Asterisk CVS-03/12/04-20:33:56, Copyright (C) 1999-2004 Digium. Written by Mark Spencer ========================================================================= Connected to Asterisk CVS-03/12/04-20:33:56 currently running on raven (pid = 25644) raven*CLI> Sip read: REGISTER sip:212.18.255.13 SIP/2.0 Via: SIP/2.0/UDP 213.137.8.59:5060;branch=z9hG4bK-hXj-31614 From: webdes1 ;tag=fTh-14319 To: webdes1 ;tag=fTh-14319 Call-ID: pXf-356@213.137.8.59 CSeq: 110 REGISTER Contact: Max-Forwards: 70 Expires: 600 User-Agent: DrayTek UA-1.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE Content-Length: 0 ^Draven*CLI> 12 headers, 0 lines ^Draven*CLI> Using latest request as basis request ^Draven*CLI> Sending to 213.137.8.59 : 5060 (non-NAT) ^Draven*CLI> Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 213.137.8.59:5060;branch=z9hG4bK-hXj-31614;received=213.137.8.59 From: webdes1 ;tag=fTh-14319 To: webdes1 ;tag=fTh-14319 Call-ID: pXf-356@213.137.8.59 CSeq: 110 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 213.137.8.59:5060 ^Draven*CLI> 11 headers, 0 lines ^Draven*CLI> Reliably Transmitting: OPTIONS sip:213.137.8.59 SIP/2.0 Via: SIP/2.0/UDP 212.18.255.13:5060;branch=z9hG4bK3c3eb661 From: "asterisk" ;tag=as39519e94 To: Contact: Call-ID: 037d1337634d007c0e40a04777660baf@212.18.255.13 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Thu, 18 Mar 2004 00:20:09 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 213.137.8.59:5060 -- Registered SIP 'webdes1' at 213.137.8.59 port 5060 expires 180 ^Draven*CLI> Transmitting (NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 213.137.8.59:5060;branch=z9hG4bK-hXj-31614;received=213.137.8.59 From: webdes1 ;tag=fTh-14319 To: webdes1 ;tag=fTh-14319 Call-ID: pXf-356@213.137.8.59 CSeq: 110 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Expires: 180 Contact: ;expires=180 Date: Thu, 18 Mar 2004 00:20:09 GMT Content-Length: 0 to 213.137.8.59:5060 ^Draven*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 212.18.255.13:5060;branch=z9hG4bK3c3eb661 From: "asterisk" ;tag=as39519e94 To: ;tag=Dcl-12215 Call-ID: 037d1337634d007c0e40a04777660baf@212.18.255.13 CSeq: 102 OPTIONS Allow: INVITE, ACK, CANCEL, OPTIONS, BYE Content-Length: 0 ^Draven*CLI> 8 headers, 0 lines ^Draven*CLI> Sip read: INVITE sip:845612@212.18.255.13 SIP/2.0 Via: SIP/2.0/UDP 213.137.8.59:5060;branch=z9hG4bK-PHH-9909 From: webdes1 ;tag=Ldp-13803 To: Call-ID: joB-27486@213.137.8.59 CSeq: 1 INVITE Contact: Max-Forwards: 70 User-Agent: DrayTek UA-1.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE Content-Type: application/sdp Content-Length: 237 v=0 o=webdes1 5972727 56415 IN IP4 213.137.8.59 s=SIP Call c=IN IP4 213.137.8.59 t=0 0 m=audio 10064 RTP/AVP 8 0 18 4 2 a=rtpmap:8 pcma/8000 a=rtpmap:0 pcmu/8000 a=rtpmap:18 G729/8000 a=rtpmap:4 g723/8000 a=rtpmap:2 g726/8000 ^Draven*CLI> 12 headers, 11 lines ^Draven*CLI> Using latest request as basis request ^Draven*CLI> Sending to 213.137.8.59 : 5060 (non-NAT) Found audio format ALAW ^Draven*CLI> Found audio format UNKN Found audio format UNKN Found audio format ULAW Found audio format GSM ^Draven*CLI> Found description format pcma Found description format pcmu Found description format G729 ^Draven*CLI> Found description format g723 Found description format g726 Capabilities: us - 258, them - 285/0, combined - 256 ^Draven*CLI> Non-codec capabilities: us - 1, them - 0, combined - 0 Looking for 845612 in default ^Draven*CLI> list_route: hop: ^Draven*CLI> Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 213.137.8.59:5060;branch=z9hG4bK-PHH-9909;received=213.137.8.59 From: webdes1 ;tag=Ldp-13803 To: ;tag=as6da0d7e7 Call-ID: joB-27486@213.137.8.59 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 213.137.8.59:5060 ^Draven*CLI> -- Executing Dial("SIP/webdes1-50dc", "IAX2/justthet:malta@magoutbound/441624845612") in new stack ^Draven*CLI> -- Called justthet:malta@magoutbound/441624845612 ^Draven*CLI> -- Call accepted by 213.166.5.130 (format ALAW) -- Format for call is ALAW ^Draven*CLI> We're at 212.18.255.13 port 18840 ^Draven*CLI> Answering with preferred capability 2 Answering with preferred capability 256 Answering with non-codec capability 1 ^Draven*CLI> Transmitting (NAT): SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 213.137.8.59:5060;branch=z9hG4bK-PHH-9909;received=213.137.8.59 From: webdes1 ;tag=Ldp-13803 To: ;tag=as6da0d7e7 Call-ID: joB-27486@213.137.8.59 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 243 v=0 o=root 27973 27973 IN IP4 212.18.255.13 s=session c=IN IP4 212.18.255.13 t=0 0 m=audio 18840 RTP/AVP 3 18 101 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - to 213.137.8.59:5060 ^Draven*CLI> -- IAX2[magoutbound]/2 is ringing ^Draven*CLI> -- IAX2[magoutbound]/2 stopped sounds ^Draven*CLI> -- IAX2[magoutbound]/2 answered SIP/webdes1-50dc ^Draven*CLI> We're at 212.18.255.13 port 18840 ^Draven*CLI> Answering with preferred capability 2 Answering with preferred capability 256 Answering with non-codec capability 1 ^Draven*CLI> Reliably Transmitting (NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 213.137.8.59:5060;branch=z9hG4bK-PHH-9909;received=213.137.8.59 From: webdes1 ;tag=Ldp-13803 To: ;tag=as6da0d7e7 Call-ID: joB-27486@213.137.8.59 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 243 v=0 o=root 27973 27974 IN IP4 212.18.255.13 s=session c=IN IP4 212.18.255.13 t=0 0 m=audio 18840 RTP/AVP 3 18 101 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - to 213.137.8.59:5060 ^Draven*CLI> Sip read: ACK sip:845612@212.18.255.13 SIP/2.0 Via: SIP/2.0/UDP 213.137.8.59:5060;branch=z9hG4bK-GHb-30362 From: webdes1 ;tag=Ldp-13803 To: ;tag=as6da0d7e7 Call-ID: joB-27486@213.137.8.59 CSeq: 1 ACK Max-Forwards: 70 User-Agent: DrayTek UA-1.0 Content-Length: 0 ^Draven*CLI> 9 headers, 0 lines ^Draven*CLI> 11 headers, 0 lines ^Draven*CLI> Reliably Transmitting: OPTIONS sip:213.137.8.59 SIP/2.0 Via: SIP/2.0/UDP 212.18.255.13:5060;branch=z9hG4bK2093e64f From: "asterisk" ;tag=as5447083a To: Contact: Call-ID: 2206175567a48a475c49724139356ff8@212.18.255.13 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Thu, 18 Mar 2004 00:21:09 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 213.137.8.59:5060 ^Draven*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 212.18.255.13:5060;branch=z9hG4bK2093e64f From: "asterisk" ;tag=as5447083a To: ;tag=DgT-29393 Call-ID: 2206175567a48a475c49724139356ff8@212.18.255.13 CSeq: 102 OPTIONS Allow: INVITE, ACK, CANCEL, OPTIONS, BYE Content-Length: 0 ^Draven*CLI> 8 headers, 0 lines ^Draven*CLI> 11 headers, 0 lines ^Draven*CLI> Reliably Transmitting: OPTIONS sip:213.137.8.59 SIP/2.0 Via: SIP/2.0/UDP 212.18.255.13:5060;branch=z9hG4bK66b388ce From: "asterisk" ;tag=as1fd959b2 To: Contact: Call-ID: 0ffe96211abc789931aebdb03ad1775b@212.18.255.13 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Thu, 18 Mar 2004 00:22:09 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 213.137.8.59:5060 ^Draven*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 212.18.255.13:5060;branch=z9hG4bK66b388ce From: "asterisk" ;tag=as1fd959b2 To: ;tag=CBf-11611 Call-ID: 0ffe96211abc789931aebdb03ad1775b@212.18.255.13 CSeq: 102 OPTIONS Allow: INVITE, ACK, CANCEL, OPTIONS, BYE Content-Length: 0 ^Draven*CLI> 8 headers, 0 lines ^Draven*CLI> -- Hungup 'IAX2[magoutbound]/2' ^Draven*CLI> == Spawn extension (default, 845612, 1) exited non-zero on 'SIP/webdes1-50dc' ^Draven*CLI> set_destination: Parsing for address/port to send to ^Draven*CLI> set_destination: set destination to 213.137.8.59, port 5060 ^Draven*CLI> Reliably Transmitting: BYE sip:webdes1@213.137.8.59 SIP/2.0 Via: SIP/2.0/UDP 212.18.255.13:5060;branch=z9hG4bK65e8b871 From: ;tag=as6da0d7e7 To: webdes1 ;tag=Ldp-13803 Contact: Call-ID: joB-27486@213.137.8.59 CSeq: 102 BYE User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 213.137.8.59:5060 ^Draven*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 212.18.255.13:5060;branch=z9hG4bK65e8b871 From: ;tag=as6da0d7e7 To: webdes1 ;tag=Ldp-13803 Call-ID: joB-27486@213.137.8.59 CSeq: 102 BYE Content-Length: 0 ^Draven*CLI> 7 headers, 0 lines Message is BYE ^Draven*CLI> 11 headers, 0 lines ^Draven*CLI> Reliably Transmitting: OPTIONS sip:213.137.8.59 SIP/2.0 Via: SIP/2.0/UDP 212.18.255.13:5060;branch=z9hG4bK3207f408 From: "asterisk" ;tag=as10f1c6fe To: Contact: Call-ID: 61830aaf0127f01202aff88977b2687a@212.18.255.13 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Thu, 18 Mar 2004 00:23:09 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 213.137.8.59:5060 ^Draven*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 212.18.255.13:5060;branch=z9hG4bK3207f408 From: "asterisk" ;tag=as10f1c6fe To: ;tag=GHH-31415 Call-ID: 61830aaf0127f01202aff88977b2687a@212.18.255.13 CSeq: 102 OPTIONS Allow: INVITE, ACK, CANCEL, OPTIONS, BYE Content-Length: 0 ^Draven*CLI> 8 headers, 0 lines ^Draven*CLI>