Sorry for the late reply pal. Been busy. I configured the CUBE as a HW MTP and it sorted it. Now no MOH even though the SIP trunk is assigned an MOH server. Im thinking maybe its because of the g729br8 codec as i dont even see CUCM invoke MOH. ... View more

Hi, There are now problems with supplemntary services.  A debug seems to show the audio inactive messages being passed to the ITSP and the ITSP sending an ok.  The CUBE then sends an ACK and the a BYE to end the call. Debug below: *May 28 15:22:22.687: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Received: INVITE sip:0824636267@192.168.244.250:5060 SIP/2.0 Via: SIP/2.0/UDP 10.128.16.21:5060;branch=z9hG4bK27bcd7da7872c From: ;tag=9a6faa78-a360-49d5-b7da-21602c33796a-44746543 To: ;tag=1F99CD74-669 Date: Tue, 28 May 2013 15:24:03 GMT Call-ID: 37657887-C6E111E2-85E0F85B-C341310C@192.168.244.250 Supported: timer,resource-priority,replaces Min-SE:  1800 Cisco-Guid: 0929196991-3336638946-2245589083-3275829516 User-Agent: Cisco-CUCM8.0 Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY CSeq: 101 INVITE Max-Forwards: 70 Contact: Expires: 180 Allow-Events: presence Supported: X-cisco-srtp-fallback Supported: Geolocation Session-Expires:  1800;refresher=uac P-Asserted-Identity: "Test JR" Remote-Party-ID: "Test JR" ;party=calling;screen=yes;privacy=off Content-Type: application/sdp Content-Length: 221 v=0 o=CiscoSystemsCCM-SIP 2000 2 IN IP4 10.128.16.21 s=SIP Call c=IN IP4 0.0.0.0 t=0 0 m=audio 30060 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=ptime:20 a=inactive a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 *May 28 15:22:22.695: //495/37626BBF85D8/SIP/Msg/ccsipDisplayMsg: Sent: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.128.16.21:5060;branch=z9hG4bK27bcd7da7872c From: ;tag=9a6faa78-a360-49d5-b7da-21602c33796a-44746543 To: ;tag=1F99CD74-669 Date: Tue, 28 May 2013 15:22:22 GMT Call-ID: 37657887-C6E111E2-85E0F85B-C341310C@192.168.244.250 CSeq: 101 INVITE Allow-Events: telephone-event Server: Cisco-SIPGateway/IOS-15.2.4.M2 Content-Length: 0 *May 28 15:22:22.695: //493/37626BBF85D8/SIP/Msg/ccsipDisplayMsg: Sent: INVITE sip:0824636267@192.168.244.10:5060;user=phone;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.244.250:5060;branch=z9hG4bK2D02489 From: ;tag=1F99CE18-20DF To: ;tag=isn3930u-CC-29 Date: Tue, 28 May 2013 15:22:22 GMT Call-ID: rwswtmxrwmxsrqxtviiu03qtuvb1u0im@SoftX3000 Supported: 100rel,timer,resource-priority,replaces,sdp-anat Min-SE:  1800 Cisco-Guid: 0929196991-3336638946-2245589083-3275829516 User-Agent: Cisco-SIPGateway/IOS-15.2.4.M2 Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER CSeq: 101 INVITE Max-Forwards: 70 Timestamp: 1369754542 Contact: Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Length: 289 v=0 o=CiscoSystemsSIP-GW-UserAgent 3683 3901 IN IP4 192.168.244.250 s=SIP Call c=IN IP4 192.168.244.250 t=0 0 m=audio 17364 RTP/AVP 18 97 c=IN IP4 192.168.244.250 a=inactive a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-15 a=ptime:20 *May 28 15:22:22.755: //493/37626BBF85D8/SIP/Msg/ccsipDisplayMsg: Received: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.244.250:5060;branch=z9hG4bK2D02489 From: ;tag=1F99CE18-20DF To: ;tag=isn3930u-CC-29 Call-ID: rwswtmxrwmxsrqxtviiu03qtuvb1u0im@SoftX3000 CSeq: 101 INVITE Timestamp: 1369754542 Contact: Content-Length: 243 Content-Type: application/sdp v=0 o=HuaweiSoftX3000 15910632 15910633 IN IP4 192.168.244.10 s=Sip Call c=IN IP4 192.168.244.10 t=0 0 m=audio 20342 RTP/AVP 18 97 a=rtpmap:18 G729/8000 a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-15 a=fmtp:18 annexb=no a=inactive *May 28 15:22:22.755: //493/37626BBF85D8/SIP/Msg/ccsipDisplayMsg: Sent: ACK sip:0824636267@192.168.244.10:5060;user=phone;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.244.250:5060;branch=z9hG4bK2D1FE From: ;tag=1F99CE18-20DF To: ;tag=isn3930u-CC-29 Date: Tue, 28 May 2013 15:22:22 GMT Call-ID: rwswtmxrwmxsrqxtviiu03qtuvb1u0im@SoftX3000 Max-Forwards: 70 CSeq: 101 ACK Allow-Events: telephone-event Content-Length: 0 *May 28 15:22:22.759: //493/37626BBF85D8/SIP/Msg/ccsipDisplayMsg: Sent: BYE sip:0824636267@192.168.244.10:5060;user=phone;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.244.250:5060;branch=z9hG4bK2D2B11 From: ;tag=1F99CE18-20DF To: ;tag=isn3930u-CC-29 Date: Tue, 28 May 2013 15:22:22 GMT Call-ID: rwswtmxrwmxsrqxtviiu03qtuvb1u0im@SoftX3000 User-Agent: Cisco-SIPGateway/IOS-15.2.4.M2 Max-Forwards: 70 Timestamp: 1369754542 CSeq: 102 BYE Reason: Q.850;cause=65 P-RTP-Stat: PS=123,OS=2460,PR=112,OR=2240,PL=0,JI=0,LA=0,DU=2 Content-Length: 0 *May 28 15:22:22.759: //495/37626BBF85D8/SIP/Msg/ccsipDisplayMsg: Sent: BYE sip:0218169019@10.128.16.21:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.244.250:5060;branch=z9hG4bK2D3792 From: ;tag=1F99CD74-669 To: ;tag=9a6faa78-a360-49d5-b7da-21602c33796a-44746543 Date: Tue, 28 May 2013 15:22:22 GMT Call-ID: 37657887-C6E111E2-85E0F85B-C341310C@192.168.244.250 User-Agent: Cisco-SIPGateway/IOS-15.2.4.M2 Max-Forwards: 70 Timestamp: 1369754542 CSeq: 102 BYE Reason: Q.850;cause=65 P-RTP-Stat: PS=112,OS=2240,PR=130,OR=2600,PL=0,JI=0,LA=0,DU=2 Content-Length: 0 *May 28 15:22:22.795: //493/37626BBF85D8/SIP/Msg/ccsipDisplayMsg: Received: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.244.250:5060;branch=z9hG4bK2D2B11 From: ;tag=1F99CE18-20DF To: ;tag=isn3930u-CC-29 Call-ID: rwswtmxrwmxsrqxtviiu03qtuvb1u0im@SoftX3000 Timestamp: 1369754542 CSeq: 102 BYE Content-Length: 0 *May 28 15:22:22.919: //495/37626BBF85D8/SIP/Msg/ccsipDisplayMsg: Received: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.244.250:5060;branch=z9hG4bK2D3792 From: ;tag=1F99CD74-669 To: ;tag=9a6faa78-a360-49d5-b7da-21602c33796a-44746543 Date: Tue, 28 May 2013 15:24:03 GMT Call-ID: 37657887-C6E111E2-85E0F85B-C341310C@192.168.244.250 CSeq: 102 BYE Content-Length: 0 Thanks Kev ... 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Aye mate, I already have. Thanks for the use of your eyes much appreciated. Cheers Kev ... View more

Hi, I realised after my last post that the sip was bound to the wrong interface and so it was in a different VLAN. I changed the bind to .300 and changed the ip on the trunk on UCM and bang its fixed. Cheers ... View more

Ive attached the config. The control and media are bound to gi0/1.300 under dial-peer 1. Cheers ... View more

Ok its connecting now at both ends but im getting one way voice called can hear caller but not the other way around. Call then drops after about 20 seconds. Debug attached. Cheers ... View more

Disabled rel1xx on dial-peer 1. Makes sense as this is the dial-peer with the incoming called number on it. Must stop the re1xx being sent from the cube when it hits this dial-peer. Im going to debug it to confirm. Cheers ... View more

Default SIP profile has rel1xx options disabled on the CUCM. Cheers ... View more

You reckon this is being passed on from the CUCM then? I was starting to have the same thoughts as ive disabled rel1xx on the dialpeer and its still the same. But i still see the 100rel required in the session progress out to the Huwawei PBX. ... View more

Hi, My config is the same as before thats the only bit that has changed but can upload it. PRACK isnt configured. It would appear it is on by default. Apparently you can disable it under voice service voip > sip > rel1xx but there are 3 options. Im going to look into each one and test accordingly. My config is the same as before thats the only bit that has changed but can upload it. Ill let you know. Cheers Kev ... View more

Hi, Ok ive done two calls. One with g729b configured on the cube and second one where the CUCM Trunk is set to use mtp with g729b. Relevent config below: voice class codec 1 codec preference 1 g729br8 codec preference 2 g729r8 codec preference 3 g711alaw ! dial-peer voice 1 voip description To/from CUCM destination-pattern .......... session protocol sipv2 session target ipv4:10.128.16.21 incoming called-number . voice-class codec 1 voice-class sip bind control source-interface GigabitEthernet0/1.200 voice-class sip bind media source-interface GigabitEthernet0/1.200 dtmf-relay rtp-nte no vad ! dial-peer voice 2 voip description outbound SIP destination-pattern .T session protocol sipv2 session target sip-server incoming called-number 9T voice-class codec 1 voice-class sip dtmf-relay force rtp-nte dtmf-relay rtp-nte The leg from ITSP to CUBE still isnt connecting and i see the invite go out with annex b to the CUCM: *May 15 18:34:12.727: //116/DFB251008145/SIP/Msg/ccsipDisplayMsg: Sent: INVITE sip:0218169019@10.128.16.21:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.60.254:5060;branch=z9hG4bK60162C Remote-Party-ID: ;party=calling;screen=no;privacy=off From: ;tag=89D3A8-1360 To: Date: Wed, 15 May 2013 18:34:12 GMT Call-ID: DFB38950-BCC411E2-814C82C4-744629B4@192.168.60.254 Supported: 100rel,timer,resource-priority,replaces,sdp-anat Min-SE:  1800 Cisco-Guid: 3753005312-3166966242-2168816324-1950755252 User-Agent: Cisco-SIPGateway/IOS-15.2.4.M2 Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER CSeq: 101 INVITE Timestamp: 1368642852 Contact: Expires: 180 Allow-Events: telephone-event Max-Forwards: 63 Content-Type: application/sdp Content-Disposition: session;handling=required Content-Length: 313 v=0 o=CiscoSystemsSIP-GW-UserAgent 6576 6664 IN IP4 192.168.60.254 s=SIP Call c=IN IP4 192.168.60.254 t=0 0 m=audio 16554 RTP/AVP 8 0 18 101 c=IN IP4 192.168.60.254 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 Thanks for you persistence with this. I have attached two logs. Cheers Kev ... View more

Hi I spoke to soon. I passed th info onto my colleague and have just tried it. The sip leg between the CUBE and the ITSP isnt connecting: Total call-legs: 2 12DA : 87 3803600ms.1 (*17:07:03.715 UTC Wed May 15 2013) +-1 pid:1 Answer 447730540752 connecting dur 00:00:00 tx:576/11520 rx:0/0 dscp:0 media:0 audio tos:0xB8 video tos:0x0 IP 192.168.244.10:20092 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g729br8 TextRelay: off Transcoded: No media inactive detected:n media contrl rcvd:n/a timestamp:n/a long duration call detected:n long duration call duration:n/a timestamp:n/a 12DA : 89 3803600ms.2 (*17:07:03.715 UTC Wed May 15 2013) +1430 pid:1 Originate 0218169019 active dur 00:00:11 tx:0/0 rx:583/11660 dscp:0 media:0 audio tos:0xB8 video tos:0x0 IP 192.168.60.11:23262 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g729br8 TextRelay: off Transcoded: No media inactive detected:n media contrl rcvd:n/a timestamp:n/a long duration call detected:n long duration call duration:n/a timestamp:n/a This happens if the g729br8 is in the voice class list or not. Ive attached a debug ccsip messages. A debug ccsip all seems to kill the router which is why the duplicate information was being shown. Cheers Kev ... View more