12-18-2013 03:38 PM - edited 03-16-2019 08:56 PM
All phones are unable to dial a single target number on the PSTN. The symptom is that it rings once and goes fast busy.
The call flow is:
Phone >>> CUCM >>> CUBE >>> Verizon SIP Trunk >>> PSTN >>> Target Number
As seen in the CUBE debug ccsip messages, the CUBE receives a "SIP/2.0 480 Temporarily unavailable" message. debug ccsip messages, dial-peer and voice class information follows:
Received:
INVITE sip:1XXX-XXX-XXXX@10.139.64.52:5060 SIP/2.0
Via: SIP/2.0/TCP 192.168.106.11:5060;branch=z9hG4bK6c4bb65f140c8
From: "" <sip:XXX-XXX-XXXX@192.168.106.11>;tag=4037968~a812c08a-f7f0-43a7-a92c-e5ac2a38867c-65745615
To: <sip:1XXX-XXX-XXXX@10.139.64.52>
Date: Wed, 18 Dec 2013 21:48:27 GMT
Call-ID: 1f068d00-2b21182b-fc6d-b6aa8c0@192.168.106.11
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM8.6
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Expires: 180
Allow-Events: presence, kpml
Supported: X-cisco-srtp-fallback,X-cisco-original-called
Call-Info: <sip:192.168.106.11:5060>;method="NOTIFY;Event=telephone-event;Duration=500"
Cisco-Guid: 0520523008-0000065536-0000067523-0191539392
Session-Expires: 1800
P-Asserted-Identity: "" <sip:XXX-XXX-XXXX@192.168.106.11>
Remote-Party-ID: "" <sip:XXX-XXX-XXXX@192.168.106.11>;party=calling;screen=yes;privacy=off
Contact: <sip:XXX-XXX-XXXX@192.168.106.11:5060;transport=tcp>
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 390
v=0
o=CiscoSystemsCCM-SIP 4037968 1 IN IP4 192.168.106.11
s=SIP Call
c=IN IP4 10.139.64.171
b=TIAS:64000
b=AS:64
t=0 0
m=audio 30688 RTP/AVP 0 8 116 18 101
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
a=ptime:20
a=rtpmap:116 iLBC/8000
a=ptime:20
a=maxptime:60
a=fmtp:116 mode=20
a=rtpmap:18 G729/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
Sent:
INVITE sip:1XXX-XXX-XXXX@10.189.135.17:5073 SIP/2.0
Via: SIP/2.0/UDP 10.139.64.52:5060;branch=z9hG4bK636B313D7
Remote-Party-ID: "" <sip:XXX-XXX-XXXX@10.139.64.52>;party=calling;screen=yes;privacy=off
From: "" <sip:XXX-XXX-XXXX@10.139.64.52>;tag=78FC5414-198D
To: <sip:1XXX-XXX-XXXX@10.189.135.17>
Date: Wed, 18 Dec 2013 21:40:10 GMT
Call-ID: CD993954-676311E3-8734DBF2-34273F6B@10.139.64.52
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 0520523008-0000065536-0000067523-0191539392
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1387402810
Contact: <sip:XXX-XXX-XXXX@10.139.64.52:5060>
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 69
Session-Expires: 1800
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 348
v=0
o=CiscoSystemsSIP-GW-UserAgent 4778 3356 IN IP4 10.139.64.52
s=SIP Call
c=IN IP4 10.139.64.52
t=0 0
m=audio 23372 RTP/AVP 0 8 116 18 101
c=IN IP4 10.139.64.52
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:116 iLBC/8000
a=fmtp:116
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
079120: Dec 18 2013 16:40:10.008: //314738/1F068D000001/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 192.168.106.11:5060;branch=z9hG4bK6c4bb65f140c8
From: "" <sip:XXX-XXX-XXXX@192.168.106.11>;tag=4037968~a812c08a-f7f0-43a7-a92c-e5ac2a38867c-65745615
To: <sip:1XXX-XXX-XXXX@10.139.64.52>
Date: Wed, 18 Dec 2013 21:40:09 GMT
Call-ID: 1f068d00-2b21182b-fc6d-b6aa8c0@192.168.106.11
CSeq: 101 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
079121: Dec 18 2013 16:40:10.080: //314739/1F068D000001/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.139.64.52:5060;branch=z9hG4bK636B313D7
From: "" <sip:XXX-XXX-XXXX@10.139.64.52>;tag=78FC5414-198D
To: <sip:1XXX-XXX-XXXX@10.189.135.17>
Call-ID: CD993954-676311E3-8734DBF2-34273F6B@10.139.64.52
CSeq: 101 INVITE
Timestamp: 1387402810
+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
079122: Dec 18 2013 16:40:11.176: //314739/1F068D000001/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.139.64.52:5060;branch=z9hG4bK636B313D7
From: "" <sip:XXX-XXX-XXXX@10.139.64.52>;tag=78FC5414-198D
To: <sip:1XXX-XXX-XXXX@10.189.135.17>;tag=182903799-1387403308449
Call-ID: CD993954-676311E3-8734DBF2-34273F6B@10.139.64.52
CSeq: 101 INVITE
Timestamp: 1387402810
Supported:
Contact: <sip:1XXX-XXX-XXXX@10.189.135.17:5073;transport=udp>
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
Content-Length: 0
+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
Sent:
SIP/2.0 180 Ringing
Via: SIP/2.0/TCP 192.168.106.11:5060;branch=z9hG4bK6c4bb65f140c8
From: "" <sip:XXX-XXX-XXXX@192.168.106.11>;tag=4037968~a812c08a-f7f0-43a7-a92c-e5ac2a38867c-65745615
To: <sip:1XXX-XXX-XXXX@10.139.64.52>;tag=78FC58A8-1B6B
Date: Wed, 18 Dec 2013 21:40:09 GMT
Call-ID: 1f068d00-2b21182b-fc6d-b6aa8c0@192.168.106.11
CSeq: 101 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Remote-Party-ID: <sip:1XXX-XXX-XXXX@10.139.64.52>;party=called;screen=no;privacy=off
Contact: <sip:1XXX-XXX-XXXX@10.139.64.52:5060;transport=tcp>
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
079128: Dec 18 2013 16:40:12.384: //314739/1F068D000001/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 480 Temporarily unavailable
Via: SIP/2.0/UDP 10.139.64.52:5060;branch=z9hG4bK636B313D7
From: "" <sip:XXX-XXX-XXXX@10.139.64.52>;tag=78FC5414-198D
To: <sip:1XXX-XXX-XXXX@10.189.135.17>;tag=182903799-1387403308449
Call-ID: CD993954-676311E3-8734DBF2-34273F6B@10.139.64.52
CSeq: 101 INVITE
Timestamp: 1387402810
Content-Length: 0
+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
Sent:
SIP/2.0 480 Temporarily Not Available
Via: SIP/2.0/TCP 192.168.106.11:5060;branch=z9hG4bK6c4bb65f140c8
From: "" <sip:XXX-XXX-XXXX@192.168.106.11>;tag=4037968~a812c08a-f7f0-43a7-a92c-e5ac2a38867c-65745615
To: <sip:1XXX-XXX-XXXX@10.139.64.52>;tag=78FC58A8-1B6B
Date: Wed, 18 Dec 2013 21:40:09 GMT
Call-ID: 1f068d00-2b21182b-fc6d-b6aa8c0@192.168.106.11
CSeq: 101 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Reason: Q.850;cause=18
Content-Length: 0
+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
079146: Dec 18 2013 16:40:12.388: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:1XXX-XXX-XXXX@10.189.135.17:5073 SIP/2.0
Via: SIP/2.0/UDP 10.139.64.52:5060;branch=z9hG4bK636B313D7
From: "" <sip:XXX-XXX-XXXX@10.139.64.52>;tag=78FC5414-198D
To: <sip:1XXX-XXX-XXXX@10.189.135.17>;tag=182903799-1387403308449
Date: Wed, 18 Dec 2013 21:40:10 GMT
Call-ID: CD993954-676311E3-8734DBF2-34273F6B@10.139.64.52
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0
+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
079147: Dec 18 2013 16:40:12.404: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:1XXX-XXX-XXXX@10.139.64.52:5060 SIP/2.0
Via: SIP/2.0/TCP 192.168.106.11:5060;branch=z9hG4bK6c4bb65f140c8
From: "" <sip:XXX-XXX-XXXX@192.168.106.11>;tag=4037968~a812c08a-f7f0-43a7-a92c-e5ac2a38867c-65745615
To: <sip:1XXX-XXX-XXXX@10.139.64.52>;tag=78FC58A8-1B6B
Date: Wed, 18 Dec 2013 21:48:27 GMT
Call-ID: 1f068d00-2b21182b-fc6d-b6aa8c0@192.168.106.11
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: presence, kpml
Content-Length: 0
+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
dial-peer voice 9100 voip
description inboubd dial-peer for outgoing calls from CUCM (11D)
preference 1
session protocol sipv2
incoming called-number ^1..........$
voice-class codec 10
dtmf-relay rtp-nte digit-drop
ip qos dscp cs5 media
ip qos dscp cs3 signaling
no vad
outbound DP
dial-peer voice 8100 voip
description outbound dial-peer for outgoing calls to Verizon (11D)
destination-pattern ^1..........$
session protocol sipv2
session target sip-server
voice-class codec 10
voice-class sip dtmf-relay force rtp-nte
voice-class sip early-offer forced
dtmf-relay rtp-nte digit-drop
ip qos dscp cs5 media
ip qos dscp cs3 signaling
no vad
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
voice class codec 1
codec preference 1 g729r8
codec preference 2 g711ulaw
voice class codec 10
codec preference 1 transparent
voice class codec 2
codec preference 1 g711ulaw
codec preference 2 g722-64
Solved! Go to Solution.
12-18-2013 03:51 PM
Open a ticket with Verizon..This is an issue on their side. They are not able to route the call to this number, they need to tell you why
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"opportunity is a haughty goddess who waste no time with those who are unprepared"
12-18-2013 03:51 PM
Open a ticket with Verizon..This is an issue on their side. They are not able to route the call to this number, they need to tell you why
Please rate all useful posts
"opportunity is a haughty goddess who waste no time with those who are unprepared"
12-18-2013 03:53 PM
Thank you! That was my conclusion as well.
Amir
12-19-2013 10:26 AM
Please see the reply from Verizon as follows. Does this make sense to you?
On the invite we are receiving, includes a codec in the SDP that is not supported over our network. Can you please remove this option and retest the call.
Codec
116 iLBC/8000 is not supported over our network and is causing the Long Distance gateway to fail the call
rtpmap:116 iLBC/8000
fmtp:116
12-19-2013 10:44 AM
Yes it does..The INVITE from CUCM has ilbc codec in it..Here is the SDP portion of the INVITE
t=0 0
m=audio 30688 RTP/AVP 0 8 116 18 101
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
a=ptime:20
a=rtpmap:116 iLBC/8000
a=ptime:20
a=maxptime:60
a=fmtp:116 mode=20
a=rtpmap:18 G729/8000
a=ptime:20
You can disable ilbc on CUCM by going to CUCM service parameters, then slect call manager> click on advanced settings and look for "iLBC Codec Enabled" and set it to false.
Please rate all useful posts
"opportunity is a haughty goddess who waste no time with those who are unprepared"
12-19-2013 10:47 AM
Ah - this will require a change management process. Do you see any downside in disabling this codec? What end points are using this codec?
Or better yet - can I disable this at the CUBE "only" so that I can first confirm that this resolves the issue? Thus not disabling it globally within CUCM.
12-19-2013 11:03 AM
Ignore my first post..Leave voice class codec 10..
voice class codec 11
codec preference 1 g711ulaw
codec preference 2 g711alaw
codec preference 3 g729r8
dial-peer voice 8100 voip
no voice-class codec 10
voice-class codec 11
Then apply this to your outbound dial-peer
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"opportunity is a haughty goddess who waste no time with those who are unprepared"
12-19-2013 01:16 PM
I created the new voice class and mapped it to the outgoing dial-peer 8100. The call was then successful.
See new voice class:
#sh run | be voice class codec 11
voice class codec 11
codec preference 1 g711ulaw
codec preference 2 g711alaw
codec preference 3 g729r8
See revised dial-peer 8100:
dial-peer voice 8100 voip
description outbound dial-peer for outgoing calls to Verizon (11D)
destination-pattern ^1..........$
session protocol sipv2
session target sip-server
voice-class codec 11
voice-class sip dtmf-relay force rtp-nte
voice-class sip early-offer forced
dtmf-relay rtp-nte digit-drop
ip qos dscp cs5 media
ip qos dscp cs3 signaling
no vad
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
My only remaining question is why did the CUBE invite NOT include the m line for g729r8?
++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
See the ccapi inout snippet showing the hit with dial-peer 8100:
++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
080927: Dec 19 2013 15:27:57.810: //316459/32C4F8800001/CCAPI/ccCallSetupRequest:
Destination=, Calling IE Present=TRUE, Mode=0,
Outgoing Dial-peer=8100, Params=0x2B912E08, Progress Indication=NULL(0)
+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
See the debug ccsip messages output showing original CUCM invite received by CUBE with 5 a line references:
+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
080907: Dec 19 2013 15:27:57.806: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:XXXXXXXXXX@10.139.64.52:5060 SIP/2.0
Via: SIP/2.0/TCP 192.168.106.11:5060;branch=z9hG4bK6d715c9c6ad1
From: "XXXXXXXXXX"
To:
Date: Thu, 19 Dec 2013 20:36:14 GMT
Call-ID: 32c4f880-2b3158be-ffc1-b6aa8c0@192.168.106.11
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM8.6
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Expires: 180
Allow-Events: presence, kpml
Supported: X-cisco-srtp-fallback,X-cisco-original-called
Call-Info: <192.168.106.11:5060>;method="NOTIFY;Event=telephone-event;Duration=500"192.168.106.11:5060>
Cisco-Guid: 0851769472-0000065536-0000068412-0191539392
Session-Expires: 1800
P-Asserted-Identity: "XXXXXXXXXX"
Remote-Party-ID: "XXXXXXX"
Contact:
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 464
v=0
o=CiscoSystemsCCM-SIP 4077346 1 IN IP4 192.168.106.11
s=SIP Call
c=IN IP4 10.139.64.52
b=TIAS:64000
b=AS:64
t=0 0
m=audio 26738 RTP/AVP 0 8 116 116 18 101
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
a=ptime:20
a=rtpmap:116 iLBC/8000
a=ptime:20
a=maxptime:60
a=fmtp:116 mode=20
a=rtpmap:116 iLBC/8000
a=ptime:30
a=maxptime:60
a=fmtp:116 mode=30
a=rtpmap:18 G729/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
See ccsip messages output showing CUBE sending invite to Verizon:
+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
Sent:
INVITE sip:13369272626@10.189.135.17:5073 SIP/2.0
Via: SIP/2.0/UDP 10.139.64.52:5060;branch=z9hG4bK63F9C611
Remote-Party-ID: "David Callahan" <3367036158>;party=calling;screen=yes;privacy=off3367036158>
From: "David Callahan" <3367036158>;tag=7DE0957C-1CAB3367036158>
To: <13369272626>13369272626>
Date: Thu, 19 Dec 2013 20:27:57 GMT
Call-ID: E1D3496E-682211E3-95F4DBF2-34273F6B@10.139.64.52
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 0851769472-0000065536-0000068412-0191539392
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1387484877
Contact: <3367036158>3367036158>
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 69
Session-Expires: 1800
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 259
v=0
o=CiscoSystemsSIP-GW-UserAgent 6966 4178 IN IP4 10.139.64.52
s=SIP Call
c=IN IP4 10.139.64.52
t=0 0
m=audio 32502 RTP/AVP 0 8 101
c=IN IP4 10.139.64.52
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
12-19-2013 02:55 PM
Glad to know its working..Dont forget to rate useful posts or mark as answered any answered question
Thats a good question..and that is strange..Try this..
voice class codec 11
codec preference 1 g729r8
codec preference 2 g711ulaw
codec preference 3 g711alaw
Then test again and see if G729 is offered in the m-line to your ITSP..If it is, revert back to the old one and test again, then enable debug ccsip all...(make sure that there is little traffic on the cube when you do this)..Send the logs to me
Please rate all useful posts
"opportunity is a haughty goddess who waste no time with those who are unprepared"
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