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Replies

CUBE SIP Trunk Outbound calls failing

Nina
Level 1
Level 1

Hello,

 

We can't seem to get the outbound calls to work on our SIP Trunk. The inbound calls have no issue.

Please take a look at the trace routes, need help..



003820: *Sep 27 12:34:02.125: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:915147047479@192.168.22.10:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.248.10:5060;branch=z9hG4bK1548106ad02277
From: <sip:9056711271@192.168.248.10>;tag=13956200~a38f0a9c-d7b3-49cb-a446-3d557da7351f-20514648
To: <sip:915147047479@192.168.22.10>
Date: Wed, 27 Sep 2017 16:31:05 GMT
Call-ID: 40c13000-9cb1d249-d91a6-af8a8c0@192.168.248.10
Supported: 100rel,timer,resource-priority,replaces
Min-SE:  1800
User-Agent: Cisco-CUCM11.5
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Expires: 180
Allow-Events: presence
Supported: X-cisco-srtp-fallback,X-cisco-original-called
Call-Info: <urn:x-cisco-remotecc:callinfo>;x-cisco-video-traffic-class=DESKTOP
Session-ID: 6a53137f00105000a000009e1ede3ddc;remote=00000000000000000000000000000000
Cisco-Guid: 1086402560-0000065536-0000000103-0184068288
Session-Expires:  1800
P-Asserted-Identity: <sip:9056711271@192.168.248.10>
Remote-Party-ID: <sip:9056711271@192.168.248.10>;party=calling;screen=yes;privacy=off
Contact: <sip:9056711271@192.168.248.10:5060>;+u.sip!devicename.ccm.cisco.com="SEP009E1EDE3DDC"
Max-Forwards: 69
Content-Length: 0


003821: *Sep 27 12:34:02.128: //543583/40C130000000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.248.10:5060;branch=z9hG4bK1548106ad02277
From: <sip:9056711271@192.168.248.10>;tag=13956200~a38f0a9c-d7b3-49cb-a446-3d557da7351f-20514648
To: <sip:915147047479@192.168.22.10>
Date: Wed, 27 Sep 2017 12:34:02 GMT
Call-ID: 40c13000-9cb1d249-d91a6-af8a8c0@192.168.248.10
CSeq: 101 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-15.5.3.S4b
Content-Length: 0


003822: *Sep 27 12:34:02.131: //543584/40C130000000/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:15147047479@199.188.188.3:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.22.10:5060;branch=z9hG4bK7D7B380
From: <sip:9056711271@hgtp.com;user=phone>;tag=21B3A136-1F4E
To: <sip:15147047479@siptrunking.bell.ca>
Date: Wed, 27 Sep 2017 12:34:02 GMT
Call-ID: 81234E75-A2D811E7-8E9C8CD2-340EA8A7@hgtp.com
Supported: 100rel,timer,resource-priority,replaces,histinfo,sdp-anat
Min-SE:  1800
Cisco-Guid: 1086402560-0000065536-0000000103-0184068288
User-Agent: Cisco-SIPGateway/IOS-15.5.3.S4b
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1506530042
Contact: <sip:9056711271;tgrp=3268488;trunk-context=siptrunking.bell.ca@192.168.22.10:5060>
History-Info: <sip:15147047479@199.188.188.3:5060>;index=1
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 68
P-Asserted-Identity: <sip:9056711271@hgtp.com;user=phone>
Session-Expires:  1800
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 250

v=0
o=CiscoSystemsSIP-GW-UserAgent 1601 6295 IN IP4 192.168.22.10
s=SIP Call
c=IN IP4 192.168.22.10
t=0 0
m=audio 13110 RTP/AVP 0 101
c=IN IP4 192.168.22.10
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

 

15 Replies 15

The issue was solved it was the customer's phone system.
the CUCM was sending a PAI field in his invite but asterisk didn't not recognize it. It was removed and all worked fine.

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