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russelln

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Everything posted by russelln

  1. Well I had a few minutes this morning and got the log out and found this error, it's a 0, v=0 o=- 1462283088 1462283088 IN IP4 127.0.0.1 s=- c=IN IP4 127.0.0.1 t=0 0 m=audio 62252 RTP/AVP 0 101 a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv [5] 20080427233651: PSTN: Response code: 200 [0] 20080427233651: PSTN: Exit: VAPI_DisableConnection Failed [5] 20080427233651: PSTN: enable_callerid 0 [9] 20080427233736: SIP Rx tls:192.168.1.101:2143: SUBSCRIBE sip:40@localhost;user=phone SIP/2.0 Via: SIP/2.0/TLS 192.168.1.101:2048;branch=z9hG4bK-z9kx70z2hq2k;rport From: <sip:40@localhost>;tag=xn7gpmapdq fxo port is locked up when I get it, I have to hard start the switch to make it go away sometimes I get two or three of these the same day and sometimes it goes 5 or 6 days before it happens, I have the log of the last month saved I have started a new one..I will try to document how often now and when it happens, as I know now kind of what is going on and where to spot in the log, the log shows a reboot, after I see it, but sometimes, some time transpires before I know it is going on... [8] 20080428131852: Packet authenticated by transport layer [7] 20080428131852: UDP: Opening socket on port 55280 [7] 20080428131852: UDP: Opening socket on port 55281 [8] 20080428131852: Could not find a trunk (1 trunks) [9] 20080428131852: Using outbound proxy sip:192.168.1.101:2143;transport=tls because of flow-label [9] 20080428131852: Last message repeated 2 times [9] 20080428131852: Resolve 6040: tls 192.168.1.101 2143 [9] 20080428131852: SIP Tx tls:192.168.1.101:2143: SIP/2.0 100 Trying by the way while this is going on you are just sitting there listening to comfort noise on the phone, and you can sit there all day if you do not know what is going on... Via: SIP/2.0/TLS 192.168.1.101:2048;branch=z9hG4bK-0tjhtw64ufxq;rport=2143 From: "40" <sip:40@localhost>;tag=r0a09ki1al To: <sip:18008212279@localhost;user=phone>;tag=71c159f893 Call-ID: 3c2a1ad432b6-ydk2soy9mx8v CSeq: 1 INVITE Content-Length: 0 [7] 20080428131852: Set packet length to 20 [6] 20080428131852: Sending RTP for 3c2a1ad432b6-ydk2soy9mx8v#71c159f893 to 192.168.1.101:57652 [1] 20080428132134: Starting up version 3.0.0.2899 [8] 20080428132134: Route: eth0 c0a80100 ffffff00 [8] 20080428132134: Route: eth1 01010100 ffffff00 [8] 20080428132134: Default Route uses 192.168.1.100 [7] 20080428132134: Found time zones AKDT AKST PDT PST MDT MST CDT CST2 EDT EST ADT AST NDT NST BST CET GMT+3 CST CAT IST AUS1 AUS2 AUS3 AUS4 AUS5 AUS6 GMT [1] 20080428132134: Working Directory is /pbx here I am trying to make a call and now I notice the fxo port is lit up, no call, and I restart the switch.. Hope this sheds some light on what is going on.. Also still have the 2 lines lighting up when a call comes and it is gotten random as to which button to hit to get the call. All the best Russell
  2. Went to spool directory, there was about 30 emails in there, I guess the one had the others hung...opened the file and changed the email server in the file, saved it and they all dissappeared to the network, was kinda cool seeing them all go through on the log.. thanks again, russell
  3. ah ha,,,thanks... made almost a week without freezing the fxo port and the phones unregistering..crashed about 2:30 this afternoon, I have it logged will peel it out and post if wanted..still have the double line ringing thing happening also...but now it answers on the first line..a second phone will continue to ring..it has to be hooked to make it stop.. thanks russell
  4. that is what I thought you were saying...except what I am saying is why is it there in first place? I removed it, it should be gone.. I was just wondering where it could be hiding in the software..I would like to not see that error.. russell
  5. I am not really clear on this response, as I am not a programming guru... I know the domain does not exist, that is why it was changed. this "smtp.mail.thesoundshop-llc.com" is a mail server I address I used originally in the admin software in the CS410, it did not work (wrong guess) email_server.bmp this "smtp.bizmail.yahoo.com" is the mail server address I replaced it with, it works now, I removed this "smtp.mail.thesoundshop-llc.com" and replaced it with this "smtp.bizmail.yahoo.com" (right one after reading POP3 info on my domain server) if I replaced one with the other, why do I keep seeing the first...in the logs? thanks russell
  6. Hi, pushed new DSP this morning, thanks for the heads up, checked the log this evening and still getting the cannont resolve error...that email server address is still hiding in there somewhere.. I will see how it went in the morning, system has not locked yet.. is this system available yet in 25 extensions? and when do you think the extended memory ones will be out? all the best, russell
  7. I am running 3.0.0.2899 thanks for the response russell
  8. Ok, Other than the 3 errors that I can reproduce at will, (logs posted under hang up detects a ring, posts 14, 15, 16) I have one other problem that happens from time to time (like every 3 to 4 days), the fxo1 port locks into the line off hook state (green light is on and unhooking the cable from the port which should create another CPC pulse does not disconnect it), when this happens only my analog phone will ring on the second line, as line one is off hook, there is no dial tone, you see the fxo2 port flashing on the switch when a call comes in, but the switch does not respond in any way other than the flashing light, I have to do a hard reset (power off and on), and the system starts working again, I have not yet caught a log of this, but will the next time it happens...it is real frustrating, as I lose the CDR report for the day, I have been logging all the call features, so I get everything that transpires during the day, except while the port is locked..therefore you lose the AA also. The phones show to stay registered, but you cannot make any calls, untill resetting the system. Any one else seeing this? I have not yet figured out how to make this happen. Also, I keep seeing this in my logs 8] 2008/04/12 12:07:35: DNS: Add dns_cname smtp.bizmail.yahoo.com smtp.bizmail.mail.yahoo4.akadns.net (ttl=1800) [8] 2008/04/12 12:07:35: DNS: Add dns_a smtp.bizmail.mail.yahoo4.akadns.net 68.142.200.11 (ttl=300) [8] 2008/04/12 12:07:36: SMTP: Connect to 68.142.200.11:25 [8] 2008/04/12 12:07:36: SMTP: Received 220 smtp106.biz.mail.mud.yahoo.com ESMTP [8] 2008/04/12 12:07:36: SMTP: Received 250-smtp106.biz.mail.mud.yahoo.com 250-AUTH LOGIN PLAIN XYMCOOKIE 250-PIPELINING 250 8BITMIME [8] 2008/04/12 12:07:36: SMTP: Received 334 VXNlcm5hbWU6 [8] 2008/04/12 12:07:36: SMTP: Received 334 UGFzc3dvcmQ6 [3] 2008/04/12 12:07:36: PSTN: Channel 0 going to IDLE [8] 2008/04/12 12:07:36: SMTP: Received 235 ok, go ahead (#2.0.0) [8] 2008/04/12 12:07:36: SMTP: Received 250 ok [8] 2008/04/12 12:07:37: Last message repeated 2 times [8] 2008/04/12 12:07:37: SMTP: Received 354 go ahead [8] 2008/04/12 12:07:37: SMTP: Received 250 ok 1208020056 qp 4422 [8] 2008/04/12 12:07:37: SMTP: Received 221 smtp106.biz.mail.mud.yahoo.com [8] 2008/04/12 12:07:37: Sucessfully sent email to <russell@thesoundshop-llc.com> [3] 2008/04/12 12:07:37: SMTP: Cannot resolve smtp.mail.thesoundshop-llc.com I keep seeing this email server(the one above) address in the logs, it is one I tried earlier and then changed to smtp.bizmail.yahoo.com which works, but nowhere in anyplace in the gui's do I use the other...it is like it is comming from a leftover uncleared data register, that did not change when I changed the email server. Where would you find this in the appliance using the SSH to clear it out...? thanks, Russell
  9. Ok now for the fun one, this one I can reproduce regulary, what I have to do is call line 2, the first call will not answer..as I have shown, then call line 2 again, that call will answer and seems to act ok, then call line 1 again, what happens is the extensions ring showing 2 lines at once, you pick up to answer the system answers on the second line, no one is there, you can put it on hold and hit the first line button that wil put the call on hold and then you can unhold it and get the call...the calling party hears music on hold first and then the hears the called party...pretty neat little game to play when this happens... seach for occurrences of "received on" within the log to see... [3] 2008/04/10 09:50:35: PSTN: Channel 0 going to RING [5] 2008/04/10 09:50:38: PSTN: eVAPI_CALLER_ID_DETECTED_EVENT [5] 2008/04/10 09:50:38: PSTN: Received on 0: Caller-ID 3255131178 [5] 2008/04/10 09:50:38: PSTN: Received on 0: Name THE SOUND SHOP [9] 2008/04/10 09:50:38: SIP Rx udp:127.0.0.1:5062: INVITE sip:3256725804@localhost;user=phone SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5062 From: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1956297539 To: <sip:3256725804@localhost;user=phone> Call-ID: baaab916@fxo Contact: <sip:127.0.0.1:5062> CSeq: 1 INVITE Content-Type: application/sdp Content-Length: 137 v=0 o=root 0 0 IN IP4 1.1.1.2 s=- c=IN IP4 1.1.1.2 t=0 0 m=audio 2078 RTP/AVP 0 101 a=rtpmap:101 telephone-event/8000 a=ptime:20 [7] 2008/04/10 09:50:38: UDP: Opening socket on port 55018 [7] 2008/04/10 09:50:38: UDP: Opening socket on port 55019 [5] 2008/04/10 09:50:38: Identify trunk (IP address/port and domain match) 5 [9] 2008/04/10 09:50:38: Resolve 425: aaaa udp 127.0.0.1 5062 [9] 2008/04/10 09:50:38: Resolve 425: a udp 127.0.0.1 5062 [9] 2008/04/10 09:50:38: Resolve 425: udp 127.0.0.1 5062 [9] 2008/04/10 09:50:38: SIP Tx udp:127.0.0.1:5062: SIP/2.0 100 Trying Via: SIP/2.0/UDP 127.0.0.1:5062 From: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1956297539 To: <sip:3256725804@localhost;user=phone>;tag=c00cbed448 Call-ID: baaab916@fxo CSeq: 1 INVITE Content-Length: 0 [5] 2008/04/10 09:50:38: PSTN: eVAPI_CALLER_ID_DETECTED_EVENT [7] 2008/04/10 09:50:38: Set packet length to 20 [6] 2008/04/10 09:50:38: Sending RTP for baaab916@fxo#c00cbed448 to 1.1.1.2:2078 [5] 2008/04/10 09:50:38: Trunk PSTN1 sends call to 72 [8] 2008/04/10 09:50:38: Play audio_moh/noise.wav [7] 2008/04/10 09:50:38: Hunt Group 72: Moving to next stage [7] 2008/04/10 09:50:38: Hunt group 72 called 2 registrations [5] 2008/04/10 09:50:38: PSTN: Received on 1: Caller-ID 3255131178 [7] 2008/04/10 09:50:38: Set packet length to 20 [9] 2008/04/10 09:50:38: Resolve 426: aaaa udp 127.0.0.1 5062 [9] 2008/04/10 09:50:38: Resolve 426: a udp 127.0.0.1 5062 [9] 2008/04/10 09:50:38: Resolve 426: udp 127.0.0.1 5062 [9] 2008/04/10 09:50:38: SIP Tx udp:127.0.0.1:5062: SIP/2.0 183 Ringing Via: SIP/2.0/UDP 127.0.0.1:5062 From: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1956297539 To: <sip:3256725804@localhost;user=phone>;tag=c00cbed448 Call-ID: baaab916@fxo CSeq: 1 INVITE Contact: <sip:3256725804@127.0.0.1:5060;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/3.0.0.2899 Content-Type: application/sdp Content-Length: 206 v=0 o=- 963854412 963854412 IN IP4 127.0.0.1 s=- c=IN IP4 127.0.0.1 t=0 0 m=audio 55018 RTP/AVP 0 101 a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv [7] 2008/04/10 09:50:38: UDP: Opening socket on port 62690 [7] 2008/04/10 09:50:38: UDP: Opening socket on port 62691 [9] 2008/04/10 09:50:38: Using outbound proxy sip:192.168.1.101:2614;transport=tls because of flow-label [9] 2008/04/10 09:50:38: Resolve 427: url sip:192.168.1.101:2614;transport=tls [9] 2008/04/10 09:50:38: Resolve 427: a tls 192.168.1.101 2614 [9] 2008/04/10 09:50:38: Resolve 427: tls 192.168.1.101 2614 [9] 2008/04/10 09:50:38: SIP Tx tls:192.168.1.101:2614: INVITE sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n SIP/2.0 Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-f9daac678e35eccefae7af0c85ba2101;rport From: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1213919521 To: <sip:3256725804@localhost;user=phone> Call-ID: be4800cd@pbx CSeq: 523 INVITE Max-Forwards: 70 Contact: <sip:40@192.168.1.100:5061;transport=tls> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pbxnsip-PBX/3.0.0.2899 Content-Type: application/sdp Content-Length: 386 v=0 o=- 1814072997 1814072997 IN IP4 192.168.1.100 s=- c=IN IP4 192.168.1.100 t=0 0 m=audio 62690 RTP/AVP 0 8 9 2 3 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:27Kzdm31wQsZGrs5Zita0i0FmlEETAAOWJjtOnpf 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] 2008/04/10 09:50:38: UDP: Opening socket on port 60046 [7] 2008/04/10 09:50:38: UDP: Opening socket on port 60047 [9] 2008/04/10 09:50:38: Using outbound proxy sip:192.168.1.102:2358;transport=tls because of flow-label [9] 2008/04/10 09:50:38: Resolve 428: url sip:192.168.1.102:2358;transport=tls [9] 2008/04/10 09:50:38: Resolve 428: a tls 192.168.1.102 2358 [9] 2008/04/10 09:50:38: Resolve 428: tls 192.168.1.102 2358 [9] 2008/04/10 09:50:38: SIP Tx tls:192.168.1.102:2358: INVITE sip:41@192.168.1.102:2060;transport=tls;line=94aa7mxt SIP/2.0 Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-c55a23ffcba2c8431c6995527c6e074e;rport From: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=829959946 To: <sip:3256725804@localhost;user=phone> Call-ID: c2b83f2d@pbx CSeq: 972 INVITE Max-Forwards: 70 Contact: <sip:41@192.168.1.100:5061;transport=tls> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pbxnsip-PBX/3.0.0.2899 Content-Type: application/sdp Content-Length: 386 v=0 o=- 1111676072 1111676072 IN IP4 192.168.1.100 s=- c=IN IP4 192.168.1.100 t=0 0 m=audio 60046 RTP/AVP 0 8 9 2 3 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:RtoOcTTgnzl68D3H8EsfiDhZA5c5WanMJShUs7L1 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 [5] 2008/04/10 09:50:38: PSTN: Received on 1: Name THE SOUND SHOP [9] 2008/04/10 09:50:38: SIP Rx udp:127.0.0.1:5062: INVITE sip:3256723475@localhost;user=phone SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5062 From: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1330573317 To: <sip:3256723475@localhost;user=phone> Call-ID: cc458e19@fxo Contact: <sip:127.0.0.1:5062> CSeq: 1 INVITE Content-Type: application/sdp Content-Length: 137 v=0 o=root 0 0 IN IP4 1.1.1.2 s=- c=IN IP4 1.1.1.2 t=0 0 m=audio 2080 RTP/AVP 0 101 a=rtpmap:101 telephone-event/8000 a=ptime:20 [7] 2008/04/10 09:50:38: UDP: Opening socket on port 55816 [7] 2008/04/10 09:50:38: UDP: Opening socket on port 55817 [5] 2008/04/10 09:50:38: Identify trunk (IP address/port and domain match) 5 [9] 2008/04/10 09:50:38: Resolve 429: aaaa udp 127.0.0.1 5062 [9] 2008/04/10 09:50:38: Resolve 429: a udp 127.0.0.1 5062 [9] 2008/04/10 09:50:38: Resolve 429: udp 127.0.0.1 5062 [9] 2008/04/10 09:50:38: SIP Tx udp:127.0.0.1:5062: SIP/2.0 100 Trying Via: SIP/2.0/UDP 127.0.0.1:5062 From: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1330573317 To: <sip:3256723475@localhost;user=phone>;tag=9f853f62b9 Call-ID: cc458e19@fxo CSeq: 1 INVITE Content-Length: 0 [7] 2008/04/10 09:50:38: Set packet length to 20 [6] 2008/04/10 09:50:38: Sending RTP for cc458e19@fxo#9f853f62b9 to 1.1.1.2:2080 [5] 2008/04/10 09:50:38: Trunk PSTN1 sends call to 72 [5] 2008/04/10 09:50:38: PSTN: Response code: 100 [5] 2008/04/10 09:50:38: PSTN: Response code: 183 [8] 2008/04/10 09:50:38: Play audio_moh/noise.wav [7] 2008/04/10 09:50:38: Hunt Group 72: Moving to next stage [7] 2008/04/10 09:50:38: Hunt group 72 called 2 registrations [7] 2008/04/10 09:50:38: Set packet length to 20 [9] 2008/04/10 09:50:38: Resolve 430: aaaa udp 127.0.0.1 5062 [9] 2008/04/10 09:50:38: Resolve 430: a udp 127.0.0.1 5062 [9] 2008/04/10 09:50:38: Resolve 430: udp 127.0.0.1 5062 [9] 2008/04/10 09:50:38: SIP Tx udp:127.0.0.1:5062: SIP/2.0 183 Ringing Via: SIP/2.0/UDP 127.0.0.1:5062 From: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1330573317 To: <sip:3256723475@localhost;user=phone>;tag=9f853f62b9 Call-ID: cc458e19@fxo CSeq: 1 INVITE Contact: <sip:3256723475@127.0.0.1:5060;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/3.0.0.2899 Content-Type: application/sdp Content-Length: 208 v=0 o=- 2066418937 2066418937 IN IP4 127.0.0.1 s=- c=IN IP4 127.0.0.1 t=0 0 m=audio 55816 RTP/AVP 0 101 a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv [7] 2008/04/10 09:50:38: UDP: Opening socket on port 57200 [7] 2008/04/10 09:50:38: UDP: Opening socket on port 57201 [9] 2008/04/10 09:50:38: Using outbound proxy sip:192.168.1.101:2614;transport=tls because of flow-label [9] 2008/04/10 09:50:38: Resolve 431: url sip:192.168.1.101:2614;transport=tls [9] 2008/04/10 09:50:38: Resolve 431: a tls 192.168.1.101 2614 [9] 2008/04/10 09:50:38: Resolve 431: tls 192.168.1.101 2614 [9] 2008/04/10 09:50:38: SIP Tx tls:192.168.1.101:2614: INVITE sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n SIP/2.0 Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-795ba654881a546cd196f5b7166c0662;rport From: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1658042394 To: <sip:3256723475@localhost;user=phone> Call-ID: ce792cdd@pbx CSeq: 9718 INVITE Max-Forwards: 70 Contact: <sip:40@192.168.1.100:5061;transport=tls> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pbxnsip-PBX/3.0.0.2899 Content-Type: application/sdp Content-Length: 386 v=0 o=- 2036358899 2036358899 IN IP4 192.168.1.100 s=- c=IN IP4 192.168.1.100 t=0 0 m=audio 57200 RTP/AVP 0 8 9 2 3 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:SFJvIVfVvQWsVrf5g6YT7Z0hvPDWo5WOmVuxCkSK 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] 2008/04/10 09:50:38: UDP: Opening socket on port 50794 [7] 2008/04/10 09:50:38: UDP: Opening socket on port 50795 [9] 2008/04/10 09:50:38: Using outbound proxy sip:192.168.1.102:2358;transport=tls because of flow-label [9] 2008/04/10 09:50:38: Resolve 432: url sip:192.168.1.102:2358;transport=tls [9] 2008/04/10 09:50:38: Resolve 432: a tls 192.168.1.102 2358 [9] 2008/04/10 09:50:38: Resolve 432: tls 192.168.1.102 2358 [9] 2008/04/10 09:50:38: SIP Tx tls:192.168.1.102:2358: INVITE sip:41@192.168.1.102:2060;transport=tls;line=94aa7mxt SIP/2.0 Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-78fdc559082cc3d548b51b53d5840b87;rport From: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=184923319 To: <sip:3256723475@localhost;user=phone> Call-ID: a7ef3565@pbx CSeq: 941 INVITE Max-Forwards: 70 Contact: <sip:41@192.168.1.100:5061;transport=tls> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pbxnsip-PBX/3.0.0.2899 Content-Type: application/sdp Content-Length: 384 v=0 o=- 150428609 150428609 IN IP4 192.168.1.100 s=- c=IN IP4 192.168.1.100 t=0 0 m=audio 50794 RTP/AVP 0 8 9 2 3 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:8edFl/syNBzvJPOSuoErFTI1Wb0z5Zqgk87i/gfQ 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] 2008/04/10 09:50:38: SIP Rx tls:192.168.1.101:2614: SIP/2.0 180 Ringing Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-f9daac678e35eccefae7af0c85ba2101;rport=5061 From: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1213919521 To: <sip:3256725804@localhost;user=phone>;tag=ve112iifw2 Call-ID: be4800cd@pbx CSeq: 523 INVITE Contact: <sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n>;flow-id=1 Require: 100rel RSeq: 1 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Content-Length: 0 [9] 2008/04/10 09:50:38: Resolve 433: url sip:192.168.1.101:2614;transport=tls [9] 2008/04/10 09:50:38: Resolve 433: a tls 192.168.1.101 2614 [9] 2008/04/10 09:50:38: Resolve 433: tls 192.168.1.101 2614 [9] 2008/04/10 09:50:38: SIP Tx tls:192.168.1.101:2614: PRACK sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n SIP/2.0 Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-72c1970c907da13fa76269fb3e59c8cd;rport From: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1213919521 To: <sip:3256725804@localhost;user=phone>;tag=ve112iifw2 Call-ID: be4800cd@pbx CSeq: 524 PRACK Max-Forwards: 70 Contact: <sip:40@192.168.1.100:5061;transport=tls> RAck: 1 523 INVITE Content-Length: 0 [5] 2008/04/10 09:50:38: PSTN: Response code: 100 [8] 2008/04/10 09:50:38: Play audio_en/ringback.wav [9] 2008/04/10 09:50:38: SIP Rx tls:192.168.1.102:2358: SIP/2.0 180 Ringing Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-c55a23ffcba2c8431c6995527c6e074e;rport=5061 From: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=829959946 To: <sip:3256725804@localhost;user=phone>;tag=t153uxg8ia Call-ID: c2b83f2d@pbx CSeq: 972 INVITE Contact: <sip:41@192.168.1.102:2060;transport=tls;line=94aa7mxt>;flow-id=1 Require: 100rel RSeq: 1 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Content-Length: 0 [9] 2008/04/10 09:50:38: Resolve 434: url sip:192.168.1.102:2358;transport=tls [9] 2008/04/10 09:50:38: Resolve 434: a tls 192.168.1.102 2358 [9] 2008/04/10 09:50:38: Resolve 434: tls 192.168.1.102 2358 [9] 2008/04/10 09:50:38: SIP Tx tls:192.168.1.102:2358: PRACK sip:41@192.168.1.102:2060;transport=tls;line=94aa7mxt SIP/2.0 Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-80fa95266c2da1afd45ffd6c51e9ebb3;rport From: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=829959946 To: <sip:3256725804@localhost;user=phone>;tag=t153uxg8ia Call-ID: c2b83f2d@pbx CSeq: 973 PRACK Max-Forwards: 70 Contact: <sip:41@192.168.1.100:5061;transport=tls> RAck: 1 972 INVITE Content-Length: 0 [8] 2008/04/10 09:50:38: Play audio_en/ringback.wav [5] 2008/04/10 09:50:38: PSTN: Response code: 183 [9] 2008/04/10 09:50:38: SIP Rx tls:192.168.1.101:2614: SIP/2.0 180 Ringing Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-795ba654881a546cd196f5b7166c0662;rport=5061 From: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1658042394 To: <sip:3256723475@localhost;user=phone>;tag=k67nty7ini Call-ID: ce792cdd@pbx CSeq: 9718 INVITE Contact: <sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n>;flow-id=1 Require: 100rel RSeq: 1 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Content-Length: 0 [9] 2008/04/10 09:50:38: Resolve 435: url sip:192.168.1.101:2614;transport=tls [9] 2008/04/10 09:50:38: Resolve 435: a tls 192.168.1.101 2614 [9] 2008/04/10 09:50:38: Resolve 435: tls 192.168.1.101 2614 [9] 2008/04/10 09:50:38: SIP Tx tls:192.168.1.101:2614: PRACK sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n SIP/2.0 Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-42c3ee21b2b22d876cfb3d1fd4fac0ea;rport From: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1658042394 To: <sip:3256723475@localhost;user=phone>;tag=k67nty7ini Call-ID: ce792cdd@pbx CSeq: 9719 PRACK Max-Forwards: 70 Contact: <sip:40@192.168.1.100:5061;transport=tls> RAck: 1 9718 INVITE Content-Length: 0 [8] 2008/04/10 09:50:38: Play audio_en/ringback.wav [9] 2008/04/10 09:50:38: SIP Rx tls:192.168.1.102:2358: SIP/2.0 180 Ringing Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-78fdc559082cc3d548b51b53d5840b87;rport=5061 From: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=184923319 To: <sip:3256723475@localhost;user=phone>;tag=l81zzl3z1b Call-ID: a7ef3565@pbx CSeq: 941 INVITE Contact: <sip:41@192.168.1.102:2060;transport=tls;line=94aa7mxt>;flow-id=1 Require: 100rel RSeq: 1 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Content-Length: 0 [9] 2008/04/10 09:50:38: Resolve 436: url sip:192.168.1.102:2358;transport=tls [9] 2008/04/10 09:50:38: Resolve 436: a tls 192.168.1.102 2358 [9] 2008/04/10 09:50:38: Resolve 436: tls 192.168.1.102 2358 [9] 2008/04/10 09:50:38: SIP Tx tls:192.168.1.102:2358: PRACK sip:41@192.168.1.102:2060;transport=tls;line=94aa7mxt SIP/2.0 Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-2ba2e9f29dcce53650d8d5f6ac5d1f23;rport From: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=184923319 To: <sip:3256723475@localhost;user=phone>;tag=l81zzl3z1b Call-ID: a7ef3565@pbx CSeq: 942 PRACK Max-Forwards: 70 Contact: <sip:41@192.168.1.100:5061;transport=tls> RAck: 1 941 INVITE Content-Length: 0 [8] 2008/04/10 09:50:38: Play audio_en/ringback.wav [9] 2008/04/10 09:50:38: SIP Rx tls:192.168.1.101:2614: SIP/2.0 200 Ok Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-72c1970c907da13fa76269fb3e59c8cd;rport=5061 From: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1213919521 To: <sip:3256725804@localhost;user=phone>;tag=ve112iifw2 Call-ID: be4800cd@pbx CSeq: 524 PRACK Contact: <sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n>;flow-id=1 Content-Length: 0 [7] 2008/04/10 09:50:38: Call be4800cd@pbx#1213919521: Clear last request [9] 2008/04/10 09:50:38: SIP Rx tls:192.168.1.101:2614: SIP/2.0 200 Ok Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-42c3ee21b2b22d876cfb3d1fd4fac0ea;rport=5061 From: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1658042394 To: <sip:3256723475@localhost;user=phone>;tag=k67nty7ini Call-ID: ce792cdd@pbx CSeq: 9719 PRACK Contact: <sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n>;flow-id=1 Content-Length: 0 [7] 2008/04/10 09:50:38: Call ce792cdd@pbx#1658042394: Clear last request [9] 2008/04/10 09:50:38: SIP Rx tls:192.168.1.102:2358: SIP/2.0 200 Ok Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-80fa95266c2da1afd45ffd6c51e9ebb3;rport=5061 From: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=829959946 To: <sip:3256725804@localhost;user=phone>;tag=t153uxg8ia Call-ID: c2b83f2d@pbx CSeq: 973 PRACK Contact: <sip:41@192.168.1.102:2060;transport=tls;line=94aa7mxt>;flow-id=1 Content-Length: 0 [7] 2008/04/10 09:50:38: Call c2b83f2d@pbx#829959946: Clear last request [9] 2008/04/10 09:50:39: SIP Rx tls:192.168.1.102:2358: SIP/2.0 200 Ok Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-2ba2e9f29dcce53650d8d5f6ac5d1f23;rport=5061 From: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=184923319 To: <sip:3256723475@localhost;user=phone>;tag=l81zzl3z1b Call-ID: a7ef3565@pbx CSeq: 942 PRACK Contact: <sip:41@192.168.1.102:2060;transport=tls;line=94aa7mxt>;flow-id=1 Content-Length: 0 [7] 2008/04/10 09:50:39: Call a7ef3565@pbx#184923319: Clear last request [9] 2008/04/10 09:50:39: SIP Tr udp:127.0.0.1:5062: SIP/2.0 183 Ringing Via: SIP/2.0/UDP 127.0.0.1:5062 From: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1956297539 To: <sip:3256725804@localhost;user=phone>;tag=c00cbed448 Call-ID: baaab916@fxo CSeq: 1 INVITE Contact: <sip:3256725804@127.0.0.1:5060;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/3.0.0.2899 Content-Type: application/sdp Content-Length: 206 v=0 o=- 963854412 963854412 IN IP4 127.0.0.1 s=- c=IN IP4 127.0.0.1 t=0 0 m=audio 55018 RTP/AVP 0 101 a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv [5] 2008/04/10 09:50:39: PSTN: Response code: 183 [9] 2008/04/10 09:50:39: SIP Tr udp:127.0.0.1:5062: SIP/2.0 183 Ringing Via: SIP/2.0/UDP 127.0.0.1:5062 From: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1330573317 To: <sip:3256723475@localhost;user=phone>;tag=9f853f62b9 Call-ID: cc458e19@fxo CSeq: 1 INVITE Contact: <sip:3256723475@127.0.0.1:5060;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/3.0.0.2899 Content-Type: application/sdp Content-Length: 208 v=0 o=- 2066418937 2066418937 IN IP4 127.0.0.1 s=- c=IN IP4 127.0.0.1 t=0 0 m=audio 55816 RTP/AVP 0 101 a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv [5] 2008/04/10 09:50:39: PSTN: Response code: 183 [3] 2008/04/10 09:50:39: PSTN: Channel 0 going to NO_RING [9] 2008/04/10 09:50:40: SIP Tr udp:127.0.0.1:5062: SIP/2.0 183 Ringing Via: SIP/2.0/UDP 127.0.0.1:5062 From: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1956297539 To: <sip:3256725804@localhost;user=phone>;tag=c00cbed448 Call-ID: baaab916@fxo CSeq: 1 INVITE Contact: <sip:3256725804@127.0.0.1:5060;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/3.0.0.2899 Content-Type: application/sdp Content-Length: 206 v=0 o=- 963854412 963854412 IN IP4 127.0.0.1 s=- c=IN IP4 127.0.0.1 t=0 0 m=audio 55018 RTP/AVP 0 101 a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv [5] 2008/04/10 09:50:40: PSTN: Response code: 183 [9] 2008/04/10 09:50:40: SIP Tr udp:127.0.0.1:5062: SIP/2.0 183 Ringing Via: SIP/2.0/UDP 127.0.0.1:5062 From: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1330573317 To: <sip:3256723475@localhost;user=phone>;tag=9f853f62b9 Call-ID: cc458e19@fxo CSeq: 1 INVITE Contact: <sip:3256723475@127.0.0.1:5060;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/3.0.0.2899 Content-Type: application/sdp Content-Length: 208 v=0 o=- 2066418937 2066418937 IN IP4 127.0.0.1 s=- c=IN IP4 127.0.0.1 t=0 0 m=audio 55816 RTP/AVP 0 101 a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv [5] 2008/04/10 09:50:40: PSTN: Response code: 183 [3] 2008/04/10 09:50:41: PSTN: Channel 0 going to RING [5] 2008/04/10 09:50:41: PSTN: Ringing, but last invite = 1 [9] 2008/04/10 09:50:42: SIP Tr udp:127.0.0.1:5062: SIP/2.0 183 Ringing Via: SIP/2.0/UDP 127.0.0.1:5062 From: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1956297539 To: <sip:3256725804@localhost;user=phone>;tag=c00cbed448 Call-ID: baaab916@fxo CSeq: 1 INVITE Contact: <sip:3256725804@127.0.0.1:5060;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/3.0.0.2899 Content-Type: application/sdp Content-Length: 206 v=0 o=- 963854412 963854412 IN IP4 127.0.0.1 s=- c=IN IP4 127.0.0.1 t=0 0 m=audio 55018 RTP/AVP 0 101 a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv [5] 2008/04/10 09:50:42: PSTN: Response code: 183 [9] 2008/04/10 09:50:42: SIP Tr udp:127.0.0.1:5062: SIP/2.0 183 Ringing Via: SIP/2.0/UDP 127.0.0.1:5062 From: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1330573317 To: <sip:3256723475@localhost;user=phone>;tag=9f853f62b9 Call-ID: cc458e19@fxo CSeq: 1 INVITE Contact: <sip:3256723475@127.0.0.1:5060;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/3.0.0.2899 Content-Type: application/sdp Content-Length: 208 v=0 o=- 2066418937 2066418937 IN IP4 127.0.0.1 s=- c=IN IP4 127.0.0.1 t=0 0 m=audio 55816 RTP/AVP 0 101 a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv [5] 2008/04/10 09:50:42: PSTN: Response code: 183 [7] 2008/04/10 09:50:43: Hunt Group 72: Moving to next stage [7] 2008/04/10 09:50:43: Hunt group 72 called 0 registrations [7] 2008/04/10 09:50:43: Hunt Group 72: Moving to next stage [7] 2008/04/10 09:50:43: Hunt group 72 called 0 registrations [9] 2008/04/10 09:50:46: SIP Tr udp:127.0.0.1:5062: SIP/2.0 183 Ringing Via: SIP/2.0/UDP 127.0.0.1:5062 From: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1956297539 To: <sip:3256725804@localhost;user=phone>;tag=c00cbed448 Call-ID: baaab916@fxo CSeq: 1 INVITE Contact: <sip:3256725804@127.0.0.1:5060;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/3.0.0.2899 Content-Type: application/sdp Content-Length: 206 v=0 o=- 963854412 963854412 IN IP4 127.0.0.1 s=- c=IN IP4 127.0.0.1 t=0 0 m=audio 55018 RTP/AVP 0 101 a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv [5] 2008/04/10 09:50:46: PSTN: Response code: 183 [3] 2008/04/10 09:50:46: PSTN: Channel 0 going to NO_RING [9] 2008/04/10 09:50:46: SIP Tr udp:127.0.0.1:5062: SIP/2.0 183 Ringing Via: SIP/2.0/UDP 127.0.0.1:5062 From: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1330573317 To: <sip:3256723475@localhost;user=phone>;tag=9f853f62b9 Call-ID: cc458e19@fxo CSeq: 1 INVITE Contact: <sip:3256723475@127.0.0.1:5060;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/3.0.0.2899 Content-Type: application/sdp Content-Length: 208 v=0 o=- 2066418937 2066418937 IN IP4 127.0.0.1 s=- c=IN IP4 127.0.0.1 t=0 0 m=audio 55816 RTP/AVP 0 101 a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv [5] 2008/04/10 09:50:46: PSTN: Response code: 183 [3] 2008/04/10 09:50:47: PSTN: Channel 0 going to RING [5] 2008/04/10 09:50:47: PSTN: Ringing, but last invite = 1 [7] 2008/04/10 09:50:48: Hunt Group 72: Moving to next stage [7] 2008/04/10 09:50:48: Hunt group 72 called 0 registrations [7] 2008/04/10 09:50:48: Hunt Group 72: Moving to next stage [7] 2008/04/10 09:50:48: Hunt group 72 called 0 registrations [9] 2008/04/10 09:50:48: SIP Rx tls:192.168.1.101:2614: SIP/2.0 200 Ok Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-f9daac678e35eccefae7af0c85ba2101;rport=5061 From: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1213919521 To: <sip:3256725804@localhost;user=phone>;tag=ve112iifw2 Call-ID: be4800cd@pbx CSeq: 523 INVITE Contact: <sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n>;flow-id=1 User-Agent: snom300/7.1.30 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, callerid Content-Type: application/sdp Content-Length: 459 v=0 o=root 72619833 72619834 IN IP4 192.168.1.101 s=call c=IN IP4 192.168.1.101 t=0 0 m=audio 63850 RTP/AVP 0 8 9 2 3 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:iQOaQK1ugUoXci63fXVnXrqLQvOnvTlwm79vmtjx 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=encryption:optional a=alt:1 0.9 : user 9kksj== 192.168.1.101 63850 a=sendrecv [7] 2008/04/10 09:50:48: Call be4800cd@pbx#1213919521: Clear last INVITE [6] 2008/04/10 09:50:48: Sending RTP for be4800cd@pbx#1213919521 to 192.168.1.101:63850 [9] 2008/04/10 09:50:48: Resolve 437: url sip:192.168.1.101:2614;transport=tls [9] 2008/04/10 09:50:48: Resolve 437: a tls 192.168.1.101 2614 [9] 2008/04/10 09:50:48: Resolve 437: tls 192.168.1.101 2614 [9] 2008/04/10 09:50:48: SIP Tx tls:192.168.1.101:2614: ACK sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n SIP/2.0 Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-f1eef136ef0e7efa6a434109f75c828d;rport From: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1213919521 To: <sip:3256725804@localhost;user=phone>;tag=ve112iifw2 Call-ID: be4800cd@pbx CSeq: 523 ACK Max-Forwards: 70 Contact: <sip:40@192.168.1.100:5061;transport=tls> Content-Length: 0 [7] 2008/04/10 09:50:48: Determine pass-through mode after receiving response [9] 2008/04/10 09:50:48: Resolve 438: tls 192.168.1.102 2358 [9] 2008/04/10 09:50:48: SIP Tx tls:192.168.1.102:2358: CANCEL sip:41@192.168.1.102:2060;transport=tls;line=94aa7mxt SIP/2.0 Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-c55a23ffcba2c8431c6995527c6e074e;rport From: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=829959946 To: <sip:3256725804@localhost;user=phone> Call-ID: c2b83f2d@pbx CSeq: 972 CANCEL Max-Forwards: 70 Reason: SIP;cause=200;text="Call completed elsewhere" Content-Length: 0 [9] 2008/04/10 09:50:48: Resolve 439: aaaa udp 127.0.0.1 5062 [9] 2008/04/10 09:50:48: Resolve 439: a udp 127.0.0.1 5062 [9] 2008/04/10 09:50:48: Resolve 439: udp 127.0.0.1 5062 [9] 2008/04/10 09:50:48: SIP Tx udp:127.0.0.1:5062: SIP/2.0 200 Ok Via: SIP/2.0/UDP 127.0.0.1:5062 From: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1956297539 To: <sip:3256725804@localhost;user=phone>;tag=c00cbed448 Call-ID: baaab916@fxo CSeq: 1 INVITE Contact: <sip:3256725804@127.0.0.1:5060;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/3.0.0.2899 Content-Type: application/sdp Content-Length: 206 v=0 o=- 963854412 963854412 IN IP4 127.0.0.1 s=- c=IN IP4 127.0.0.1 t=0 0 m=audio 55018 RTP/AVP 0 101 a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv [5] 2008/04/10 09:50:48: PSTN: Response code: 200 [9] 2008/04/10 09:50:48: SIP Rx udp:127.0.0.1:5062: ACK sip:3256725804@localhost;user=phone SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5062 From: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1956297539 To: <sip:3256725804@localhost;user=phone> Call-ID: baaab916@fxo Contact: <sip:127.0.0.1:5062> CSeq: 1 ACK Content-Length: 0 [5] 2008/04/10 09:50:48: PSTN: RTP destination=100007f [5] 2008/04/10 09:50:48: PSTN: RTP destination=55018 [9] 2008/04/10 09:50:48: SIP Rx tls:192.168.1.102:2358: SIP/2.0 200 OK Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-c55a23ffcba2c8431c6995527c6e074e;rport=5061 From: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=829959946 To: <sip:3256725804@localhost;user=phone>;tag=t153uxg8ia Call-ID: c2b83f2d@pbx CSeq: 972 CANCEL Content-Length: 0 [7] 2008/04/10 09:50:48: Call c2b83f2d@pbx#829959946: Clear last request [9] 2008/04/10 09:50:48: SIP Rx tls:192.168.1.102:2358: SIP/2.0 487 Request Terminated Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-c55a23ffcba2c8431c6995527c6e074e;rport=5061 From: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=829959946 To: <sip:3256725804@localhost;user=phone>;tag=t153uxg8ia Call-ID: c2b83f2d@pbx CSeq: 972 INVITE Contact: <sip:41@192.168.1.102:2060;transport=tls;line=94aa7mxt>;flow-id=1 Content-Length: 0 [7] 2008/04/10 09:50:48: Call c2b83f2d@pbx#829959946: Clear last INVITE [9] 2008/04/10 09:50:48: Resolve 440: url sip:192.168.1.102:2358;transport=tls [9] 2008/04/10 09:50:48: Resolve 440: a tls 192.168.1.102 2358 [9] 2008/04/10 09:50:48: Resolve 440: tls 192.168.1.102 2358 [9] 2008/04/10 09:50:48: SIP Tx tls:192.168.1.102:2358: ACK sip:41@192.168.1.102:2060;transport=tls;line=94aa7mxt SIP/2.0 Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-c55a23ffcba2c8431c6995527c6e074e;rport From: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=829959946 To: <sip:3256725804@localhost;user=phone>;tag=t153uxg8ia Call-ID: c2b83f2d@pbx CSeq: 972 ACK Max-Forwards: 70 Contact: <sip:41@192.168.1.100:5061;transport=tls> Content-Length: 0 [5] 2008/04/10 09:50:48: INVITE Response: Terminate c2b83f2d@pbx [7] 2008/04/10 09:50:48: Other Ports: 5 [7] 2008/04/10 09:50:48: Call Port: a7ef3565@pbx#184923319 [7] 2008/04/10 09:50:48: Call Port: baaab916@fxo#c00cbed448 [7] 2008/04/10 09:50:48: Call Port: be4800cd@pbx#1213919521 [7] 2008/04/10 09:50:48: Call Port: cc458e19@fxo#9f853f62b9 [7] 2008/04/10 09:50:48: Call Port: ce792cdd@pbx#1658042394 [5] 2008/04/10 09:50:48: PSTN: RTP OOB codec=101 [6] 2008/04/10 09:50:48: PSTN: Start call on 0 [5] 2008/04/10 09:50:48: PSTN: Channel 0 goes offhook [3] 2008/04/10 09:50:48: PSTN: Channel 0 going to TALKING [5] 2008/04/10 09:50:48: PSTN: Country Code set to 64 [5] 2008/04/10 09:50:48: PSTN: Tone Detection set to 0 [9] 2008/04/10 09:50:52: SIP Rx tls:192.168.1.101:2614: INVITE sip:40@192.168.1.100:5061;transport=tls SIP/2.0 Via: SIP/2.0/TLS 192.168.1.101:2060;branch=z9hG4bK-7d3wqn17yyn0;rport From: <sip:3256725804@localhost;user=phone>;tag=ve112iifw2 To: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1213919521 Call-ID: be4800cd@pbx CSeq: 1 INVITE Max-Forwards: 70 Contact: <sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n>;flow-id=1 P-Key-Flags: keys="3" User-Agent: snom300/7.1.30 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, callerid Session-Expires: 3600;refresher=uas Min-SE: 90 Proxy-Require: buttons Content-Type: application/sdp Content-Length: 521 v=0 o=root 72619833 72619835 IN IP4 192.168.1.101 s=call c=IN IP4 192.168.1.101 t=0 0 m=audio 63850 RTP/AVP 9 0 8 2 3 18 4 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:iQOaQK1ugUoXci63fXVnXrqLQvOnvTlwm79vmtjx a=rtpmap:9 g722/8000 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:18 g729/8000 a=rtpmap:4 g723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=encryption:optional a=alt:1 0.9 : user 9kksj== 192.168.1.101 63850 a=sendonly [7] 2008/04/10 09:50:52: Set packet length to 20 [9] 2008/04/10 09:50:52: Resolve 441: tls 192.168.1.101 2614 [9] 2008/04/10 09:50:52: SIP Tx tls:192.168.1.101:2614: SIP/2.0 200 Ok Via: SIP/2.0/TLS 192.168.1.101:2060;branch=z9hG4bK-7d3wqn17yyn0;rport=2614 From: <sip:3256725804@localhost;user=phone>;tag=ve112iifw2 To: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1213919521 Call-ID: be4800cd@pbx CSeq: 1 INVITE Contact: <sip:40@192.168.1.100:5061;transport=tls> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pbxnsip-PBX/3.0.0.2899 Content-Type: application/sdp Content-Length: 398 v=0 o=- 1814072997 1814072997 IN IP4 192.168.1.100 s=- c=IN IP4 192.168.1.100 t=0 0 m=audio 62690 RTP/AVP 0 8 9 2 3 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:27Kzdm31wQsZGrs5Zita0i0FmlEETAAOWJjtOnpf 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=ptime:20 a=recvonly [9] 2008/04/10 09:50:52: SIP Rx tls:192.168.1.101:2614: SIP/2.0 200 Ok Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-795ba654881a546cd196f5b7166c0662;rport=5061 From: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1658042394 To: <sip:3256723475@localhost;user=phone>;tag=k67nty7ini Call-ID: ce792cdd@pbx CSeq: 9718 INVITE Contact: <sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n>;flow-id=1 User-Agent: snom300/7.1.30 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, callerid Content-Type: application/sdp Content-Length: 463 v=0 o=root 1101773985 1101773986 IN IP4 192.168.1.101 s=call c=IN IP4 192.168.1.101 t=0 0 m=audio 54456 RTP/AVP 0 8 9 2 3 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:6EntKzyU6HUFgzR0HQ1lQeVZ/vormeVvpmowjswY 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=encryption:optional a=alt:1 0.9 : user 9kksj== 192.168.1.101 54456 a=sendrecv [7] 2008/04/10 09:50:52: Call ce792cdd@pbx#1658042394: Clear last INVITE [6] 2008/04/10 09:50:52: Sending RTP for ce792cdd@pbx#1658042394 to 192.168.1.101:54456 [9] 2008/04/10 09:50:52: Resolve 442: url sip:192.168.1.101:2614;transport=tls [9] 2008/04/10 09:50:52: Resolve 442: a tls 192.168.1.101 2614 [9] 2008/04/10 09:50:52: Resolve 442: tls 192.168.1.101 2614 [9] 2008/04/10 09:50:52: SIP Tx tls:192.168.1.101:2614: ACK sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n SIP/2.0 Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-0be30940ce01eeca9c4803e1291e794f;rport From: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1658042394 To: <sip:3256723475@localhost;user=phone>;tag=k67nty7ini Call-ID: ce792cdd@pbx CSeq: 9718 ACK Max-Forwards: 70 Contact: <sip:40@192.168.1.100:5061;transport=tls> Content-Length: 0 [7] 2008/04/10 09:50:52: Determine pass-through mode after receiving response [9] 2008/04/10 09:50:52: Resolve 443: tls 192.168.1.102 2358 [9] 2008/04/10 09:50:52: SIP Tx tls:192.168.1.102:2358: CANCEL sip:41@192.168.1.102:2060;transport=tls;line=94aa7mxt SIP/2.0 Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-78fdc559082cc3d548b51b53d5840b87;rport From: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=184923319 To: <sip:3256723475@localhost;user=phone> Call-ID: a7ef3565@pbx CSeq: 941 CANCEL Max-Forwards: 70 Reason: SIP;cause=200;text="Call completed elsewhere" Content-Length: 0 [9] 2008/04/10 09:50:52: Resolve 444: aaaa udp 127.0.0.1 5062 [9] 2008/04/10 09:50:52: Resolve 444: a udp 127.0.0.1 5062 [9] 2008/04/10 09:50:52: Resolve 444: udp 127.0.0.1 5062 [9] 2008/04/10 09:50:52: SIP Tx udp:127.0.0.1:5062: SIP/2.0 200 Ok Via: SIP/2.0/UDP 127.0.0.1:5062 From: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1330573317 To: <sip:3256723475@localhost;user=phone>;tag=9f853f62b9 Call-ID: cc458e19@fxo CSeq: 1 INVITE Contact: <sip:3256723475@127.0.0.1:5060;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/3.0.0.2899 Content-Type: application/sdp Content-Length: 208 v=0 o=- 2066418937 2066418937 IN IP4 127.0.0.1 s=- c=IN IP4 127.0.0.1 t=0 0 m=audio 55816 RTP/AVP 0 101 a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv [9] 2008/04/10 09:50:52: SIP Rx tls:192.168.1.101:2614: ACK sip:40@192.168.1.100:5061;transport=tls SIP/2.0 Via: SIP/2.0/TLS 192.168.1.101:2060;branch=z9hG4bK-fwh4dc3rv8kq;rport From: <sip:3256725804@localhost;user=phone>;tag=ve112iifw2 To: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1213919521 Call-ID: be4800cd@pbx CSeq: 1 ACK Max-Forwards: 70 Contact: <sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n>;flow-id=1 Proxy-Require: buttons Content-Length: 0 [5] 2008/04/10 09:50:52: PSTN: Response code: 200 [9] 2008/04/10 09:50:52: SIP Rx tls:192.168.1.102:2358: SIP/2.0 200 OK Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-78fdc559082cc3d548b51b53d5840b87;rport=5061 From: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=184923319 To: <sip:3256723475@localhost;user=phone>;tag=l81zzl3z1b Call-ID: a7ef3565@pbx CSeq: 941 CANCEL Content-Length: 0 [7] 2008/04/10 09:50:52: Call a7ef3565@pbx#184923319: Clear last request [9] 2008/04/10 09:50:52: SIP Rx tls:192.168.1.102:2358: SIP/2.0 487 Request Terminated Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-78fdc559082cc3d548b51b53d5840b87;rport=5061 From: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=184923319 To: <sip:3256723475@localhost;user=phone>;tag=l81zzl3z1b Call-ID: a7ef3565@pbx CSeq: 941 INVITE Contact: <sip:41@192.168.1.102:2060;transport=tls;line=94aa7mxt>;flow-id=1 Content-Length: 0 [7] 2008/04/10 09:50:52: Call a7ef3565@pbx#184923319: Clear last INVITE [9] 2008/04/10 09:50:52: Resolve 445: url sip:192.168.1.102:2358;transport=tls [9] 2008/04/10 09:50:52: Resolve 445: a tls 192.168.1.102 2358 [9] 2008/04/10 09:50:52: Resolve 445: tls 192.168.1.102 2358 [9] 2008/04/10 09:50:52: SIP Tx tls:192.168.1.102:2358: ACK sip:41@192.168.1.102:2060;transport=tls;line=94aa7mxt SIP/2.0 Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-78fdc559082cc3d548b51b53d5840b87;rport From: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=184923319 To: <sip:3256723475@localhost;user=phone>;tag=l81zzl3z1b Call-ID: a7ef3565@pbx CSeq: 941 ACK Max-Forwards: 70 Contact: <sip:41@192.168.1.100:5061;transport=tls> Content-Length: 0 [5] 2008/04/10 09:50:52: INVITE Response: Terminate a7ef3565@pbx [7] 2008/04/10 09:50:52: Other Ports: 4 [7] 2008/04/10 09:50:52: Call Port: baaab916@fxo#c00cbed448 [7] 2008/04/10 09:50:52: Call Port: be4800cd@pbx#1213919521 [7] 2008/04/10 09:50:52: Call Port: cc458e19@fxo#9f853f62b9 [7] 2008/04/10 09:50:52: Call Port: ce792cdd@pbx#1658042394 [9] 2008/04/10 09:50:52: SIP Tr udp:127.0.0.1:5062: SIP/2.0 200 Ok Via: SIP/2.0/UDP 127.0.0.1:5062 From: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1330573317 To: <sip:3256723475@localhost;user=phone>;tag=9f853f62b9 Call-ID: cc458e19@fxo CSeq: 1 INVITE Contact: <sip:3256723475@127.0.0.1:5060;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/3.0.0.2899 Content-Type: application/sdp Content-Length: 208 v=0 o=- 2066418937 2066418937 IN IP4 127.0.0.1 s=- c=IN IP4 127.0.0.1 t=0 0 m=audio 55816 RTP/AVP 0 101 a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv [5] 2008/04/10 09:50:52: PSTN: Response code: 200 [9] 2008/04/10 09:50:52: SIP Rx tls:192.168.1.102:2358: REGISTER sip:localhost SIP/2.0 Via: SIP/2.0/TLS 192.168.1.102:2060;branch=z9hG4bK-wn0vp59hpqv9;rport From: "41" <sip:41@localhost>;tag=gmz36br37m To: "41" <sip:41@localhost> Call-ID: 3c26701aaf59-kw566buonz1c CSeq: 14235 REGISTER Max-Forwards: 70 Contact: <sip:41@192.168.1.102:2060;transport=tls;line=94aa7mxt>;flow-id=1;q=1.0;+sip.instance="<urn:uuid:ec9ea11d-0cc3-4c4e-b306-6953a0386af5>" Contact: <http://192.168.1.102:80> Contact: <https://192.168.1.102:443> User-Agent: snom300/7.1.30 Supported: gruu Allow-Events: dialog X-Real-IP: 192.168.1.102 Proxy-Require: buttons Expires: 3600 Content-Length: 0 [8] 2008/04/10 09:50:52: Packet authenticated by transport layer [9] 2008/04/10 09:50:52: Resolve 446: tls 192.168.1.102 2358 [9] 2008/04/10 09:50:52: SIP Tx tls:192.168.1.102:2358: SIP/2.0 200 Ok Via: SIP/2.0/TLS 192.168.1.102:2060;branch=z9hG4bK-wn0vp59hpqv9;rport=2358 From: "41" <sip:41@localhost>;tag=gmz36br37m To: "41" <sip:41@localhost>;tag=b4f17a5688 Call-ID: 3c26701aaf59-kw566buonz1c CSeq: 14235 REGISTER Contact: <sip:41@192.168.1.102:2060;transport=tls;line=94aa7mxt>;expires=178 Contact: <http://192.168.1.102:80>;expires=178 Contact: <https://192.168.1.102:443>;expires=178 Content-Length: 0 [9] 2008/04/10 09:50:53: SIP Tr udp:127.0.0.1:5062: SIP/2.0 200 Ok Via: SIP/2.0/UDP 127.0.0.1:5062 From: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1330573317 To: <sip:3256723475@localhost;user=phone>;tag=9f853f62b9 Call-ID: cc458e19@fxo CSeq: 1 INVITE Contact: <sip:3256723475@127.0.0.1:5060;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/3.0.0.2899 Content-Type: application/sdp Content-Length: 208 v=0 o=- 2066418937 2066418937 IN IP4 127.0.0.1 s=- c=IN IP4 127.0.0.1 t=0 0 m=audio 55816 RTP/AVP 0 101 a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv [5] 2008/04/10 09:50:53: PSTN: Response code: 200 [9] 2008/04/10 09:50:55: SIP Tr udp:127.0.0.1:5062: SIP/2.0 200 Ok Via: SIP/2.0/UDP 127.0.0.1:5062 From: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1330573317 To: <sip:3256723475@localhost;user=phone>;tag=9f853f62b9 Call-ID: cc458e19@fxo CSeq: 1 INVITE Contact: <sip:3256723475@127.0.0.1:5060;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/3.0.0.2899 Content-Type: application/sdp Content-Length: 208 v=0 o=- 2066418937 2066418937 IN IP4 127.0.0.1 s=- c=IN IP4 127.0.0.1 t=0 0 m=audio 55816 RTP/AVP 0 101 a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv [5] 2008/04/10 09:50:55: PSTN: Response code: 200 [9] 2008/04/10 09:50:59: SIP Rx tls:192.168.1.101:2614: INVITE sip:40@192.168.1.100:5061;transport=tls SIP/2.0 Via: SIP/2.0/TLS 192.168.1.101:2060;branch=z9hG4bK-dqgcmhhzt4ao;rport From: <sip:3256723475@localhost;user=phone>;tag=k67nty7ini To: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1658042394 Call-ID: ce792cdd@pbx CSeq: 1 INVITE Max-Forwards: 70 Contact: <sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n>;flow-id=1 P-Key-Flags: keys="3" User-Agent: snom300/7.1.30 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, callerid Session-Expires: 3600;refresher=uas Min-SE: 90 Proxy-Require: buttons Content-Type: application/sdp Content-Length: 525 v=0 o=root 1101773985 1101773987 IN IP4 192.168.1.101 s=call c=IN IP4 192.168.1.101 t=0 0 m=audio 54456 RTP/AVP 9 0 8 2 3 18 4 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:6EntKzyU6HUFgzR0HQ1lQeVZ/vormeVvpmowjswY a=rtpmap:9 g722/8000 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:18 g729/8000 a=rtpmap:4 g723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=encryption:optional a=alt:1 0.9 : user 9kksj== 192.168.1.101 54456 a=sendonly [7] 2008/04/10 09:50:59: Set packet length to 20 [9] 2008/04/10 09:50:59: Resolve 447: tls 192.168.1.101 2614 [9] 2008/04/10 09:50:59: SIP Tx tls:192.168.1.101:2614: SIP/2.0 200 Ok Via: SIP/2.0/TLS 192.168.1.101:2060;branch=z9hG4bK-dqgcmhhzt4ao;rport=2614 From: <sip:3256723475@localhost;user=phone>;tag=k67nty7ini To: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1658042394 Call-ID: ce792cdd@pbx CSeq: 1 INVITE Contact: <sip:40@192.168.1.100:5061;transport=tls> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pbxnsip-PBX/3.0.0.2899 Content-Type: application/sdp Content-Length: 398 v=0 o=- 2036358899 2036358899 IN IP4 192.168.1.100 s=- c=IN IP4 192.168.1.100 t=0 0 m=audio 57200 RTP/AVP 0 8 9 2 3 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:SFJvIVfVvQWsVrf5g6YT7Z0hvPDWo5WOmVuxCkSK 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=ptime:20 a=recvonly [9] 2008/04/10 09:50:59: SIP Rx tls:192.168.1.101:2614: ACK sip:40@192.168.1.100:5061;transport=tls SIP/2.0 Via: SIP/2.0/TLS 192.168.1.101:2060;branch=z9hG4bK-wuzuqb8xkevm;rport From: <sip:3256723475@localhost;user=phone>;tag=k67nty7ini To: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1658042394 Call-ID: ce792cdd@pbx CSeq: 1 ACK Max-Forwards: 70 Contact: <sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n>;flow-id=1 Proxy-Require: buttons Content-Length: 0 [9] 2008/04/10 09:50:59: SIP Rx tls:192.168.1.101:2614: SUBSCRIBE sip:40@localhost;user=phone SIP/2.0 Via: SIP/2.0/TLS 192.168.1.101:2060;branch=z9hG4bK-89yq4h62rmed;rport From: <sip:40@localhost>;tag=vgds1qnpk0 To: <sip:40@localhost;user=phone>;tag=9d4243196b Call-ID: 3c2674105300-yc4ja694jzkk CSeq: 13678 SUBSCRIBE Max-Forwards: 70 Contact: <sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n>;flow-id=1 Event: message-summary Accept: application/simple-message-summary User-Agent: snom300/7.1.30 Proxy-Require: buttons Expires: 3600 Content-Length: 0 [8] 2008/04/10 09:50:59: Packet authenticated by transport layer [9] 2008/04/10 09:50:59: Resolve 448: tls 192.168.1.101 2614 [9] 2008/04/10 09:50:59: SIP Tx tls:192.168.1.101:2614: SIP/2.0 200 Ok Via: SIP/2.0/TLS 192.168.1.101:2060;branch=z9hG4bK-89yq4h62rmed;rport=2614 From: <sip:40@localhost>;tag=vgds1qnpk0 To: <sip:40@localhost;user=phone>;tag=9d4243196b Call-ID: 3c2674105300-yc4ja694jzkk CSeq: 13678 SUBSCRIBE Contact: <sip:192.168.1.100:5061;transport=tls> Expires: 182 Content-Length: 0 [9] 2008/04/10 09:50:59: SIP Tr udp:127.0.0.1:5062: SIP/2.0 200 Ok Via: SIP/2.0/UDP 127.0.0.1:5062 From: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1330573317 To: <sip:3256723475@localhost;user=phone>;tag=9f853f62b9 Call-ID: cc458e19@fxo CSeq: 1 INVITE Contact: <sip:3256723475@127.0.0.1:5060;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/3.0.0.2899 Content-Type: application/sdp Content-Length: 208 v=0 o=- 2066418937 2066418937 IN IP4 127.0.0.1 s=- c=IN IP4 127.0.0.1 t=0 0 m=audio 55816 RTP/AVP 0 101 a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv [5] 2008/04/10 09:50:59: PSTN: Response code: 200 [9] 2008/04/10 09:51:04: SIP Rx tls:192.168.1.101:2614: INVITE sip:40@192.168.1.100:5061;transport=tls SIP/2.0 Via: SIP/2.0/TLS 192.168.1.101:2060;branch=z9hG4bK-0ox4u2ne519o;rport From: <sip:3256725804@localhost;user=phone>;tag=ve112iifw2 To: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1213919521 Call-ID: be4800cd@pbx CSeq: 2 INVITE Max-Forwards: 70 Contact: <sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n>;flow-id=1 P-Key-Flags: keys="3" User-Agent: snom300/7.1.30 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, callerid Session-Expires: 3600;refresher=uas Min-SE: 90 Proxy-Require: buttons Content-Type: application/sdp Content-Length: 521 v=0 o=root 72619833 72619836 IN IP4 192.168.1.101 s=call c=IN IP4 192.168.1.101 t=0 0 m=audio 63850 RTP/AVP 9 0 8 2 3 18 4 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:iQOaQK1ugUoXci63fXVnXrqLQvOnvTlwm79vmtjx a=rtpmap:9 g722/8000 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:18 g729/8000 a=rtpmap:4 g723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=encryption:optional a=alt:1 0.9 : user 9kksj== 192.168.1.101 63850 a=sendrecv [7] 2008/04/10 09:51:04: Set packet length to 20 [9] 2008/04/10 09:51:04: Resolve 449: tls 192.168.1.101 2614 [9] 2008/04/10 09:51:04: SIP Tx tls:192.168.1.101:2614: SIP/2.0 200 Ok Via: SIP/2.0/TLS 192.168.1.101:2060;branch=z9hG4bK-0ox4u2ne519o;rport=2614 From: <sip:3256725804@localhost;user=phone>;tag=ve112iifw2 To: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1213919521 Call-ID: be4800cd@pbx CSeq: 2 INVITE Contact: <sip:40@192.168.1.100:5061;transport=tls> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pbxnsip-PBX/3.0.0.2899 Content-Type: application/sdp Content-Length: 398 v=0 o=- 1814072997 1814072998 IN IP4 192.168.1.100 s=- c=IN IP4 192.168.1.100 t=0 0 m=audio 62690 RTP/AVP 0 8 9 2 3 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:uKJD47d2GRAzTPXN7IibzoainkabVkoqCYHu2afW 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=ptime:20 a=sendrecv [9] 2008/04/10 09:51:05: SIP Rx tls:192.168.1.101:2614: ACK sip:40@192.168.1.100:5061;transport=tls SIP/2.0 Via: SIP/2.0/TLS 192.168.1.101:2060;branch=z9hG4bK-617u8w78jtdr;rport From: <sip:3256725804@localhost;user=phone>;tag=ve112iifw2 To: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1213919521 Call-ID: be4800cd@pbx CSeq: 2 ACK Max-Forwards: 70 Contact: <sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n>;flow-id=1 Proxy-Require: buttons Content-Length: 0 [9] 2008/04/10 09:51:06: SIP Rx tls:192.168.1.101:2614: REGISTER sip:localhost SIP/2.0 Via: SIP/2.0/TLS 192.168.1.101:2060;branch=z9hG4bK-d4l0jblczcy7;rport From: "40" <sip:40@localhost>;tag=aje04aahuz To: "40" <sip:40@localhost> Call-ID: 3c26741018ce-d9hh4ncip2ol CSeq: 14230 REGISTER Max-Forwards: 70 Contact: <sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n>;flow-id=1;q=1.0;+sip.instance="<urn:uuid:822ed566-4660-4bfb-90c3-2b5411c69eb0>" Contact: <http://192.168.1.101:80> Contact: <https://192.168.1.101:443> User-Agent: snom300/7.1.30 Supported: gruu Allow-Events: dialog X-Real-IP: 192.168.1.101 Proxy-Require: buttons Expires: 3600 Content-Length: 0 [8] 2008/04/10 09:51:06: Packet authenticated by transport layer [9] 2008/04/10 09:51:06: Resolve 450: tls 192.168.1.101 2614 [9] 2008/04/10 09:51:06: SIP Tx tls:192.168.1.101:2614: SIP/2.0 200 Ok Via: SIP/2.0/TLS 192.168.1.101:2060;branch=z9hG4bK-d4l0jblczcy7;rport=2614 From: "40" <sip:40@localhost>;tag=aje04aahuz To: "40" <sip:40@localhost>;tag=c02c2dcda3 Call-ID: 3c26741018ce-d9hh4ncip2ol CSeq: 14230 REGISTER Contact: <sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n>;expires=180 Contact: <http://192.168.1.101:80>;expires=180 Contact: <https://192.168.1.101:443>;expires=180 Content-Length: 0 [5] 2008/04/10 09:51:06: PSTN: Busy Tone detected on 0 [9] 2008/04/10 09:51:07: SIP Tr udp:127.0.0.1:5062: SIP/2.0 200 Ok Via: SIP/2.0/UDP 127.0.0.1:5062 From: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1330573317 To: <sip:3256723475@localhost;user=phone>;tag=9f853f62b9 Call-ID: cc458e19@fxo CSeq: 1 INVITE Contact: <sip:3256723475@127.0.0.1:5060;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/3.0.0.2899 Content-Type: application/sdp Content-Length: 208 v=0 o=- 2066418937 2066418937 IN IP4 127.0.0.1 s=- c=IN IP4 127.0.0.1 t=0 0 m=audio 55816 RTP/AVP 0 101 a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv [5] 2008/04/10 09:51:07: PSTN: Response code: 200 [9] 2008/04/10 09:51:12: SIP Rx tls:192.168.1.101:2614: INVITE sip:40@192.168.1.100:5061;transport=tls SIP/2.0 Via: SIP/2.0/TLS 192.168.1.101:2060;branch=z9hG4bK-84aeuxcd83pr;rport From: <sip:3256725804@localhost;user=phone>;tag=ve112iifw2 To: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1213919521 Call-ID: be4800cd@pbx CSeq: 3 INVITE Max-Forwards: 70 Contact: <sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n>;flow-id=1 P-Key-Flags: keys="3" User-Agent: snom300/7.1.30 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, callerid Session-Expires: 3600;refresher=uas Min-SE: 90 Proxy-Require: buttons Content-Type: application/sdp Content-Length: 521 v=0 o=root 72619833 72619837 IN IP4 192.168.1.101 s=call c=IN IP4 192.168.1.101 t=0 0 m=audio 63850 RTP/AVP 9 0 8 2 3 18 4 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:iQOaQK1ugUoXci63fXVnXrqLQvOnvTlwm79vmtjx a=rtpmap:9 g722/8000 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:18 g729/8000 a=rtpmap:4 g723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=encryption:optional a=alt:1 0.9 : user 9kksj== 192.168.1.101 63850 a=sendonly [7] 2008/04/10 09:51:12: Set packet length to 20 [9] 2008/04/10 09:51:12: Resolve 451: tls 192.168.1.101 2614 [9] 2008/04/10 09:51:12: SIP Tx tls:192.168.1.101:2614: SIP/2.0 200 Ok Via: SIP/2.0/TLS 192.168.1.101:2060;branch=z9hG4bK-84aeuxcd83pr;rport=2614 From: <sip:3256725804@localhost;user=phone>;tag=ve112iifw2 To: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1213919521 Call-ID: be4800cd@pbx CSeq: 3 INVITE Contact: <sip:40@192.168.1.100:5061;transport=tls> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pbxnsip-PBX/3.0.0.2899 Content-Type: application/sdp Content-Length: 398 v=0 o=- 1814072997 1814072998 IN IP4 192.168.1.100 s=- c=IN IP4 192.168.1.100 t=0 0 m=audio 62690 RTP/AVP 0 8 9 2 3 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:uKJD47d2GRAzTPXN7IibzoainkabVkoqCYHu2afW 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=ptime:20 a=recvonly [9] 2008/04/10 09:51:12: SIP Rx tls:192.168.1.101:2614: ACK sip:40@192.168.1.100:5061;transport=tls SIP/2.0 Via: SIP/2.0/TLS 192.168.1.101:2060;branch=z9hG4bK-g5kd9vd2hgs8;rport From: <sip:3256725804@localhost;user=phone>;tag=ve112iifw2 To: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1213919521 Call-ID: be4800cd@pbx CSeq: 3 ACK Max-Forwards: 70 Contact: <sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n>;flow-id=1 Proxy-Require: buttons Content-Length: 0 [9] 2008/04/10 09:51:19: SIP Rx tls:192.168.1.101:2614: INVITE sip:40@192.168.1.100:5061;transport=tls SIP/2.0 Via: SIP/2.0/TLS 192.168.1.101:2060;branch=z9hG4bK-rcnakugffl5x;rport From: <sip:3256723475@localhost;user=phone>;tag=k67nty7ini To: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1658042394 Call-ID: ce792cdd@pbx CSeq: 2 INVITE Max-Forwards: 70 Contact: <sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n>;flow-id=1 P-Key-Flags: keys="3" User-Agent: snom300/7.1.30 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, callerid Session-Expires: 3600;refresher=uas Min-SE: 90 Proxy-Require: buttons Content-Type: application/sdp Content-Length: 525 v=0 o=root 1101773985 1101773988 IN IP4 192.168.1.101 s=call c=IN IP4 192.168.1.101 t=0 0 m=audio 54456 RTP/AVP 9 0 8 2 3 18 4 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:6EntKzyU6HUFgzR0HQ1lQeVZ/vormeVvpmowjswY a=rtpmap:9 g722/8000 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:18 g729/8000 a=rtpmap:4 g723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=encryption:optional a=alt:1 0.9 : user 9kksj== 192.168.1.101 54456 a=sendrecv [7] 2008/04/10 09:51:19: Set packet length to 20 [9] 2008/04/10 09:51:19: Resolve 452: tls 192.168.1.101 2614 [9] 2008/04/10 09:51:19: SIP Tx tls:192.168.1.101:2614: SIP/2.0 200 Ok Via: SIP/2.0/TLS 192.168.1.101:2060;branch=z9hG4bK-rcnakugffl5x;rport=2614 From: <sip:3256723475@localhost;user=phone>;tag=k67nty7ini To: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1658042394 Call-ID: ce792cdd@pbx CSeq: 2 INVITE Contact: <sip:40@192.168.1.100:5061;transport=tls> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pbxnsip-PBX/3.0.0.2899 Content-Type: application/sdp Content-Length: 398 v=0 o=- 2036358899 2036358900 IN IP4 192.168.1.100 s=- c=IN IP4 192.168.1.100 t=0 0 m=audio 57200 RTP/AVP 0 8 9 2 3 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:X3jfQy9VXD+IqDRVlL3wYkOSAIktV9RYYFVGOf0c 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=ptime:20 a=sendrecv [9] 2008/04/10 09:51:19: SIP Rx tls:192.168.1.101:2614: ACK sip:40@192.168.1.100:5061;transport=tls SIP/2.0 Via: SIP/2.0/TLS 192.168.1.101:2060;branch=z9hG4bK-2to6qqqswo3y;rport From: <sip:3256723475@localhost;user=phone>;tag=k67nty7ini To: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1658042394 Call-ID: ce792cdd@pbx CSeq: 2 ACK Max-Forwards: 70 Contact: <sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n>;flow-id=1 Proxy-Require: buttons Content-Length: 0 [3] 2008/04/10 09:51:20: SMTP: Cannot resolve smtp.mail.thesoundshop-llc.com [9] 2008/04/10 09:51:31: SIP Rx tls:192.168.1.101:2614: BYE sip:40@192.168.1.100:5061;transport=tls SIP/2.0 Via: SIP/2.0/TLS 192.168.1.101:2060;branch=z9hG4bK-fk7owztky7tb;rport From: <sip:3256723475@localhost;user=phone>;tag=k67nty7ini To: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1658042394 Call-ID: ce792cdd@pbx CSeq: 3 BYE Max-Forwards: 70 Contact: <sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n>;flow-id=1 User-Agent: snom300/7.1.30 RTP-RxStat: Total_Rx_Pkts=19,Rx_Pkts=12,Rx_Pkts_Lost=0,Remote_Rx_Pkts_Lost=0 RTP-TxStat: Total_Tx_Pkts=968,Tx_Pkts=615,Remote_Tx_Pkts=0 Proxy-Require: buttons Content-Length: 0 [9] 2008/04/10 09:51:31: Resolve 453: tls 192.168.1.101 2614 [9] 2008/04/10 09:51:31: SIP Tx tls:192.168.1.101:2614: SIP/2.0 200 Ok Via: SIP/2.0/TLS 192.168.1.101:2060;branch=z9hG4bK-fk7owztky7tb;rport=2614 From: <sip:3256723475@localhost;user=phone>;tag=k67nty7ini To: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1658042394 Call-ID: ce792cdd@pbx CSeq: 3 BYE Contact: <sip:40@192.168.1.100:5061;transport=tls> User-Agent: pbxnsip-PBX/3.0.0.2899 RTP-RxStat: Dur=53,Pkt=971,Oct=170896,Underun=1946 RTP-TxStat: Dur=40,Pkt=29,Oct=3556 Content-Length: 0 [7] 2008/04/10 09:51:31: Other Ports: 3 [7] 2008/04/10 09:51:31: Call Port: baaab916@fxo#c00cbed448 [7] 2008/04/10 09:51:31: Call Port: be4800cd@pbx#1213919521 [7] 2008/04/10 09:51:31: Call Port: cc458e19@fxo#9f853f62b9 [8] 2008/04/10 09:51:31: SMTP: Connect to 68.142.200.11:25 [9] 2008/04/10 09:51:31: Resolve 454: url sip:127.0.0.1:5062 [9] 2008/04/10 09:51:31: Resolve 454: udp 127.0.0.1 5062 [9] 2008/04/10 09:51:31: SIP Tx udp:127.0.0.1:5062: BYE sip:127.0.0.1:5062 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK-29d2771eb98b126bb76f455f73d2498d;rport From: <sip:3256723475@localhost;user=phone>;tag=9f853f62b9 To: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1330573317 Call-ID: cc458e19@fxo CSeq: 16490 BYE Max-Forwards: 70 Contact: <sip:3256723475@127.0.0.1:5060;transport=udp> RTP-RxStat: Dur=53,Pkt=0,Oct=0,Underun=12 RTP-TxStat: Dur=40,Pkt=2628,Oct=452016 Content-Length: 0 [9] 2008/04/10 09:51:31: SIP Rx udp:127.0.0.1:5062: SIP/2.0 404 Not found Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK-29d2771eb98b126bb76f455f73d2498d;rport From: <sip:3256723475@localhost;user=phone>;tag=9f853f62b9 To: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1330573317 Call-ID: cc458e19@fxo CSeq: 16490 BYE Content-Length: 0 [7] 2008/04/10 09:51:31: Call cc458e19@fxo#9f853f62b9: Clear last request [5] 2008/04/10 09:51:31: BYE Response: Terminate cc458e19@fxo [7] 2008/04/10 09:51:31: Other Ports: 2 [7] 2008/04/10 09:51:31: Call Port: baaab916@fxo#c00cbed448 [7] 2008/04/10 09:51:31: Call Port: be4800cd@pbx#1213919521 [8] 2008/04/10 09:51:31: SMTP: Received 220 smtp104.biz.mail.mud.yahoo.com ESMTP [8] 2008/04/10 09:51:31: SMTP: Received 250-smtp104.biz.mail.mud.yahoo.com 250-AUTH LOGIN PLAIN XYMCOOKIE 250-PIPELINING 250 8BITMIME [8] 2008/04/10 09:51:32: SMTP: Received 334 VXNlcm5hbWU6 [8] 2008/04/10 09:51:32: SMTP: Received 334 UGFzc3dvcmQ6 [8] 2008/04/10 09:51:32: SMTP: Received 235 ok, go ahead (#2.0.0) [8] 2008/04/10 09:51:32: SMTP: Received 250 ok [8] 2008/04/10 09:51:32: Last message repeated 2 times [8] 2008/04/10 09:51:32: SMTP: Received 354 go ahead [8] 2008/04/10 09:51:33: SMTP: Received 250 ok 1207839092 qp 10292 [8] 2008/04/10 09:51:33: SMTP: Received 221 smtp104.biz.mail.mud.yahoo.com [8] 2008/04/10 09:51:33: Sucessfully sent email to <russell@thesoundshop-llc.com> [3] 2008/04/10 09:51:33: SMTP: Cannot resolve smtp.mail.thesoundshop-llc.com [9] 2008/04/10 09:51:34: SIP Rx tls:192.168.1.101:2614: INVITE sip:40@192.168.1.100:5061;transport=tls SIP/2.0 Via: SIP/2.0/TLS 192.168.1.101:2060;branch=z9hG4bK-dzad3hlzhe4z;rport From: <sip:3256725804@localhost;user=phone>;tag=ve112iifw2 To: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1213919521 Call-ID: be4800cd@pbx CSeq: 4 INVITE Max-Forwards: 70 Contact: <sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n>;flow-id=1 P-Key-Flags: keys="3" User-Agent: snom300/7.1.30 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, callerid Session-Expires: 3600;refresher=uas Min-SE: 90 Proxy-Require: buttons Content-Type: application/sdp Content-Length: 521 v=0 o=root 72619833 72619838 IN IP4 192.168.1.101 s=call c=IN IP4 192.168.1.101 t=0 0 m=audio 63850 RTP/AVP 9 0 8 2 3 18 4 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:iQOaQK1ugUoXci63fXVnXrqLQvOnvTlwm79vmtjx a=rtpmap:9 g722/8000 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:18 g729/8000 a=rtpmap:4 g723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=encryption:optional a=alt:1 0.9 : user 9kksj== 192.168.1.101 63850 a=sendrecv [7] 2008/04/10 09:51:34: Set packet length to 20 [9] 2008/04/10 09:51:34: Resolve 455: tls 192.168.1.101 2614 [9] 2008/04/10 09:51:34: SIP Tx tls:192.168.1.101:2614: SIP/2.0 200 Ok Via: SIP/2.0/TLS 192.168.1.101:2060;branch=z9hG4bK-dzad3hlzhe4z;rport=2614 From: <sip:3256725804@localhost;user=phone>;tag=ve112iifw2 To: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1213919521 Call-ID: be4800cd@pbx CSeq: 4 INVITE Contact: <sip:40@192.168.1.100:5061;transport=tls> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pbxnsip-PBX/3.0.0.2899 Content-Type: application/sdp Content-Length: 398 v=0 o=- 1814072997 1814072999 IN IP4 192.168.1.100 s=- c=IN IP4 192.168.1.100 t=0 0 m=audio 62690 RTP/AVP 0 8 9 2 3 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:+wKRozbmN/PWmTdomsCW8ZTuUeo0i+dQYEPkFOOn 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=ptime:20 a=sendrecv [9] 2008/04/10 09:51:34: SIP Rx tls:192.168.1.101:2614: ACK sip:40@192.168.1.100:5061;transport=tls SIP/2.0 Via: SIP/2.0/TLS 192.168.1.101:2060;branch=z9hG4bK-mhb8iebg71pe;rport From: <sip:3256725804@localhost;user=phone>;tag=ve112iifw2 To: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1213919521 Call-ID: be4800cd@pbx CSeq: 4 ACK Max-Forwards: 70 Contact: <sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n>;flow-id=1 Proxy-Require: buttons Content-Length: 0 [5] 2008/04/10 09:51:42: PSTN: Busy Tone detected on 0 [9] 2008/04/10 09:51:43: SIP Rx tls:192.168.1.101:2614: BYE sip:40@192.168.1.100:5061;transport=tls SIP/2.0 Via: SIP/2.0/TLS 192.168.1.101:2060;branch=z9hG4bK-5fu8xaouui7x;rport From: <sip:3256725804@localhost;user=phone>;tag=ve112iifw2 To: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1213919521 Call-ID: be4800cd@pbx CSeq: 5 BYE Max-Forwards: 70 Contact: <sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n>;flow-id=1 User-Agent: snom300/7.1.30 RTP-RxStat: Total_Rx_Pkts=1015,Rx_Pkts=469,Rx_Pkts_Lost=0,Remote_Rx_Pkts_Lost=0 RTP-TxStat: Total_Tx_Pkts=1016,Tx_Pkts=471,Remote_Tx_Pkts=0 Proxy-Require: buttons Content-Length: 0 [9] 2008/04/10 09:51:43: Resolve 456: tls 192.168.1.101 2614 [9] 2008/04/10 09:51:43: SIP Tx tls:192.168.1.101:2614: SIP/2.0 200 Ok Via: SIP/2.0/TLS 192.168.1.101:2060;branch=z9hG4bK-5fu8xaouui7x;rport=2614 From: <sip:3256725804@localhost;user=phone>;tag=ve112iifw2 To: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1213919521 Call-ID: be4800cd@pbx CSeq: 5 BYE Contact: <sip:40@192.168.1.100:5061;transport=tls> User-Agent: pbxnsip-PBX/3.0.0.2899 RTP-RxStat: Dur=65,Pkt=1016,Oct=178816,Underun=6 RTP-TxStat: Dur=55,Pkt=1043,Oct=180816 Content-Length: 0 [7] 2008/04/10 09:51:43: Other Ports: 1 [7] 2008/04/10 09:51:43: Call Port: baaab916@fxo#c00cbed448 [8] 2008/04/10 09:51:44: SMTP: Connect to 68.142.200.11:25 [9] 2008/04/10 09:51:44: Resolve 457: url sip:127.0.0.1:5062 [9] 2008/04/10 09:51:44: Resolve 457: udp 127.0.0.1 5062 [9] 2008/04/10 09:51:44: SIP Tx udp:127.0.0.1:5062: BYE sip:127.0.0.1:5062 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK-073f078b6f7a519cac4f081ed18ed0ec;rport From: <sip:3256725804@localhost;user=phone>;tag=c00cbed448 To: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1956297539 Call-ID: baaab916@fxo CSeq: 24189 BYE Max-Forwards: 70 Contact: <sip:3256725804@127.0.0.1:5060;transport=udp> RTP-RxStat: Dur=66,Pkt=2754,Oct=473688,Underun=40 RTP-TxStat: Dur=55,Pkt=3233,Oct=556076 Content-Length: 0 [5] 2008/04/10 09:51:44: PSTN: Received BYE message on channel 0 [9] 2008/04/10 09:51:44: SIP Rx udp:127.0.0.1:5062: SIP/2.0 200 Ok Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK-073f078b6f7a519cac4f081ed18ed0ec;rport From: <sip:3256725804@localhost;user=phone>;tag=c00cbed448 To: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1956297539 Call-ID: baaab916@fxo CSeq: 24189 BYE Content-Length: 0 [7] 2008/04/10 09:51:44: Call baaab916@fxo#c00cbed448: Clear last request [5] 2008/04/10 09:51:44: BYE Response: Terminate baaab916@fxo [3] 2008/04/10 09:51:44: PSTN: Channel 0: Hangup [5] 2008/04/10 09:51:44: PSTN: Channel 0 goes onhook [5] 2008/04/10 09:51:44: PSTN: enable_callerid 0 [3] 2008/04/10 09:51:44: PSTN: Channel 0 going to GO_ONHOOK [8] 2008/04/10 09:51:44: SMTP: Received 220 smtp108.biz.mail.mud.yahoo.com ESMTP [8] 2008/04/10 09:51:44: SMTP: Received 250-smtp108.biz.mail.mud.yahoo.com 250-AUTH LOGIN PLAIN XYMCOOKIE 250-PIPELINING 250 8BITMIME [8] 2008/04/10 09:51:44: SMTP: Received 334 VXNlcm5hbWU6 [8] 2008/04/10 09:51:44: SMTP: Received 334 UGFzc3dvcmQ6 [8] 2008/04/10 09:51:44: SMTP: Received 235 ok, go ahead (#2.0.0) [8] 2008/04/10 09:51:44: SMTP: Received 250 ok [8] 2008/04/10 09:51:44: Last message repeated 2 times [8] 2008/04/10 09:51:44: SMTP: Received 354 go ahead [3] 2008/04/10 09:51:45: PSTN: Channel 0 going to IDLE [8] 2008/04/10 09:51:45: SMTP: Received 250 ok 1207839104 qp 3575
  10. Ok so the next call comes in ok, and then after that call, I start calling line one, and the next 3 times I try the system it works fine, then I try line 2 again and the phones do not ring at all, but the weird thing I see in this log is the caller ID message.. I was calling from my cell phone but I see a call I received earlier in the caller ID.. [3] 2008/04/10 09:43:19: PSTN: Channel 1 going to RING [9] 2008/04/10 09:43:20: SIP Rx tls:192.168.1.102:2358: REGISTER sip:localhost SIP/2.0 Via: SIP/2.0/TLS 192.168.1.102:2060;branch=z9hG4bK-tiae9flqybwy;rport From: "41" <sip:41@localhost>;tag=6h8z2ywbws To: "41" <sip:41@localhost> Call-ID: 3c26701aaf59-kw566buonz1c CSeq: 14230 REGISTER Max-Forwards: 70 Contact: <sip:41@192.168.1.102:2060;transport=tls;line=94aa7mxt>;flow-id=1;q=1.0;+sip.instance="<urn:uuid:ec9ea11d-0cc3-4c4e-b306-6953a0386af5>" Contact: <http://192.168.1.102:80> Contact: <https://192.168.1.102:443> User-Agent: snom300/7.1.30 Supported: gruu Allow-Events: dialog X-Real-IP: 192.168.1.102 Proxy-Require: buttons Expires: 3600 Content-Length: 0 [8] 2008/04/10 09:43:20: Packet authenticated by transport layer [9] 2008/04/10 09:43:20: Resolve 397: tls 192.168.1.102 2358 [9] 2008/04/10 09:43:20: SIP Tx tls:192.168.1.102:2358: SIP/2.0 200 Ok Via: SIP/2.0/TLS 192.168.1.102:2060;branch=z9hG4bK-tiae9flqybwy;rport=2358 From: "41" <sip:41@localhost>;tag=6h8z2ywbws To: "41" <sip:41@localhost>;tag=b4f17a5688 Call-ID: 3c26701aaf59-kw566buonz1c CSeq: 14230 REGISTER Contact: <sip:41@192.168.1.102:2060;transport=tls;line=94aa7mxt>;expires=182 Contact: <http://192.168.1.102:80>;expires=182 Contact: <https://192.168.1.102:443>;expires=182 Content-Length: 0 [5] 2008/04/10 09:43:22: PSTN: eVAPI_CALLER_ID_DETECTED_EVENT [5] 2008/04/10 09:43:22: PSTN: Received on 1: Caller-ID 3255131178 [5] 2008/04/10 09:43:22: PSTN: Received on 1: Name THE SOUND SHOP [8] 2008/04/10 09:43:22: PSTN: Received Caller-ID on channel 1, but already sent INVITE [3] 2008/04/10 09:43:23: PSTN: Channel 1 going to NO_RING [3] 2008/04/10 09:43:25: PSTN: Channel 1 going to RING [5] 2008/04/10 09:43:25: PSTN: Ringing, but last invite = 1 [9] 2008/04/10 09:43:30: SIP Rx tls:192.168.1.101:2614: SUBSCRIBE sip:40@localhost;user=phone SIP/2.0 Via: SIP/2.0/TLS 192.168.1.101:2060;branch=z9hG4bK-c4rumwbxh7tm;rport From: <sip:40@localhost>;tag=vgds1qnpk0 To: <sip:40@localhost;user=phone>;tag=9d4243196b Call-ID: 3c2674105300-yc4ja694jzkk CSeq: 13673 SUBSCRIBE Max-Forwards: 70 Contact: <sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n>;flow-id=1 Event: message-summary Accept: application/simple-message-summary User-Agent: snom300/7.1.30 Proxy-Require: buttons Expires: 3600 Content-Length: 0 [8] 2008/04/10 09:43:30: Packet authenticated by transport layer [3] 2008/04/10 09:43:30: PSTN: Channel 1 going to NO_RING [9] 2008/04/10 09:43:30: Resolve 398: tls 192.168.1.101 2614 [9] 2008/04/10 09:43:30: SIP Tx tls:192.168.1.101:2614: SIP/2.0 200 Ok Via: SIP/2.0/TLS 192.168.1.101:2060;branch=z9hG4bK-c4rumwbxh7tm;rport=2614 From: <sip:40@localhost>;tag=vgds1qnpk0 To: <sip:40@localhost;user=phone>;tag=9d4243196b Call-ID: 3c2674105300-yc4ja694jzkk CSeq: 13673 SUBSCRIBE Contact: <sip:192.168.1.100:5061;transport=tls> Expires: 180 Content-Length: 0 [3] 2008/04/10 09:43:31: PSTN: Channel 1 going to RING [5] 2008/04/10 09:43:31: PSTN: Ringing, but last invite = 1 [9] 2008/04/10 09:43:32: SIP Rx tls:192.168.1.101:2614: REGISTER sip:localhost SIP/2.0 Via: SIP/2.0/TLS 192.168.1.101:2060;branch=z9hG4bK-pfh0d9vyfsmq;rport From: "40" <sip:40@localhost>;tag=jpos857gfh To: "40" <sip:40@localhost> Call-ID: 3c26741018ce-d9hh4ncip2ol CSeq: 14225 REGISTER Max-Forwards: 70 Contact: <sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n>;flow-id=1;q=1.0;+sip.instance="<urn:uuid:822ed566-4660-4bfb-90c3-2b5411c69eb0>" Contact: <http://192.168.1.101:80> Contact: <https://192.168.1.101:443> User-Agent: snom300/7.1.30 Supported: gruu Allow-Events: dialog X-Real-IP: 192.168.1.101 Proxy-Require: buttons Expires: 3600 Content-Length: 0 [8] 2008/04/10 09:43:32: Packet authenticated by transport layer [9] 2008/04/10 09:43:32: Resolve 399: tls 192.168.1.101 2614 [9] 2008/04/10 09:43:32: SIP Tx tls:192.168.1.101:2614: SIP/2.0 200 Ok Via: SIP/2.0/TLS 192.168.1.101:2060;branch=z9hG4bK-pfh0d9vyfsmq;rport=2614 From: "40" <sip:40@localhost>;tag=jpos857gfh To: "40" <sip:40@localhost>;tag=c02c2dcda3 Call-ID: 3c26741018ce-d9hh4ncip2ol CSeq: 14225 REGISTER Contact: <sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n>;expires=182 Contact: <http://192.168.1.101:80>;expires=182 Contact: <https://192.168.1.101:443>;expires=182 Content-Length: 0 [3] 2008/04/10 09:43:36: PSTN: Channel 1 going to NO_RING [9] 2008/04/10 09:43:42: SIP Rx udp:127.0.0.1:5062: CANCEL sip:3256723475@localhost;user=phone SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5062 From: "ABILEN IND SCH " <sip:3256771444@localhost;user=phone>;tag=1059961393 To: <sip:3256723475@localhost;user=phone> Call-ID: 83a5d578@fxo Contact: <sip:127.0.0.1:5062> CSeq: 1 CANCEL Content-Length: 0 [9] 2008/04/10 09:43:42: Resolve 400: aaaa udp 127.0.0.1 5062 [9] 2008/04/10 09:43:42: Resolve 400: a udp 127.0.0.1 5062 [9] 2008/04/10 09:43:42: Resolve 400: udp 127.0.0.1 5062 [9] 2008/04/10 09:43:42: SIP Tx udp:127.0.0.1:5062: SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP 127.0.0.1:5062 From: "ABILEN IND SCH " <sip:3256771444@localhost;user=phone>;tag=1059961393 To: <sip:3256723475@localhost;user=phone> Call-ID: 83a5d578@fxo CSeq: 1 CANCEL Content-Length: 0 [5] 2008/04/10 09:43:42: PSTN: Timeout without ring on 1, going to idle [3] 2008/04/10 09:43:42: PSTN: Channel 1 going to IDLE [5] 2008/04/10 09:43:42: PSTN: Response code: 481 [3] 2008/04/10 09:43:56: SMTP: Cannot resolve smtp.mail.thesoundshop-llc.com
  11. Allrighty, next call comes in on first FXO port, extension rings, answer extension,no one at other end, analog line keeps ringing, answer call on analog phone, but I get caller id on SNOM phone. [5] 2008/04/10 09:13:05: PSTN: Received on 0: Name ABILEN IND SCH [8] 2008/04/10 09:13:05: PSTN: Received Caller-ID on channel 0, but already sent INVITE [5] 2008/04/10 09:13:05: PSTN: Response code: 100 [5] 2008/04/10 09:13:05: PSTN: Response code: 183 [9] 2008/04/10 09:13:05: SIP Rx tls:192.168.1.101:2614: SIP/2.0 180 Ringing Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-a6612d6557f6f20952121e2346f72850;rport=5061 From: "ABILEN IND SCH " <sip:3256771444@localhost;user=phone>;tag=217507709 To: <sip:3256723475@localhost;user=phone>;tag=ooxg71bo4d Call-ID: a4449186@pbx CSeq: 29409 INVITE Contact: <sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n>;flow-id=1 Require: 100rel RSeq: 1 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Content-Length: 0 [9] 2008/04/10 09:13:05: Resolve 254: url sip:192.168.1.101:2614;transport=tls [9] 2008/04/10 09:13:05: Resolve 254: a tls 192.168.1.101 2614 [9] 2008/04/10 09:13:05: Resolve 254: tls 192.168.1.101 2614 [9] 2008/04/10 09:13:05: SIP Tx tls:192.168.1.101:2614: PRACK sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n SIP/2.0 Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-2a83d7a5343447c2ba9c9f230e923d20;rport From: "ABILEN IND SCH " <sip:3256771444@localhost;user=phone>;tag=217507709 To: <sip:3256723475@localhost;user=phone>;tag=ooxg71bo4d Call-ID: a4449186@pbx CSeq: 29410 PRACK Max-Forwards: 70 Contact: <sip:40@192.168.1.100:5061;transport=tls> RAck: 1 29409 INVITE Content-Length: 0 [8] 2008/04/10 09:13:05: Play audio_en/ringback.wav [9] 2008/04/10 09:13:05: SIP Rx tls:192.168.1.102:2358: SIP/2.0 180 Ringing Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-5ded06c142d0cd523a5e87c446470647;rport=5061 From: "ABILEN IND SCH " <sip:3256771444@localhost;user=phone>;tag=1195090579 To: <sip:3256723475@localhost;user=phone>;tag=s2wpjj2zgg Call-ID: 5f095176@pbx CSeq: 30726 INVITE Contact: <sip:41@192.168.1.102:2060;transport=tls;line=94aa7mxt>;flow-id=1 Require: 100rel RSeq: 1 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Content-Length: 0 [9] 2008/04/10 09:13:05: Resolve 255: url sip:192.168.1.102:2358;transport=tls [9] 2008/04/10 09:13:05: Resolve 255: a tls 192.168.1.102 2358 [9] 2008/04/10 09:13:05: Resolve 255: tls 192.168.1.102 2358 [9] 2008/04/10 09:13:05: SIP Tx tls:192.168.1.102:2358: PRACK sip:41@192.168.1.102:2060;transport=tls;line=94aa7mxt SIP/2.0 Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-bf69f588cbba0daaa1a352ca0e0a68ba;rport From: "ABILEN IND SCH " <sip:3256771444@localhost;user=phone>;tag=1195090579 To: <sip:3256723475@localhost;user=phone>;tag=s2wpjj2zgg Call-ID: 5f095176@pbx CSeq: 30727 PRACK Max-Forwards: 70 Contact: <sip:41@192.168.1.100:5061;transport=tls> RAck: 1 30726 INVITE Content-Length: 0 [8] 2008/04/10 09:13:05: Play audio_en/ringback.wav [9] 2008/04/10 09:13:05: SIP Rx tls:192.168.1.101:2614: SIP/2.0 200 Ok Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-2a83d7a5343447c2ba9c9f230e923d20;rport=5061 From: "ABILEN IND SCH " <sip:3256771444@localhost;user=phone>;tag=217507709 To: <sip:3256723475@localhost;user=phone>;tag=ooxg71bo4d Call-ID: a4449186@pbx CSeq: 29410 PRACK Contact: <sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n>;flow-id=1 Content-Length: 0 [7] 2008/04/10 09:13:05: Call a4449186@pbx#217507709: Clear last request [9] 2008/04/10 09:13:05: SIP Rx tls:192.168.1.102:2358: SIP/2.0 200 Ok Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-bf69f588cbba0daaa1a352ca0e0a68ba;rport=5061 From: "ABILEN IND SCH " <sip:3256771444@localhost;user=phone>;tag=1195090579 To: <sip:3256723475@localhost;user=phone>;tag=s2wpjj2zgg Call-ID: 5f095176@pbx CSeq: 30727 PRACK Contact: <sip:41@192.168.1.102:2060;transport=tls;line=94aa7mxt>;flow-id=1 Content-Length: 0 [7] 2008/04/10 09:13:05: Call 5f095176@pbx#1195090579: Clear last request [9] 2008/04/10 09:13:06: SIP Tr udp:127.0.0.1:5062: SIP/2.0 183 Ringing Via: SIP/2.0/UDP 127.0.0.1:5062 From: "ABILEN IND SCH " <sip:3256771444@localhost;user=phone>;tag=1059961393 To: <sip:3256723475@localhost;user=phone>;tag=c8db8ae26e Call-ID: 83a5d578@fxo CSeq: 1 INVITE Contact: <sip:3256723475@127.0.0.1:5060;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/3.0.0.2899 Content-Type: application/sdp Content-Length: 208 v=0 o=- 1076076106 1076076106 IN IP4 127.0.0.1 s=- c=IN IP4 127.0.0.1 t=0 0 m=audio 62278 RTP/AVP 0 101 a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv [5] 2008/04/10 09:13:06: PSTN: Response code: 183 [3] 2008/04/10 09:13:06: PSTN: Channel 0 going to NO_RING [9] 2008/04/10 09:13:07: SIP Tr udp:127.0.0.1:5062: SIP/2.0 183 Ringing Via: SIP/2.0/UDP 127.0.0.1:5062 From: "ABILEN IND SCH " <sip:3256771444@localhost;user=phone>;tag=1059961393 To: <sip:3256723475@localhost;user=phone>;tag=c8db8ae26e Call-ID: 83a5d578@fxo CSeq: 1 INVITE Contact: <sip:3256723475@127.0.0.1:5060;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/3.0.0.2899 Content-Type: application/sdp Content-Length: 208 v=0 o=- 1076076106 1076076106 IN IP4 127.0.0.1 s=- c=IN IP4 127.0.0.1 t=0 0 m=audio 62278 RTP/AVP 0 101 a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv [5] 2008/04/10 09:13:07: PSTN: Response code: 183 [3] 2008/04/10 09:13:08: PSTN: Channel 0 going to RING [5] 2008/04/10 09:13:08: PSTN: Ringing, but last invite = 1 [9] 2008/04/10 09:13:08: SIP Rx tls:192.168.1.102:2358: SIP/2.0 200 Ok Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-5ded06c142d0cd523a5e87c446470647;rport=5061 From: "ABILEN IND SCH " <sip:3256771444@localhost;user=phone>;tag=1195090579 To: <sip:3256723475@localhost;user=phone>;tag=s2wpjj2zgg Call-ID: 5f095176@pbx CSeq: 30726 INVITE Contact: <sip:41@192.168.1.102:2060;transport=tls;line=94aa7mxt>;flow-id=1 User-Agent: snom300/7.1.30 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, callerid Content-Type: application/sdp Content-Length: 463 v=0 o=root 1135634967 1135634968 IN IP4 192.168.1.102 s=call c=IN IP4 192.168.1.102 t=0 0 m=audio 50046 RTP/AVP 0 8 9 2 3 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:4PlRLbprmuw+qpsW9CkbT2QqYXuCwIiXZ6f9Olzd 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=encryption:optional a=alt:1 0.9 : user 9kksj== 192.168.1.102 50046 a=sendrecv [7] 2008/04/10 09:13:08: Call 5f095176@pbx#1195090579: Clear last INVITE [6] 2008/04/10 09:13:08: Sending RTP for 5f095176@pbx#1195090579 to 192.168.1.102:50046 [9] 2008/04/10 09:13:08: Resolve 256: url sip:192.168.1.102:2358;transport=tls [9] 2008/04/10 09:13:08: Resolve 256: a tls 192.168.1.102 2358 [9] 2008/04/10 09:13:08: Resolve 256: tls 192.168.1.102 2358 [9] 2008/04/10 09:13:08: SIP Tx tls:192.168.1.102:2358: ACK sip:41@192.168.1.102:2060;transport=tls;line=94aa7mxt SIP/2.0 Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-c9784da620ae5c4c196f13aa887103f9;rport From: "ABILEN IND SCH " <sip:3256771444@localhost;user=phone>;tag=1195090579 To: <sip:3256723475@localhost;user=phone>;tag=s2wpjj2zgg Call-ID: 5f095176@pbx CSeq: 30726 ACK Max-Forwards: 70 Contact: <sip:41@192.168.1.100:5061;transport=tls> Content-Length: 0 [7] 2008/04/10 09:13:08: Determine pass-through mode after receiving response [9] 2008/04/10 09:13:08: Resolve 257: tls 192.168.1.101 2614 [9] 2008/04/10 09:13:08: SIP Tx tls:192.168.1.101:2614: CANCEL sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n SIP/2.0 Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-a6612d6557f6f20952121e2346f72850;rport From: "ABILEN IND SCH " <sip:3256771444@localhost;user=phone>;tag=217507709 To: <sip:3256723475@localhost;user=phone> Call-ID: a4449186@pbx CSeq: 29409 CANCEL Max-Forwards: 70 Reason: SIP;cause=200;text="Call completed elsewhere" Content-Length: 0 [9] 2008/04/10 09:13:08: Resolve 258: aaaa udp 127.0.0.1 5062 [9] 2008/04/10 09:13:08: Resolve 258: a udp 127.0.0.1 5062 [9] 2008/04/10 09:13:08: Resolve 258: udp 127.0.0.1 5062 [9] 2008/04/10 09:13:08: SIP Tx udp:127.0.0.1:5062: SIP/2.0 200 Ok Via: SIP/2.0/UDP 127.0.0.1:5062 From: "ABILEN IND SCH " <sip:3256771444@localhost;user=phone>;tag=1059961393 To: <sip:3256723475@localhost;user=phone>;tag=c8db8ae26e Call-ID: 83a5d578@fxo CSeq: 1 INVITE Contact: <sip:3256723475@127.0.0.1:5060;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/3.0.0.2899 Content-Type: application/sdp Content-Length: 208 v=0 o=- 1076076106 1076076106 IN IP4 127.0.0.1 s=- c=IN IP4 127.0.0.1 t=0 0 m=audio 62278 RTP/AVP 0 101 a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv [5] 2008/04/10 09:13:08: PSTN: Response code: 200 [9] 2008/04/10 09:13:09: SIP Rx tls:192.168.1.101:2614: SIP/2.0 200 OK Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-a6612d6557f6f20952121e2346f72850;rport=5061 From: "ABILEN IND SCH " <sip:3256771444@localhost;user=phone>;tag=217507709 To: <sip:3256723475@localhost;user=phone>;tag=ooxg71bo4d Call-ID: a4449186@pbx CSeq: 29409 CANCEL Content-Length: 0 [7] 2008/04/10 09:13:09: Call a4449186@pbx#217507709: Clear last request [9] 2008/04/10 09:13:09: SIP Rx tls:192.168.1.101:2614: SIP/2.0 487 Request Terminated Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-a6612d6557f6f20952121e2346f72850;rport=5061 From: "ABILEN IND SCH " <sip:3256771444@localhost;user=phone>;tag=217507709 To: <sip:3256723475@localhost;user=phone>;tag=ooxg71bo4d Call-ID: a4449186@pbx CSeq: 29409 INVITE Contact: <sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n>;flow-id=1 Content-Length: 0 [7] 2008/04/10 09:13:09: Call a4449186@pbx#217507709: Clear last INVITE [9] 2008/04/10 09:13:09: Resolve 259: url sip:192.168.1.101:2614;transport=tls [9] 2008/04/10 09:13:09: Resolve 259: a tls 192.168.1.101 2614 [9] 2008/04/10 09:13:09: Resolve 259: tls 192.168.1.101 2614 [9] 2008/04/10 09:13:09: SIP Tx tls:192.168.1.101:2614: ACK sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n SIP/2.0 Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-a6612d6557f6f20952121e2346f72850;rport From: "ABILEN IND SCH " <sip:3256771444@localhost;user=phone>;tag=217507709 To: <sip:3256723475@localhost;user=phone>;tag=ooxg71bo4d Call-ID: a4449186@pbx CSeq: 29409 ACK Max-Forwards: 70 Contact: <sip:40@192.168.1.100:5061;transport=tls> Content-Length: 0 [3] 2008/04/10 09:13:09: SMTP: Cannot resolve smtp.mail.thesoundshop-llc.com [5] 2008/04/10 09:13:09: INVITE Response: Terminate a4449186@pbx [7] 2008/04/10 09:13:09: Other Ports: 2 [7] 2008/04/10 09:13:09: Call Port: 5f095176@pbx#1195090579 [7] 2008/04/10 09:13:09: Call Port: 83a5d578@fxo#c8db8ae26e [9] 2008/04/10 09:13:09: SIP Tr udp:127.0.0.1:5062: SIP/2.0 200 Ok Via: SIP/2.0/UDP 127.0.0.1:5062 From: "ABILEN IND SCH " <sip:3256771444@localhost;user=phone>;tag=1059961393 To: <sip:3256723475@localhost;user=phone>;tag=c8db8ae26e Call-ID: 83a5d578@fxo CSeq: 1 INVITE Contact: <sip:3256723475@127.0.0.1:5060;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/3.0.0.2899 Content-Type: application/sdp Content-Length: 208 v=0 o=- 1076076106 1076076106 IN IP4 127.0.0.1 s=- c=IN IP4 127.0.0.1 t=0 0 m=audio 62278 RTP/AVP 0 101 a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv [5] 2008/04/10 09:13:09: PSTN: Response code: 200 [9] 2008/04/10 09:13:10: SIP Tr udp:127.0.0.1:5062: SIP/2.0 200 Ok Via: SIP/2.0/UDP 127.0.0.1:5062 From: "ABILEN IND SCH " <sip:3256771444@localhost;user=phone>;tag=1059961393 To: <sip:3256723475@localhost;user=phone>;tag=c8db8ae26e Call-ID: 83a5d578@fxo CSeq: 1 INVITE Contact: <sip:3256723475@127.0.0.1:5060;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/3.0.0.2899 Content-Type: application/sdp Content-Length: 208 v=0 o=- 1076076106 1076076106 IN IP4 127.0.0.1 s=- c=IN IP4 127.0.0.1 t=0 0 m=audio 62278 RTP/AVP 0 101 a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv [5] 2008/04/10 09:13:10: PSTN: Response code: 200 [9] 2008/04/10 09:13:12: SIP Tr udp:127.0.0.1:5062: SIP/2.0 200 Ok Via: SIP/2.0/UDP 127.0.0.1:5062 From: "ABILEN IND SCH " <sip:3256771444@localhost;user=phone>;tag=1059961393 To: <sip:3256723475@localhost;user=phone>;tag=c8db8ae26e Call-ID: 83a5d578@fxo CSeq: 1 INVITE Contact: <sip:3256723475@127.0.0.1:5060;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/3.0.0.2899 Content-Type: application/sdp Content-Length: 208 v=0 o=- 1076076106 1076076106 IN IP4 127.0.0.1 s=- c=IN IP4 127.0.0.1 t=0 0 m=audio 62278 RTP/AVP 0 101 a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv [5] 2008/04/10 09:13:12: PSTN: Response code: 200 [3] 2008/04/10 09:13:13: PSTN: Channel 0 going to NO_RING [3] 2008/04/10 09:13:14: PSTN: Channel 0 going to RING [5] 2008/04/10 09:13:14: PSTN: Ringing, but last invite = 1 [9] 2008/04/10 09:13:15: SIP Rx tls:192.168.1.102:2358: REGISTER sip:localhost SIP/2.0 Via: SIP/2.0/TLS 192.168.1.102:2060;branch=z9hG4bK-iu3rw72wtsv7;rport From: "41" <sip:41@localhost>;tag=6tk3q6i8fp To: "41" <sip:41@localhost> Call-ID: 3c26701aaf59-kw566buonz1c CSeq: 14210 REGISTER Max-Forwards: 70 Contact: <sip:41@192.168.1.102:2060;transport=tls;line=94aa7mxt>;flow-id=1;q=1.0;+sip.instance="<urn:uuid:ec9ea11d-0cc3-4c4e-b306-6953a0386af5>" Contact: <http://192.168.1.102:80> Contact: <https://192.168.1.102:443> User-Agent: snom300/7.1.30 Supported: gruu Allow-Events: dialog X-Real-IP: 192.168.1.102 Proxy-Require: buttons Expires: 3600 Content-Length: 0 [8] 2008/04/10 09:13:15: Packet authenticated by transport layer [9] 2008/04/10 09:13:15: Resolve 260: tls 192.168.1.102 2358 [9] 2008/04/10 09:13:15: SIP Tx tls:192.168.1.102:2358: SIP/2.0 200 Ok Via: SIP/2.0/TLS 192.168.1.102:2060;branch=z9hG4bK-iu3rw72wtsv7;rport=2358 From: "41" <sip:41@localhost>;tag=6tk3q6i8fp To: "41" <sip:41@localhost>;tag=b4f17a5688 Call-ID: 3c26701aaf59-kw566buonz1c CSeq: 14210 REGISTER Contact: <sip:41@192.168.1.102:2060;transport=tls;line=94aa7mxt>;expires=180 Contact: <http://192.168.1.102:80>;expires=180 Contact: <https://192.168.1.102:443>;expires=180 Content-Length: 0 [9] 2008/04/10 09:13:16: SIP Tr udp:127.0.0.1:5062: SIP/2.0 200 Ok Via: SIP/2.0/UDP 127.0.0.1:5062 From: "ABILEN IND SCH " <sip:3256771444@localhost;user=phone>;tag=1059961393 To: <sip:3256723475@localhost;user=phone>;tag=c8db8ae26e Call-ID: 83a5d578@fxo CSeq: 1 INVITE Contact: <sip:3256723475@127.0.0.1:5060;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/3.0.0.2899 Content-Type: application/sdp Content-Length: 208 v=0 o=- 1076076106 1076076106 IN IP4 127.0.0.1 s=- c=IN IP4 127.0.0.1 t=0 0 m=audio 62278 RTP/AVP 0 101 a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv [5] 2008/04/10 09:13:16: PSTN: Response code: 200 [9] 2008/04/10 09:13:17: SIP Rx tls:192.168.1.102:2358: BYE sip:41@192.168.1.100:5061;transport=tls SIP/2.0 Via: SIP/2.0/TLS 192.168.1.102:2060;branch=z9hG4bK-f2webjjtdm4t;rport From: <sip:3256723475@localhost;user=phone>;tag=s2wpjj2zgg To: "ABILEN IND SCH " <sip:3256771444@localhost;user=phone>;tag=1195090579 Call-ID: 5f095176@pbx CSeq: 1 BYE Max-Forwards: 70 Contact: <sip:41@192.168.1.102:2060;transport=tls;line=94aa7mxt>;flow-id=1 User-Agent: snom300/7.1.30 RTP-RxStat: Total_Rx_Pkts=11,Rx_Pkts=11,Rx_Pkts_Lost=0,Remote_Rx_Pkts_Lost=0 RTP-TxStat: Total_Tx_Pkts=430,Tx_Pkts=430,Remote_Tx_Pkts=0 Proxy-Require: buttons Content-Length: 0 [9] 2008/04/10 09:13:17: Resolve 261: tls 192.168.1.102 2358 [9] 2008/04/10 09:13:17: SIP Tx tls:192.168.1.102:2358: SIP/2.0 200 Ok Via: SIP/2.0/TLS 192.168.1.102:2060;branch=z9hG4bK-f2webjjtdm4t;rport=2358 From: <sip:3256723475@localhost;user=phone>;tag=s2wpjj2zgg To: "ABILEN IND SCH " <sip:3256771444@localhost;user=phone>;tag=1195090579 Call-ID: 5f095176@pbx CSeq: 1 BYE Contact: <sip:41@192.168.1.100:5061;transport=tls> User-Agent: pbxnsip-PBX/3.0.0.2899 RTP-RxStat: Dur=12,Pkt=437,Oct=76912,Underun=884 RTP-TxStat: Dur=9,Pkt=11,Oct=1936 Content-Length: 0 [7] 2008/04/10 09:13:17: Other Ports: 1 [7] 2008/04/10 09:13:17: Call Port: 83a5d578@fxo#c8db8ae26e [8] 2008/04/10 09:13:17: SMTP: Connect to 68.142.200.11:25 [9] 2008/04/10 09:13:17: Resolve 262: url sip:127.0.0.1:5062 [9] 2008/04/10 09:13:17: Resolve 262: udp 127.0.0.1 5062 [9] 2008/04/10 09:13:17: SIP Tx udp:127.0.0.1:5062: BYE sip:127.0.0.1:5062 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK-352849f79ced1ffa776bcbafa30d129a;rport From: <sip:3256723475@localhost;user=phone>;tag=c8db8ae26e To: "ABILEN IND SCH " <sip:3256771444@localhost;user=phone>;tag=1059961393 Call-ID: 83a5d578@fxo CSeq: 24806 BYE Max-Forwards: 70 Contact: <sip:3256723475@127.0.0.1:5060;transport=udp> RTP-RxStat: Dur=12,Pkt=0,Oct=0,Underun=10 RTP-TxStat: Dur=9,Pkt=615,Oct=105780 Content-Length: 0 [9] 2008/04/10 09:13:17: SIP Rx udp:127.0.0.1:5062: SIP/2.0 404 Not found Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK-352849f79ced1ffa776bcbafa30d129a;rport From: <sip:3256723475@localhost;user=phone>;tag=c8db8ae26e To: "ABILEN IND SCH " <sip:3256771444@localhost;user=phone>;tag=1059961393 Call-ID: 83a5d578@fxo CSeq: 24806 BYE Content-Length: 0 [7] 2008/04/10 09:13:17: Call 83a5d578@fxo#c8db8ae26e: Clear last request [5] 2008/04/10 09:13:17: BYE Response: Terminate 83a5d578@fxo [8] 2008/04/10 09:13:18: SMTP: Received 220 smtp100.biz.mail.mud.yahoo.com ESMTP [8] 2008/04/10 09:13:18: SMTP: Received 250-smtp100.biz.mail.mud.yahoo.com 250-AUTH LOGIN PLAIN XYMCOOKIE 250-PIPELINING 250 8BITMIME [8] 2008/04/10 09:13:18: SMTP: Received 334 VXNlcm5hbWU6 [8] 2008/04/10 09:13:18: SMTP: Received 334 UGFzc3dvcmQ6 [8] 2008/04/10 09:13:18: SMTP: Received 235 ok, go ahead (#2.0.0) [8] 2008/04/10 09:13:18: SMTP: Received 250 ok [8] 2008/04/10 09:13:18: Last message repeated 2 times [8] 2008/04/10 09:13:18: SMTP: Received 354 go ahead [3] 2008/04/10 09:13:19: PSTN: Channel 0 going to NO_RING [8] 2008/04/10 09:13:19: SMTP: Received 250 ok 1207836798 qp 19516 [8] 2008/04/10 09:13:19: SMTP: Received 221 smtp100.biz.mail.mud.yahoo.com [8] 2008/04/10 09:13:19: Sucessfully sent email to <russell@thesoundshop-llc.com> [3] 2008/04/10 09:13:19: SMTP: Cannot resolve smtp.mail.thesoundshop-llc.com [9] 2008/04/10 09:13:24: SIP Tr udp:127.0.0.1:5062: SIP/2.0 200 Ok Via: SIP/2.0/UDP 127.0.0.1:5062 From: "ABILEN IND SCH " <sip:3256771444@localhost;user=phone>;tag=1059961393 To: <sip:3256723475@localhost;user=phone>;tag=c8db8ae26e Call-ID: 83a5d578@fxo CSeq: 1 INVITE Contact: <sip:3256723475@127.0.0.1:5060;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/3.0.0.2899 Content-Type: application/sdp Content-Length: 208 v=0 o=- 1076076106 1076076106 IN IP4 127.0.0.1 s=- c=IN IP4 127.0.0.1 t=0 0 m=audio 62278 RTP/AVP 0 101 a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv [5] 2008/04/10 09:13:24: PSTN: Response code: 200 [9] 2008/04/10 09:13:25: SIP Rx udp:127.0.0.1:5062: CANCEL sip:3256725804@localhost;user=phone SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5062 From: "Anonymous" <sip:anonymous@localhost;user=phone>;tag=233665123 To: <sip:3256725804@localhost;user=phone> Call-ID: 3f9a62d7@fxo Contact: <sip:127.0.0.1:5062> CSeq: 1 CANCEL Content-Length: 0 [9] 2008/04/10 09:13:25: Resolve 263: aaaa udp 127.0.0.1 5062 [9] 2008/04/10 09:13:25: Resolve 263: a udp 127.0.0.1 5062 [9] 2008/04/10 09:13:25: Resolve 263: udp 127.0.0.1 5062 [9] 2008/04/10 09:13:25: SIP Tx udp:127.0.0.1:5062: SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP 127.0.0.1:5062 From: "Anonymous" <sip:anonymous@localhost;user=phone>;tag=233665123 To: <sip:3256725804@localhost;user=phone> Call-ID: 3f9a62d7@fxo CSeq: 1 CANCEL Content-Length: 0 [5] 2008/04/10 09:13:25: PSTN: Timeout without ring on 0, going to idle [3] 2008/04/10 09:13:25: PSTN: Channel 0 going to IDLE [5] 2008/04/10 09:13:25: PSTN: Response code: 481 [9] 2008/04/10 09:13:30: SIP Rx tls:192.168.1.101:2614: SUBSCRIBE sip:40@localhost;user=phone SIP/2.0 Via: SIP/2.0/TLS 192.168.1.101:2060;branch=z9hG4bK-p04p5na21bhq;rport From: <sip:40@localhost>;tag=vgds1qnpk0 To: <sip:40@localhost;user=phone>;tag=9d4243196b Call-ID: 3c2674105300-yc4ja694jzkk CSeq: 13653 SUBSCRIBE Max-Forwards: 70 Contact: <sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n>;flow-id=1 Event: message-summary Accept: application/simple-message-summary User-Agent: snom300/7.1.30 Proxy-Require: buttons Expires: 3600 Content-Length: 0 [8] 2008/04/10 09:13:30: Packet authenticated by transport layer [9] 2008/04/10 09:13:30: Resolve 264: tls 192.168.1.101 2614 [9] 2008/04/10 09:13:30: SIP Tx tls:192.168.1.101:2614: SIP/2.0 200 Ok Via: SIP/2.0/TLS 192.168.1.101:2060;branch=z9hG4bK-p04p5na21bhq;rport=2614 From: <sip:40@localhost>;tag=vgds1qnpk0 To: <sip:40@localhost;user=phone>;tag=9d4243196b Call-ID: 3c2674105300-yc4ja694jzkk CSeq: 13653 SUBSCRIBE Contact: <sip:192.168.1.100:5061;transport=tls> Expires: 181 Content-Length: 0 [9] 2008/04/10 09:13:34: SIP Rx tls:192.168.1.101:2614: REGISTER sip:localhost SIP/2.0 Via: SIP/2.0/TLS 192.168.1.101:2060;branch=z9hG4bK-9lc6y7ibd7v2;rport From: "40" <sip:40@localhost>;tag=wlaz8y5hn3 To: "40" <sip:40@localhost> Call-ID: 3c26741018ce-d9hh4ncip2ol CSeq: 14205 REGISTER Max-Forwards: 70 Contact: <sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n>;flow-id=1;q=1.0;+sip.instance="<urn:uuid:822ed566-4660-4bfb-90c3-2b5411c69eb0>" Contact: <http://192.168.1.101:80> Contact: <https://192.168.1.101:443> User-Agent: snom300/7.1.30 Supported: gruu Allow-Events: dialog X-Real-IP: 192.168.1.101 Proxy-Require: buttons Expires: 3600 Content-Length: 0 [8] 2008/04/10 09:13:34: Packet authenticated by transport layer [9] 2008/04/10 09:13:34: Resolve 265: tls 192.168.1.101 2614 [9] 2008/04/10 09:13:34: SIP Tx tls:192.168.1.101:2614: SIP/2.0 200 Ok Via: SIP/2.0/TLS 192.168.1.101:2060;branch=z9hG4bK-9lc6y7ibd7v2;rport=2614 From: "40" <sip:40@localhost>;tag=wlaz8y5hn3 To: "40" <sip:40@localhost>;tag=c02c2dcda3 Call-ID: 3c26741018ce-d9hh4ncip2ol CSeq: 14205 REGISTER Contact: <sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n>;expires=181 Contact: <http://192.168.1.101:80>;expires=181 Contact: <https://192.168.1.101:443>;expires=181 Content-Length: 0 [3] 2008/04/10 09:14:19: SMTP: Cannot resolve smtp.mail.thesoundshop-llc.com [8] 2008/04/10 09:14:19: DNS: dns_cname smtp.bizmail.yahoo.com expired [8] 2008/04/10 09:14:27: DNS: dns_a smtp.bizmail.mail.yahoo4.akadns.net expired [9] 2008/04/10 09:14:45: SIP Rx tls:192.168.1.102:2358: REGISTER sip:localhost SIP/2.0 Via: SIP/2.0/TLS 192.168.1.102:2060;branch=z9hG4bK-2jfbug3q53c1;rport From: "41" <sip:41@localhost>;tag=dvtmvo1rcr To: "41" <sip:41@localhost> Call-ID: 3c26701aaf59-kw566buonz1c CSeq: 14211 REGISTER Max-Forwards: 70 Contact: <sip:41@192.168.1.102:2060;transport=tls;line=94aa7mxt>;flow-id=1;q=1.0;+sip.instance="<urn:uuid:ec9ea11d-0cc3-4c4e-b306-6953a0386af5>" Contact: <http://192.168.1.102:80> Contact: <https://192.168.1.102:443> User-Agent: snom300/7.1.30 Supported: gruu Allow-Events: dialog X-Real-IP: 192.168.1.102 Proxy-Require: buttons Expires: 3600 Content-Length: 0 [8] 2008/04/10 09:14:45: Packet authenticated by transport layer [9] 2008/04/10 09:14:45: Resolve 266: tls 192.168.1.102 2358 [9] 2008/04/10 09:14:45: SIP Tx tls:192.168.1.102:2358: SIP/2.0 200 Ok Via: SIP/2.0/TLS 192.168.1.102:2060;branch=z9hG4bK-2jfbug3q53c1;rport=2358 From: "41" <sip:41@localhost>;tag=dvtmvo1rcr To: "41" <sip:41@localhost>;tag=b4f17a5688 Call-ID: 3c26701aaf59-kw566buonz1c CSeq: 14211 REGISTER Contact: <sip:41@192.168.1.102:2060;transport=tls;line=94aa7mxt>;expires=179 Contact: <http://192.168.1.102:80>;expires=179 Contact: <https://192.168.1.102:443>;expires=179 Content-Length: 0 [9] 2008/04/10 09:15:00: SIP Rx tls:192.168.1.101:2614: SUBSCRIBE sip:40@localhost;user=phone SIP/2.0 Via: SIP/2.0/TLS 192.168.1.101:2060;branch=z9hG4bK-6i9c74bg97n7;rport From: <sip:40@localhost>;tag=vgds1qnpk0 To: <sip:40@localhost;user=phone>;tag=9d4243196b Call-ID: 3c2674105300-yc4ja694jzkk CSeq: 13654 SUBSCRIBE Max-Forwards: 70 Contact: <sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n>;flow-id=1 Event: message-summary Accept: application/simple-message-summary User-Agent: snom300/7.1.30 Proxy-Require: buttons Expires: 3600 Content-Length: 0 [8] 2008/04/10 09:15:00: Packet authenticated by transport layer [9] 2008/04/10 09:15:00: Resolve 267: tls 192.168.1.101 2614 [9] 2008/04/10 09:15:00: SIP Tx tls:192.168.1.101:2614: SIP/2.0 200 Ok Via: SIP/2.0/TLS 192.168.1.101:2060;branch=z9hG4bK-6i9c74bg97n7;rport=2614 From: <sip:40@localhost>;tag=vgds1qnpk0 To: <sip:40@localhost;user=phone>;tag=9d4243196b Call-ID: 3c2674105300-yc4ja694jzkk CSeq: 13654 SUBSCRIBE Contact: <sip:192.168.1.100:5061;transport=tls> Expires: 178 Content-Length: 0 [9] 2008/04/10 09:15:05: SIP Rx tls:192.168.1.101:2614: REGISTER sip:localhost SIP/2.0 Via: SIP/2.0/TLS 192.168.1.101:2060;branch=z9hG4bK-96iva23npt5i;rport From: "40" <sip:40@localhost>;tag=y1r4lu2urn To: "40" <sip:40@localhost> Call-ID: 3c26741018ce-d9hh4ncip2ol CSeq: 14206 REGISTER Max-Forwards: 70 Contact: <sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n>;flow-id=1;q=1.0;+sip.instance="<urn:uuid:822ed566-4660-4bfb-90c3-2b5411c69eb0>" Contact: <http://192.168.1.101:80> Contact: <https://192.168.1.101:443> User-Agent: snom300/7.1.30 Supported: gruu Allow-Events: dialog X-Real-IP: 192.168.1.101 Proxy-Require: buttons Expires: 3600 Content-Length: 0 [8] 2008/04/10 09:15:05: Packet authenticated by transport layer [9] 2008/04/10 09:15:05: Resolve 268: tls 192.168.1.101 2614 [9] 2008/04/10 09:15:05: SIP Tx tls:192.168.1.101:2614: SIP/2.0 200 Ok Via: SIP/2.0/TLS 192.168.1.101:2060;branch=z9hG4bK-96iva23npt5i;rport=2614 From: "40" <sip:40@localhost>;tag=y1r4lu2urn To: "40" <sip:40@localhost>;tag=c02c2dcda3 Call-ID: 3c26741018ce-d9hh4ncip2ol CSeq: 14206 REGISTER Contact: <sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n>;expires=179 Contact: <http://192.168.1.101:80>;expires=179 Contact: <https://192.168.1.101:443>;expires=179 Content-Length: 0 [3] 2008/04/10 09:15:19: SMTP: Cannot resolve smtp.mail.thesoundshop-llc.com [9] 2008/04/10 09:16:15: SIP Rx tls:192.168.1.102:2358: REGISTER sip:localhost SIP/2.0 Via: SIP/2.0/TLS 192.168.1.102:2060;branch=z9hG4bK-w66jgcmtd3gi;rport From: "41" <sip:41@localhost>;tag=azhcqtj84e To: "41" <sip:41@localhost> Call-ID: 3c26701aaf59-kw566buonz1c CSeq: 14212 REGISTER Max-Forwards: 70 Contact: <sip:41@192.168.1.102:2060;transport=tls;line=94aa7mxt>;flow-id=1;q=1.0;+sip.instance="<urn:uuid:ec9ea11d-0cc3-4c4e-b306-6953a0386af5>" Contact: <http://192.168.1.102:80> Contact: <https://192.168.1.102:443> User-Agent: snom300/7.1.30 Supported: gruu Allow-Events: dialog X-Real-IP: 192.168.1.102 Proxy-Require: buttons Expires: 3600 Content-Length: 0 [8] 2008/04/10 09:16:15: Packet authenticated by transport layer [9] 2008/04/10 09:16:15: Resolve 269: tls 192.168.1.102 2358 [9] 2008/04/10 09:16:15: SIP Tx tls:192.168.1.102:2358: SIP/2.0 200 Ok Via: SIP/2.0/TLS 192.168.1.102:2060;branch=z9hG4bK-w66jgcmtd3gi;rport=2358 From: "41" <sip:41@localhost>;tag=azhcqtj84e To: "41" <sip:41@localhost>;tag=b4f17a5688 Call-ID: 3c26701aaf59-kw566buonz1c CSeq: 14212 REGISTER Contact: <sip:41@192.168.1.102:2060;transport=tls;line=94aa7mxt>;expires=181 Contact: <http://192.168.1.102:80>;expires=181 Contact: <https://192.168.1.102:443>;expires=181 Content-Length: 0 [3] 2008/04/10 09:16:19: SMTP: Cannot resolve smtp.mail.thesoundshop-llc.com [9] 2008/04/10 09:16:29: SIP Rx tls:192.168.1.101:2614: SUBSCRIBE sip:40@localhost;user=phone SIP/2.0 Via: SIP/2.0/TLS 192.168.1.101:2060;branch=z9hG4bK-26t04w8d2q9c;rport From: <sip:40@localhost>;tag=vgds1qnpk0 To: <sip:40@localhost;user=phone>;tag=9d4243196b Call-ID: 3c2674105300-yc4ja694jzkk CSeq: 13655 SUBSCRIBE Max-Forwards: 70 Contact: <sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n>;flow-id=1 Event: message-summary Accept: application/simple-message-summary User-Agent: snom300/7.1.30 Proxy-Require: buttons Expires: 3600 Content-Length: 0 [8] 2008/04/10 09:16:29: Packet authenticated by transport layer [9] 2008/04/10 09:16:29: Resolve 270: tls 192.168.1.101 2614 [9] 2008/04/10 09:16:29: SIP Tx tls:192.168.1.101:2614: SIP/2.0 200 Ok Via: SIP/2.0/TLS 192.168.1.101:2060;branch=z9hG4bK-26t04w8d2q9c;rport=2614 From: <sip:40@localhost>;tag=vgds1qnpk0 To: <sip:40@localhost;user=phone>;tag=9d4243196b Call-ID: 3c2674105300-yc4ja694jzkk CSeq: 13655 SUBSCRIBE Contact: <sip:192.168.1.100:5061;transport=tls> Expires: 179 Content-Length: 0 [9] 2008/04/10 09:16:34: SIP Rx tls:192.168.1.101:2614: REGISTER sip:localhost SIP/2.0 Via: SIP/2.0/TLS 192.168.1.101:2060;branch=z9hG4bK-fy6nyde880e1;rport From: "40" <sip:40@localhost>;tag=ocf1tlbn9m To: "40" <sip:40@localhost> Call-ID: 3c26741018ce-d9hh4ncip2ol CSeq: 14207 REGISTER Max-Forwards: 70 Contact: <sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n>;flow-id=1;q=1.0;+sip.instance="<urn:uuid:822ed566-4660-4bfb-90c3-2b5411c69eb0>" Contact: <http://192.168.1.101:80> Contact: <https://192.168.1.101:443> User-Agent: snom300/7.1.30 Supported: gruu Allow-Events: dialog X-Real-IP: 192.168.1.101 Proxy-Require: buttons Expires: 3600 Content-Length: 0 [8] 2008/04/10 09:16:34: Packet authenticated by transport layer [9] 2008/04/10 09:16:34: Resolve 271: tls 192.168.1.101 2614 [9] 2008/04/10 09:16:34: SIP Tx tls:192.168.1.101:2614: SIP/2.0 200 Ok Via: SIP/2.0/TLS 192.168.1.101:2060;branch=z9hG4bK-fy6nyde880e1;rport=2614 From: "40" <sip:40@localhost>;tag=ocf1tlbn9m To: "40" <sip:40@localhost>;tag=c02c2dcda3 Call-ID: 3c26741018ce-d9hh4ncip2ol CSeq: 14207 REGISTER Contact: <sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n>;expires=180 Contact: <http://192.168.1.101:80>;expires=180 Contact: <https://192.168.1.101:443>;expires=180 Content-Length: 0 [3] 2008/04/10 09:17:19: SMTP: Cannot resolve smtp.mail.thesoundshop-llc.com [9] 2008/04/10 09:17:45: SIP Rx tls:192.168.1.102:2358: REGISTER sip:localhost SIP/2.0 Via: SIP/2.0/TLS 192.168.1.102:2060;branch=z9hG4bK-lz58s2vskjak;rport From: "41" <sip:41@localhost>;tag=khr4qcyiyl To: "41" <sip:41@localhost> Call-ID: 3c26701aaf59-kw566buonz1c CSeq: 14213 REGISTER Max-Forwards: 70 Contact: <sip:41@192.168.1.102:2060;transport=tls;line=94aa7mxt>;flow-id=1;q=1.0;+sip.instance="<urn:uuid:ec9ea11d-0cc3-4c4e-b306-6953a0386af5>" Contact: <http://192.168.1.102:80> Contact: <https://192.168.1.102:443> User-Agent: snom300/7.1.30 Supported: gruu Allow-Events: dialog X-Real-IP: 192.168.1.102 Proxy-Require: buttons Expires: 3600 Content-Length: 0 [8] 2008/04/10 09:17:45: Packet authenticated by transport layer [9] 2008/04/10 09:17:45: Resolve 272: tls 192.168.1.102 2358 [9] 2008/04/10 09:17:45: SIP Tx tls:192.168.1.102:2358: SIP/2.0 200 Ok Via: SIP/2.0/TLS 192.168.1.102:2060;branch=z9hG4bK-lz58s2vskjak;rport=2358 From: "41" <sip:41@localhost>;tag=khr4qcyiyl To: "41" <sip:41@localhost>;tag=b4f17a5688 Call-ID: 3c26701aaf59-kw566buonz1c CSeq: 14213 REGISTER Contact: <sip:41@192.168.1.102:2060;transport=tls;line=94aa7mxt>;expires=180 Contact: <http://192.168.1.102:80>;expires=180 Contact: <https://192.168.1.102:443>;expires=180 Content-Length: 0 [9] 2008/04/10 09:17:59: SIP Rx tls:192.168.1.101:2614: SUBSCRIBE sip:40@localhost;user=phone SIP/2.0 Via: SIP/2.0/TLS 192.168.1.101:2060;branch=z9hG4bK-xv30apapngx6;rport From: <sip:40@localhost>;tag=vgds1qnpk0 To: <sip:40@localhost;user=phone>;tag=9d4243196b Call-ID: 3c2674105300-yc4ja694jzkk CSeq: 13656 SUBSCRIBE Max-Forwards: 70 Contact: <sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n>;flow-id=1 Event: message-summary Accept: application/simple-message-summary User-Agent: snom300/7.1.30 Proxy-Require: buttons Expires: 3600 Content-Length: 0 [8] 2008/04/10 09:17:59: Packet authenticated by transport layer [9] 2008/04/10 09:17:59: Resolve 273: tls 192.168.1.101 2614 [9] 2008/04/10 09:17:59: SIP Tx tls:192.168.1.101:2614: SIP/2.0 200 Ok Via: SIP/2.0/TLS 192.168.1.101:2060;branch=z9hG4bK-xv30apapngx6;rport=2614 From: <sip:40@localhost>;tag=vgds1qnpk0 To: <sip:40@localhost;user=phone>;tag=9d4243196b Call-ID: 3c2674105300-yc4ja694jzkk CSeq: 13656 SUBSCRIBE Contact: <sip:192.168.1.100:5061;transport=tls> Expires: 181 Content-Length: 0 [9] 2008/04/10 09:18:04: SIP Rx tls:192.168.1.101:2614: REGISTER sip:localhost SIP/2.0 Via: SIP/2.0/TLS 192.168.1.101:2060;branch=z9hG4bK-9c7slmor056o;rport From: "40" <sip:40@localhost>;tag=y7yyjg703x To: "40" <sip:40@localhost> Call-ID: 3c26741018ce-d9hh4ncip2ol CSeq: 14208 REGISTER Max-Forwards: 70 Contact: <sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n>;flow-id=1;q=1.0;+sip.instance="<urn:uuid:822ed566-4660-4bfb-90c3-2b5411c69eb0>" Contact: <http://192.168.1.101:80> Contact: <https://192.168.1.101:443> User-Agent: snom300/7.1.30 Supported: gruu Allow-Events: dialog X-Real-IP: 192.168.1.101 Proxy-Require: buttons Expires: 3600 Content-Length: 0 [8] 2008/04/10 09:18:04: Packet authenticated by transport layer [9] 2008/04/10 09:18:04: Resolve 274: tls 192.168.1.101 2614 [9] 2008/04/10 09:18:04: SIP Tx tls:192.168.1.101:2614: SIP/2.0 200 Ok Via: SIP/2.0/TLS 192.168.1.101:2060;branch=z9hG4bK-9c7slmor056o;rport=2614 From: "40" <sip:40@localhost>;tag=y7yyjg703x To: "40" <sip:40@localhost>;tag=c02c2dcda3 Call-ID: 3c26741018ce-d9hh4ncip2ol CSeq: 14208 REGISTER Contact: <sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n>;expires=178 Contact: <http://192.168.1.101:80>;expires=178 Contact: <https://192.168.1.101:443>;expires=178 Content-Length: 0 [3] 2008/04/10 09:18:19: SMTP: Cannot resolve smtp.mail.thesoundshop-llc.com [9] 2008/04/10 09:19:15: SIP Rx tls:192.168.1.102:2358: REGISTER sip:localhost SIP/2.0 Via: SIP/2.0/TLS 192.168.1.102:2060;branch=z9hG4bK-a3n6w796cqnc;rport From: "41" <sip:41@localhost>;tag=0ziehgzq3g To: "41" <sip:41@localhost> Call-ID: 3c26701aaf59-kw566buonz1c CSeq: 14214 REGISTER Max-Forwards: 70 Contact: <sip:41@192.168.1.102:2060;transport=tls;line=94aa7mxt>;flow-id=1;q=1.0;+sip.instance="<urn:uuid:ec9ea11d-0cc3-4c4e-b306-6953a0386af5>" Contact: <http://192.168.1.102:80> Contact: <https://192.168.1.102:443> User-Agent: snom300/7.1.30 Supported: gruu Allow-Events: dialog X-Real-IP: 192.168.1.102 Proxy-Require: buttons Expires: 3600 Content-Length: 0 [8] 2008/04/10 09:19:15: Packet authenticated by transport layer [9] 2008/04/10 09:19:15: Resolve 275: tls 192.168.1.102 2358 [9] 2008/04/10 09:19:15: SIP Tx tls:192.168.1.102:2358: SIP/2.0 200 Ok Via: SIP/2.0/TLS 192.168.1.102:2060;branch=z9hG4bK-a3n6w796cqnc;rport=2358 From: "41" <sip:41@localhost>;tag=0ziehgzq3g To: "41" <sip:41@localhost>;tag=b4f17a5688 Call-ID: 3c26701aaf59-kw566buonz1c CSeq: 14214 REGISTER Contact: <sip:41@192.168.1.102:2060;transport=tls;line=94aa7mxt>;expires=180 Contact: <http://192.168.1.102:80>;expires=180 Contact: <https://192.168.1.102:443>;expires=180 Content-Length: 0 [3] 2008/04/10 09:19:19: SMTP: Cannot resolve smtp.mail.thesoundshop-llc.com
  12. OK, got some time this morning here is the first log, call comes in on second fxo port, system rings extension, answer extension, no one at other end, line keeps ringing...answer call on backup phone...I have on system.. [9] 2008/04/10 09:11:44: SIP Rx tls:192.168.1.102:2358: REGISTER sip:localhost SIP/2.0 Via: SIP/2.0/TLS 192.168.1.102:2060;branch=z9hG4bK-0j4n97biwih4;rport From: "41" <sip:41@localhost>;tag=8d3szdg5q4 To: "41" <sip:41@localhost> Call-ID: 3c26701aaf59-kw566buonz1c CSeq: 14209 REGISTER Max-Forwards: 70 Contact: <sip:41@192.168.1.102:2060;transport=tls;line=94aa7mxt>;flow-id=1;q=1.0;+sip.instance="<urn:uuid:ec9ea11d-0cc3-4c4e-b306-6953a0386af5>" Contact: <http://192.168.1.102:80> Contact: <https://192.168.1.102:443> User-Agent: snom300/7.1.30 Supported: gruu Allow-Events: dialog X-Real-IP: 192.168.1.102 Proxy-Require: buttons Expires: 3600 Content-Length: 0 [8] 2008/04/10 09:11:44: Packet authenticated by transport layer [9] 2008/04/10 09:11:44: Resolve 234: tls 192.168.1.102 2358 [9] 2008/04/10 09:11:44: SIP Tx tls:192.168.1.102:2358: SIP/2.0 200 Ok Via: SIP/2.0/TLS 192.168.1.102:2060;branch=z9hG4bK-0j4n97biwih4;rport=2358 From: "41" <sip:41@localhost>;tag=8d3szdg5q4 To: "41" <sip:41@localhost>;tag=b4f17a5688 Call-ID: 3c26701aaf59-kw566buonz1c CSeq: 14209 REGISTER Contact: <sip:41@192.168.1.102:2060;transport=tls;line=94aa7mxt>;expires=182 Contact: <http://192.168.1.102:80>;expires=182 Contact: <https://192.168.1.102:443>;expires=182 Content-Length: 0 [3] 2008/04/10 09:11:46: PSTN: Channel 1 going to RING [5] 2008/04/10 09:11:49: PSTN: eVAPI_CALLER_ID_DETECTED_EVENT [9] 2008/04/10 09:11:49: PSTN: Caller-ID: Received unknown tag: 01 08 30 34 31 30 30 39 71 c5 12 a4 68 93 a6 4d 4d 9c 33 31 31 37 38 07 0f 54 48 45 20 53 4f 55 4e 44 20 53 c8 3d 41 [9] 2008/04/10 09:11:49: SIP Rx udp:127.0.0.1:5062: INVITE sip:3256725804@localhost;user=phone SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5062 From: "Anonymous" <sip:anonymous@localhost;user=phone>;tag=233665123 To: <sip:3256725804@localhost;user=phone> Call-ID: 3f9a62d7@fxo Contact: <sip:127.0.0.1:5062> CSeq: 1 INVITE Content-Type: application/sdp Content-Length: 137 v=0 o=root 0 0 IN IP4 1.1.1.2 s=- c=IN IP4 1.1.1.2 t=0 0 m=audio 2062 RTP/AVP 0 101 a=rtpmap:101 telephone-event/8000 a=ptime:20 [7] 2008/04/10 09:11:49: UDP: Opening socket on port 53392 [7] 2008/04/10 09:11:49: UDP: Opening socket on port 53393 [5] 2008/04/10 09:11:49: Identify trunk (IP address/port and domain match) 5 [9] 2008/04/10 09:11:49: Resolve 235: aaaa udp 127.0.0.1 5062 [9] 2008/04/10 09:11:49: Resolve 235: a udp 127.0.0.1 5062 [9] 2008/04/10 09:11:49: Resolve 235: udp 127.0.0.1 5062 [9] 2008/04/10 09:11:49: SIP Tx udp:127.0.0.1:5062: SIP/2.0 100 Trying Via: SIP/2.0/UDP 127.0.0.1:5062 From: "Anonymous" <sip:anonymous@localhost;user=phone>;tag=233665123 To: <sip:3256725804@localhost;user=phone>;tag=28741e4a25 Call-ID: 3f9a62d7@fxo CSeq: 1 INVITE Content-Length: 0 [5] 2008/04/10 09:11:49: PSTN: eVAPI_CALLER_ID_DETECTED_EVENT [7] 2008/04/10 09:11:49: Set packet length to 20 [6] 2008/04/10 09:11:49: Sending RTP for 3f9a62d7@fxo#28741e4a25 to 1.1.1.2:2062 [5] 2008/04/10 09:11:49: PSTN: Response code: 100 [5] 2008/04/10 09:11:49: Trunk PSTN1 sends call to 72 [8] 2008/04/10 09:11:49: Play audio_moh/noise.wav [7] 2008/04/10 09:11:49: Hunt Group 72: Moving to next stage [7] 2008/04/10 09:11:49: Hunt group 72 called 2 registrations [5] 2008/04/10 09:11:49: PSTN: Received on 1: Caller-ID 3255131178 [7] 2008/04/10 09:11:49: Set packet length to 20 [9] 2008/04/10 09:11:49: Resolve 236: aaaa udp 127.0.0.1 5062 [9] 2008/04/10 09:11:49: Resolve 236: a udp 127.0.0.1 5062 [9] 2008/04/10 09:11:49: Resolve 236: udp 127.0.0.1 5062 [9] 2008/04/10 09:11:49: SIP Tx udp:127.0.0.1:5062: SIP/2.0 183 Ringing Via: SIP/2.0/UDP 127.0.0.1:5062 From: "Anonymous" <sip:anonymous@localhost;user=phone>;tag=233665123 To: <sip:3256725804@localhost;user=phone>;tag=28741e4a25 Call-ID: 3f9a62d7@fxo CSeq: 1 INVITE Contact: <sip:3256725804@127.0.0.1:5060;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/3.0.0.2899 Content-Type: application/sdp Content-Length: 208 v=0 o=- 1380716876 1380716876 IN IP4 127.0.0.1 s=- c=IN IP4 127.0.0.1 t=0 0 m=audio 53392 RTP/AVP 0 101 a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv [7] 2008/04/10 09:11:49: UDP: Opening socket on port 56164 [7] 2008/04/10 09:11:49: UDP: Opening socket on port 56165 [9] 2008/04/10 09:11:49: Using outbound proxy sip:192.168.1.101:2614;transport=tls because of flow-label [9] 2008/04/10 09:11:49: Resolve 237: url sip:192.168.1.101:2614;transport=tls [9] 2008/04/10 09:11:49: Resolve 237: a tls 192.168.1.101 2614 [9] 2008/04/10 09:11:49: Resolve 237: tls 192.168.1.101 2614 [9] 2008/04/10 09:11:49: SIP Tx tls:192.168.1.101:2614: INVITE sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n SIP/2.0 Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-cc71df1ba31471f6fbbb0389c307fd35;rport From: "Anonymous" <sip:anonymous@localhost;user=phone>;tag=986462258 To: <sip:3256725804@localhost;user=phone> Call-ID: f7b7fdc2@pbx CSeq: 27585 INVITE Max-Forwards: 70 Contact: <sip:40@192.168.1.100:5061;transport=tls> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pbxnsip-PBX/3.0.0.2899 Content-Type: application/sdp Content-Length: 386 v=0 o=- 1711529881 1711529881 IN IP4 192.168.1.100 s=- c=IN IP4 192.168.1.100 t=0 0 m=audio 56164 RTP/AVP 0 8 9 2 3 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:7PwZcsJzAakf+GjMlE7/1g4IsOOEaMLJKYZNjrdR 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] 2008/04/10 09:11:49: UDP: Opening socket on port 53014 [7] 2008/04/10 09:11:49: UDP: Opening socket on port 53015 [9] 2008/04/10 09:11:49: Using outbound proxy sip:192.168.1.102:2358;transport=tls because of flow-label [9] 2008/04/10 09:11:49: Resolve 238: url sip:192.168.1.102:2358;transport=tls [9] 2008/04/10 09:11:49: Resolve 238: a tls 192.168.1.102 2358 [9] 2008/04/10 09:11:49: Resolve 238: tls 192.168.1.102 2358 [9] 2008/04/10 09:11:49: SIP Tx tls:192.168.1.102:2358: INVITE sip:41@192.168.1.102:2060;transport=tls;line=94aa7mxt SIP/2.0 Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-0b25baeffb1448798c8837752400d61f;rport From: "Anonymous" <sip:anonymous@localhost;user=phone>;tag=1946255428 To: <sip:3256725804@localhost;user=phone> Call-ID: 2f9a5c8b@pbx CSeq: 13845 INVITE Max-Forwards: 70 Contact: <sip:41@192.168.1.100:5061;transport=tls> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pbxnsip-PBX/3.0.0.2899 Content-Type: application/sdp Content-Length: 384 v=0 o=- 607823118 607823118 IN IP4 192.168.1.100 s=- c=IN IP4 192.168.1.100 t=0 0 m=audio 53014 RTP/AVP 0 8 9 2 3 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:IaCicJ94fqcoYSuRI/S6qUFJYJJwA9+xGO9mLqg5 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 [5] 2008/04/10 09:11:49: PSTN: Received on 1: Name THE SOUND SHOP [8] 2008/04/10 09:11:49: PSTN: Received Caller-ID on channel 1, but already sent INVITE [5] 2008/04/10 09:11:49: PSTN: Response code: 183 [9] 2008/04/10 09:11:49: SIP Rx tls:192.168.1.101:2614: SIP/2.0 180 Ringing Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-cc71df1ba31471f6fbbb0389c307fd35;rport=5061 From: "Anonymous" <sip:anonymous@localhost;user=phone>;tag=986462258 To: <sip:3256725804@localhost;user=phone>;tag=5dp6j7fb67 Call-ID: f7b7fdc2@pbx CSeq: 27585 INVITE Contact: <sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n>;flow-id=1 Require: 100rel RSeq: 1 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Content-Length: 0 [9] 2008/04/10 09:11:49: Resolve 239: url sip:192.168.1.101:2614;transport=tls [9] 2008/04/10 09:11:49: Resolve 239: a tls 192.168.1.101 2614 [9] 2008/04/10 09:11:49: Resolve 239: tls 192.168.1.101 2614 [9] 2008/04/10 09:11:49: SIP Tx tls:192.168.1.101:2614: PRACK sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n SIP/2.0 Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-51248bc93881184e8df761caa954c81c;rport From: "Anonymous" <sip:anonymous@localhost;user=phone>;tag=986462258 To: <sip:3256725804@localhost;user=phone>;tag=5dp6j7fb67 Call-ID: f7b7fdc2@pbx CSeq: 27586 PRACK Max-Forwards: 70 Contact: <sip:40@192.168.1.100:5061;transport=tls> RAck: 1 27585 INVITE Content-Length: 0 [8] 2008/04/10 09:11:49: Play audio_en/ringback.wav [9] 2008/04/10 09:11:49: SIP Rx tls:192.168.1.102:2358: SIP/2.0 180 Ringing Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-0b25baeffb1448798c8837752400d61f;rport=5061 From: "Anonymous" <sip:anonymous@localhost;user=phone>;tag=1946255428 To: <sip:3256725804@localhost;user=phone>;tag=wobtjxxe3t Call-ID: 2f9a5c8b@pbx CSeq: 13845 INVITE Contact: <sip:41@192.168.1.102:2060;transport=tls;line=94aa7mxt>;flow-id=1 Require: 100rel RSeq: 1 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Content-Length: 0 [9] 2008/04/10 09:11:49: Resolve 240: url sip:192.168.1.102:2358;transport=tls [9] 2008/04/10 09:11:49: Resolve 240: a tls 192.168.1.102 2358 [9] 2008/04/10 09:11:49: Resolve 240: tls 192.168.1.102 2358 [9] 2008/04/10 09:11:49: SIP Tx tls:192.168.1.102:2358: PRACK sip:41@192.168.1.102:2060;transport=tls;line=94aa7mxt SIP/2.0 Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-44400d04a1fdd63afa2589dcd4f0f273;rport From: "Anonymous" <sip:anonymous@localhost;user=phone>;tag=1946255428 To: <sip:3256725804@localhost;user=phone>;tag=wobtjxxe3t Call-ID: 2f9a5c8b@pbx CSeq: 13846 PRACK Max-Forwards: 70 Contact: <sip:41@192.168.1.100:5061;transport=tls> RAck: 1 13845 INVITE Content-Length: 0 [8] 2008/04/10 09:11:49: Play audio_en/ringback.wav [9] 2008/04/10 09:11:49: SIP Rx tls:192.168.1.101:2614: SIP/2.0 200 Ok Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-51248bc93881184e8df761caa954c81c;rport=5061 From: "Anonymous" <sip:anonymous@localhost;user=phone>;tag=986462258 To: <sip:3256725804@localhost;user=phone>;tag=5dp6j7fb67 Call-ID: f7b7fdc2@pbx CSeq: 27586 PRACK Contact: <sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n>;flow-id=1 Content-Length: 0 [7] 2008/04/10 09:11:49: Call f7b7fdc2@pbx#986462258: Clear last request [9] 2008/04/10 09:11:49: SIP Rx tls:192.168.1.102:2358: SIP/2.0 200 Ok Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-44400d04a1fdd63afa2589dcd4f0f273;rport=5061 From: "Anonymous" <sip:anonymous@localhost;user=phone>;tag=1946255428 To: <sip:3256725804@localhost;user=phone>;tag=wobtjxxe3t Call-ID: 2f9a5c8b@pbx CSeq: 13846 PRACK Contact: <sip:41@192.168.1.102:2060;transport=tls;line=94aa7mxt>;flow-id=1 Content-Length: 0 [7] 2008/04/10 09:11:49: Call 2f9a5c8b@pbx#1946255428: Clear last request [9] 2008/04/10 09:11:50: SIP Tr udp:127.0.0.1:5062: SIP/2.0 183 Ringing Via: SIP/2.0/UDP 127.0.0.1:5062 From: "Anonymous" <sip:anonymous@localhost;user=phone>;tag=233665123 To: <sip:3256725804@localhost;user=phone>;tag=28741e4a25 Call-ID: 3f9a62d7@fxo CSeq: 1 INVITE Contact: <sip:3256725804@127.0.0.1:5060;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/3.0.0.2899 Content-Type: application/sdp Content-Length: 208 v=0 o=- 1380716876 1380716876 IN IP4 127.0.0.1 s=- c=IN IP4 127.0.0.1 t=0 0 m=audio 53392 RTP/AVP 0 101 a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv [5] 2008/04/10 09:11:50: PSTN: Response code: 183 [3] 2008/04/10 09:11:51: PSTN: Channel 1 going to NO_RING [9] 2008/04/10 09:11:51: SIP Tr udp:127.0.0.1:5062: SIP/2.0 183 Ringing Via: SIP/2.0/UDP 127.0.0.1:5062 From: "Anonymous" <sip:anonymous@localhost;user=phone>;tag=233665123 To: <sip:3256725804@localhost;user=phone>;tag=28741e4a25 Call-ID: 3f9a62d7@fxo CSeq: 1 INVITE Contact: <sip:3256725804@127.0.0.1:5060;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/3.0.0.2899 Content-Type: application/sdp Content-Length: 208 v=0 o=- 1380716876 1380716876 IN IP4 127.0.0.1 s=- c=IN IP4 127.0.0.1 t=0 0 m=audio 53392 RTP/AVP 0 101 a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv [5] 2008/04/10 09:11:51: PSTN: Response code: 183 [3] 2008/04/10 09:11:52: PSTN: Channel 1 going to RING [5] 2008/04/10 09:11:52: PSTN: Ringing, but last invite = 1 [9] 2008/04/10 09:11:52: SIP Rx tls:192.168.1.101:2614: SIP/2.0 200 Ok Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-cc71df1ba31471f6fbbb0389c307fd35;rport=5061 From: "Anonymous" <sip:anonymous@localhost;user=phone>;tag=986462258 To: <sip:3256725804@localhost;user=phone>;tag=5dp6j7fb67 Call-ID: f7b7fdc2@pbx CSeq: 27585 INVITE Contact: <sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n>;flow-id=1 User-Agent: snom300/7.1.30 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, callerid Content-Type: application/sdp Content-Length: 463 v=0 o=root 1435449537 1435449538 IN IP4 192.168.1.101 s=call c=IN IP4 192.168.1.101 t=0 0 m=audio 52728 RTP/AVP 0 8 9 2 3 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:4L+puWgqA3EYj56N47G+I0A/8cIGl8ZkqvFD0jO2 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=encryption:optional a=alt:1 0.9 : user 9kksj== 192.168.1.101 52728 a=sendrecv [7] 2008/04/10 09:11:52: Call f7b7fdc2@pbx#986462258: Clear last INVITE [6] 2008/04/10 09:11:52: Sending RTP for f7b7fdc2@pbx#986462258 to 192.168.1.101:52728 [9] 2008/04/10 09:11:52: Resolve 241: url sip:192.168.1.101:2614;transport=tls [9] 2008/04/10 09:11:52: Resolve 241: a tls 192.168.1.101 2614 [9] 2008/04/10 09:11:52: Resolve 241: tls 192.168.1.101 2614 [9] 2008/04/10 09:11:52: SIP Tx tls:192.168.1.101:2614: ACK sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n SIP/2.0 Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-c07f3f53965b11f12d9c1f4ac143483d;rport From: "Anonymous" <sip:anonymous@localhost;user=phone>;tag=986462258 To: <sip:3256725804@localhost;user=phone>;tag=5dp6j7fb67 Call-ID: f7b7fdc2@pbx CSeq: 27585 ACK Max-Forwards: 70 Contact: <sip:40@192.168.1.100:5061;transport=tls> Content-Length: 0 [7] 2008/04/10 09:11:52: Determine pass-through mode after receiving response [9] 2008/04/10 09:11:53: Resolve 242: tls 192.168.1.102 2358 [9] 2008/04/10 09:11:53: SIP Tx tls:192.168.1.102:2358: CANCEL sip:41@192.168.1.102:2060;transport=tls;line=94aa7mxt SIP/2.0 Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-0b25baeffb1448798c8837752400d61f;rport From: "Anonymous" <sip:anonymous@localhost;user=phone>;tag=1946255428 To: <sip:3256725804@localhost;user=phone> Call-ID: 2f9a5c8b@pbx CSeq: 13845 CANCEL Max-Forwards: 70 Reason: SIP;cause=200;text="Call completed elsewhere" Content-Length: 0 [9] 2008/04/10 09:11:53: Resolve 243: aaaa udp 127.0.0.1 5062 [9] 2008/04/10 09:11:53: Resolve 243: a udp 127.0.0.1 5062 [9] 2008/04/10 09:11:53: Resolve 243: udp 127.0.0.1 5062 [9] 2008/04/10 09:11:53: SIP Tx udp:127.0.0.1:5062: SIP/2.0 200 Ok Via: SIP/2.0/UDP 127.0.0.1:5062 From: "Anonymous" <sip:anonymous@localhost;user=phone>;tag=233665123 To: <sip:3256725804@localhost;user=phone>;tag=28741e4a25 Call-ID: 3f9a62d7@fxo CSeq: 1 INVITE Contact: <sip:3256725804@127.0.0.1:5060;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/3.0.0.2899 Content-Type: application/sdp Content-Length: 208 v=0 o=- 1380716876 1380716876 IN IP4 127.0.0.1 s=- c=IN IP4 127.0.0.1 t=0 0 m=audio 53392 RTP/AVP 0 101 a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv [5] 2008/04/10 09:11:53: PSTN: Response code: 200 [9] 2008/04/10 09:11:53: SIP Rx tls:192.168.1.102:2358: SIP/2.0 200 OK Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-0b25baeffb1448798c8837752400d61f;rport=5061 From: "Anonymous" <sip:anonymous@localhost;user=phone>;tag=1946255428 To: <sip:3256725804@localhost;user=phone>;tag=wobtjxxe3t Call-ID: 2f9a5c8b@pbx CSeq: 13845 CANCEL Content-Length: 0 [7] 2008/04/10 09:11:53: Call 2f9a5c8b@pbx#1946255428: Clear last request [9] 2008/04/10 09:11:53: SIP Rx tls:192.168.1.102:2358: SIP/2.0 487 Request Terminated Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-0b25baeffb1448798c8837752400d61f;rport=5061 From: "Anonymous" <sip:anonymous@localhost;user=phone>;tag=1946255428 To: <sip:3256725804@localhost;user=phone>;tag=wobtjxxe3t Call-ID: 2f9a5c8b@pbx CSeq: 13845 INVITE Contact: <sip:41@192.168.1.102:2060;transport=tls;line=94aa7mxt>;flow-id=1 Content-Length: 0 [7] 2008/04/10 09:11:53: Call 2f9a5c8b@pbx#1946255428: Clear last INVITE [9] 2008/04/10 09:11:53: Resolve 244: url sip:192.168.1.102:2358;transport=tls [9] 2008/04/10 09:11:53: Resolve 244: a tls 192.168.1.102 2358 [9] 2008/04/10 09:11:53: Resolve 244: tls 192.168.1.102 2358 [9] 2008/04/10 09:11:53: SIP Tx tls:192.168.1.102:2358: ACK sip:41@192.168.1.102:2060;transport=tls;line=94aa7mxt SIP/2.0 Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-0b25baeffb1448798c8837752400d61f;rport From: "Anonymous" <sip:anonymous@localhost;user=phone>;tag=1946255428 To: <sip:3256725804@localhost;user=phone>;tag=wobtjxxe3t Call-ID: 2f9a5c8b@pbx CSeq: 13845 ACK Max-Forwards: 70 Contact: <sip:41@192.168.1.100:5061;transport=tls> Content-Length: 0 [5] 2008/04/10 09:11:53: INVITE Response: Terminate 2f9a5c8b@pbx [7] 2008/04/10 09:11:53: Other Ports: 2 [7] 2008/04/10 09:11:53: Call Port: 3f9a62d7@fxo#28741e4a25 [7] 2008/04/10 09:11:53: Call Port: f7b7fdc2@pbx#986462258 [9] 2008/04/10 09:11:53: SIP Tr udp:127.0.0.1:5062: SIP/2.0 200 Ok Via: SIP/2.0/UDP 127.0.0.1:5062 From: "Anonymous" <sip:anonymous@localhost;user=phone>;tag=233665123 To: <sip:3256725804@localhost;user=phone>;tag=28741e4a25 Call-ID: 3f9a62d7@fxo CSeq: 1 INVITE Contact: <sip:3256725804@127.0.0.1:5060;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/3.0.0.2899 Content-Type: application/sdp Content-Length: 208 v=0 o=- 1380716876 1380716876 IN IP4 127.0.0.1 s=- c=IN IP4 127.0.0.1 t=0 0 m=audio 53392 RTP/AVP 0 101 a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv [5] 2008/04/10 09:11:53: PSTN: Response code: 200 [9] 2008/04/10 09:11:54: SIP Tr udp:127.0.0.1:5062: SIP/2.0 200 Ok Via: SIP/2.0/UDP 127.0.0.1:5062 From: "Anonymous" <sip:anonymous@localhost;user=phone>;tag=233665123 To: <sip:3256725804@localhost;user=phone>;tag=28741e4a25 Call-ID: 3f9a62d7@fxo CSeq: 1 INVITE Contact: <sip:3256725804@127.0.0.1:5060;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/3.0.0.2899 Content-Type: application/sdp Content-Length: 208 v=0 o=- 1380716876 1380716876 IN IP4 127.0.0.1 s=- c=IN IP4 127.0.0.1 t=0 0 m=audio 53392 RTP/AVP 0 101 a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv [5] 2008/04/10 09:11:54: PSTN: Response code: 200 [9] 2008/04/10 09:11:56: SIP Tr udp:127.0.0.1:5062: SIP/2.0 200 Ok Via: SIP/2.0/UDP 127.0.0.1:5062 From: "Anonymous" <sip:anonymous@localhost;user=phone>;tag=233665123 To: <sip:3256725804@localhost;user=phone>;tag=28741e4a25 Call-ID: 3f9a62d7@fxo CSeq: 1 INVITE Contact: <sip:3256725804@127.0.0.1:5060;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/3.0.0.2899 Content-Type: application/sdp Content-Length: 208 v=0 o=- 1380716876 1380716876 IN IP4 127.0.0.1 s=- c=IN IP4 127.0.0.1 t=0 0 m=audio 53392 RTP/AVP 0 101 a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv [5] 2008/04/10 09:11:56: PSTN: Response code: 200 [3] 2008/04/10 09:11:57: PSTN: Channel 1 going to NO_RING [3] 2008/04/10 09:11:58: PSTN: Channel 1 going to RING [5] 2008/04/10 09:11:58: PSTN: Ringing, but last invite = 1 [9] 2008/04/10 09:12:00: SIP Tr udp:127.0.0.1:5062: SIP/2.0 200 Ok Via: SIP/2.0/UDP 127.0.0.1:5062 From: "Anonymous" <sip:anonymous@localhost;user=phone>;tag=233665123 To: <sip:3256725804@localhost;user=phone>;tag=28741e4a25 Call-ID: 3f9a62d7@fxo CSeq: 1 INVITE Contact: <sip:3256725804@127.0.0.1:5060;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/3.0.0.2899 Content-Type: application/sdp Content-Length: 208 v=0 o=- 1380716876 1380716876 IN IP4 127.0.0.1 s=- c=IN IP4 127.0.0.1 t=0 0 m=audio 53392 RTP/AVP 0 101 a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv [5] 2008/04/10 09:12:00: PSTN: Response code: 200 [9] 2008/04/10 09:12:01: SIP Rx tls:192.168.1.101:2614: SUBSCRIBE sip:40@localhost;user=phone SIP/2.0 Via: SIP/2.0/TLS 192.168.1.101:2060;branch=z9hG4bK-qqrcaqbwme2v;rport From: <sip:40@localhost>;tag=vgds1qnpk0 To: <sip:40@localhost;user=phone>;tag=9d4243196b Call-ID: 3c2674105300-yc4ja694jzkk CSeq: 13652 SUBSCRIBE Max-Forwards: 70 Contact: <sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n>;flow-id=1 Event: message-summary Accept: application/simple-message-summary User-Agent: snom300/7.1.30 Proxy-Require: buttons Expires: 3600 Content-Length: 0 [8] 2008/04/10 09:12:01: Packet authenticated by transport layer [9] 2008/04/10 09:12:01: Resolve 245: tls 192.168.1.101 2614 [9] 2008/04/10 09:12:01: SIP Tx tls:192.168.1.101:2614: SIP/2.0 200 Ok Via: SIP/2.0/TLS 192.168.1.101:2060;branch=z9hG4bK-qqrcaqbwme2v;rport=2614 From: <sip:40@localhost>;tag=vgds1qnpk0 To: <sip:40@localhost;user=phone>;tag=9d4243196b Call-ID: 3c2674105300-yc4ja694jzkk CSeq: 13652 SUBSCRIBE Contact: <sip:192.168.1.100:5061;transport=tls> Expires: 179 Content-Length: 0 [3] 2008/04/10 09:12:03: PSTN: Channel 1 going to NO_RING [9] 2008/04/10 09:12:03: SIP Rx tls:192.168.1.101:2614: REGISTER sip:localhost SIP/2.0 Via: SIP/2.0/TLS 192.168.1.101:2060;branch=z9hG4bK-1jx7405cbw2m;rport From: "40" <sip:40@localhost>;tag=lrws3sy5w8 To: "40" <sip:40@localhost> Call-ID: 3c26741018ce-d9hh4ncip2ol CSeq: 14204 REGISTER Max-Forwards: 70 Contact: <sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n>;flow-id=1;q=1.0;+sip.instance="<urn:uuid:822ed566-4660-4bfb-90c3-2b5411c69eb0>" Contact: <http://192.168.1.101:80> Contact: <https://192.168.1.101:443> User-Agent: snom300/7.1.30 Supported: gruu Allow-Events: dialog X-Real-IP: 192.168.1.101 Proxy-Require: buttons Expires: 3600 Content-Length: 0 [8] 2008/04/10 09:12:03: Packet authenticated by transport layer [9] 2008/04/10 09:12:03: Resolve 246: tls 192.168.1.101 2614 [9] 2008/04/10 09:12:03: SIP Tx tls:192.168.1.101:2614: SIP/2.0 200 Ok Via: SIP/2.0/TLS 192.168.1.101:2060;branch=z9hG4bK-1jx7405cbw2m;rport=2614 From: "40" <sip:40@localhost>;tag=lrws3sy5w8 To: "40" <sip:40@localhost>;tag=c02c2dcda3 Call-ID: 3c26741018ce-d9hh4ncip2ol CSeq: 14204 REGISTER Contact: <sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n>;expires=182 Contact: <http://192.168.1.101:80>;expires=182 Contact: <https://192.168.1.101:443>;expires=182 Content-Length: 0 [3] 2008/04/10 09:12:04: PSTN: Channel 1 going to RING [5] 2008/04/10 09:12:04: PSTN: Ringing, but last invite = 1 [3] 2008/04/10 09:12:05: SMTP: Cannot resolve smtp.mail.thesoundshop-llc.com [9] 2008/04/10 09:12:07: SIP Rx tls:192.168.1.101:2614: BYE sip:40@192.168.1.100:5061;transport=tls SIP/2.0 Via: SIP/2.0/TLS 192.168.1.101:2060;branch=z9hG4bK-ehgsa1xb4ej8;rport From: <sip:3256725804@localhost;user=phone>;tag=5dp6j7fb67 To: "Anonymous" <sip:anonymous@localhost;user=phone>;tag=986462258 Call-ID: f7b7fdc2@pbx CSeq: 1 BYE Max-Forwards: 70 Contact: <sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n>;flow-id=1 User-Agent: snom300/7.1.30 RTP-RxStat: Total_Rx_Pkts=17,Rx_Pkts=17,Rx_Pkts_Lost=0,Remote_Rx_Pkts_Lost=0 RTP-TxStat: Total_Tx_Pkts=712,Tx_Pkts=712,Remote_Tx_Pkts=0 Proxy-Require: buttons Content-Length: 0 [9] 2008/04/10 09:12:07: Resolve 247: tls 192.168.1.101 2614 [9] 2008/04/10 09:12:07: SIP Tx tls:192.168.1.101:2614: SIP/2.0 200 Ok Via: SIP/2.0/TLS 192.168.1.101:2060;branch=z9hG4bK-ehgsa1xb4ej8;rport=2614 From: <sip:3256725804@localhost;user=phone>;tag=5dp6j7fb67 To: "Anonymous" <sip:anonymous@localhost;user=phone>;tag=986462258 Call-ID: f7b7fdc2@pbx CSeq: 1 BYE Contact: <sip:40@192.168.1.100:5061;transport=tls> User-Agent: pbxnsip-PBX/3.0.0.2899 RTP-RxStat: Dur=18,Pkt=716,Oct=126016,Underun=1422 RTP-TxStat: Dur=14,Pkt=17,Oct=2992 Content-Length: 0 [7] 2008/04/10 09:12:07: Other Ports: 1 [7] 2008/04/10 09:12:07: Call Port: 3f9a62d7@fxo#28741e4a25 [9] 2008/04/10 09:12:07: Resolve 248: url sip:127.0.0.1:5062 [9] 2008/04/10 09:12:07: Resolve 248: udp 127.0.0.1 5062 [9] 2008/04/10 09:12:07: SIP Tx udp:127.0.0.1:5062: BYE sip:127.0.0.1:5062 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK-438247cad4e43aea9daa827aa236b1dd;rport From: <sip:3256725804@localhost;user=phone>;tag=28741e4a25 To: "Anonymous" <sip:anonymous@localhost;user=phone>;tag=233665123 Call-ID: 3f9a62d7@fxo CSeq: 23963 BYE Max-Forwards: 70 Contact: <sip:3256725804@127.0.0.1:5060;transport=udp> RTP-RxStat: Dur=18,Pkt=0,Oct=0,Underun=6 RTP-TxStat: Dur=14,Pkt=882,Oct=151704 Content-Length: 0 [8] 2008/04/10 09:12:07: DNS: Add dns_a smtp.bizmail.mail.yahoo4.akadns.net 68.142.200.11 (ttl=140) [9] 2008/04/10 09:12:07: SIP Rx udp:127.0.0.1:5062: SIP/2.0 404 Not found Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK-438247cad4e43aea9daa827aa236b1dd;rport From: <sip:3256725804@localhost;user=phone>;tag=28741e4a25 To: "Anonymous" <sip:anonymous@localhost;user=phone>;tag=233665123 Call-ID: 3f9a62d7@fxo CSeq: 23963 BYE Content-Length: 0 [7] 2008/04/10 09:12:07: Call 3f9a62d7@fxo#28741e4a25: Clear last request [5] 2008/04/10 09:12:07: BYE Response: Terminate 3f9a62d7@fxo [8] 2008/04/10 09:12:07: SMTP: Connect to 68.142.200.11:25 [8] 2008/04/10 09:12:07: SMTP: Received 220 smtp100.biz.mail.mud.yahoo.com ESMTP [8] 2008/04/10 09:12:08: SMTP: Received 250-smtp100.biz.mail.mud.yahoo.com 250-AUTH LOGIN PLAIN XYMCOOKIE 250-PIPELINING 250 8BITMIME [8] 2008/04/10 09:12:08: SMTP: Received 334 VXNlcm5hbWU6 [8] 2008/04/10 09:12:08: SMTP: Received 334 UGFzc3dvcmQ6 [3] 2008/04/10 09:12:08: PSTN: Channel 1 going to NO_RING [8] 2008/04/10 09:12:08: SMTP: Received 235 ok, go ahead (#2.0.0) [8] 2008/04/10 09:12:08: SMTP: Received 250 ok [8] 2008/04/10 09:12:08: Last message repeated 2 times [8] 2008/04/10 09:12:08: SMTP: Received 354 go ahead [9] 2008/04/10 09:12:08: SIP Tr udp:127.0.0.1:5062: SIP/2.0 200 Ok Via: SIP/2.0/UDP 127.0.0.1:5062 From: "Anonymous" <sip:anonymous@localhost;user=phone>;tag=233665123 To: <sip:3256725804@localhost;user=phone>;tag=28741e4a25 Call-ID: 3f9a62d7@fxo CSeq: 1 INVITE Contact: <sip:3256725804@127.0.0.1:5060;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/3.0.0.2899 Content-Type: application/sdp Content-Length: 208 v=0 o=- 1380716876 1380716876 IN IP4 127.0.0.1 s=- c=IN IP4 127.0.0.1 t=0 0 m=audio 53392 RTP/AVP 0 101 a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv [5] 2008/04/10 09:12:08: PSTN: Response code: 200 [8] 2008/04/10 09:12:09: SMTP: Received 250 ok 1207836728 qp 18839 [8] 2008/04/10 09:12:09: SMTP: Received 221 smtp100.biz.mail.mud.yahoo.com [8] 2008/04/10 09:12:09: Sucessfully sent email to <russell@thesoundshop-llc.com> [3] 2008/04/10 09:12:09: SMTP: Cannot resolve smtp.mail.thesoundshop-llc.com [9] 2008/04/10 09:12:14: SIP Rx udp:127.0.0.1:5062: CANCEL sip:3256723475@localhost;user=phone SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5062 From: "Anonymous" <sip:anonymous@localhost;user=phone>;tag=1804289383 To: <sip:3256723475@localhost;user=phone> Call-ID: 6931fac9@fxo Contact: <sip:127.0.0.1:5062> CSeq: 1 CANCEL Content-Length: 0 [9] 2008/04/10 09:12:14: Resolve 249: aaaa udp 127.0.0.1 5062 [9] 2008/04/10 09:12:14: Resolve 249: a udp 127.0.0.1 5062 [9] 2008/04/10 09:12:14: Resolve 249: udp 127.0.0.1 5062 [9] 2008/04/10 09:12:14: SIP Tx udp:127.0.0.1:5062: SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP 127.0.0.1:5062 From: "Anonymous" <sip:anonymous@localhost;user=phone>;tag=1804289383 To: <sip:3256723475@localhost;user=phone> Call-ID: 6931fac9@fxo CSeq: 1 CANCEL Content-Length: 0 [5] 2008/04/10 09:12:14: PSTN: Timeout without ring on 1, going to idle [3] 2008/04/10 09:12:14: PSTN: Channel 1 going to IDLE [5] 2008/04/10 09:12:14: PSTN: Response code: 481 [5] 2008/04/10 09:12:27: PSTN: Tone 34 detected on 1 [5] 2008/04/10 09:12:27: PSTN: Tone 255 detected on 1 [5] 2008/04/10 09:12:32: PSTN: Tone 34 detected on 1 [5] 2008/04/10 09:12:32: PSTN: Tone 255 detected on 1
  13. thanks, loaded new version, here is a log of one call causing 2 channels to ring, this is just logged from the pstn side, if you want me capture one with everything let me know..I will try to grab it.. without caller id it worked great.... [3] 2008/03/26 10:51:55: PSTN: Channel 0 going to RING [5] 2008/03/26 10:51:59: PSTN: eVAPI_CALLER_ID_DETECTED_EVENT [5] 2008/03/26 10:51:59: PSTN: Received on 0: Caller-ID 3255131178 [5] 2008/03/26 10:51:59: PSTN: Received on 0: Name THE SOUND SHOP [7] 2008/03/26 10:51:59: UDP: Opening socket on port 53590 [7] 2008/03/26 10:51:59: UDP: Opening socket on port 53591 [5] 2008/03/26 10:51:59: Identify trunk (IP address/port and domain match) 5 [5] 2008/03/26 10:51:59: PSTN: eVAPI_CALLER_ID_DETECTED_EVENT [5] 2008/03/26 10:51:59: PSTN: Received on 1: Caller-ID 3255131178 [7] 2008/03/26 10:51:59: UDP: Opening socket on port 62300 [7] 2008/03/26 10:51:59: UDP: Opening socket on port 62301 [7] 2008/03/26 10:51:59: UDP: Opening socket on port 53548 [7] 2008/03/26 10:51:59: UDP: Opening socket on port 53549 [5] 2008/03/26 10:51:59: PSTN: Received on 1: Name THE SOUND SHOP [7] 2008/03/26 10:51:59: UDP: Opening socket on port 56692 [7] 2008/03/26 10:51:59: UDP: Opening socket on port 56693 [5] 2008/03/26 10:51:59: Identify trunk (IP address/port and domain match) 5 [5] 2008/03/26 10:51:59: PSTN: Response code: 100 [5] 2008/03/26 10:51:59: PSTN: Response code: 183 [5] 2008/03/26 10:51:59: PSTN: Response code: 100 [7] 2008/03/26 10:51:59: UDP: Opening socket on port 63712 [7] 2008/03/26 10:51:59: UDP: Opening socket on port 63713 [7] 2008/03/26 10:51:59: UDP: Opening socket on port 51518 [7] 2008/03/26 10:51:59: UDP: Opening socket on port 51519 [5] 2008/03/26 10:51:59: PSTN: Response code: 183 [5] 2008/03/26 10:52:00: Last message repeated 3 times [3] 2008/03/26 10:52:00: PSTN: Channel 0 going to NO_RING [5] 2008/03/26 10:52:00: PSTN: Response code: 183 [5] 2008/03/26 10:52:02: Last message repeated 2 times [3] 2008/03/26 10:52:02: PSTN: Channel 0 going to RING [5] 2008/03/26 10:52:02: PSTN: Ringing, but last invite = 1 [5] 2008/03/26 10:52:02: PSTN: Response code: 183 [5] 2008/03/26 10:52:06: Last message repeated 3 times [3] 2008/03/26 10:52:06: PSTN: Channel 0 going to NO_RING [5] 2008/03/26 10:52:06: PSTN: Response code: 183 [5] 2008/03/26 10:52:07: PSTN: Response code: 200 [5] 2008/03/26 10:52:07: PSTN: RTP destination=100007f [5] 2008/03/26 10:52:07: PSTN: RTP destination=53590 [5] 2008/03/26 10:52:07: PSTN: RTP OOB codec=101 [6] 2008/03/26 10:52:07: PSTN: Start call on 0 [5] 2008/03/26 10:52:07: PSTN: Channel 0 goes offhook [3] 2008/03/26 10:52:07: PSTN: Channel 0 going to TALKING [5] 2008/03/26 10:52:07: PSTN: Country Code set to 64 [5] 2008/03/26 10:52:07: PSTN: Tone Detection set to 0 [7] 2008/03/26 10:52:14: UDP: Opening socket on port 57756 [7] 2008/03/26 10:52:14: UDP: Opening socket on port 57757 [5] 2008/03/26 10:52:14: PSTN: Response code: 183 [5] 2008/03/26 10:52:24: Last message repeated 6 times [5] 2008/03/26 10:52:24: PSTN: Received BYE message on channel 0 [3] 2008/03/26 10:52:24: PSTN: Channel 0: Hangup [5] 2008/03/26 10:52:24: PSTN: Channel 0 goes onhook [5] 2008/03/26 10:52:24: PSTN: enable_callerid 0 [3] 2008/03/26 10:52:24: PSTN: Channel 0 going to GO_ONHOOK [3] 2008/03/26 10:52:25: PSTN: Channel 0 going to IDLE [5] 2008/03/26 10:52:27: PSTN: Response code: 200
  14. Well you would think.. maybe the processor catches the caller ID right after the ring , you see the no ring after the caller id, (i actually get the phone number in the email trace, it sends a missed called and leaves it in the phone also) but this is what bothering me, is I can see the fxo port flashing, but the phones don't ring, you see the port flashing while this happening [3] 2008/03/25 21:02:52: PSTN: Channel 1 going to NO_RING [3] 2008/03/25 21:02:54: PSTN: Channel 1 going to RING [5] 2008/03/25 21:02:54: PSTN: Ringing, but last invite = 1 [3] 2008/03/25 21:02:58: PSTN: Channel 1 going to NO_RING [3] 2008/03/25 21:03:00: PSTN: Channel 1 going to RING [5] 2008/03/25 21:03:00: PSTN: Ringing, but last invite = 1 [3] 2008/03/25 21:03:04: PSTN: Channel 1 going to NO_RING [3] 2008/03/25 21:03:06: PSTN: Channel 1 going to RING [5] 2008/03/25 21:03:06: PSTN: Ringing, but last invite = 1 I was just wondering what the "last invite =1" thing is? Also where can I find some info on the response and tone codes? thanks russelln
  15. I spent a little time with the box tonight, here is a log of me hanging up a call right as a second call comes in, just logging the pstn port, I can see the call ringing the second PSTN port, but the phones never ring...I get the caller id info, but no call...the first call was disconnected from the other end by myself, I had called myself with my cell phone.. 5] 2008/03/25 21:02:14: PSTN: Tone Detection set to 64 [3] 2008/03/25 21:02:47: PSTN: Channel 1 going to RING [5] 2008/03/25 21:02:51: PSTN: eVAPI_CALLER_ID_DETECTED_EVENT [5] 2008/03/25 21:02:51: PSTN: Received on 1 Caller-ID 3256900821 [5] 2008/03/25 21:02:51: PSTN: Received on 1 Name JACKS DEBORAH [5] 2008/03/25 21:02:51: PSTN: Received LoopInterrupt (remote hung up) signal on channel 0 [3] 2008/03/25 21:02:51: PSTN: Channel 0: Hangup [5] 2008/03/25 21:02:51: PSTN: Channel 0 goes onhook [5] 2008/03/25 21:02:51: PSTN: enable_callerid 0 [3] 2008/03/25 21:02:51: PSTN: Channel 0 going to GO_ONHOOK [5] 2008/03/25 21:02:51: PSTN: Response code: 200 [3] 2008/03/25 21:02:52: PSTN: Channel 0 going to IDLE [3] 2008/03/25 21:02:52: PSTN: Channel 1 going to NO_RING [3] 2008/03/25 21:02:54: PSTN: Channel 1 going to RING [5] 2008/03/25 21:02:54: PSTN: Ringing, but last invite = 1 [3] 2008/03/25 21:02:58: PSTN: Channel 1 going to NO_RING [3] 2008/03/25 21:03:00: PSTN: Channel 1 going to RING [5] 2008/03/25 21:03:00: PSTN: Ringing, but last invite = 1 [3] 2008/03/25 21:03:04: PSTN: Channel 1 going to NO_RING [3] 2008/03/25 21:03:06: PSTN: Channel 1 going to RING [5] 2008/03/25 21:03:06: PSTN: Ringing, but last invite = 1 [3] 2008/03/25 21:03:10: PSTN: Channel 1 going to NO_RING [5] 2008/03/25 21:03:11: PSTN: Tone 16 detected on 65 [5] 2008/03/25 21:03:14: PSTN: Tone 255 detected on 65 [5] 2008/03/25 21:03:14: PSTN: Tone 34 detected on 65 [5] 2008/03/25 21:03:15: PSTN: Tone 255 detected on 65 [5] 2008/03/25 21:03:16: PSTN: Timeout without ring on 1, going to idle [3] 2008/03/25 21:03:16: PSTN: Channel 1 going to IDLE [5] 2008/03/25 21:03:16: PSTN: Response code: 481 [5] 2008/03/25 21:03:19: PSTN: Tone 34 detected on 65 [5] 2008/03/25 21:03:20: PSTN: Tone 255 detected on 65 [5] 2008/03/25 21:03:24: PSTN: Tone 34 detected on 65 [5] 2008/03/25 21:03:25: PSTN: Tone 255 detected on 65 [5] 2008/03/25 21:03:29: PSTN: Tone 34 detected on 65 [5] 2008/03/25 21:03:30: PSTN: Tone 255 detected on 65 [5] 2008/03/25 21:03:34: PSTN: Tone 34 detected on 65 [5] 2008/03/25 21:03:35: PSTN: Tone 255 detected on 65 [5] 2008/03/25 21:03:39: PSTN: Tone 34 detected on 65 [5] 2008/03/25 21:03:40: PSTN: Tone 255 detected on 65 [5] 2008/03/25 21:03:44: PSTN: Tone 34 detected on 65 [5] 2008/03/25 21:03:45: PSTN: Tone 255 detected on 65 [5] 2008/03/25 21:03:49: PSTN: Tone 34 detected on 65 [5] 2008/03/25 21:03:50: PSTN: Tone 255 detected on 65 Copyright © 2005-2008 pbxnsip Inc. | Home | Help | Logout
  16. Hi All, Lines hangging fxo ports, I just had caller id turned on on my 2 lines, the 2 days we went with no caller ID the world was great, now I am having some new weirdness, I have not caught the time when the line hangs but it does not happen all the time, It is just that I have a port in use and no calls are being made,,I have noticed now that calls comming from some different area codes will cause both lines to ring..and I can answer the call and the other line will keep ringing but no one there, also, the opposite will happen, and it connects the second line..anyways when I get a chance I will log it and study them to see what the pstn port is doing..
  17. Well luckily I had saved PBX file structure before updating system, pushed old file structure back in, recovered liscene key and saved key in safe place, to see if happens again, pushed new software, system came up properly, removed huge log file from box 180mb and all other items in logs that did not look like needed, the system lost liscene when i was looking at date, system was changing date at noon to next day and on eastern time, made these changes and saved when system crashed and lost liscene...now have made time change again, did not lockup this time and rebooted still have liscene... phones registered again... things back to normal, maybe....
  18. Been working till today when license dissappeared!!! box currently showing no liscene and will not register phones...
  19. 2 days no problems, voice mail working, caller id working....using SNOM 300 phones with latest firmware.... hope to get a sip trunk going soon. like the new look of the new version..
  20. Used switch all day, no problems (yeah!!)using 2 extensions, added back in service flag tonight and auto attendant tonight, will add more extensions in the next week, plan is for 9, next is to install sip trunk... all the best everyone.... russelln
  21. No I really do not have the time, It is that I have a investment in this product and and I want to see it sucessful, also I wanted to fully make sure that I understood the problem as well as everybody else, because I might be wrong.(which I am alot)..the are of engineering is making sure everything is correct, I feel....better to have trouble now before I have 200 of these in the feild and wondering what the heck is going on when the phone co makes a little change that is still withing the specs of the analog system...thank you for your suggestions, because opening the box helped alot, (I started in this world as board level repair person, just did not think to look in the box),because when I knew the controller could detect a loop loss as little as 1ms, I figured it was a software issue with respect to programmer's logic towards how he understood the how the phone network behaves, the scope pictures showed me it was the box making the decesion, which I was not really sure of till that time.. all the best russelln
  22. set cpc to 400ms, left everything else the same.. I heard a busy signal, I can dial back in on my own lines, called in and out have not dropped a call yet.. thankyou,thankyou,thankyou,thankyou,thankyou,thankyou,thankyou,thankyou,thankyou,thankyou,thankyou,thankyou,thankyou,thankyou,thankyou,thankyou,thankyou,thankyou,thankyou,thankyou,thankyou,thankyou,thankyou,thankyou,thankyou,thankyou,thankyou,thankyou,thankyou,thankyou,thankyou,thankyou,thankyou,thankyou,thankyou,thankyou,thankyou,thankyou,thankyou,thankyou,thankyou,thankyou,thankyou,thankyou,thankyou,thankyou,thankyou,thankyou,thankyou,thankyou,thankyou,thankyou,thankyou,thankyou,thankyou,thankyou,thankyou,thankyou,thankyou,thankyou,thankyou,thankyou,thankyou,thankyou,thankyou,thankyou,thankyou,thankyou,thankyou,thankyou,thankyou,thankyou, russell
  23. I just downloaded 2450 for the CS410, it comes up as 2448 on the webpage, and still shows as 2448 in the status on the box... is the file different or was in not renamed? thanks russelln
  24. Well finally got rigged up to measure record the pulses on the box, this first picture is of the box capturing the line, I see this pulse right as the dialing tones string are sent out, this pulse is always the same, and the box never dissconnects on this pulse..it lasts about 70ms, scope time was set to 50ms... the next picture is the box dropping the line just called, it sometimes connects, so I guess this is where the margin is, on where the box deceides when to dissconnect, you can see the line going back to the connected call state, after the pulse, but then a decesion is made to dissconnect the line from the the voltage going to max again...you can see the pulse time is about 80ms and at about 120ms the call is dissconnected, these always look about the same. now we call a busy line, I have never heard a busy signal, all pulses coming from a busy signal are always longer than when the line is not busy, this looks like it is about 110ms, but again you see the line recover and a decesion is made to dissconnect the call...75ms later. Now I managed to get a call connected ( because it does connect about 50% of the time) and had the other party hang up, I held the line for the CO CPC pulse and here it comes about 30 seconds after and we see the pulse is about 850ms, it took two frames to show the time as it is much longer...than the little blips I get making calls, according to what i have seen ,it looks like the box makes the decesion around 60 ms or so to consider the call dissconnected, because 70ms is too long... It seems like you could use the debounce function in the port controller chip to check for no current and then create a loop to count against the processor clock in the cpu, if there is no debounce at that time then the line is at off hook you would not send the loop interrupt to the 127 server, and you could make the count adjustable for different CPC times..would really be nice to be able to increase the CPC time... all the best russelln
  25. Hi, I tried to push 2.1.6.2448 into a white box, which I upgraded to 2.1.5.2357 using the SSH, i used the image upload in the Win Browser...it acts like it is taking it but, always reverts back to 2.1.5.2357, will this unit take the new firmware? thanks russelln
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