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Community Member

Outbound Call Failure - SIP Trunk

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

Everyone's tags (2)
1 ACCEPTED SOLUTION

Accepted Solutions
VIP Super Bronze

Outbound Call Failure - SIP Trunk

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"

Please rate all useful posts "The essence of christianity is not the enthronement but the obliteration of self --William Barclay"
8 REPLIES
VIP Super Bronze

Outbound Call Failure - SIP Trunk

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"

Please rate all useful posts "The essence of christianity is not the enthronement but the obliteration of self --William Barclay"
Community Member

Outbound Call Failure - SIP Trunk

Thank you!  That was my conclusion as well. 

Amir

Community Member

Outbound Call Failure - SIP Trunk

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

VIP Super Bronze

Outbound Call Failure - SIP Trunk

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"

Please rate all useful posts "The essence of christianity is not the enthronement but the obliteration of self --William Barclay"
Community Member

Re: Outbound Call Failure - SIP Trunk

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.

VIP Super Bronze

Outbound Call Failure - SIP Trunk

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

Please rate all useful posts

"opportunity is a haughty goddess who waste no time with those who are unprepared"

Please rate all useful posts "The essence of christianity is not the enthronement but the obliteration of self --William Barclay"
Community Member

Re: Outbound Call Failure - SIP Trunk

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" ;tag=4077346~a812c08a-f7f0-43a7-a92c-e5ac2a38867c-65761788

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"

Cisco-Guid: 0851769472-0000065536-0000068412-0191539392

Session-Expires:  1800

P-Asserted-Identity: "XXXXXXXXXX"

Remote-Party-ID: "XXXXXXX" ;party=calling;screen=yes;privacy=off

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=off

From: "David Callahan" <3367036158>;tag=7DE0957C-1CAB

To: <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>

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

VIP Super Bronze

Outbound Call Failure - SIP Trunk

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"

Please rate all useful posts "The essence of christianity is not the enthronement but the obliteration of self --William Barclay"
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