|
It seems that Microsoft RTC has some
problems with originated calls from Asterisk. If I execute Manager API
originate application, with SIP channel as parameter, the Microsoft RTC
softphone will start to ring after a couple of seconds delay, but nothing more
happens after when I answer – there is no second call to an extension. When I looked through the sip debug, I
noticed that Microsoft RTC fails to properly respond to INVITE messages (I have
attached the sip debug). Asterisk has to retransmit INVITE message for 6 times
and even then the RTC still doesn't respond in a proper time. However, if I do
direct call to that problematic Microsoft RTC based softphone, everything works
fine, eventhough very same INVITE messages are being transmited to it from
Asterisk. Does anyone have any ideas for a
workaround? Regards, Alex |
Reliably Transmitting (no NAT) to 111.111.111.50:16666: INVITE sip:111.111.111.50:16666 SIP/2.0
Via: SIP/2.0/UDP 111.111.111.8:5060;branch=z9hG4bK360502ae;rport From: "asterisk" <sip:[EMAIL PROTECTED]>;tag=as348de10b To: <sip:111.111.111.50:16666> Contact: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 13 Jun 2006 07:25:35 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 242 v=0 o=root 10295 10295 IN IP4 111.111.111.8 s=session c=IN IP4 111.111.111.8 t=0 0 m=audio 12742 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- [Kasterisk*CLI> Retransmitting #1 (no NAT) to 111.111.111.50:16666: INVITE sip:111.111.111.50:16666 SIP/2.0 Via: SIP/2.0/UDP 111.111.111.8:5060;branch=z9hG4bK360502ae;rport From: "asterisk" <sip:[EMAIL PROTECTED]>;tag=as348de10b To: <sip:111.111.111.50:16666> Contact: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 13 Jun 2006 07:25:35 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 242 v=0 o=root 10295 10295 IN IP4 111.111.111.8 s=session c=IN IP4 111.111.111.8 t=0 0 m=audio 12742 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Retransmitting #2 (no NAT) to 111.111.111.50:16666: INVITE sip:111.111.111.50:16666 SIP/2.0 Via: SIP/2.0/UDP 111.111.111.8:5060;branch=z9hG4bK360502ae;rport From: "asterisk" <sip:[EMAIL PROTECTED]>;tag=as348de10b To: <sip:111.111.111.50:16666> Contact: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 13 Jun 2006 07:25:35 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 242 v=0 o=root 10295 10295 IN IP4 111.111.111.8 s=session c=IN IP4 111.111.111.8 t=0 0 m=audio 12742 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- [Kasterisk*CLI> Retransmitting #3 (no NAT) to 111.111.111.50:16666: INVITE sip:111.111.111.50:16666 SIP/2.0 Via: SIP/2.0/UDP 111.111.111.8:5060;branch=z9hG4bK360502ae;rport From: "asterisk" <sip:[EMAIL PROTECTED]>;tag=as348de10b To: <sip:111.111.111.50:16666> Contact: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 13 Jun 2006 07:25:35 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 242 v=0 o=root 10295 10295 IN IP4 111.111.111.8 s=session c=IN IP4 111.111.111.8 t=0 0 m=audio 12742 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- [Kasterisk*CLI> Retransmitting #4 (no NAT) to 111.111.111.50:16666: INVITE sip:111.111.111.50:16666 SIP/2.0 Via: SIP/2.0/UDP 111.111.111.8:5060;branch=z9hG4bK360502ae;rport From: "asterisk" <sip:[EMAIL PROTECTED]>;tag=as348de10b To: <sip:111.111.111.50:16666> Contact: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 13 Jun 2006 07:25:35 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 242 v=0 o=root 10295 10295 IN IP4 111.111.111.8 s=session c=IN IP4 111.111.111.8 t=0 0 m=audio 12742 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- [Kasterisk*CLI> Retransmitting #5 (no NAT) to 111.111.111.50:16666: INVITE sip:111.111.111.50:16666 SIP/2.0 Via: SIP/2.0/UDP 111.111.111.8:5060;branch=z9hG4bK360502ae;rport From: "asterisk" <sip:[EMAIL PROTECTED]>;tag=as348de10b To: <sip:111.111.111.50:16666> Contact: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 13 Jun 2006 07:25:35 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 242 v=0 o=root 10295 10295 IN IP4 111.111.111.8 s=session c=IN IP4 111.111.111.8 t=0 0 m=audio 12742 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- [Kasterisk*CLI> Retransmitting #6 (no NAT) to 111.111.111.50:16666: INVITE sip:111.111.111.50:16666 SIP/2.0 Via: SIP/2.0/UDP 111.111.111.8:5060;branch=z9hG4bK360502ae;rport From: "asterisk" <sip:[EMAIL PROTECTED]>;tag=as348de10b To: <sip:111.111.111.50:16666> Contact: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 13 Jun 2006 07:25:35 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 242 v=0 o=root 10295 10295 IN IP4 111.111.111.8 s=session c=IN IP4 111.111.111.8 t=0 0 m=audio 12742 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- [Kasterisk*CLI> Jun 13 09:25:36 WARNING[10305]: chan_sip.c:1217 retrans_pkt: Maximum retries exceeded on transmission [EMAIL PROTECTED] for seqno 102 (Critical Request) Jun 13 09:25:36 WARNING[10305]: chan_sip.c:1234 retrans_pkt: Hanging up call [EMAIL PROTECTED] - no reply to our critical packet. [Kasterisk*CLI> <-- SIP read from 111.111.111.50:1380: SIP/2.0 100 Trying Via: SIP/2.0/UDP 111.111.111.8:5060;branch=z9hG4bK360502ae;rport From: "asterisk" <sip:[EMAIL PROTECTED]>;tag=as348de10b To: <sip:111.111.111.50:16666>;tag=60ddafdb3c924f2f87bcd1fe186f7e7f Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: RTC/1.2 Content-Length: 0
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