Kristan Posted December 19, 2007 Report Share Posted December 19, 2007 Hi All, We have a customer with a Mitel 3300ICP who would to add a remote office with PBXnSIP. We've managed to get some of the mitel setup done (my god it makes you appreciate how simple PBXnSIP is to get running!) but are falling down with the audio. If I make a call from the mitel to an unregistered extension on the PBX (to get voicemail), after the initial invite/trying, the PBX sends the 200 OK and gets this back in response : [7] 2007/12/19 10:31:34: SIP Rx udp:192.168.10.2:5060: ACK sip:602@192.168.10.240:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.10.2:5060;branch=z9hG4bK281759984-57239807 Max-Forwards: 70 From: "xxx" <sip:228@192.168.10.2>;tag=MitelICP_281599984-57239806 To: <sip:602@192.168.10.240>;tag=b0bae1689f Call-ID: 281599984-57239804 CSeq: 1 ACK Contact: "xxx" <sip:228@192.168.10.2:5060;transport=udp> Content-Type: application/sdp Content-Length: 185 v=0 o=- 14633181335402739584 14633181335402739584 IN IP4 192.168.10.3 s=- c=IN IP4 192.168.10.3 t=0 0 m=audio 20199 RTP/AVP 8 96 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 [6] 2007/12/19 10:31:34: Sending RTP for 281599984-57239804#b0bae1689f to 192.168.10.3:20199 [8] 2007/12/19 10:31:34: No codec available for sending Mitel is on 192.168.10.2, PBX is on 192.168.10.240 - I've no idea what 192.168.10.3 is, I presume it's the IP of the phone on the mitel side making a call. Now PBXnSIP offers all it's supported codecs, but the mitel isn't negotiating any. If I do the same with one of our asterisk boxes, it looks like it just sends PCMU regardless of what the mitel says (which seems to work). Any ideas??? Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted December 19, 2007 Report Share Posted December 19, 2007 Hmm, codec 8 is alaw, which is supported by the PBX... Looks like the initial INVITE does not contain a SDP, so the 200 Ok of the PBX must the the SDP offer. This is legal, but unusual. It would be interesting to see the whole INVITE/200/ACK sequence. Quote Link to comment Share on other sites More sharing options...
Kristan Posted December 20, 2007 Author Report Share Posted December 20, 2007 Your wish is my command: [7] 2007/12/19 10:31:34: SIP Rx udp:192.168.10.2:5060: INVITE sip:602@192.168.10.240:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.10.2:5060;branch=z9hG4bK281599984-57239805 Route: <sip:192.168.10.240:5060;transport=udp;lr> Max-Forwards: 70 Allow: INVITE,BYE,CANCEL,ACK,INFO,PRACK,COMET,OPTIONS,SUBSCRIBE,NOTIFY,MESSAGE,REFER,REGISTER,UPDATE Supported: timer,replaces From: "xxx" <sip:228@192.168.10.2>;tag=MitelICP_281599984-57239806 To: <sip:602@192.168.10.240> Call-ID: 281599984-57239804 CSeq: 1 INVITE Min-SE: 90 Session-Expires: 90;Refresher=uas Contact: "xxx" <sip:228@192.168.10.2:5060;transport=udp> P-Asserted-Identity: "xxx" <sip:228@192.168.10.2:5060;transport=udp> Content-Length: 0 [7] 2007/12/19 10:31:34: UDP: Opening socket on port 61252 [7] 2007/12/19 10:31:34: UDP: Opening socket on port 61253 [5] 2007/12/19 10:31:34: Identify trunk (IP address/port and domain match) 2 [9] 2007/12/19 10:31:34: Resolve destination 69745: aaaa udp 192.168.10.2 5060 [9] 2007/12/19 10:31:34: Resolve destination 69745: a udp 192.168.10.2 5060 [9] 2007/12/19 10:31:34: Resolve destination 69745: udp 192.168.10.2 5060 [7] 2007/12/19 10:31:34: SIP Tx udp:192.168.10.2:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.10.2:5060;branch=z9hG4bK281599984-57239805 From: "xxx" <sip:228@192.168.10.2>;tag=MitelICP_281599984-57239806 To: <sip:602@192.168.10.240>;tag=b0bae1689f Call-ID: 281599984-57239804 CSeq: 1 INVITE Content-Length: 0 [5] 2007/12/19 10:31:34: Trunk Mitel Trunk sends call to 602 [7] 2007/12/19 10:31:34: Calling extension 602 [8] 2007/12/19 10:31:34: Play audio_en/mb_this_is_the_mailbox_of.wav audio_en/bi_6.wav audio_en/bi_0.wav audio_en/bi_2.wav audio_en/mb_leave_msg_after_tone.wav audio_moh/mb_beep.wav [9] 2007/12/19 10:31:34: Resolve destination 69746: aaaa udp 192.168.10.2 5060 [9] 2007/12/19 10:31:34: Resolve destination 69746: a udp 192.168.10.2 5060 [9] 2007/12/19 10:31:34: Resolve destination 69746: udp 192.168.10.2 5060 [7] 2007/12/19 10:31:34: SIP Tx udp:192.168.10.2:5060: SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.10.2:5060;branch=z9hG4bK281599984-57239805 From: "xxx" <sip:228@192.168.10.2>;tag=MitelICP_281599984-57239806 To: <sip:602@192.168.10.240>;tag=b0bae1689f Call-ID: 281599984-57239804 CSeq: 1 INVITE Contact: <sip:602@192.168.10.240:5060;transport=udp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pbxnsip-PBX/2.1.2.2292 Content-Type: application/sdp Content-Length: 378 v=0 o=- 17142 17142 IN IP4 192.168.10.240 s=- c=IN IP4 192.168.10.240 t=0 0 m=audio 61252 RTP/AVP 0 8 9 2 3 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:lGnVrHxY4kJK6730M7LF2LB4T9urAIiLWKyeeTzH a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv [9] 2007/12/19 10:31:34: Resolve destination 69747: aaaa udp 192.168.10.2 5060 [9] 2007/12/19 10:31:34: Resolve destination 69747: a udp 192.168.10.2 5060 [9] 2007/12/19 10:31:34: Resolve destination 69747: udp 192.168.10.2 5060 [7] 2007/12/19 10:31:34: SIP Tx udp:192.168.10.2:5060: SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.10.2:5060;branch=z9hG4bK281599984-57239805 From: "xxx" <sip:228@192.168.10.2>;tag=MitelICP_281599984-57239806 To: <sip:602@192.168.10.240>;tag=b0bae1689f Call-ID: 281599984-57239804 CSeq: 1 INVITE Contact: <sip:602@192.168.10.240:5060;transport=udp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pbxnsip-PBX/2.1.2.2292 Content-Type: application/sdp Content-Length: 378 v=0 o=- 17142 17142 IN IP4 192.168.10.240 s=- c=IN IP4 192.168.10.240 t=0 0 m=audio 61252 RTP/AVP 0 8 9 2 3 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:lGnVrHxY4kJK6730M7LF2LB4T9urAIiLWKyeeTzH a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv [7] 2007/12/19 10:31:34: SIP Rx udp:192.168.10.2:5060: ACK sip:602@192.168.10.240:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.10.2:5060;branch=z9hG4bK281759984-57239807 Max-Forwards: 70 From: "xxx" <sip:228@192.168.10.2>;tag=MitelICP_281599984-57239806 To: <sip:602@192.168.10.240>;tag=b0bae1689f Call-ID: 281599984-57239804 CSeq: 1 ACK Contact: "xxx" <sip:228@192.168.10.2:5060;transport=udp> Content-Type: application/sdp Content-Length: 185 v=0 o=- 14633181335402739584 14633181335402739584 IN IP4 192.168.10.3 s=- c=IN IP4 192.168.10.3 t=0 0 m=audio 20199 RTP/AVP 8 96 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 [6] 2007/12/19 10:31:34: Sending RTP for 281599984-57239804#b0bae1689f to 192.168.10.3:20199 [8] 2007/12/19 10:31:34: No codec available for sending There's an option on the Mitel SIP trunk to force SDP in the initial invite, but if I do that the Mitel comes back and says "Error" Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted December 20, 2007 Report Share Posted December 20, 2007 What is very strange is that the PBX sends a SRTP key in the offer via an insecure connection. I guess we need to test the cases again where the initial INVITE does not contain SDP... Quote Link to comment Share on other sites More sharing options...
Kristan Posted December 21, 2007 Author Report Share Posted December 21, 2007 What is very strange is that the PBX sends a SRTP key in the offer via an insecure connection. I guess we need to test the cases again where the initial INVITE does not contain SDP... I wondered that too. The whole thing is very strange, those Mitels as monster systems, I'll take PBXnSIP anyday Quote Link to comment Share on other sites More sharing options...
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