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Outgoing Calls SiP Cube

martynch1
Beginner
Beginner

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

If I call my mobile from the office, I see exactly that same sequence.  We receive 183/SDP from the carrier but we don't get 200 OK until the call is answered.  If I just let it ring I hear inband ringback.   Hears an example where I let it ring for a bit then cancel from the caller's side.  Like the example posted the 183/SDP has "Require: 100rel" and "RSeq: 5344" on the message sent to CUCM, but not the message received from the ITSP.

So I really think that sequence is OK, and the fact that he hears the message saying that the call can't be placed also suggests the same.  The audio/RTP path must in fact be established.  Unless the message is a CUCM one, but it doesn't sound that way.

In this screenshot I've cropped the address off the top but from left to right they are CUCM, CUBE Inside, CUBE Outside and ITSP Proxy.

 

 

Thanks, are you using the same call flow diagram as Ayodeji Okanlawon, if so what's the name of it?

 

Thanks

"TranslatorX" https://translatorx.org/

Free download and can read and filter various sorts of debugs, and also CUCM traces.  Very easy to use.

@TONY SMITH 

No the sequence is not okay. SIP does not work the way you have described..

For media to be established in SIP, a final response must be sent. For a successful call, that would be a 200 OK

SIP is an Offer/Answer Model and in this flow, here a view of what is going on...

 

++ 

1. CUBE sends INVITE with Offer to ITSP
2. ITSP sends an answer With 183 SDP ( Offer/Answer completed here )
to complette the exchange ITSP must send 200 OK

3. CUBE sends OFFER with 183 SDP to CUCM ( offer contains all the SDP received from ITSP)

4. CUCM sends Answer in PRACK to CUBE ( offer/answer model complete also)

++

Based on SIP RFC:

https://tools.ietf.org/html/draft-rosenberg-sip-early-media-00

 

Current implementations support early media through the 183 response code, which was first described in a now-expired Internet Draft. When the called party wishes to send early media to the caller, the called party sends a 183 response to the caller. That response contains SDP. When the caller receives the 183, it suppresses any local alerting of the user (for example, audible ringtones or a pop-up window), and begins playing out media that it receives. The SDP in the 183  provides an address to which RTCP packets can be sent.

If the call is ultimately rejected, the called party generates a non-2xx final response

However, if the call is accepted, the called party generates a 2xx (generally, with the same SDP as was present in the 183), and sends that to the caller. Media transmission continues as before.

NB: That media starts playing at the 183 SDP ( But 200 OK) must follow to complete the transaction/dialog

 

So in this case, the lack of 200 OK is the issue here...

And I have so many logs showing 200 OK followed by 183 Session progress, this is the right sequence

 

NB: the 487 received is also a valid response. This is the final response to the SIP INVITE and a 487 is valid however this may not be what is expected here. So the ONUS is on the ITSP to explain why a 487 is sent in stead of a 200 OK

Please rate all useful posts


@Ayodeji Okanlawon wrote:

@TONY SMITH 

No the sequence is not okay. SIP does not work the way you have described..

For media to be established in SIP, a final response must be sent. For a successful call, that would be a 200 OK

 


I can only restate that in the example I posted audio was established following the 183/SDP, with no 200 OK received. The attached screenshot is from another gateway where I grabbed a Wireshark so as well hearing ringback, I can see the RTP in the capture and indeed play it back.  That is from a customer installation so I would have to blank out quite a bit before posting any more detail, but please let me know if you would like me to do that.  To summarise the RTP started immediately after 183/SDP, indeed it was the very next packet in the capture, and continued unbroken when 200 OK was received a few seconds later when the call was picked up.

The behaviour may well vary between providers, and maybe with CUBE configuration as well.  I have seen traces where 183/SDP was followed by PRAC and then OK, although in that case the OK was a response to the PRACK and not to the Invite.

I have also seen cases where an inband announcement for call failure has been sent following an OK for the Invite - unhelpful as the call appears to have completed normally when you just look at the SIP.

Screenshot_97.png

@TONY SMITH 

This call flow you posted is the right one. 

When we get 183 with SDP, media flows immediately however 200 OK must still follow for a normal call to proceed. That is the final response to the INVITE. In all cases regardless of ITSP, this is by the book ie RFC. 

In cases where you don't get 200 OK, you must still get a final response ie 487. And  its totally possible to play an announcement using 183 with SDP, followed by a 487...

I can only assume that you didn't wait long enough for the 200 OK Or something else is going on..

Please rate all useful posts


@Ayodeji Okanlawon wrote:

@TONY SMITH 

This call flow you posted is the right one. 

<snip>

I can only assume that you didn't wait long enough for the 200 OK Or something else is going on..

Sounds like we're in agreement, which is good.  And yes in that first trace I deliberately did not wait too long but cancelled before the call connected, purely to demonstrate two way RTP without a 200 OK.

Hi all, this is the response from the ITSP, they are placing the issues firmly on our side.

 

Thanks

 

You do not set up an RTP stream, and the lack of that is what kills the call

 

“our 183 indicates that we want early media” – not strictly correct.

Your INVITE with SDP indicates that YOU want early media, which our 183 “agrees” with.

 

The issue is no RTP steam from yourself


@martynch1 wrote:

“our 183 indicates that we want early media” – not strictly correct.

Your INVITE with SDP indicates that YOU want early media, which our 183 “agrees” with.

They're correct regarding 183.  The 183 response, with SDP, is an example of "Early Media" meaning audio is carried before the call connects.

Can you just fully confirm that the "Sorry the service you require can not be connected" message is heard at the caller's end, ie on your CUCM phone?  If so then clearly RTP of some sort is indeed passing.  You may have to resort to a Wireshark capture to prove it if they are as obstinate as some providers.

The second line is incorrect.  INVITE with SDP is "Early Offer", not the same as Early Media.   Early Offer means the call's audio parameters are included in the initial Invite rather than provided later only when the call connects (Delayed Offer).  You Invite from CUCM to the gateway, the first Invite in your debug, is Delayed Offer.

Delayed Offer does not prevent Early Media.  Again in your debug your CUCM Invite is DO, no SDP included.  Then CUCM receives the 183/SDP (Early Media) and responds with PRACK to provide its SDP.

Just like @TONY SMITH has mentioned, if you hear prompt about the service, then there is media passing through. Take a packet capture from CUBE send it over. Escalate the issue. Most times you are dealing with leve1/level 2 engineers who do not know the complexity of SIP solution.

in your packet capture you should see media flowing through after you receive 183 Session progress from ITSP

Please rate all useful posts

Hi guys, sorry for the delay, please find attached my trace from the cube, I've filtered on tcp any eq 5060 - hope that's ok.

I will step through myself to try and look into the issue, I see that we are now getting a 500 internal error, now sure why or if it's our 3rd party maintainers have been looking at this too for us as we have a ticket raised with them.

 

Looking forward to your thoughts as I'm gaining quite a bit of new knowledge here, thanks all.

 

Martyn

You will need to refine your filtering as that capture only shows packets from 10.012.30 to 10.0.12.17, which I think is your CUBE to your CUCM.   It doesn't even show the reply.

Filtering for TCP port 5060 will only show SIP signalling, and only if it uses TCP.  Your ITSP dial peers are specifically configured for UDP.   In any case you want to see the RTP as well as the signalling, which is UDP but with dynamically chosen port numbers.

Now you've got the packet capture at least working, I'd suggest filtering for anything going between your CUBE and the ITSP end point.

By the way that 500 error sent back to Callmanager is not necessarily a different symptom.  It's probably just the CUBE's way of telling CUCM that it's received an error from the provider.

I think we need to concentrate on the CUBE <> ITSP leg.

Well Thursday afternoon and all of Friday was wiped out because we had our 3rd party maintainers trying to help us out, its a service we do pay or after all, wasted two day as we have gone backwards, we now can not receive calls never mind make them, our test bed as now closed. We got close at one point when they put the full e164 on the phone itself, we could make an outgoing call, answer it and have a conversation, soon after that when they had played with out CUCM and cube - Nothing!

 

Here are my traces from the cube both outgoing and incoming, not sure if it shows where we are failing, today I will mostly be checking the configs before they were changed to see if I can get back to where we were on Thursday.

We are now struggling because we don't have the ITSP test bed to use.

 

Thanks all for your support

Martyn

Your captures still only show conversation between CUCM and your CUBE.  We really need to see the conversation between CUBE and ITSP as well.

However looking at what you say, and your "outbound-working" capture it looks like the main difference is your calling number, which was an internal number "45000" in all the failed call examples, but "+441158045000" in your working example.

You say your maintainer did this by putting E164 number as the actual DN in CUCM.  Of course that's one way to do it, but not necessarily the best way.  

Can you tell us a bit more about your numbering plan and the CUCM configuration?  Making a wild guess the most common practice would be to have the internal number "45000" as the DN, external number mask converting that to the most commonly used national format for the location (for example mask of "011580XXXXX" would convert to "01158045000").

If your ITSP wants numbers in E164 format you also have the option of converting with a voice translation rule on the gateway itself.   We can help you with that.

Hi all, thanks for the replies and sorry fro my delay in responding, our test bed was taken off line by our ITSP so unable to peruse this right now, we've asked them for an extension to testing and if approved I will get a full capture over to you.

 

Regards

Martyn

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