== Parsing '/etc/asterisk/asterisk.conf': Found == Parsing '/etc/asterisk/extconfig.conf': Found Asterisk SVN-trunk-r7618M, Copyright (C) 1999 - 2005 Digium, Inc. and others. Created by Mark Spencer Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'show license' for details. ========================================================================= Connected to Asterisk SVN-trunk-r7618M currently running on astlap (pid = 12021) astlap*CLI> Verbosity was 3 and is now 4 astlap*CLI> set debug 10 astlap*CLI> Core debug was 0 and is now 10 astlap*CLI> set bveverbose 10 astlap*CLI> Verbosity was 4 and is now 10 astlap*CLI> sip debug astlap*CLI> SIP Debugging enabled astlap*CLI> <-- SIP read from 192.168.254.1:8917: REGISTER sip:cojensen.net SIP/2.0 Via: SIP/2.0/UDP 192.168.254.60:8917;branch=z9hG4bK-d87543-b87b1065292f3c40-1--d87543-;rport Max-Forwards: 70 Contact: To: "Kai-Uwe Jensen" From: "Kai-Uwe Jensen";tag=6b1c673d Call-ID: 1d3f56532d28e464@TUFETUFOMi5hbWVyaWNhcy5ocHFjb3JwLm5ldA.. CSeq: 1 REGISTER Expires: 3600 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Supported: eventlist User-Agent: eyeBeam release 3010n stamp 19039 Content-Length: 0 --- (13 headers 0 lines)--- Using latest REGISTER request as basis request Sending to 192.168.254.60 : 8917 (non-NAT) Transmitting (NAT) to 192.168.254.1:8917: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.254.60:8917;branch=z9hG4bK-d87543-b87b1065292f3c40-1--d87543-;received=192.168.254.1;rport=8917 From: "Kai-Uwe Jensen";tag=6b1c673d To: "Kai-Uwe Jensen" Call-ID: 1d3f56532d28e464@TUFETUFOMi5hbWVyaWNhcy5ocHFjb3JwLm5ldA.. CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- Transmitting (NAT) to 192.168.254.1:8917: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.254.60:8917;branch=z9hG4bK-d87543-b87b1065292f3c40-1--d87543-;received=192.168.254.1;rport=8917 From: "Kai-Uwe Jensen";tag=6b1c673d To: "Kai-Uwe Jensen";tag=as58010d8c Call-ID: 1d3f56532d28e464@TUFETUFOMi5hbWVyaWNhcy5ocHFjb3JwLm5ldA.. CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: WWW-Authenticate: Digest realm="asterisk", nonce="0bbe8a7d" Content-Length: 0 --- Scheduling destruction of call '1d3f56532d28e464@TUFETUFOMi5hbWVyaWNhcy5ocHFjb3JwLm5ldA..' in 15000 ms astlap*CLI> <-- SIP read from 192.168.254.1:8917: REGISTER sip:cojensen.net SIP/2.0 Via: SIP/2.0/UDP 192.168.254.60:8917;branch=z9hG4bK-d87543-416fc76189275470-1--d87543-;rport Max-Forwards: 70 Contact: To: "Kai-Uwe Jensen" From: "Kai-Uwe Jensen";tag=6b1c673d Call-ID: 1d3f56532d28e464@TUFETUFOMi5hbWVyaWNhcy5ocHFjb3JwLm5ldA.. CSeq: 2 REGISTER Expires: 3600 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Supported: eventlist User-Agent: eyeBeam release 3010n stamp 19039 Authorization: Digest username="ext2003",realm="asterisk",nonce="0bbe8a7d",uri="sip:cojensen.net",response="f032757f99ec153c9e3bcdaf17d96683",algorithm=MD5 Content-Length: 0 --- (14 headers 0 lines)--- Using latest REGISTER request as basis request Sending to 192.168.254.60 : 8917 (NAT) Transmitting (NAT) to 192.168.254.1:8917: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.254.60:8917;branch=z9hG4bK-d87543-416fc76189275470-1--d87543-;received=192.168.254.1;rport=8917 From: "Kai-Uwe Jensen";tag=6b1c673d To: "Kai-Uwe Jensen" Call-ID: 1d3f56532d28e464@TUFETUFOMi5hbWVyaWNhcy5ocHFjb3JwLm5ldA.. CSeq: 2 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- astlap*CLI> 12 headers, 0 lines astlap*CLI> Reliably Transmitting (no NAT) to 192.168.254.1:8917: OPTIONS sip:ext2003@192.168.254.60:8917 SIP/2.0 Via: SIP/2.0/UDP 192.168.254.250:5060;branch=z9hG4bK4d503b1e;rport From: "Asterisk" ;tag=as218dfaf6 To: Contact: Call-ID: 321315ca2f6dd3d253295eb630d76198@192.168.254.250 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Sat, 24 Dec 2005 20:39:01 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- -- Registered SIP 'ext2003' at 192.168.254.1 port 8917 expires 3600 -- Saved useragent "eyeBeam release 3010n stamp 19039" for peer ext2003 Transmitting (NAT) to 192.168.254.1:8917: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.254.60:8917;branch=z9hG4bK-d87543-416fc76189275470-1--d87543-;received=192.168.254.1;rport=8917 From: "Kai-Uwe Jensen";tag=6b1c673d To: "Kai-Uwe Jensen";tag=as58010d8c Call-ID: 1d3f56532d28e464@TUFETUFOMi5hbWVyaWNhcy5ocHFjb3JwLm5ldA.. CSeq: 2 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Expires: 3600 Contact: ;expires=3600 Date: Sat, 24 Dec 2005 20:39:01 GMT Content-Length: 0 --- Scheduling destruction of call '1d3f56532d28e464@TUFETUFOMi5hbWVyaWNhcy5ocHFjb3JwLm5ldA..' in 15000 ms astlap*CLI> <-- SIP read from 192.168.254.1:8917: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.254.250:5060;branch=z9hG4bK4d503b1e;rport=5060;received=67.174.106.30 Contact: To: ;tag=597b2a14 From: "Asterisk";tag=as218dfaf6 Call-ID: 321315ca2f6dd3d253295eb630d76198@192.168.254.250 CSeq: 102 OPTIONS Accept: application/sdp Accept-Language: en Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO S astlap*CLI> upported: eventlist User-Agent: eyeBeam release 3010n stamp 19039 Content-Length: 0 --- (13 headers 0 lines)--- Destroying call '321315ca2f6dd3d253295eb630d76198@192.168.254.250' astlap*CLI> 12 headers, 3 lines Reliably Transmitting (NAT) to 192.168.254.1:8917: NOTIFY sip:ext2003@192.168.254.60:8917 SIP/2.0 Via: SIP/2.0/UDP 192.168.254.250:5060;branch=z9hG4bK71999702;rport From: "Asterisk" ;tag=as1e95841b To: Contact: Call-ID: 174331503c8666a15333539d595d128f@192.168.254.250 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 80 Messages-Waiting: no Message-Account: sip:asterisk@ Voice-Message: 0/0 (0/0) --- Scheduling destruction of call '174331503c8666a15333539d595d128f@192.168.254.250' in 15000 ms astlap*CLI> <-- SIP read from 192.168.254.1:8917: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.254.250:5060;branch=z9hG4bK71999702;rport=5060;received=67.174.106.30 Contact: To: ;tag=fd6dc502 From: "Asterisk";tag=as1e95841b Call-ID: 174331503c8666a15333539d595d128f@192.168.254.250 CSeq: 102 NOTIFY User-Agent: eyeBeam release 3010n stamp 19039 Content-Length: 0 --- (9 headers 0 lines)--- Destroying call '174331503c8666a15333539d595d128f@192.168.254.250' astlap*CLI> Destroying call '7ec44bce3ac5a65a1e70384c6dbe2282@192.168.254.250' astlap*CLI> Destroying call '704a6f430de6c4940708d0313b20cf31@192.168.254.250' astlap*CLI> Destroying call '06e58d0f318a0d3f7c0269614a478651@192.168.254.250' astlap*CLI> <-- SIP read from 192.168.254.1:8917: astlap*CLI> --- (0 headers 0 lines) Nat keepalive --- astlap*CLI> <-- SIP read from 192.168.254.252:5060: INVITE sip:2600@192.168.254.250;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.254.252:5060;branch=z9hG4bK2853007a132E9BD From: "Kai-Uwe Jensen" ;tag=E4DDD358-75027071 To: CSeq: 1 INVITE Call-ID: ecf5e574-b09a8c3e-3d378bcf@192.168.254.252 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.6.2.0041 Supported: 100rel,replace Allow-Events: talk,hold,conference Max-Forwards: 70 Content-Type: application/sdp Content-Length: 243 v=0 o=- 1135456524 1135456524 IN IP4 192.168.254.252 s=Polycom IP Phone c=IN IP4 192.168.254.252 t=0 0 m=audio 2226 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 --- (14 headers 10 lines)--- Using INVITE request as basis request - ecf5e574-b09a8c3e-3d378bcf@192.168.254.252 Sending to 192.168.254.252 : 5060 (non-NAT) Reliably Transmitting (no NAT) to 192.168.254.252:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.254.252:5060;branch=z9hG4bK2853007a132E9BD;received=192.168.254.252 From: "Kai-Uwe Jensen" ;tag=E4DDD358-75027071 To: ;tag=as445c1d14 Call-ID: ecf5e574-b09a8c3e-3d378bcf@192.168.254.252 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Proxy-Authenticate: Digest realm="asterisk", nonce="7ad42d8c" Content-Length: 0 --- Scheduling destruction of call 'ecf5e574-b09a8c3e-3d378bcf@192.168.254.252' in 15000 ms Found user 'ext2006' astlap*CLI> <-- SIP read from 192.168.254.252:5060: ACK sip:2600@192.168.254.250 SIP/2.0 Via: SIP/2.0/UDP 192.168.254.252:5060;branch=z9hG4bK2853007a132E9BD From: "Kai-Uwe Jensen" ;tag=E4DDD358-75027071 To: ;tag=as445c1d14 CSeq: 1 ACK Call-ID: ecf5e574-b09a8c3e-3d378bcf@192.168.254.252 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.6.2.0041 Max-Forwards: 70 Content-Length: 0 --- (11 headers 0 lines)--- astlap*CLI> <-- SIP read from 192.168.254.252:5060: INVITE sip:2600@192.168.254.250;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.254.252:5060;branch=z9hG4bKd6213c23808A0D42 From: "Kai-Uwe Jensen" ;tag=E4DDD358-75027071 To: CSeq: 2 INVITE Call-ID: ecf5e574-b09a8c3e-3d378bcf@192.168.254.252 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.6.2.0041 Supported: 100rel,replace Allow-Events: talk,hold,conference Proxy-Authorization: Digest username="ext2006", realm="asterisk", nonce="7ad42d8c", uri="sip:2600@192.168.254.250;user=phone", response="004c334c04e2d2af1498025dd3620875", algorithm=MD5 Max-Forwards: 70 Content-Type: application/sdp Content-Length: 243 v=0 o=- 1135456524 1135456524 IN IP4 192.168.254.252 s=Polycom IP Phone c=IN IP4 192.168.254.252 t=0 0 m=audio 2226 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 --- (15 headers 10 lines)--- Using INVITE request as basis request - ecf5e574-b09a8c3e-3d378bcf@192.168.254.252 Sending to 192.168.254.252 : 5060 (non-NAT) Found user 'ext2006' Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 192.168.254.252:2226 Peer video RTP is at port 192.168.254.252:65535 Found description format PCMU Found description format PCMA Found description format G729 Found description format telephone-event Capabilities: us - 0x104 (ulaw|g729), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x104 (ulaw|g729) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for 2600 in home (domain 192.168.254.250) list_route: hop: astlap*CLI> Transmitting (no NAT) to 192.168.254.252:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.254.252:5060;branch=z9hG4bKd6213c23808A0D42;received=192.168.254.252 From: "Kai-Uwe Jensen" ;tag=E4DDD358-75027071 To: Call-ID: ecf5e574-b09a8c3e-3d378bcf@192.168.254.252 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- astlap*CLI> -- Executing Goto("SIP/ext2006-c324", "confline|s|1") in new stack astlap*CLI> -- Goto (confline,s,1) astlap*CLI> -- Executing Answer("SIP/ext2006-c324", "") in new stack astlap*CLI> We're at 192.168.254.250 port 9012 astlap*CLI> Video is at 192.168.254.250 port 9044 astlap*CLI> Adding codec 0x4 (ulaw) to SDP astlap*CLI> Adding codec 0x100 (g729) to SDP astlap*CLI> Adding non-codec 0x1 (telephone-event) to SDP astlap*CLI> Reliably Transmitting (no NAT) to 192.168.254.252:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.254.252:5060;branch=z9hG4bKd6213c23808A0D42;received=192.168.254.252 From: "Kai-Uwe Jensen" ;tag=E4DDD358-75027071 To: ;tag=as219a39e0 Call-ID: ecf5e574-b09a8c3e-3d378bcf@192.168.254.252 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 268 v=0 o=root 12021 12021 IN IP4 192.168.254.250 s=session c=IN IP4 192.168.254.250 t=0 0 m=audio 9012 RTP/AVP 0 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- astlap*CLI> -- Executing Wait("SIP/ext2006-c324", "1") in new stack astlap*CLI> <-- SIP read from 192.168.254.252:5060: ACK sip:2600@192.168.254.250 SIP/2.0 Via: SIP/2.0/UDP 192.168.254.252:5060;branch=z9hG4bK4a84a786FB8DD299 From: "Kai-Uwe Jensen" ;tag=E4DDD358-75027071 To: ;tag=as219a39e0 CSeq: 2 ACK Call-ID: ecf5e574-b09a8c3e-3d378bcf@192.168.254.252 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.6.2.0041 Max-Forwards: 70 Content-Length: 0 --- (11 headers 0 lines)--- astlap*CLI> -- Executing BackGround("SIP/ext2006-c324", "enter-conf-call-number") in new stack astlap*CLI> -- Playing 'enter-conf-call-number' (language 'en') astlap*CLI> == CDR updated on SIP/ext2006-c324 -- Executing Goto("SIP/ext2006-c324", "8987762|1") in new stack -- Goto (confline,8987762,1) -- Executing Wait("SIP/ext2006-c324", "1") in new stack astlap*CLI> Destroying call '1d3f56532d28e464@TUFETUFOMi5hbWVyaWNhcy5ocHFjb3JwLm5ldA..' astlap*CLI> -- Executing MeetMe("SIP/ext2006-c324", "8987762|MPxwsvn") in new stack -- Created MeetMe conference 1023 for conference '8987762' -- Playing 'conf-waitforleader' (language 'en') astlap*CLI> <-- SIP read from 192.168.254.1:8917: --- (0 headers 0 lines) Nat keepalive --- astlap*CLI> -- Started music on hold, class 'default', on SIP/ext2006-c324 astlap*CLI> <-- SIP read from 192.168.254.1:8917: INVITE sip:2600@cojensen.net SIP/2.0 Via: SIP/2.0/UDP 192.168.254.60:8917;branch=z9hG4bK-d87543-df09b343d8437430-1--d87543-;rport Max-Forwards: 70 Contact: To: From: "Kai-Uwe Jensen";tag=a5645b0a Call-ID: cf6b472bf7571c6e@TUFETUFOMi5hbWVyaWNhcy5ocHFjb3JwLm5ldA.. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Supported: eventlist User-Agent: eyeBeam release 3010n stamp 19039 Content-Length: 336 v=0 o=- 258245085 258245130 IN IP4 192.168.254.60 s=eyeBeam c=IN IP4 192.168.254.60 t=0 0 m=audio 9080 RTP/AVP 0 18 3 97 101 a=alt:1 1 : 7B7E75E3 0000007D 192.168.254.60 9080 a=fmtp:101 0-15 a=rtpmap:0 pcmu/8000 a=rtpmap:18 g729/8000 a=rtpmap:3 gsm/8000 a=rtpmap:97 speex/8000 a=rtpmap:101 telephone-event/8000 a=sendrecv --- (13 headers 14 lines)--- Using INVITE request as basis request - cf6b472bf7571c6e@TUFETUFOMi5hbWVyaWNhcy5ocHFjb3JwLm5ldA.. Sending to 192.168.254.60 : 8917 (non-NAT) Reliably Transmitting (NAT) to 192.168.254.1:8917: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.254.60:8917;branch=z9hG4bK-d87543-df09b343d8437430-1--d87543-;received=192.168.254.1;rport=8917 From: "Kai-Uwe Jensen";tag=a5645b0a To: ;tag=as5393dad8 Call-ID: cf6b472bf7571c6e@TUFETUFOMi5hbWVyaWNhcy5ocHFjb3JwLm5ldA.. CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Proxy-Authenticate: Digest realm="asterisk", nonce="147b6f01" Content-Length: 0 --- Scheduling destruction of call 'cf6b472bf7571c6e@TUFETUFOMi5hbWVyaWNhcy5ocHFjb3JwLm5ldA..' in 15000 ms Found user 'ext2003' astlap*CLI> <-- SIP read from 192.168.254.1:8917: ACK sip:2600@cojensen.net SIP/2.0 Via: SIP/2.0/UDP 192.168.254.60:8917;branch=z9hG4bK-d87543-df09b343d8437430-1--d87543-;rport To: ;tag=as5393dad8 From: "Kai-Uwe Jensen";tag=a5645b0a Call-ID: cf6b472bf7571c6e@TUFETUFOMi5hbWVyaWNhcy5ocHFjb3JwLm5ldA.. CSeq: 1 ACK Content-Length: 0 --- (7 headers 0 lines)--- astlap*CLI> <-- SIP read from 192.168.254.1:8917: INVITE sip:2600@cojensen.net SIP/2.0 Via: SIP/2.0/UDP 192.168.254.60:8917;branch=z9hG4bK-d87543-674fb83cb5299930-1--d87543-;rport Max-Forwards: 70 Contact: To: From: "Kai-Uwe Jensen";tag=a5645b0a Call-ID: cf6b472bf7571c6e@TUFETUFOMi5hbWVyaWNhcy5ocHFjb3JwLm5ldA.. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Proxy-Authorization: Digest username="ext2003",realm="asterisk",nonce="147b6f01",uri="sip:2600@cojensen.net",response="769c34158eb53d42735e8780dcc113ba",algorithm=MD5 Supported: eventlist User-Agent: eyeBeam release 3010n stamp 19039 Content-Length: 336 v=0 o=- 258245085 258245130 IN IP4 192.168.254.60 s=eyeBeam c=IN IP4 192.168.254.60 t=0 0 m=audio 9080 RTP/AVP 0 18 3 97 101 a=alt:1 1 : 7B7E75E3 0000007D 192.168.254.60 9080 a=fmtp:101 0-15 a=rtpmap:0 pcmu/8000 a=rtpmap:18 g729/8000 a=rtpmap:3 gsm/8000 a=rtpmap:97 speex/8000 a=rtpmap:101 telephone-event/8000 a=sendrecv --- (14 headers 14 lines)--- Using INVITE request as basis request - cf6b472bf7571c6e@TUFETUFOMi5hbWVyaWNhcy5ocHFjb3JwLm5ldA.. Sending to 192.168.254.60 : 8917 (NAT) Found user 'ext2003' Found RTP audio format 0 Found RTP audio format 18 Found RTP audio format 3 Found RTP audio format 97 Found RTP audio format 101 Peer audio RTP is at port 192.168.254.60:9080 Peer video RTP is at port 192.168.254.60:65535 Found description format pcmu Found description format g729 Found description format gsm Found description format speex Found description format telephone-event Capabilities: us - 0x1c0104 (ulaw|g729|h261|h263|h263p), peer - audio=0x306 (gsm|ulaw|g729|speex)/video=0x0 (nothing), combined - 0x104 (ulaw|g729) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for 2600 in dial-by-ip (domain cojensen.net) list_route: hop: Transmitting (NAT) to 192.168.254.1:8917: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.254.60:8917;branch=z9hG4bK-d87543-674fb83cb5299930-1--d87543-;received=192.168.254.1;rport=8917 From: "Kai-Uwe Jensen";tag=a5645b0a To: Call-ID: cf6b472bf7571c6e@TUFETUFOMi5hbWVyaWNhcy5ocHFjb3JwLm5ldA.. CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- astlap*CLI> -- Executing NoOp("SIP/ext2003-a44b", "cojensen.net|2600") in new stack astlap*CLI> -- Executing NoOp("SIP/ext2003-a44b", "cojensen.net") in new stack astlap*CLI> -- Executing GotoIf("SIP/ext2003-a44b", "0?home|2600|1:4") in new stack astlap*CLI> -- Goto (dial-by-ip,2600,4) astlap*CLI> -- Executing GotoIf("SIP/ext2003-a44b", "0?home|2600|1:5") in new stack astlap*CLI> -- Goto (dial-by-ip,2600,5) astlap*CLI> -- Executing GotoIf("SIP/ext2003-a44b", "1?home|2600|1:6") in new stack astlap*CLI> -- Goto (home,2600,1) astlap*CLI> -- Executing Goto("SIP/ext2003-a44b", "confline|s|1") in new stack astlap*CLI> -- Goto (confline,s,1) astlap*CLI> -- Executing Answer("SIP/ext2003-a44b", "") in new stack astlap*CLI> We're at 192.168.254.250 port 9010 astlap*CLI> Video is at 192.168.254.250 port 9028 astlap*CLI> Adding codec 0x4 (ulaw) to SDP astlap*CLI> Adding codec 0x100 (g729) to SDP astlap*CLI> Adding codec 0x40000 (h261) to SDP astlap*CLI> Adding codec 0x80000 (h263) to SDP astlap*CLI> Adding codec 0x100000 (h263p) to SDP astlap*CLI> Adding non-codec 0x1 (telephone-event) to SDP astlap*CLI> Reliably Transmitting (NAT) to 192.168.254.1:8917: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.254.60:8917;branch=z9hG4bK-d87543-674fb83cb5299930-1--d87543-;received=192.168.254.1;rport=8917 From: "Kai-Uwe Jensen";tag=a5645b0a To: ;tag=as49a86cce Call-ID: cf6b472bf7571c6e@TUFETUFOMi5hbWVyaWNhcy5ocHFjb3JwLm5ldA.. CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: astlap*CLI> Content-Type: application/sdp Content-Length: 268 v=0 o=root 12021 12021 IN IP4 192.168.254.250 s=session c=IN IP4 192.168.254.250 t=0 0 m=audio 9010 RTP/AVP 0 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- astlap*CLI> -- Executing Wait("SIP/ext2003-a44b", "1") in new stack astlap*CLI> <-- SIP read from 192.168.254.60:8917: ACK sip:2600@192.168.254.250 SIP/2.0 Via: SIP/2.0/UDP 192.168.254.60:8917;branch=z9hG4bK-d87543-bf1a0d365b14433a-1--d87543-;rport Max-Forwards: 70 Contact: To: ;tag=as49a86cce From: "Kai-Uwe Jensen";tag=a5645b0a Call-ID: cf6b472bf7571c6e@TUFETUFOMi5hbWVyaWNhcy5ocHFjb3JwLm5ldA.. CSeq: 2 ACK User-Agent: eyeBeam release 3010n stamp 19039 Content-Length: 0 --- (10 headers 0 lines)--- astlap*CLI> -- Executing BackGround("SIP/ext2003-a44b", "enter-conf-call-number") in new stack -- Playing 'enter-conf-call-number' (language 'en') astlap*CLI> <-- SIP read from 192.168.254.1:8917: --- (0 headers 0 lines) Nat keepalive --- astlap*CLI> Executing last minute cleanups