213.213.213.213:5060
> INVITE sip:123456789 at 213.213.213.213:5060;transport=udp SIP/2.0.
> Max-Forwards: 10.
> Record-Route:
> <sip:123.123.123.123;r2=on;lr=on;ftag=3686E06C-518BBB1F000519FA-062EA700;ngcplb=yes>.
> Record-Route:
> <sip:127.0.0.1;r2=on;lr=on;ftag=3686E06C-518BBB1F000519FA-062EA700;ngcplb=yes>.
> Via: SIP/2.0/UDP
> 123.123.123.123;branch=z9hG4bK1bf6.45addc0c691ddcb65ab53185af6baca1.0.
> Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bKOqRJLadA;rport=5080.
> From: <sip:9030 at 123.123.123.123>;tag=3686E06C-518BBB1F000519FA-062EA700.
> To: <sip:123456789 at 213.213.213.213>.
> CSeq: 10 INVITE.
> Call-ID: 243cdb514476fa870cabdef15e9c406f at 123.123.123.123_b2b-1.
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO.
> Supported: replaces.
> P-Asserted-Identity: <sip:9030 at 123.123.123.123>.
> Content-Type: application/sdp.
> Content-Length: 252.
> Contact: <sip:ngcp-lb at 123.123.123.123:5060;ngcpct='sip:127.0.0.1:5080'>.
> .
> v=0.
> o=root 13171 13172 IN IP4 123.123.123.123.
> s=session.
> c=IN IP4 123.123.123.123.
> t=0 0.
> m=audio 32354 RTP/AVP 18 101.
> 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=sendrecv.
> a=rtcp:32355.
>
>
>
> On Fri, May 10, 2013 at 12:21 AM, Kevin Masse <kmasse at questblue.com>wrote:
>
>> Good morning, if you are looking to force the outbound caller ID of the
>> subscriber make sure you have the DID in the e.164 number field.****
>>
>> That should force the outbound caller ID to be that number.****
>>
>> ** **
>>
>> For clients that have multiple outbound caller ID leave out the e.164
>> subscriber number and place all the numbers in the alias area instead.***
>> *
>>
>> ** **
>>
>> Kevin****
>>
>> ** **
>>
>> ** **
>>
>> *From:* spce-user-bounces at lists.sipwise.com [mailto:
>> spce-user-bounces at lists.sipwise.com] *On Behalf Of *Barry Flanagan
>> *Sent:* Thursday, May 09, 2013 10:18 AM
>> *To:* Martin Wong
>> *Cc:* spce-user at lists.sipwise.com
>> *Subject:* Re: [Spce-user] Force CLI****
>>
>> ** **
>>
>> On 9 May 2013 13:30, Martin Wong <martin.wong at binaryelements.com.au>
>> wrote:****
>>
>> Hi,****
>>
>> ** **
>>
>> I am testing a few subscribers. ****
>>
>> ** **
>>
>> One is from a normal sip phone ... the fromuser at xx.xx.xx.xx is the
>> correct CLI set, therefore the caller ID is correctly displayed at the end
>> point device as it goes through the downstream provider.****
>>
>> ** **
>>
>> The other is an asterisk box. Even though the CLI is set in the SPCE
>> portal, it comes up as something else in the fromuser at xx.xx.xx.xx****
>>
>> ** **
>>
>> I would like to force the CLI no matter what the asterisk box is giving
>> out.****
>>
>> ** **
>>
>> ** **
>>
>> Hi ****
>>
>> ** **
>>
>> I think you want to set user_cli in the subscriber's preferences:****
>>
>> ** **
>>
>> user_cli: SIP username (the localpart of the whole SIP URI, eg., "user"
>> of SIP URI "user at example.com"). "user-provided calling line
>> identification" - specifies the SIP username that is used for outgoing
>> calls. If set, this is put in the SIP "From" header (as user-provided
>> calling number) if a client sends a CLI which is not allowed by
>> "allowed_clis" or if "allowed_clis" is not set.****
>>
>> ** **
>>
>> ** **
>>
>> Hope this helps.****
>>
>> ** **
>>
>> -Barry****
>>
>> ** **
>>
>
>
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