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martynch1
Beginner

Outgoing Calls SiP Cube

Wonder if you could help me out here, I'm currently setting up a new cube but hitting problems placing outgoing calls, incoming calls are working fine, if I place an outgoing call to my mobile number I get a message (I can hear it) Sorry the service you require can not be connected.  Below you will find my running config stripped and a few debugs.

 

Debug ccsip call

Jul 13 07:16:08.075: //96612/B2C67D800000/SIP/Call/sipSPICallInfo: 
The Call Setup Information is:
Call Control Block (CCB) : 0x0x7F0626233170
State of The Call        : STATE_DEAD
TCP Sockets Used         : NO
Calling Number           : 45000
Called Number            : 07973xxxxxx
Source IP Address (Sig  ): 10.2.251.57
Destn SIP Req Addr:Port  : 82.16.19.2:5060
Destn SIP Resp Addr:Port : 82.16.19.2:5060
Destination Name         : 82.16.19.2

Jul 13 07:16:08.075: //96612/B2C67D800000/SIP/Call/sipSPIMediaCallInfo: 
Number of Media Streams: 1
Media Stream             : 1
Negotiated Codec         : g711alaw
Negotiated Codec Bytes   : 160
Nego. Codec payload      : 8 (tx), 8 (rx)
Negotiated Dtmf-relay    : 6
Dtmf-relay Payload       : 101 (tx), 101 (rx)
Source IP Address (Media): 10.2.251.57
Source IP Port    (Media): 8190
Destn  IP Address (Media): 82.16.19.18
Destn  IP Port    (Media): 10100
Orig Destn IP Address:Port (Media): [ - ]:0

Jul 13 07:16:08.075: //96612/B2C67D800000/SIP/Call/sipSPICallInfo: 
Disconnect Cause (CC)    : 16
Disconnect Cause (SIP)   : 487

Jul 13 07:16:08.076: //96616/B2C67D800000/SIP/Call/sipSPICallInfo: 
The Call Setup Information is:
Call Control Block (CCB) : 0x0x7F0626241C80
State of The Call        : STATE_DEAD
TCP Sockets Used         : YES
Calling Number           : 45000
Called Number            : 07973xxxxxx
Source IP Address (Sig  ): 10.2.251.57
Destn SIP Req Addr:Port  : :0
Destn SIP Resp Addr:Port : :0
Destination Name         : 

Jul 13 07:16:08.077: //96616/B2C67D800000/SIP/Call/sipSPIMediaCallInfo: 
Number of Media Streams: 1
Media Stream             : 1
Negotiated Codec         : No Codec   
Negotiated Codec Bytes   : 0
Nego. Codec payload      : 255 (tx), 255 (rx)
Negotiated Dtmf-relay    : 0
Dtmf-relay Payload       : 0 (tx), 0 (rx)
Source IP Address (Media): 10.2.251.57
Source IP Port    (Media): 8192
Destn  IP Address (Media):  - 
Destn  IP Port    (Media): 0
Orig Destn IP Address:Port (Media): [ - ]:0

Jul 13 07:16:08.077: //96616/B2C67D800000/SIP/Call/sipSPICallInfo: 
Disconnect Cause (CC)    : 188
Disconnect Cause (SIP)   : 200

Jul 13 07:16:08.080: //96611/B2C67D800000/SIP/Call/sipSPICallInfo: 
The Call Setup Information is:
Call Control Block (CCB) : 0x0x7F062621D0D8
State of The Call        : STATE_DEAD
TCP Sockets Used         : YES
Calling Number           : 45000
Called Number            : 07973xxxxxx
Source IP Address (Sig  ): 10.0.12.30
Destn SIP Req Addr:Port  : 10.0.12.17:5060
Destn SIP Resp Addr:Port : 10.0.12.17:34867
Destination Name         : 10.0.12.17

Jul 13 07:16:08.080: //96611/B2C67D800000/SIP/Call/sipSPIMediaCallInfo: 
Number of Media Streams: 1
Media Stream             : 1
Negotiated Codec         : g711alaw
Negotiated Codec Bytes   : 160
Nego. Codec payload      : 8 (tx), 8 (rx)
Negotiated Dtmf-relay    : 6
Dtmf-relay Payload       : 101 (tx), 101 (rx)
Source IP Address (Media): 10.0.12.30
Source IP Port    (Media): 8188
Destn  IP Address (Media): 172.30.152.11
Destn  IP Port    (Media): 32362
Orig Destn IP Address:Port (Media): [ - ]:0

Jul 13 07:16:08.080: //96611/B2C67D800000/SIP/Call/sipSPICallInfo: 
Disconnect Cause (CC)    : 38
Disconnect Cause (SIP)   : 503
Debug ccsip message

Jul 13 07:10:30.638: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received: 
INVITE sip:07973xxxxxx@10.0.12.30:5060 SIP/2.0
Via: SIP/2.0/TCP 10.0.12.17:5060;branch=z9hG4bK1eead65cae8f3
From: "VM SIP TEST 2" <sip:45000@10.0.12.17>;tag=404459~9c2c55f8-f6da-4001-942f-f8cb268e11d0-49787915
To: <sip:0797xxxxxx@10.0.12.30>
Date: Mon, 13 Jul 2020 07:10:30 GMT
Call-ID: ee147c80-f0c108e6-15b0f-110c000a@10.0.12.17
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=VIDEO_UNSPECIFIED
Session-ID: 30bca1117401e23b8117f8034def3ba0;remote=00000000000000000000000000000000
Cisco-Guid: 3994320000-0000065536-0000004169-0285999114
Session-Expires:  1800
P-Asserted-Identity: "VM SIP TEST 2" <sip:45000@10.0.12.17>
Contact: <sip:45000@10.0.12.17:5060;transport=tcp>
Max-Forwards: 70
Content-Length: 0


Jul 13 07:10:30.642: //96514/EE147C800000/SIP/Msg/ccsipDisplayMsg:
Sent: 
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 10.0.12.17:5060;branch=z9hG4bK1eead65cae8f3
From: "VM SIP TEST 2" <sip:45000@10.0.12.17>;tag=404459~9c2c55f8-f6da-4001-942f-f8cb268e11d0-49787915
To: <sip:07973xxxxxx@10.0.12.30>
Date: Mon, 13 Jul 2020 07:10:30 GMT
Call-ID: ee147c80-f0c108e6-15b0f-110c000a@10.0.12.17
CSeq: 101 INVITE
Allow-Events: kpml, telephone-event
Server: Cisco-SIPGateway/IOS-16.6.6
Session-ID: 00000000000000000000000000000000;remote=30bca1117401e23b8117f8034def3ba0
Content-Length: 0


Jul 13 07:10:30.645: //96515/EE147C800000/SIP/Msg/ccsipDisplayMsg:
Sent: 
INVITE sip:07973xxxxxx@82.16.19.2:5060 SIP/2.0
Via: SIP/2.0/UDP 10.2.251.57:5060;branch=z9hG4bK12084228B
From: "VM SIP TEST 2" <sip:45000@10.2.251.57>;tag=14803119-1814
To: <sip:07973xxxxxx@82.16.19.2>
Date: Mon, 13 Jul 2020 07:10:30 GMT
Call-ID: C544F9D6-C40E11EA-B9A3B5A2-101A3230@10.2.251.57
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE:  1800
Cisco-Guid: 3994320000-0000065536-0000004169-0285999114
User-Agent: Cisco-SIPGateway/IOS-16.6.6
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1594624230
Contact: <sip:45000@10.2.251.57:5060>
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 69
P-Asserted-Identity: "VM SIP TEST 2" <sip:45000@10.2.251.57>
Session-ID: 30bca1117401e23b8117f8034def3ba0;remote=00000000000000000000000000000000
Session-Expires:  1800
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 243

v=0
o=CiscoSystemsSIP-GW-UserAgent 1948 8484 IN IP4 10.2.251.57
s=SIP Call
c=IN IP4 10.2.251.57
t=0 0
m=audio 8184 RTP/AVP 8 101
c=IN IP4 10.2.251.57
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

Jul 13 07:10:30.649: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received: 
OPTIONS sip:10.0.12.30:5060 SIP/2.0
Via: SIP/2.0/TCP 10.0.12.17:5060;branch=z9hG4bK1eeae2d555626
From: <sip:10.0.12.17>;tag=524611185
To: <sip:10.0.12.30>
Date: Mon, 13 Jul 2020 07:10:30 GMT
Call-ID: ee147c80-f0c108e6-15b10-110c000a@10.0.12.17
User-Agent: Cisco-CUCM11.5
CSeq: 101 OPTIONS
Contact: <sip:10.0.12.17:5060;transport=tcp>
Max-Forwards: 0
Content-Length: 0


Jul 13 07:10:30.651: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent: 
SIP/2.0 200 OK
Via: SIP/2.0/TCP 10.0.12.17:5060;branch=z9hG4bK1eeae2d555626
From: <sip:10.0.12.17>;tag=524611185
To: <sip:10.0.12.30>;tag=1480311F-2001
Date: Mon, 13 Jul 2020 07:10:30 GMT
Call-ID: ee147c80-f0c108e6-15b10-110c000a@10.0.12.17
Server: Cisco-SIPGateway/IOS-16.6.6
CSeq: 101 OPTIONS
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: kpml, telephone-event
Accept: application/sdp
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Content-Type: application/sdp
Content-Length: 163

v=0
o=CiscoSystemsSIP-GW-UserAgent 8832 2086 IN IP4 10.0.12.30
s=SIP Call
c=IN IP4 10.0.12.30
t=0 0
m=audio 0 RTP/AVP 18 0 8 9 4 2 15 3
c=IN IP4 10.0.12.30

Jul 13 07:10:30.654: //96515/EE147C800000/SIP/Msg/ccsipDisplayMsg:
Received: 
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.2.251.57:5060;branch=z9hG4bK12084228B
From: "VM SIP TEST 2" <sip:45000@10.2.251.57>;tag=14803119-1814
To: <sip:07973xxxxxx@82.16.19.2>
Call-ID: C544F9D6-C40E11EA-B9A3B5A2-101A3230@10.2.251.57
CSeq: 101 INVITE
Timestamp: 1594624230
Content-Length: 0


Jul 13 07:10:30.662: //96515/EE147C800000/SIP/Msg/ccsipDisplayMsg:
Received: 
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.2.251.57:5060;branch=z9hG4bK12084228B
Proxy-Authenticate: Digest realm="Realm",nonce="MTU5NDYyMDIyMTg3NjU3ZTRlZGJjZTRiNTdlYjQ4OGU5NTFiMzQyY2Q1Yzc5",stale=false,algorithm=MD5,qop="auth"
To: <sip:07973xxxxxx@82.16.19.2>;tag=3803613030-698069643
From: "VM SIP TEST 2" <sip:45000@10.2.251.57;user=phone>;tag=14803119-1814
Call-ID: C544F9D6-C40E11EA-B9A3B5A2-101A3230@10.2.251.57
CSeq: 101 INVITE
Allow: UPDATE,PRACK,INFO,NOTIFY,OPTIONS,BYE,INVITE,ACK,CANCEL
Contact: <sip:07973xxxxxx@82.16.19.2:5060>
Content-Length: 0


Jul 13 07:10:30.663: //96515/EE147C800000/SIP/Msg/ccsipDisplayMsg:
Sent: 
ACK sip:07973xxxxxx@82.16.19.2:5060 SIP/2.0
Via: SIP/2.0/UDP 10.2.251.57:5060;branch=z9hG4bK12084228B
From: "VM SIP TEST 2" <sip:45000@10.2.251.57>;tag=14803119-1814
To: <sip:07973xxxxxx@82.16.19.2>;tag=3803613030-698069643
Date: Mon, 13 Jul 2020 07:10:30 GMT
Call-ID: C544F9D6-C40E11EA-B9A3B5A2-101A3230@10.2.251.57
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Session-ID: ;remote=
Content-Length: 0


Jul 13 07:10:30.664: //96515/EE147C800000/SIP/Msg/ccsipDisplayMsg:
Sent: 
INVITE sip:07973xxxxxx@82.16.19.2:5060 SIP/2.0
Via: SIP/2.0/UDP 10.2.251.57:5060;branch=z9hG4bK12085A0C
From: "VM SIP TEST 2" <sip:45000@10.2.251.57>;tag=14803119-1814
To: <sip:07973xxxxxx@82.16.19.2>
Date: Mon, 13 Jul 2020 07:10:30 GMT
Call-ID: C544F9D6-C40E11EA-B9A3B5A2-101A3230@10.2.251.57
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE:  1800
Cisco-Guid: 3994320000-0000065536-0000004169-0285999114
User-Agent: Cisco-SIPGateway/IOS-16.6.6
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 102 INVITE
Timestamp: 1594624230
Contact: <sip:45000@10.2.251.57:5060>
Expires: 180
Allow-Events: telephone-event
Proxy-Authorization: Digest username="onboard09",realm="Realm",uri="sip:07973xxxxxx@82.16.19.2:5060",response="ad85f4c9c9878a2375cec50674659789",nonce="MTU5NDYyMDIyMTg3NjU3ZTRlZGJjZTRiNTdlYjQ4OGU5NTFiMzQyY2Q1Yzc5",cnonce="E6D38045",qop=auth,algorithm=MD5,nc=00000001
Max-Forwards: 69
P-Asserted-Identity: "VM SIP TEST 2" <sip:45000@10.2.251.57>
Session-ID: 30bca1117401e23b8117f8034def3ba0;remote=00000000000000000000000000000000
Session-Expires:  1800
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 243

v=0
o=CiscoSystemsSIP-GW-UserAgent 1948 8484 IN IP4 10.2.251.57
s=SIP Call
c=IN IP4 10.2.251.57
t=0 0
m=audio 8184 RTP/AVP 8 101
c=IN IP4 10.2.251.57
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

Jul 13 07:10:30.673: //96515/EE147C800000/SIP/Msg/ccsipDisplayMsg:
Received: 
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.2.251.57:5060;branch=z9hG4bK12085A0C
From: "VM SIP TEST 2" <sip:45000@10.2.251.57>;tag=14803119-1814
To: <sip:07973xxxxxx@82.16.19.2>
Call-ID: C544F9D6-C40E11EA-B9A3B5A2-101A3230@10.2.251.57
CSeq: 102 INVITE
Timestamp: 1594624230
Content-Length: 0


Jul 13 07:10:30.725: //96515/EE147C800000/SIP/Msg/ccsipDisplayMsg:
Received: 
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.2.251.57:5060;branch=z9hG4bK12085A0C
To: <sip:07973xxxxxx@82.16.19.2>;tag=3803613030-1020988772
From: "VM SIP TEST 2" <sip:45000@10.2.251.57;user=phone>;tag=14803119-1814
Call-ID: C544F9D6-C40E11EA-B9A3B5A2-101A3230@10.2.251.57
CSeq: 102 INVITE
Allow: UPDATE,PRACK,INFO,NOTIFY,OPTIONS,BYE,INVITE,ACK,CANCEL
Contact: <sip:07973xxxxxx@82.16.19.2:5060>
Content-Type: application/sdp
Content-Length: 252

v=0
o=brnt-voiponboardsbc1-a 201206121 201206121 IN IP4 82.16.19.2
s=sip call
c=IN IP4 82.16.19.18
t=0 0
m=audio 10098 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=silenceSupp:off - - - -

Jul 13 07:10:30.728: //96514/EE147C800000/SIP/Msg/ccsipDisplayMsg:
Sent: 
SIP/2.0 183 Session Progress
Via: SIP/2.0/TCP 10.0.12.17:5060;branch=z9hG4bK1eead65cae8f3
From: "VM SIP TEST 2" <sip:45000@10.0.12.17>;tag=404459~9c2c55f8-f6da-4001-942f-f8cb268e11d0-49787915
To: <sip:07973xxxxxx@10.0.12.30>;tag=1480316D-1AD7
Date: Mon, 13 Jul 2020 07:10:30 GMT
Call-ID: ee147c80-f0c108e6-15b0f-110c000a@10.0.12.17
CSeq: 101 INVITE
Require: 100rel
RSeq: 8573
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: kpml, telephone-event
Contact: <sip:07973xxxxxx@10.0.12.30:5060;transport=tcp>
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-16.6.6
Session-ID: 58d9216120e7570f97bc8825f965a0f7;remote=30bca1117401e23b8117f8034def3ba0
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 240

v=0
o=CiscoSystemsSIP-GW-UserAgent 3781 3104 IN IP4 10.0.12.30
s=SIP Call
c=IN IP4 10.0.12.30
t=0 0
m=audio 8182 RTP/AVP 8 101
c=IN IP4 10.0.12.30
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

Jul 13 07:10:30.793: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received: 
PRACK sip:07973xxxxxx@10.0.12.30:5060;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.0.12.17:5060;branch=z9hG4bK1eeaf1d80c380
From: "VM SIP TEST 2" <sip:45000@10.0.12.17>;tag=404459~9c2c55f8-f6da-4001-942f-f8cb268e11d0-49787915
To: <sip:07973xxxxxx@10.0.12.30>;tag=1480316D-1AD7
Date: Mon, 13 Jul 2020 07:10:30 GMT
Call-ID: ee147c80-f0c108e6-15b0f-110c000a@10.0.12.17
User-Agent: Cisco-CUCM11.5
CSeq: 102 PRACK
RAck: 8573 101 INVITE
Allow-Events: presence
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 245

v=0
o=CiscoSystemsCCM-SIP 404459 1 IN IP4 10.0.12.17
s=SIP Call
c=IN IP4 172.30.152.11
b=TIAS:64000
b=CT:64
b=AS:64
t=0 0
m=audio 32116 RTP/AVP 8 101
a=ptime:20
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

Jul 13 07:10:30.796: //96514/EE147C800000/SIP/Msg/ccsipDisplayMsg:
Sent: 
SIP/2.0 200 OK
Via: SIP/2.0/TCP 10.0.12.17:5060;branch=z9hG4bK1eeaf1d80c380
From: "VM SIP TEST 2" <sip:45000@10.0.12.17>;tag=404459~9c2c55f8-f6da-4001-942f-f8cb268e11d0-49787915
To: <sip:07973xxxxxx@10.0.12.30>;tag=1480316D-1AD7
Date: Mon, 13 Jul 2020 07:10:30 GMT
Call-ID: ee147c80-f0c108e6-15b0f-110c000a@10.0.12.17
Server: Cisco-SIPGateway/IOS-16.6.6
CSeq: 102 PRACK
Session-ID: 58d9216120e7570f97bc8825f965a0f7;remote=30bca1117401e23b8117f8034def3ba0
Content-Length: 0


Jul 13 07:10:31.209: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received: 
OPTIONS sip:10.0.12.30:5060 SIP/2.0
Via: SIP/2.0/TCP 10.0.12.16:5060;branch=z9hG4bK6bae22f24ecca
From: <sip:10.0.12.16>;tag=2084978717
To: <sip:10.0.12.30>
Date: Mon, 13 Jul 2020 07:10:31 GMT
Call-ID: eead1300-f0c108e7-4aa6e-100c000a@10.0.12.16
User-Agent: Cisco-CUCM11.5
CSeq: 101 OPTIONS
Contact: <sip:10.0.12.16:5060;transport=tcp>
Max-Forwards: 0
Content-Length: 0


Jul 13 07:10:31.211: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent: 
SIP/2.0 200 OK
Via: SIP/2.0/TCP 10.0.12.16:5060;branch=z9hG4bK6bae22f24ecca
From: <sip:10.0.12.16>;tag=2084978717
To: <sip:10.0.12.30>;tag=14803350-A2F
Date: Mon, 13 Jul 2020 07:10:31 GMT
Call-ID: eead1300-f0c108e7-4aa6e-100c000a@10.0.12.16
Server: Cisco-SIPGateway/IOS-16.6.6
CSeq: 101 OPTIONS
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: kpml, telephone-event
Accept: application/sdp
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Content-Type: application/sdp
Content-Length: 163

v=0
o=CiscoSystemsSIP-GW-UserAgent 4174 9701 IN IP4 10.0.12.30
s=SIP Call
c=IN IP4 10.0.12.30
t=0 0
m=audio 0 RTP/AVP 18 0 8 9 4 2 15 3
c=IN IP4 10.0.12.30

Jul 13 07:10:37.860: //96515/EE147C800000/SIP/Msg/ccsipDisplayMsg:
Received: 
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 10.2.251.57:5060;branch=z9hG4bK12085A0C
To: <sip:0797xxxxxx@82.16.19.2>;tag=3803613030-1020988772
From: "VM SIP TEST 2" <sip:45000@10.2.251.57;user=phone>;tag=14803119-1814
Call-ID: C544F9D6-C40E11EA-B9A3B5A2-101A3230@10.2.251.57
CSeq: 102 INVITE
Allow: UPDATE,PRACK,INFO,NOTIFY,OPTIONS,BYE,INVITE,ACK,CANCEL
Contact: <sip:07973xxxxxx@82.16.19.2:5060>
Content-Length: 0


Jul 13 07:10:37.862: //96515/EE147C800000/SIP/Msg/ccsipDisplayMsg:
Sent: 
ACK sip:07973xxxxxx@82.16.19.2:5060 SIP/2.0
Via: SIP/2.0/UDP 10.2.251.57:5060;branch=z9hG4bK12085A0C
From: "VM SIP TEST 2" <sip:45000@10.2.251.57>;tag=14803119-1814
To: <sip:07973xxxxxx@82.16.19.2>;tag=3803613030-1020988772
Date: Mon, 13 Jul 2020 07:10:30 GMT
Call-ID: C544F9D6-C40E11EA-B9A3B5A2-101A3230@10.2.251.57
Max-Forwards: 70
CSeq: 102 ACK
Allow-Events: telephone-event
Content-Length: 0


Jul 13 07:10:37.865: //96514/EE147C800000/SIP/Msg/ccsipDisplayMsg:
Sent: 
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/TCP 10.0.12.17:5060;branch=z9hG4bK1eead65cae8f3
From: "VM SIP TEST 2" <sip:45000@10.0.12.17>;tag=404459~9c2c55f8-f6da-4001-942f-f8cb268e11d0-49787915
To: <sip:07973xxxxxx@10.0.12.30>;tag=1480316D-1AD7
Date: Mon, 13 Jul 2020 07:10:30 GMT
Call-ID: ee147c80-f0c108e6-15b0f-110c000a@10.0.12.17
CSeq: 101 INVITE
Allow-Events: kpml, telephone-event
Server: Cisco-SIPGateway/IOS-16.6.6
Reason: Q.850;cause=0
Session-ID: 58d9216120e7570f97bc8825f965a0f7;remote=30bca1117401e23b8117f8034def3ba0
Session-Expires:  1800
Content-Length: 0


Jul 13 07:10:37.867: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received: 
ACK sip:07973xxxxxx@10.0.12.30:5060 SIP/2.0
Via: SIP/2.0/TCP 10.0.12.17:5060;branch=z9hG4bK1eead65cae8f3
From: "VM SIP TEST 2" <sip:45000@10.0.12.17>;tag=404459~9c2c55f8-f6da-4001-942f-f8cb268e11d0-49787915
To: <sip:07973xxxxxx@10.0.12.30>;tag=1480316D-1AD7
Date: Mon, 13 Jul 2020 07:10:30 GMT
Call-ID: ee147c80-f0c108e6-15b0f-110c000a@10.0.12.17
User-Agent: Cisco-CUCM11.5
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: presence
Content-Length: 0
Debug ccisp error

SIP: (96454) Group (a= group line) attribute, level 65535 instance 1 not found.
SIP: (96454) Group (a= group line) attribute, level 65535 instance 1 not found.
SIP: (96454) Group (a= group line) attribute, level 65535 instance 1 not found.
SIP: (96454) Group (a= group line) attribute, level 65535 instance 1 not found.
SIP: Attribute mid, level 1 instance 1 not found.
SIP: setup attribute, level 1 instance 1 not found.
SIP: connection attribute, level 1 instance 1 not found.
SIP: Attribute label, level 1 instance 1 not found.
SIP: a=framerate attribute, level 1 instance 1 not found.
Jul 13 07:07:03.183: //-1/xxxxxxxxxxxx/SIP/Error/voipCodec_to_rtpAvpCodec: 
 Unexpected VoIPCodec Type :No Codec   
SIP: (96453) Group (a= group line) attribute, level 65535 instance 1 not found.
SIP: (96453) Group (a= group line) attribute, level 65535 instance 1 not found.
Jul 13 07:07:03.184: //96454/72B2CB000000/SIP/Error/sipSPIGetNewLocalMediaDirection: 
 Unknown media direction (0) from remote end
SIP: (96453) Group (a= group line) attribute, level 65535 instance 1 not found.
Jul 13 07:07:03.225: //-1/xxxxxxxxxxxx/SIP/Error/httpish_msg_free: 
 Freeing NULL pointer!
SIP: Attribute mid, level 1 instance 1 not found.
Jul 13 07:07:03.227: //96453/72B2CB000000/SIP/Error/sipSPIGetNewLocalMediaDirection: 
 Unknown media direction (0) from remote end
SIP: setup attribute, level 1 instance 1 not found.
SIP: connection attribute, level 1 instance 1 not found.
SIP: Attribute label, level 1 instance 1 not found.
SIP: a=framerate attribute, level 1 instance 1 not found.
Jul 13 07:07:10.158: //96453/72B2CB000000/SIP/Error/sipSPIFlushDeferredQueue: 
 Invalid deferredQueue
Jul 13 07:07:10.165: //-1/xxxxxxxxxxxx/SIP/Error/sipSPIRemoveBranchName: 
 invalid ccb, bName or branch list for sipSPIRemoveBranchName
Jul 13 07:07:10.169: //-1/xxxxxxxxxxxx/SIP/Error/ccsip_iwf_process_event: 
 Dead CCB
Jul 13 07:07:10.169: //96454/72B2CB000000/SIP/Error/ccsip_offer_ans_md_invite_ack_answer_sent_hdlr: 
 Unable to send ACK sent peer event

And finally my running config

no voice hunt unassigned-number
no voice hunt no-response
no voice hunt dest-out-of-order
no voice hunt invalid-number
voice call send-alert
voice rtp send-recv
!
voice service voip
 mode border-element license capacity 200
 allow-connections sip to sip
 redirect ip2ip
 fax protocol pass-through g711alaw
 sip
  session refresh
  header-passing
  subscription maximum originate 2
  asserted-id pai
  privacy pstn
  options-ping 60
  no update-callerid
  early-offer forced
  midcall-signaling passthru
  privacy-policy passthru
!
!     
voice class e164-pattern-map 9
 description *** PSTN Numbers Outbound ***
  e164 +800T
  e164 +44T
  e164 ^1800[01]$
  e164 ^999$
  e164 ^11[68]...$
  e164 ^1..$
  e164 ^[01]T
  e164 ^[02]T
  e164 ^[2]T
 !
!
voice class dpg 90
 description *** SBC DPG ***
 dial-peer 90 preference 1
 dial-peer 91 preference 2
!
voice class dpg 1000
 description *** CUCM DPG ***
 dial-peer 1000 preference 1
 dial-peer 1001 preference 2
!
voice class sip-options-keepalive 1
 description ** global options pings settings **
 up-interval 30
 retry 3
 transport udp
!
voice translation-rule 1
 rule 1 /^\([127-9]......\)$/ /0115\1/
!
!
voice translation-profile BlockedNumbers
 translate calling 777
!
voice translation-profile OutboundCalledPartyforShortDIAL
 translate called 1
!
interface GigabitEthernet0/0/0
 ip address 10.2.251.57 255.255.255.252
 no ip redirects
 no ip proxy-arp
 media-type rj45
 negotiation auto
!
interface GigabitEthernet0/0/1
 ip address 10.0.12.30 255.255.255.0
 no ip redirects
 no ip proxy-arp
 negotiation auto
!
!
mgcp behavior rsip-range tgcp-only
mgcp behavior comedia-role none
mgcp behavior comedia-check-media-src disable
mgcp behavior comedia-sdp-force disable
!
mgcp profile default
!
!
!
!
dial-peer voice 9 voip
 description ** INBOUND Dial-Peer from SIP Platform **
 translation-profile incoming BlockedNumbers
 call-block disconnect-cause incoming unassigned-number
 session protocol sipv2
 session transport udp
 destination dpg 1000
 incoming uri via ITSP
 voice-class codec 1  
 voice-class sip bind control source-interface GigabitEthernet0/0/0
 voice-class sip bind media source-interface GigabitEthernet0/0/0
 dtmf-relay rtp-nte
 ip qos dscp cs3 signaling
 no vad
!         
dial-peer voice 10 voip
 description *** CUCM to CUBE (inbound) ***
 translation-profile incoming OutboundCalledPartyforShortDIAL
 session protocol sipv2
 session transport tcp
 destination dpg 90
 incoming uri via CUCM
 voice-class codec 1  
 voice-class sip bind control source-interface GigabitEthernet0/0/1
 voice-class sip bind media source-interface GigabitEthernet0/0/1
 dtmf-relay rtp-nte sip-kpml
 no vad
!
dial-peer voice 1000 voip
 description *** CUBE to CUCM LOX-SUB 1 (outbound) ***
 preference 1
 destination-pattern 69...
 session protocol sipv2
 session target ipv4:10.0.12.17
 session transport tcp
 voice-class sip options-keepalive profile 1
 voice-class sip bind control source-interface GigabitEthernet0/0/1
 voice-class sip bind media source-interface GigabitEthernet0/0/1
 dtmf-relay rtp-nte sip-kpml
 codec g711alaw
 ip qos dscp cs3 signaling
 no vad
!
dial-peer voice 90 voip
 description *** SBC3 Birmingham - Preferred Route ***
 preference 1
 destination-pattern .T
 session protocol sipv2
 session target ipv4:xx.xx.xx.xx
 session transport udp
 voice-class sip options-ping 60
 voice-class sip options-keepalive retry 3
 voice-class sip bind control source-interface GigabitEthernet0/0/0
 voice-class sip bind media source-interface GigabitEthernet0/0/0
 dtmf-relay rtp-nte
 dtmf-interworking rtp-nte
 codec g711alaw
 ip qos dscp cs3 signaling
 no vad
!
dial-peer voice 1001 voip
 description *** CUBE to CUCM WTG-SUB 1(outbound) ***
 preference 2
 destination-pattern 69...
 session protocol sipv2
 session target ipv4:SUBSCRIBER
 session transport tcp
 voice-class sip options-keepalive profile 1
 voice-class sip bind control source-interface GigabitEthernet0/0/1
 voice-class sip bind media source-interface GigabitEthernet0/0/1
 dtmf-relay rtp-nte sip-kpml
 codec g711alaw
 ip qos dscp cs3 signaling
 no vad
!
dial-peer voice 1002 voip
 description *** CUBE to CUCM PUB (outbound) ***
 preference 3
 destination-pattern 69...
 session protocol sipv2
 session target ipv4:PUBLISHER
 session transport tcp
 voice-class sip options-keepalive profile 1
 voice-class sip bind control source-interface GigabitEthernet0/0/1
 voice-class sip bind media source-interface GigabitEthernet0/0/1
 dtmf-relay rtp-nte sip-kpml
 codec g711alaw
 ip qos dscp cs3 signaling
 no vad
!
!
dial-peer hunt 1
sip-ua 
 authentication username XXXXX password 7 XXXXX
 no remote-party-id
 aaa username proxy-auth
 retry invite 4
 timers options 600
 reason-header override
 connection-reuse

Thanks in advance

 

Martyn

 

31 REPLIES 31
Mike_Brezicky
Cisco Employee

is this a new telco service or just moving to another CUBE?

I suspect the error you are getting is due to the SIP/2.0 503 Service Unavailable message.

The connection looks generally happy. Codec negotiation looks good. Its communication or you would not be getting the proxy auth request. If this is new service, are you sure telco has completed provisioning? And if it was pre-existing, does telco have any ip trust list in which they need to whitelist this new CUBE?

Hi and thanks for the response, its an excising CUCM moving services from ISDN to SiP, I'm currently testing with the telco for sign off but they are saying that its a miss-configuration from my-side. I will ask them to double check the configuration their side because as you mention it all looks fine cube side.

 

Again thanks for your response and I will let you know the outcome.

 

Martyn

So the sequence is a normal outbound call setup, gets as far as 183 proceeding at which point I assume you hear the announcement.  Then the carrier send you a 487 termination, and your gateway sends 503 Service Unavailable to CUCM.

I think it's possible that the carrier doesn't like your calling number.  Many want to see a valid number, ie one of the numbers that route into the trunk.  Especially on an authenticated trunk.

From: "VM SIP TEST 2" <sip:45000@10.2.251.57>;tag=14803119-1814

By the way is that only a partial configuration?  I see you're matching inbound by URI but I can't see the actual definitions for "ITSP" or "CUCM" (coincidentally the exact names that I use in that context).  Routing by DPG you need bombproof inbound dial peer matching.

 

Thanks, let me get back to the carrier with that question, what definitions are missing for my "ITSP" or "CUCM" as this is the majority of my configuration.

 

Thanks  

In your inbound dial peers you have configured them to match based on the VIA header.  For example ...

dial-peer voice 10 voip
 description *** CUCM to CUBE (inbound) ***
 translation-profile incoming OutboundCalledPartyforShortDIAL
 session protocol sipv2
 session transport tcp
 destination dpg 90
 incoming uri via CUCM
 voice-class codec 1  
 voice-class sip bind control source-interface GigabitEthernet0/0/1
 voice-class sip bind media source-interface GigabitEthernet0/0/1
 dtmf-relay rtp-nte sip-kpml
 no vad

That's saying it will match if the VIA header matches one of the parameters defined in a class named "CUCM".   For example if you were just matching by IP address ..

voice class uri CUCM sip
 host ipv4:192.168.21.10
 host ipv4:192.168.21.11

Apologies, yes I have our cluster

voice class uri CUCM sip
host ipv4:10.0.12.16
host ipv4:10.0.12.17
host ipv4:10.20.12.16

 

Ive been going trough my debugs which may be showing something strange, this is me calling my mobile number from 45000

 

Jul 14 15:51:05.981: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received: 
INVITE sip:07973xxxxxx@10.0.12.30:5060 SIP/2.0
Via: SIP/2.0/TCP 10.0.12.17:5060;branch=z9hG4bK60b01ac8f481
From: "SiP Test 45000" <sip:45000@10.0.12.17>;tag=67135~9c2c55f8-f6da-4001-942f-f8cb268e11d0-36791270
To: <sip:07973xxxxxx@10.0.12.30>
Date: Tue, 14 Jul 2020 15:51:05 GMT
Call-ID: d2010000-f0d1d469-3f0d-110c000a@10.0.12.17
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: 4fe824fef3d83993017e11e24aa67130;remote=00000000000000000000000000000000
Cisco-Guid: 3523280896-0000065536-0000000923-0285999114
Session-Expires:  1800
P-Asserted-Identity: "SiP Test 45000" <sip:45000@10.0.12.17>
Contact: <sip:45000@10.0.12.17:5060;transport=tcp>
Max-Forwards: 69
Content-Length: 0

As you can see I start with 

CSeq: 101 INVITE

Then we sip 100 trying followed by Sip 407 Proxy auth and Ack

 

Jul 14 15:51:05.986: //3692/D20100000000/SIP/Msg/ccsipDisplayMsg:
Sent: 
INVITE sip:07973xxxxxx@82.16.19.2:5060 SIP/2.0
Via: SIP/2.0/UDP 10.2.251.57:5060;branch=z9hG4bK91B145F
From: "SiP Test 45000" <sip:45000@10.2.251.57>;tag=13D3EBA-26AE
To: <sip:07973xxxxxx@82.16.19.2>
Date: Tue, 14 Jul 2020 15:51:05 GMT
Call-ID: A9658480-C52011EA-8EF3AD4F-A1EA3405@10.2.251.57
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE:  1800
Cisco-Guid: 3523280896-0000065536-0000000923-0285999114
User-Agent: Cisco-SIPGateway/IOS-16.6.6
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1594741865
Contact: <sip:45000@10.2.251.57:5060>
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 68
P-Asserted-Identity: "SiP Test 45000" <sip:45000@10.2.251.57>
Session-ID: 4fe824fef3d83993017e11e24aa67130;remote=00000000000000000000000000000000
Session-Expires:  1800
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 243

v=0
o=CiscoSystemsSIP-GW-UserAgent 2503 7226 IN IP4 10.2.251.57
s=SIP Call
c=IN IP4 10.2.251.57
t=0 0
m=audio 8122 RTP/AVP 8 101
c=IN IP4 10.2.251.57
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

Jul 14 15:51:05.987: //3691/D20100000000/SIP/Msg/ccsipDisplayMsg:
Sent: 
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 10.0.12.17:5060;branch=z9hG4bK60b01ac8f481
From: "SiP Test 45000" <sip:45000@10.0.12.17>;tag=67135~9c2c55f8-f6da-4001-942f-f8cb268e11d0-36791270
To: <sip:07973xxxxxx@10.0.12.30>
Date: Tue, 14 Jul 2020 15:51:05 GMT
Call-ID: d2010000-f0d1d469-3f0d-110c000a@10.0.12.17
CSeq: 101 INVITE
Allow-Events: kpml, telephone-event
Server: Cisco-SIPGateway/IOS-16.6.6
Session-ID: 00000000000000000000000000000000;remote=4fe824fef3d83993017e11e24aa67130
Content-Length: 0


Jul 14 15:51:05.996: //3692/D20100000000/SIP/Msg/ccsipDisplayMsg:
Received: 
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.2.251.57:5060;branch=z9hG4bK91B145F
From: "SiP Test 45000" <sip:45000@10.2.251.57>;tag=13D3EBA-26AE
To: <sip:07973xxxxxx@82.16.19.2>
Call-ID: A9658480-C52011EA-8EF3AD4F-A1EA3405@10.2.251.57
CSeq: 101 INVITE
Timestamp: 1594741865
Content-Length: 0

Jul 14 15:51:06.004: //3692/D20100000000/SIP/Msg/ccsipDisplayMsg:
Received: 
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.2.251.57:5060;branch=z9hG4bK91B145F
Proxy-Authenticate: Digest realm="Realm",nonce="MTU5NDczMjQzMzQ1NTkwMWE1ZGY0ZTMyOGUzNmYzNzE4ZjYzN2ViNzc1NzZl",stale=false,algorithm=MD5,qop="auth"
To: <sip:07973xxxxxx@82.16.19.2>;tag=3803730665-1501893535
From: "SiP Test 45000" <sip:45000@10.2.251.57;user=phone>;tag=13D3EBA-26AE
Call-ID: A9658480-C52011EA-8EF3AD4F-A1EA3405@10.2.251.57
CSeq: 101 INVITE
Allow: UPDATE,PRACK,INFO,NOTIFY,OPTIONS,BYE,INVITE,ACK,CANCEL
Contact: <sip:07973xxxxxx@82.16.19.2:5060>
Content-Length: 0

Jul 14 15:51:06.005: //3692/D20100000000/SIP/Msg/ccsipDisplayMsg:
Sent: 
ACK sip:07973xxxxxx@82.16.19.2:5060 SIP/2.0
Via: SIP/2.0/UDP 10.2.251.57:5060;branch=z9hG4bK91B145F
From: "SiP Test 45000" <sip:45000@10.2.251.57>;tag=13D3EBA-26AE
To: <sip:07973xxxxxx@82.16.19.2>;tag=3803730665-1501893535
Date: Tue, 14 Jul 2020 15:51:05 GMT
Call-ID: A9658480-C52011EA-8EF3AD4F-A1EA3405@10.2.251.57
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Session-ID: ;remote=
Content-Length: 0

Which gives us our CSeq: 101 Ack

Our CSeq is incremented once we have been authorised to CSeq: 102 INVITE

Jul 14 15:51:06.006: //3692/D20100000000/SIP/Msg/ccsipDisplayMsg:
Sent: 
INVITE sip:07973xxxxxx@82.16.19.2:5060 SIP/2.0
Via: SIP/2.0/UDP 10.2.251.57:5060;branch=z9hG4bK91C18C
From: "SiP Test 45000" <sip:45000@10.2.251.57>;tag=13D3EBA-26AE
To: <sip:07973xxxxxx@82.16.19.2>
Date: Tue, 14 Jul 2020 15:51:06 GMT
Call-ID: A9658480-C52011EA-8EF3AD4F-A1EA3405@10.2.251.57
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE:  1800
Cisco-Guid: 3523280896-0000065536-0000000923-0285999114
User-Agent: Cisco-SIPGateway/IOS-16.6.6
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 102 INVITE
Timestamp: 1594741866
Contact: <sip:45000@10.2.251.57:5060>
Expires: 180
Allow-Events: telephone-event
Proxy-Authorization: Digest username="onboard08",realm="Realm",uri="sip:07973xxxxxx@82.16.19.2:5060",response="1cea26f27f5360fd654f3f0ca1b19e8f",nonce="MTU5NDczMjQzMzQ1NTkwMWE1ZGY0ZTMyOGUzNmYzNzE4ZjYzN2ViNzc1NzZl",cnonce="FF513110",qop=auth,algorithm=MD5,nc=00000001
Max-Forwards: 68
P-Asserted-Identity: "SiP Test 45000" <sip:45000@10.2.251.57>
Session-ID: 4fe824fef3d83993017e11e24aa67130;remote=00000000000000000000000000000000
Session-Expires:  1800
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 243

We then moving onto 183 session progress

v=0
o=CiscoSystemsSIP-GW-UserAgent 2503 7226 IN IP4 10.2.251.57
s=SIP Call
c=IN IP4 10.2.251.57
t=0 0
m=audio 8122 RTP/AVP 8 101
c=IN IP4 10.2.251.57
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

Jul 14 15:51:06.015: //3692/D20100000000/SIP/Msg/ccsipDisplayMsg:
Received: 
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.2.251.57:5060;branch=z9hG4bK91C18C
From: "SiP Test 45000" <sip:45000@10.2.251.57>;tag=13D3EBA-26AE
To: <sip:07973xxxxxx@82.16.19.2>
Call-ID: A9658480-C52011EA-8EF3AD4F-A1EA3405@10.2.251.57
CSeq: 102 INVITE
Timestamp: 1594741866
Content-Length: 0

Jul 14 15:51:06.074: //3692/D20100000000/SIP/Msg/ccsipDisplayMsg:
Received: 
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.2.251.57:5060;branch=z9hG4bK91C18C
To: <sip:07973xxxxxx@82.16.19.2>;tag=3803730666-1791299518
From: "SiP Test 45000" <sip:45000@10.2.251.57;user=phone>;tag=13D3EBA-26AE
Call-ID: A9658480-C52011EA-8EF3AD4F-A1EA3405@10.2.251.57
CSeq: 102 INVITE
Allow: UPDATE,PRACK,INFO,NOTIFY,OPTIONS,BYE,INVITE,ACK,CANCEL
Contact: <sip:07973xxxxxx@82.16.19.2:5060>
Content-Type: application/sdp
Content-Length: 252


I then see this

 

v=0
o=CiscoSystemsSIP-GW-UserAgent 3057 3584 IN IP4 10.0.12.30
s=SIP Call
c=IN IP4 10.0.12.30
t=0 0
m=audio 8120 RTP/AVP 8 101
c=IN IP4 10.0.12.30
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

Jul 14 15:51:06.086: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received: 
PRACK sip:07973xxxxxx@10.0.12.30:5060;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.0.12.17:5060;branch=z9hG4bK60b1101193f8
From: "SiP Test 45000" <sip:45000@10.0.12.17>;tag=67135~9c2c55f8-f6da-4001-942f-f8cb268e11d0-36791270
To: <sip:07973xxxxxx@10.0.12.30>;tag=13D3F15-1794
Date: Tue, 14 Jul 2020 15:51:05 GMT
Call-ID: d2010000-f0d1d469-3f0d-110c000a@10.0.12.17
User-Agent: Cisco-CUCM11.5
CSeq: 102 PRACK
RAck: 5309 101 INVITE
Allow-Events: presence
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 223

Is this sequence correct:
CSeq: 102 PRACK
RAck: 5309 101 INVITE

 

After I see my CSeq 102 PRACK and ACK

v=0
o=CiscoSystemsCCM-SIP 67135 1 IN IP4 10.0.12.17
s=SIP Call
c=IN IP4 172.19.105.56
b=TIAS:64000
b=AS:64
t=0 0
m=audio 27026 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

Jul 14 15:51:06.088: //3691/D20100000000/SIP/Msg/ccsipDisplayMsg:
Sent: 
SIP/2.0 200 OK
Via: SIP/2.0/TCP 10.0.12.17:5060;branch=z9hG4bK60b1101193f8
From: "SiP Test 45000" <sip:45000@10.0.12.17>;tag=67135~9c2c55f8-f6da-4001-942f-f8cb268e11d0-36791270
To: <sip:07973xxxxxx@10.0.12.30>;tag=13D3F15-1794
Date: Tue, 14 Jul 2020 15:51:06 GMT
Call-ID: d2010000-f0d1d469-3f0d-110c000a@10.0.12.17
Server: Cisco-SIPGateway/IOS-16.6.6
CSeq: 102 PRACK
Session-ID: f8a7a6d78edc579ab2ba35900b096ba7;remote=4fe824fef3d83993017e11e24aa67130
Content-Length: 0

Jul 14 15:51:13.210: //3692/D20100000000/SIP/Msg/ccsipDisplayMsg:
Received: 
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 10.2.251.57:5060;branch=z9hG4bK91C18C
To: <sip:07973xxxxxx@82.16.19.2>;tag=3803730666-1791299518
From: "SiP Test 45000" <sip:45000@10.2.251.57;user=phone>;tag=13D3EBA-26AE
Call-ID: A9658480-C52011EA-8EF3AD4F-A1EA3405@10.2.251.57
CSeq: 102 INVITE
Allow: UPDATE,PRACK,INFO,NOTIFY,OPTIONS,BYE,INVITE,ACK,CANCEL
Contact: <sip:07973xxxxxx@82.16.19.2:5060>
Content-Length: 0

Jul 14 15:51:13.211: //3692/D20100000000/SIP/Msg/ccsipDisplayMsg:
Sent: 
ACK sip:07973xxxxxx@82.16.19.2:5060 SIP/2.0
Via: SIP/2.0/UDP 10.2.251.57:5060;branch=z9hG4bK91C18C
From: "SiP Test 45000" <sip:45000@10.2.251.57>;tag=13D3EBA-26AE
To: <sip:07973xxxxxx@82.16.19.2>;tag=3803730666-1791299518
Date: Tue, 14 Jul 2020 15:51:06 GMT
Call-ID: A9658480-C52011EA-8EF3AD4F-A1EA3405@10.2.251.57
Max-Forwards: 70
CSeq: 102 ACK
Allow-Events: telephone-event
Content-Length: 0

Followed by 

Jul 14 15:51:13.214: //3691/D20100000000/SIP/Msg/ccsipDisplayMsg:
Sent: 
SIP/2.0 500 Internal Server Error
Via: SIP/2.0/TCP 10.0.12.17:5060;branch=z9hG4bK60b01ac8f481
From: "SiP Test 45000" <sip:45000@10.0.12.17>;tag=67135~9c2c55f8-f6da-4001-942f-f8cb268e11d0-36791270
To: <sip:07973xxxxxx@10.0.12.30>;tag=13D3F15-1794
Date: Tue, 14 Jul 2020 15:51:06 GMT
Call-ID: d2010000-f0d1d469-3f0d-110c000a@10.0.12.17
CSeq: 101 INVITE
Allow-Events: kpml, telephone-event
Server: Cisco-SIPGateway/IOS-16.6.6
Reason: Q.850;cause=16
Session-ID: f8a7a6d78edc579ab2ba35900b096ba7;remote=4fe824fef3d83993017e11e24aa67130
Session-Expires:  1800
Content-Length: 0

Jul 14 15:51:13.216: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received: 
ACK sip:07973xxxxxx@10.0.12.30:5060 SIP/2.0
Via: SIP/2.0/TCP 10.0.12.17:5060;branch=z9hG4bK60b01ac8f481
From: "SiP Test 45000" <sip:45000@10.0.12.17>;tag=67135~9c2c55f8-f6da-4001-942f-f8cb268e11d0-36791270
To: <sip:07973xxxxxx@10.0.12.30>;tag=13D3F15-1794
Date: Tue, 14 Jul 2020 15:51:05 GMT
Call-ID: d2010000-f0d1d469-3f0d-110c000a@10.0.12.17
User-Agent: Cisco-CUCM11.5
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: presence
Content-Length: 0

Which as you can see is CSeq: 101 again

 

Hope you can help me out further

 

Thanks

Noticed in your PAI header (in INVITE) for outgoing call leg to ITSP, you just have your extension 'SiP Test 45000" in there. ITSP would want some sort of authentication before they can route the call for customer. I think you're going to need some NR /trunk number in PAI in OUTGOING SIP invite (CallID:C544F9D6-C40E11EA-B9A3B5A2-101A3230) ... Secondly your ITSP is sending you 183 ringing whereas they should ideally be sending 200 OK.  

 

You'd have to create a SIP profile on your CUBE to modify the PAI header if it's that.

Don't want to hijack your troubleshooting as it could be so many things but PAI being sent in INVITE could be one.  

I've mentioned the 183 ringing and they say that is OK, as this is a new cube and we are just testing the service at the minute, what would need to be in the profile to modify the modify the PAI header

 

Thanks

voice class sip-profiles 1
request INVITE sip-header P-Asserted-Identity modify "(.*)" "P-Asserted-Identity: <sip:XXXXX@yourSIPRealm>"

XXXXX being your authentication user ID. 

 

Apply the SIP profile on ITSP facing dialpeer (outgoing)
voice-class sip profiles 1


That  could be just one of the reasons ITSP could be not letting the call thru. Another test could be to update the matching route pattern in CUCM 'Calling Party Transform Mask' field with valid DDI on your SIP range. 

@Nabeel Minhas If Authentication / Authorization mismatch case, the Telco would not allow to place the call. The 183 session progress would also not seen in the logs that is shared by @martynch1


i think @martynch1 you need to share the configuration and the expected call flow  with expected matching dial-peers. 
peers here can have a look. I will try best if available tomorrow to check and suggest.

*** Please rate helpful post. Please mark as answer if it solves your problem/query.
regards, Ritesh Desai

I think that the 183 early media is sending the announcement, the "your call can't be completed" message.  That would be consistent with the carrier following that with a termination.  A Wireshark capture would confirm, you can listen to the captured RTP.

I still think the calling number could be an issue.  Many SIP providers have their own requirement for the presented calling number, and these vary from provider to provider.  The OP earlier showed a debug of a successful outbound call, in that call the calling number was presented in full E164 format.

@Nabeel Minhas When telecom is sending 183 Ringing which means telecom provider is configured with EARLY MEDIA so that ringback is played. These is normal and should not be issue in my opinion.

*** Please rate helpful post. Please mark as answer if it solves your problem/query.
regards, Ritesh Desai

The sequence number (CSeq) in a Request is it's own label, so other devices can distinguish a repeat or retry of the same request vs a new request.   In a Response the CSeq indicates which request the Response is responding to.  So for example you send one Invite CSeq 101, then the authenticated Invite is CSeq 102.  Further down you can see that the 183 Response includes "CSeq: 102 INVITE" to show which Invite it's related to.

The 183 with early media is perfectly normal.  All it means is "the call's not yet connected but there's some audio".  In this case presumably the audio is the announcement rejecting the call. 

Before we go into detail about how to change headers, has the ITSP actually said how they want the header formatted and what they want it to contain?   It's impossible to guess, there is no common convention.  Some want the "Contact" header modifier, some want some extra parameter added, etc etc. 

Unless someone is using the same carrier as you.

If you want to try things a random an easy test would be to set external number mask of something so you put out a valid DDI in place of the internal number of "45000"

Hi,

First of all let me say well done for attempting to break down the logs. I am always a big fan of people who make efforts and show diligence in what they do.

I would like to correct a few things on your analysis, but before I do, let me get to the issue you are facing first.

Here is a sip ladder of your trace (next time please attach the logs rather)forumsCapture.PNG

 

From the ladder, you can see where the issue is.

1. at 7:10:30 your ITSP sends a 183 Session progress with SDP ( indicating that they want to do early media)

2. At 7:10:37 ( 7s after), your get a request terminated from them.

Normally you should get a 200 OK with the same SDP parameter as the 183 Session progress.

Question then is why...did we not get a 200 OK.

My guess is that they want an ACK of the 183 session progress even though they have not requested a PRACK. This is where you need to go back to them and ask Why they are not sending a 200 OK. Do they want PRACK for 183 if they do they need to include "Require: 100rel" in their SDP

 

++ Back to your PRACK analysis ++

PRACK uses a combination of Rseq and Rack numbers to identify the exact PRACK messages.

In this log,

CUBE sends PRACK to CUCM

Sent:
SIP/2.0 183 Session Progress
Via: SIP/2.0/TCP 10.0.12.17:5060;branch=z9hG4bK1eead65cae8f3
From: "VM SIP TEST 2" <sip:45000@10.0.12.17>;tag=404459~9c2c55f8-f6da-4001-942f-f8cb268e11d0-49787915
To: <sip:07973xxxxxx@10.0.12.30>;tag=1480316D-1AD7
Date: Mon, 13 Jul 2020 07:10:30 GMT
Call-ID: ee147c80-f0c108e6-15b0f-110c000a@10.0.12.17
CSeq: 101 INVITE
Require: 100rel
RSeq: 8573

 

++ CUCM sends PRACK with Rack of the Rseq number ++

Jul 13 07:10:30.793: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
PRACK sip:07973xxxxxx@10.0.12.30:5060;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.0.12.17:5060;branch=z9hG4bK1eeaf1d80c380
From: "VM SIP TEST 2" <sip:45000@10.0.12.17>;tag=404459~9c2c55f8-f6da-4001-942f-f8cb268e11d0-49787915
To: <sip:07973xxxxxx@10.0.12.30>;tag=1480316D-1AD7
Date: Mon, 13 Jul 2020 07:10:30 GMT
Call-ID: ee147c80-f0c108e6-15b0f-110c000a@10.0.12.17
User-Agent: Cisco-CUCM11.5
CSeq: 102 PRACK
RAck: 8573 101 INVITE

 

++ CUBE sends a 200 OK to complete PRACK ++

Jul 13 07:10:30.796: //96514/EE147C800000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/TCP 10.0.12.17:5060;branch=z9hG4bK1eeaf1d80c380
From: "VM SIP TEST 2" <sip:45000@10.0.12.17>;tag=404459~9c2c55f8-f6da-4001-942f-f8cb268e11d0-49787915
To: <sip:07973xxxxxx@10.0.12.30>;tag=1480316D-1AD7
Date: Mon, 13 Jul 2020 07:10:30 GMT
Call-ID: ee147c80-f0c108e6-15b0f-110c000a@10.0.12.17
Server: Cisco-SIPGateway/IOS-16.6.6
CSeq: 102 PRACK

 

++ In your log example, you are confusing the dialog between CUCM and CUBE and that between CUBE and  ITSP ++

This PRACK is from CUCM and The INVITE to CUCM is Cseq 101 and the 183 Session progress has Rseq 5309 for PRACK

Jul 14 15:51:06.086: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received: 
PRACK sip:07973xxxxxx@10.0.12.30:5060;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.0.12.17:5060;branch=z9hG4bK60b1101193f8
From: "SiP Test 45000" <sip:45000@10.0.12.17>;tag=67135~9c2c55f8-f6da-4001-942f-f8cb268e11d0-36791270
To: <sip:07973xxxxxx@10.0.12.30>;tag=13D3F15-1794
Date: Tue, 14 Jul 2020 15:51:05 GMT
Call-ID: d2010000-f0d1d469-3f0d-110c000a@10.0.12.17
User-Agent: Cisco-CUCM11.5
CSeq: 102 PRACK
RAck: 5309 101 INVITE
Allow-Events: presence
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 223

 

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