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polycom2080
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Hi

 

 

I have 2 PBX setup with 4.0.0.3344 (Linux) and when I call from 1st PBX to the 2nd PBX Thu a 3rd party the calls fail

 

Is there a way to change what the PBX should send or a way to get the PBX to except that invite?

 

Below are 3 call traces

 

 

I think that the invite message is not working

 

 

 

 

On the pbx it shows

[5] 2010/04/15 18:02:38: Received incoming call without trunk information and user has not been found

 

Please advise

 

 

FAILED INVITE

 

U 2010/04/15 21:37:44.629126 192.168.1.52:5060 -> 123.123.123.123:5060

INVITE sip:5184441212@123.123.123.123;user=phone SIP/2.0..Record-Route: <si

p:15184441212@66.23.129.250:5060;nat=yes;ftag=1132153492;lr=on>..Via: SIP/2

.0/UDP 66.23.129.250:5060;branch=z9hG4bK5838.1e43bd02.0..v: SIP/2.0/UDP 25.25.25.25:5060;branch=z9hG4bK-b54847acbaf80a8b4f411bc4c398aa8f;rport=506

0..f: <sip:Testnet@66.23.129.250>;tag=1132153492..t: <sip:15184441212@66.

23.129.250;user=phone>..i: 1d7e6a73@pbx..CSeq: 7638 INVITE..Max-Forwards: 1

6..m: <sip:Testnet@45.45.45.45:5060;transport=udp>..Supported: 100rel

, replaces, norefersub..Allow-Events: refer..Allow: INVITE, ACK, CANCEL, BY

E, REFER, PRACK, INFO, UPDATE..Accept: application/sdp..User-Agent: Test-

pbx/4.0.0.3344..Remote-Party-ID: "hello test" <sip:15182323675@66.23.129.25

0;user=phone>;party=calling;screen=yes..c: application/sdp..l: 460....v=0..

o=- 1571734621 1571734621 IN IP4 45.45.45.45..s=-..c=IN IP4 173.203.141

.235..t=0 0..m=audio 59574 RTP/AVP 0 8 9 18 2 3 101..a=rtpmap:0 pcmu/8000..

a=rtpmap:8 pcma/8000..a=rtpmap:9 g722/8000..a=rtpmap:18 g729/8000..a=fmtp:1

8 annexb=no..a=rtpmap:2 g726-32/8000..a=rtpmap:3 gsm/8000..a=rtpmap:101 tel

ephone-event/8000..a=fmtp:101 0-16..a=rtcp-xr:pkt-loss-rle pkt-dup-rle pkt-

rcpt-times rcvr-rtt=sender stat-summary=loss,dup,jitt voip-metrics..a=sendr

ecv..

 

WORKING INVITE FROM PSTN

 

U 2010/04/15 21:44:39.486648 192.168.1.50:5060 -> 123.123.123.123:5060

INVITE sip:5184441212@123.123.123.123 SIP/2.0..Record-Route: <sip:+15184441

212@66.23.129.253:5060;nat=yes;ftag=gK0b6aa1e6;lr=on>..Via: SIP/2.0/UDP 66.

23.129.253:5060;branch=z9hG4bK33f5.ca589696.0..Via: SIP/2.0/UDP :

5060;branch=z9hG4bK0bB38b50b3acf3c2c80..From: "hello test" <sip:+1212344486

6@:5060>;tag=gK0b6aa1e6..To: <sip:+15184441212@66.23.129.253:5060

>..Call-ID: 2014037460_81379626@..CSeq: 22842 INVITE..Max-Forward

s: 16..Allow: INVITE,ACK,CANCEL,BYE,PRACK,UPDATE..Accept: application/sdp,

application/isup, application/dtmf, application/dtmf-relay, multipart/mixe

d..Contact: "hello test" <sip:+15182323675@:5060>..Remote-Party-I

D: "hello test" <sip:+15182323675@:5060>;privacy=off..Supported:

100rel..Content-Length: 302..Content-Disposition: session; handling=option

al..Content-Type: application/sdp....v=0..o=Sonus_UAC 19978 7401 IN IP4 4.5

5.5.163..s=SIP Media Capabilities..c=IN IP4 4.55.5.130..t=0 0..m=audio 1222

2 RTP/AVP 0 8 18 101..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:

18 G729/8000..a=fmtp:18 annexb=no..a=rtpmap:101 telephone-event/8000..a=fmt

p:101 0-15..a=sendrecv..a=maxptime:20..

 

 

WORKING INVITE FROM NV PBX

 

U 2010/04/15 21:44:17.787501 192.168.1.50:5060 -> 123.123.123.123:5060

INVITE sip:5184441212@123.123.123.123 SIP/2.0..Record-Route: <sip:151844412

12@66.23.129.253:5060;nat=yes;ftag=615-zultys--10359999711066_1703381798-10

66;lr=on>..Via: SIP/2.0/UDP 66.23.129.253:5060;branch=z9hG4bK5bb4.5369c037.

0..Via: SIP/2.0/UDP 66.23.136.140:5060;branch=z9hG4bK1703546294-1066..Max-F

orwards: 16..Allow: INVITE,ACK,CANCEL,BYE,REGISTER,OPTIONS,NOTIFY,SUBSCRIBE

,REFER,MESSAGE,PRACK..Zultys-Data: mx_call_id=100.44;..User-Agent: Zultys M

X250 v5.2.10 build 1..From: "7034392702" <sip:5184392702@hq.test.com>;

tag=615-zultys--10359999711066_1703381798-1066..To: sip:15184441212@66.23.1

29.253..Call-ID: 1703381373-1066..CSeq: 2 INVITE..Contact: sip:5184392702@6

6.23.136.140:5060..Content-Type: application/sdp..Supported: 100rel..Conten

t-Length: 268..Remote-Party-ID: <sip:5184392702@test.com>;party=callin

g;screen=no;privacy=off....v=0..o=MX250-5.2.10-1 1271367857 0 IN IP4 66.23.

136.140..s=-..c=IN IP4 66.23.136.140..t=0 0..m=audio 21114 RTP/AVP 0 8 18 1

01..a=rtpmap:0 PCMU/8000/1..a=rtpmap:8 PCMA/8000/1..a=rtpmap:18 G729/8000/1

..a=fmtp:18 annexb=no..a=rtpmap:101 telephone-event/8000..a=ptime:20..

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Is there a way to change what the PBX should send or a way to get the PBX to except that invite?

 

Well the SIP messages look like sphagetti in this port, very hard to read.

 

You should also pay attention to the messages in the log about identifiying trunks. In the cases that you send traffic from one PBX to the other, there are subtle cases when the PBX believes the call does not call from a trunk, but a local user. Especially when the user names overlap that is a common problem.

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