Incoming calls comms using wrong SIP

[assers001]

Hello

I have two SIP trunks from the same SIP provider. and i am facing issue with incoming calls as they targeting the wrong SIP link .

Noted in sip/sdp massage from my VGW to ISP contains the wrong SIP address (c=IN) ,dial-peer and sip bind  configuration confirmed .

Can any one explain whey this happen  ? and how we can fix it ?

BR

Ahmed

[Roger Kallberg]

What are you using to match the inbound dial peer?

[assers001]

translation-profile incoming
incoming called-number
incoming uri

[Roger Kallberg]

A translation-profile isn’t a valid match for a dial peer, it’s something used based on another match criteria. Can you please share your dial peer configuration and also the voice class used for the URI match?

[Roger Kallberg]

Also if you have the inbound invite for both SIP trunks that would be of help.

[assers001]

hello

I think due to having 2 SIP from same provider cause conflict between them . I configure sip 2 with the same configuration of SIP 1 the only different is translation-profile .

dial-peer voice 301 voip
description **Incoming Call from SIP 3**
translation-profile incoming IN-ISP3
session protocol sipv2
incoming called-number 0112......
incoming uri to ISPDNS3
voice-class codec 3
voice-class sip dtmf-relay force rtp-nte
voice-class sip bind control source-interface GigabitEthernet0/0/x
voice-class sip bind media source-interface GigabitEthernet0/0/x
no vad

voice class uri IN-ISP3 sip
host dns:isp.ss

------------------------------------------------------------
INVITE Sip 1
------------------------------------------------------------

eceived:
INVITE sip:+923112233445@sbc-trunking.isp.ss SIP/2.0
Via: SIP/2.0/UDP 192.168.63.20:5060;branch=z9hG4bK5386uu2020scfuc27mn0.1
From: "0512345678"<sip:0512345678@10.0.0.20;user=phone>;tag=1205736008-1736616608309-
To: "923112233445 923112233445"<sip:+923112233445@isp.ss>
Call-ID: BW2030083091101251603699408@10.0.0.20
CSeq: 724910363 INVITE
Contact: <sip:192.168.63.20:5060;transport=udp>
Supported: 100rel
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
Recv-Info: x-broadworks-client-session-info
Accept: application/media_control+xml,application/sdp,multipart/mixed
Max-Forwards: 69
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 429
Route: <sip:+923112233445;tgrp=170_grpMain_tgrp;trunk-context=isp.ss@ss-kusv:5060;lr>
v=0
o=BroadWorks 45221141 1 IN IP4 192.168.63.52
s=-
c=IN IP4 192.168.63.52
t=0 0
m=audio 22620 RTP/AVP 108 102 8 0 18 116
a=3gOoBTC
a=ptime:20
a=maxptime:20
a=rtpmap:108 AMR/8000
a=fmtp:108 mode-set=0,2,4,7;mode-change-neighbor=1;mode-change-period=2
a=rtpmap:102 AMR/8000
a=fmtp:102 mode-set=0,2,4,7
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:116 telephone-event/8000
a=sendrecv

------------------------------------------------------------
INVITE Sip 2
------------------------------------------------------------

eceived:
INVITE sip:+923112432110@sbc-trunking.isp.ss SIP/2.0
Via: SIP/2.0/UDP 192.168.63.4:5060;branch=z9hG4bKrfsb621010r1h8s37ls0.1
From: "0512345678"<sip:0512345678@10.0.0.20;user=phone>;tag=183506376-1736616634304-
To: "923112432110 923112432110"<sip:+923112432110@isp.ss>
Call-ID: BW203034304110125-363642613@10.0.0.20
CSeq: 724923361 INVITE
Contact: <sip:192.168.63.4:5060;transport=udp>
Supported: 100rel
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
Recv-Info: x-broadworks-client-session-info
Accept: application/media_control+xml,application/sdp,multipart/mixed
Max-Forwards: 69
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 429
Route: <sip:+923112432110;tgrp=760_grpMain_tgrp;trunk-context=isp.ss@sa-760.KUSV:5060;lr>
v=0
o=BroadWorks 45222059 1 IN IP4 192.168.63.36
s=-
c=IN IP4 192.168.63.36
t=0 0
m=audio 24464 RTP/AVP 108 102 8 0 18 116
a=3gOoBTC
a=ptime:20
a=maxptime:20
a=rtpmap:108 AMR/8000
a=fmtp:108 mode-set=0,2,4,7;mode-change-neighbor=1;mode-change-period=2
a=rtpmap:102 AMR/8000
a=fmtp:102 mode-set=0,2,4,7
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:116 telephone-event/8000
a=sendrecv

[Roger Kallberg]

From what I can tell you get different information in the VIA header.

Invite 1
Via: SIP/2.0/UDP 192.168.63.20:5060;branch=z9hG4bK5386uu2020scfuc27mn0.1

Invite 2
Via: SIP/2.0/UDP 192.168.63.4:5060;branch=z9hG4bKrfsb621010r1h8s37ls0.1

Create two voice class uri to match the IP in each of the VIA headers and uses these on each of the specific inbound dial peers to match.

For specific information on this please see this document. Explain Cisco IOS and IOS XE Call Routing

[Roger Kallberg]

Something like this should work.

voice class uri SIP1 sip
 host ipv4:192.168.63.20
!
voice class uri SIP2 sip
 host ipv4:192.168.63.4
!
dial-peer voice 301 voip
 no incoming called-number 0112......
 no incoming uri to ISPDNS3
 incoming uri via SIP2
!
Repeate the dial peer configuration for the other inbound DP to use the other voice class uri (SIP1) to match on VIA header information

[assers001]

Hi 

Below are signaling IPs of the same provider , I will test and let you know .

192.168.63.4 192.168.63.20

[Roger Kallberg]

If the connections uses different signaling IPs then the configuration that I suggested should work for you. Please try it out and let us know.

[Roger Kallberg]

Did you get anywhere with this?

[assers001]

HI 

The ISP use the same signaling IPs on both link and same domain . still not tested but i will update you soon .