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Beginner

outgoing call not work on SIP trunk

Hi everybody
I need help to make calls via my SIP trunk.
There is no problem with incoming calls
But outgoing calls fail
I have an ISR4321 with CME 12.6
I really need help.

here is the debug message.
debug ccsip messages

 

Jun 22 20:57:03.356: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:22548010@192.168.20.254;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.20.1:5060;branch=z9hG4bK63074bd8
From: "KASSIME CISSE" <sip:802@192.168.20.254>;tag=cc70ed342a760047600c4647-23270c5c
To: <sip:22548010@192.168.20.254>
Call-ID: cc70ed34-2a76001f-7985b008-231cda15@192.168.20.1
Max-Forwards: 70
Session-ID: 3ffabd7700105000a000cc70ed342a76;remote=00000000000000000000000000000000
Date: Sat, 22 Jun 2019 20:57:02 GMT
CSeq: 101 INVITE
User-Agent: Cisco-CP8845/12.1.1
Contact: <sip:259FF-1495@192.168.20.1:5060;transport=udp>;+u.sip!devicename.ccm.cisco.com="SEPCC70ED342A76";video
Expires: 180
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE,INFO
Remote-Party-ID: "KASSIME CISSE" <sip:802@192.168.20.254>;party=calling;id-type=subscriber;privacy=off;screen=yes
Supported: replaces,join,sdp-anat,norefersub,resource-priority,extended-refer,X-cisco-callinfo,X-cisco-serviceuri,X-cisco-escapecodes,X-cisco-service-control,X-cisco-srtp-fallback,X-cisco-monrec,X-cisco-config,X-cisco-sis-7.0.0,X-cisco-xsi-8.5.1
Allow-Events: kpml,dialog
Recv-Info: conference
Recv-Info: x-cisco-conference
Content-Length: 1172
Content-Type: application/sdp
Content-Disposition: session;handling=optional

v=0
o=Cisco-SIPUA 6982 0 IN IP4 192.168.20.1
s=SIP Call
b=AS:4064
t=0 0
m=audio 31292 RTP/AVP 8 0 116 18 101
c=IN IP4 192.168.20.1
b=TIAS:64000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:116 iLBC/8000
a=fmtp:116 mode=20
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
m=video 25376 RTP/AVP 100 126 97
c=IN IP4 192.168.20.1
b=TIAS:4000000
a=rtpmap:100 H264/90000
a=fmtp:100 profile-level-id=640C16;packetization-mode=1;level-asymmetry-allowed=1;max-mbps=108000;max-fs=3600;max-rcmd-nalu-size=256000
a=imageattr:* recv [x=800,y=480,q=0.60] [x=1280,y=720,q=0.50]
a=rtpmap:126 H264/90000
a=fmtp:126 profile-level-id=428016;packetization-mode=1;level-asymmetry-allowed=1;max-mbps=108000;max-fs=3600;max-rcmd-nalu-size=256000
a=imageattr:* recv [x=800,y=480,q=0.60] [x=1280,y=720,q=0.50]
a=rtpmap:97 H264/90000
a=fmtp:97 profile-level-id=428016;packetization-mode=0;level-asymmetry-allowed=1;max-mbps=108000;max-fs=3600;max-rcmd-nalu-size=256000
a=imageattr:* recv [x=800,y=480,q=0.60] [x=1280,y=720,q=0.50]
a=rtcp-fb:* nack pli
a=rtcp-fb:* ccm fir
a=rtcp-fb:* ccm tmmbr
a=recvonly

Jun 22 20:57:03.361: //336/1EF731E18186/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.20.1:5060;branch=z9hG4bK63074bd8
From: "KASSIME CISSE" <sip:802@192.168.20.254>;tag=cc70ed342a760047600c4647-23270c5c
To: <sip:22548010@192.168.20.254>
Date: Sat, 22 Jun 2019 20:57:03 GMT
Call-ID: cc70ed34-2a76001f-7985b008-231cda15@192.168.20.1
CSeq: 101 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-16.11.1a
Session-ID: 00000000000000000000000000000000;remote=3ffabd7700105000a000cc70ed342a76
Content-Length: 0


Jun 22 20:57:03.376: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
NOTIFY sip:22548010@192.168.20.254:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.20.1:5060;branch=z9hG4bK2d6b333a
To: <sip:22548010@192.168.20.254>;tag=BA6B44-1DD4
From: <sip:802@192.168.20.1>;tag=cc70ed342a7600485f9319d1-6c4501c7
Call-ID: 1EF7A6B3-946711E9-818BC64C-E539A4E5@192.168.20.254
Session-ID: 3e48838f00105000a000cc70ed342a76;remote=00000000000000000000000000000000
Date: Sat, 22 Jun 2019 20:57:02 GMT
CSeq: 1000 NOTIFY
Event: kpml
Subscription-State: active; expires=7200
Max-Forwards: 70
Contact: <sip:259FF-1495@192.168.20.1:5060;transport=udp>;+u.sip!devicename.ccm.cisco.com="SEPCC70ED342A76"
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE
Content-Length: 0


Jun 22 20:57:13.385: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
NOTIFY sip:22548010@192.168.20.254:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.20.1:5060;branch=z9hG4bK060e5ae0
To: <sip:22548010@192.168.20.254>;tag=BA6B44-1DD4
From: <sip:802@192.168.20.1>;tag=cc70ed342a7600485f9319d1-6c4501c7
Call-ID: 1EF7A6B3-946711E9-818BC64C-E539A4E5@192.168.20.254
Session-ID: 3e48838f00105000a000cc70ed342a76;remote=00000000000000000000000000000000
Date: Sat, 22 Jun 2019 20:57:12 GMT
CSeq: 1001 NOTIFY
Event: kpml
Subscription-State: terminated; reason=timeout
Max-Forwards: 70
Contact: <sip:259FF-1495@192.168.20.1:5060;transport=udp>;+u.sip!devicename.ccm.cisco.com="SEPCC70ED342A76"
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE
Content-Length: 218
Content-Type: application/kpml-response+xml
Content-Disposition: session;handling=required

<?xml version="1.0" encoding="UTF-8"?>
<kpml-response xmlns="urn:ietf:params:xml:ns:kpml-response" version="1.0" code="487" text="Subscription Exp" suppressed="false" forced_flush="false" digits="" tag="Backspace OK"/>
Jun 22 20:57:13.386: //338/1EF731E18186/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:22548010@obusiness.ci SIP/2.0
Via: SIP/2.0/UDP 192.168.1.254:5060;branch=z9hG4bK1AC194A
From: "KASSIME CISSE" <sip:21598453@obusiness.ci>;tag=BA926B-9B6
To: <sip:22548010@obusiness.ci>
Date: Sat, 22 Jun 2019 20:57:13 GMT
Call-ID: 24F02042-946711E9-818DC64C-E539A4E5@192.168.1.254
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 0519516641-2489782761-2173093452-3845760229
User-Agent: Cisco-SIPGateway/IOS-16.11.1a
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1561237033
Contact: <sip:21598453@192.168.1.254:5060>
Expires: 360
Allow-Events: telephone-event
Max-Forwards: 69
Session-ID: 3ffabd7700105000a000cc70ed342a76;remote=00000000000000000000000000000000
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 726

v=0
o=CiscoSystemsSIP-GW-UserAgent 527 5959 IN IP4 192.168.1.254
s=SIP Call
c=IN IP4 192.168.1.254
t=0 0
m=audio 8312 RTP/AVP 8 0 18 101
c=IN IP4 192.168.1.254
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/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
m=video 8314 RTP/AVP 98 98 99
c=IN IP4 192.168.1.254
b=TIAS:4000000
a=rtpmap:98 H264/90000
a=fmtp:98 profile-level-id=640C16;packetization-mode=1;max-mbps=108000;max-fs=3600
a=rtpmap:98 H264/90000
a=fmtp:98 profile-level-id=428016;packetization-mode=1;max-mbps=108000;max-fs=3600
a=rtpmap:99 H264/90000
a=fmtp:99 profile-level-id=428016;packetization-mode=0;max-mbps=108000;max-fs=3600
a=recvonly

Jun 22 20:57:13.396: //338/1EF731E18186/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.254:5060;branch=z9hG4bK1AC194A;received=160.120.165.64
Call-ID: 24F02042-946711E9-818DC64C-E539A4E5@192.168.1.254
From: "KASSIME CISSE"<sip:21598453@obusiness.ci>;tag=BA926B-9B6
To: <sip:22548010@obusiness.ci>
CSeq: 101 INVITE
Content-Length: 0


Jun 22 20:57:13.397: //338/1EF731E18186/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 192.168.1.254:5060;branch=z9hG4bK1AC194A
Call-ID: 24F02042-946711E9-818DC64C-E539A4E5@192.168.1.254
From: "KASSIME CISSE"<sip:21598453@obusiness.ci>;tag=BA926B-9B6
To: <sip:22548010@obusiness.ci>;tag=sbc05111q7dq440
CSeq: 101 INVITE
Content-Length: 0


Jun 22 20:57:13.399: //338/1EF731E18186/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:22548010@obusiness.ci SIP/2.0
Via: SIP/2.0/UDP 192.168.1.254:5060;branch=z9hG4bK1AC194A
From: "KASSIME CISSE" <sip:21598453@obusiness.ci>;tag=BA926B-9B6
To: <sip:22548010@obusiness.ci>;tag=sbc05111q7dq440
Date: Sat, 22 Jun 2019 20:57:13 GMT
Call-ID: 24F02042-946711E9-818DC64C-E539A4E5@192.168.1.254
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Session-ID: ;remote=
Content-Length: 0


Jun 22 20:57:13.400: //336/1EF731E18186/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 192.168.20.1:5060;branch=z9hG4bK63074bd8
From: "KASSIME CISSE" <sip:802@192.168.20.254>;tag=cc70ed342a760047600c4647-23270c5c
To: <sip:22548010@192.168.20.254>;tag=BA927A-1234
Date: Sat, 22 Jun 2019 20:57:03 GMT
Call-ID: cc70ed34-2a76001f-7985b008-231cda15@192.168.20.1
CSeq: 101 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-16.11.1a
Reason: Q.850;cause=57
Session-ID: 00000000000000000000000000000000;remote=3ffabd7700105000a000cc70ed342a76
Content-Length: 0


Jun 22 20:57:13.403: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:22548010@192.168.20.254;user=phone SIP/2.0

Via: SIP/2.0/UDP 192.168.20.1:5060;branch=z9hG4bK63074bd8
From: "KASSIME CISSE" <sip:802@192.168.20.254>;tag=cc70ed342a760047600c4647-23270c5c
To: <sip:22548010@192.168.20.254>;tag=BA927A-1234
Call-ID: cc70ed34-2a76001f-7985b008-231cda15@192.168.20.1
Session-ID: 3ffabd7700105000a000cc70ed342a76;remote=00000000000000000000000000000000
Max-Forwards: 70
Date: Sat, 22 Jun 2019 20:57:12 GMT
CSeq: 101 ACK
Content-Length: 0

1 ACCEPTED SOLUTION

Accepted Solutions
Beginner

Re: outgoing call not work on SIP trunk

Hi Florentgoue,

 

By saying calls, do you refer audio only calls or video Calls ?

Also, does incoming calls means Audio as well as Video Calls ?

 

As per the SIP debugs, you are attempting to place a video call:

m=audio 8312 RTP/AVP 8 0 18 101
c=IN IP4 192.168.1.254
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/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
m=video 8314 RTP/AVP 98 98 99
c=IN IP4 192.168.1.254
b=TIAS:4000000
a=rtpmap:98 H264/90000
a=fmtp:98 profile-level-id=640C16;packetization-mode=1;max-mbps=108000;max-fs=3600
a=rtpmap:98 H264/90000
a=fmtp:98 profile-level-id=428016;packetization-mode=1;max-mbps=108000;max-fs=3600
a=rtpmap:99 H264/90000
a=fmtp:99 profile-level-id=428016;packetization-mode=0;max-mbps=108000;max-fs=3600
a=recvonly

 

Please ensure that your ITSP circuit / trunk is provisioned for video calls; else video calls will always fail if ITSP circuit is just provisioned for Audio only calls.

 

Did you tried to place an audio only call ? If not, then please attempt to place one to check if it succeeds.

 

HTH

Rishabh

6 REPLIES 6
Frequent Contributor

Re: outgoing call not work on SIP trunk

Hi,

You get a 403 forbidden, so the provide receives the INVITE, but does not allow it

Contact your provider and ask why

 

JH

Rising star

Re: outgoing call not work on SIP trunk

Hi,

In the logs, I see Reason: Q.850;cause=57 in forbidden message. This is mainly related to codec mismatch or calling numbers allowed on SIP Trunk or Username and Password configuration under sip-ua. Make sure you have all the settings configured correctly. Also, please post your complete CME configuration. 

 

 

Please rate helpful posts and if applicable mark "Accept as a Solution".
Thanks, Vaijanath S.
Highlighted
Rising star

Re: outgoing call not work on SIP trunk

You need to check with the Provider for more info on the error as ISP is dropping the call.

*** Please rate helpful post; Mark "Accept as a Solution" if applicable

Thanks,
Haris

Re: outgoing call not work on SIP trunk

Since you are receiving "SIP/2.0 403 Forbidden", you should contact your Provider and cross-check that you are using the correct/same codec.

Please always Rate Helpful Posts and/or Mark Solutions as Accepted
Beginner

Re: outgoing call not work on SIP trunk

Hi Florentgoue,

 

By saying calls, do you refer audio only calls or video Calls ?

Also, does incoming calls means Audio as well as Video Calls ?

 

As per the SIP debugs, you are attempting to place a video call:

m=audio 8312 RTP/AVP 8 0 18 101
c=IN IP4 192.168.1.254
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/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
m=video 8314 RTP/AVP 98 98 99
c=IN IP4 192.168.1.254
b=TIAS:4000000
a=rtpmap:98 H264/90000
a=fmtp:98 profile-level-id=640C16;packetization-mode=1;max-mbps=108000;max-fs=3600
a=rtpmap:98 H264/90000
a=fmtp:98 profile-level-id=428016;packetization-mode=1;max-mbps=108000;max-fs=3600
a=rtpmap:99 H264/90000
a=fmtp:99 profile-level-id=428016;packetization-mode=0;max-mbps=108000;max-fs=3600
a=recvonly

 

Please ensure that your ITSP circuit / trunk is provisioned for video calls; else video calls will always fail if ITSP circuit is just provisioned for Audio only calls.

 

Did you tried to place an audio only call ? If not, then please attempt to place one to check if it succeeds.

 

HTH

Rishabh

Beginner

Re: outgoing call not work on SIP trunk

Hello everyone, the problem has been fixed.

1- the provider does not allow Video calls while I use cp8845 phones that make video calls by default.

solution:

force the audio call on the SIP trunk In the dial-peer configuration mode       

voice-class sip audio forced

https://community.cisco.com/t5/ip-telephony-and-phones/cisco-8845-call-failure-over-sip-trunk-when-video-enable/td-p/3779399

 

2- The provider does not accept the name and internal number of the caller in the request.

solution:

- translation internal number in DID number

- normalization Invite request : change the caller name by his DID.

 

Thank you

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