[Asterisk-Users] SIP, Microsoft RTC, and Originate problem
Asterisk
asterisk at abraxas.si
Wed Jun 14 00:43:01 MST 2006
Skipped content of type multipart/alternative-------------- next part --------------
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:asterisk at 111.111.111.8>;tag=as348de10b
To: <sip:111.111.111.50:16666>
Contact: <sip:asterisk at 111.111.111.8>
Call-ID: 104d16a9260f074f7d3dba3235b1870a at 111.111.111.8
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:asterisk at 111.111.111.8>;tag=as348de10b
To: <sip:111.111.111.50:16666>
Contact: <sip:asterisk at 111.111.111.8>
Call-ID: 104d16a9260f074f7d3dba3235b1870a at 111.111.111.8
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:asterisk at 111.111.111.8>;tag=as348de10b
To: <sip:111.111.111.50:16666>
Contact: <sip:asterisk at 111.111.111.8>
Call-ID: 104d16a9260f074f7d3dba3235b1870a at 111.111.111.8
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:asterisk at 111.111.111.8>;tag=as348de10b
To: <sip:111.111.111.50:16666>
Contact: <sip:asterisk at 111.111.111.8>
Call-ID: 104d16a9260f074f7d3dba3235b1870a at 111.111.111.8
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:asterisk at 111.111.111.8>;tag=as348de10b
To: <sip:111.111.111.50:16666>
Contact: <sip:asterisk at 111.111.111.8>
Call-ID: 104d16a9260f074f7d3dba3235b1870a at 111.111.111.8
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:asterisk at 111.111.111.8>;tag=as348de10b
To: <sip:111.111.111.50:16666>
Contact: <sip:asterisk at 111.111.111.8>
Call-ID: 104d16a9260f074f7d3dba3235b1870a at 111.111.111.8
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:asterisk at 111.111.111.8>;tag=as348de10b
To: <sip:111.111.111.50:16666>
Contact: <sip:asterisk at 111.111.111.8>
Call-ID: 104d16a9260f074f7d3dba3235b1870a at 111.111.111.8
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 104d16a9260f074f7d3dba3235b1870a at 111.111.111.8 for seqno 102 (Critical Request)
Jun 13 09:25:36 WARNING[10305]: chan_sip.c:1234 retrans_pkt: Hanging up call 104d16a9260f074f7d3dba3235b1870a at 111.111.111.8 - 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:asterisk at 111.111.111.8>;tag=as348de10b
To: <sip:111.111.111.50:16666>;tag=60ddafdb3c924f2f87bcd1fe186f7e7f
Call-ID: 104d16a9260f074f7d3dba3235b1870a at 111.111.111.8
CSeq: 102 INVITE
User-Agent: RTC/1.2
Content-Length: 0
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