смотрю логи звонка:
| Код: |
| -- Executing [s@macro-dialout-trunk:30] Dial("SIP/107-0000002e", "datacard/r2/+79138QWERTY,300,") in new stack -- Called datacard/r2/+79138QWERTY -- Datacard/datacard1-b108 is making progress passing it to SIP/107-0000002e -- Datacard/datacard1-b108 answered SIP/107-0000002e -- Executing [h@macro-dialout-trunk:1] Macro("SIP/107-0000002e", "hangupcall,") in new stack -- Executing [s@macro-hangupcall:1] GotoIf("SIP/107-0000002e", "1?theend") in new stack -- Goto (macro-hangupcall,s,3) -- Executing [s@macro-hangupcall:3] ExecIf("SIP/107-0000002e", "0?Set(CDR(recordingfile)=)") in new stack -- Executing [s@macro-hangupcall:4] Hangup("SIP/107-0000002e", "") in new stack == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'SIP/107-0000002e' in macro 'hangupcall' == Spawn extension (macro-dialout-trunk, h, 1) exited non-zero on 'SIP/107-0000002e' == Spawn extension (macro-dialout-trunk, s, 30) exited non-zero on 'SIP/107-0000002e' in macro 'dialout-trunk' == Spawn extension (from-internal, 89138QWERTY, 5) exited non-zero on 'SIP/107-0000002e' == MixMonitor close filestream == End MixMonitor Recording SIP/107-0000002e |
а вот лог с этого же телефона, только старый астер:
| Код: |
| -- Executing [s@macro-dialout-trunk:28] Dial("SIP/107-00000030", "datacard/i:35244504742XXXX/+79138QWERTY,300,") in new stack -- Called i:35244504742XXXX/+79138QWERTY -- Datacard/MTS-b9b3 is making progress passing it to SIP/107-00000030 -- Datacard/MTS-b9b3 answered SIP/107-00000030 -- Executing [h@macro-dialout-trunk:1] Macro("SIP/107-00000030", "hangupcall,") in new stack -- Executing [s@macro-hangupcall:1] GotoIf("SIP/107-00000030", "1?theend") in new stack -- Goto (macro-hangupcall,s,3) -- Executing [s@macro-hangupcall:3] ExecIf("SIP/107-00000030", "0?Set(CDR(recordingfile)=)") in new stack -- Executing [s@macro-hangupcall:4] Hangup("SIP/107-00000030", "") in new stack == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'SIP/107-00000030' in macro 'hangupcall' == Spawn extension (macro-dialout-trunk, h, 1) exited non-zero on 'SIP/107-00000030' == Spawn extension (macro-dialout-trunk, s, 28) exited non-zero on 'SIP/107-00000030' in macro 'dialout-trunk' == Spawn extension (from-internal, 89138QWERTY, 5) exited non-zero on 'SIP/107-00000030' -- Executing [h@from-internal:1] Hangup("SIP/107-00000030", "") in new stack == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/107-00000030' == MixMonitor close filestream == End MixMonitor Recording SIP/107-00000030 |
видно что в первом случае отсутствует строка -- Executing [h@from-internal:1] Hangup("SIP/107-00000030", "") in new stack
как добиться чтобы она появилась?
| Код: |
| [from-test] exten => 500,1,Answer() exten => 500,2,Playback(agent-pass) ;exten => 500,n,Echo() exten => 500,3,Wait(5) exten => 500,4,Hangup() |
| Код: |
| -- Executing [500@from-test:1] Answer("SIP/107-0000000a", "") in new stack -- Executing [500@from-test:2] Playback("SIP/107-0000000a", "agent-pass") in new stack -- Playing 'agent-pass.slin' (language 'ru') [2012-08-20 13:07:02] NOTICE[21134]: channel.c:4176 __ast_read: Dropping incompatible voice frame on SIP/107-0000000a of format alaw since our native format has changed to 0x4 (ulaw) -- Executing [500@from-test:3] Wait("SIP/107-0000000a", "5") in new stack -- Executing [500@from-test:4] Hangup("SIP/107-0000000a", "") in new stack == Spawn extension (from-test, 500, 4) exited non-zero on 'SIP/107-0000000a' |
телефон не понимает что разговор закончился
_________________
Внимание! Свет в конце тоннеля может быть светом фар приближающегося поезда!
Ubuntu 10.04/12.04 - Asterisk 1.8.11.0-rc2/1.8.14.1/1.8.17.0/10.10.0
_________________
Внимание! Свет в конце тоннеля может быть светом фар приближающегося поезда!
Ubuntu 10.04/12.04 - Asterisk 1.8.11.0-rc2/1.8.14.1/1.8.17.0/10.10.0
какие конкретно логи глянуть?
Leon77
linksys spa942
ну и звоним с 107 на 500
| Код: |
| Contact: "107" User-Agent: Linksys/SPA942-6.1.5(a) Content-Length: 0 --- (10 headers 0 lines) --- INVITE sip:500@192.168.33.29 SIP/2.0 Via: SIP/2.0/UDP 192.168.33.24:5060;branch=z9hG4bK-48a8bf6b From: "107" ;tag=2199e732c462449fo3 To: Call-ID: 3fdc0cee-ea01fa13@192.168.33.24 CSeq: 102 INVITE Max-Forwards: 70 Authorization: Digest username="107",realm="asterisk",nonce="326bd2e3",uri="sip:500@192.168.33.29",algorithm=MD5,response="e3efcc0cac73e415cc197e619a72de83" Contact: "107" Expires: 240 User-Agent: Linksys/SPA942-6.1.5(a) Content-Length: 397 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces Content-Type: application/sdp v=0 o=- 139178 139178 IN IP4 192.168.33.24 s=- c=IN IP4 192.168.33.24 t=0 0 m=audio 16404 RTP/AVP 0 2 4 8 18 96 97 98 101 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729a/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv --- (15 headers 18 lines) --- Sending to 192.168.33.24:5060 (no NAT) Using INVITE request as basis request - 3fdc0cee-ea01fa13@192.168.33.24 Found peer '107' for '107' from 192.168.33.24:5060 == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 Found RTP audio format 0 Found RTP audio format 2 Found RTP audio format 4 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 96 Found RTP audio format 97 ound RTP audio format 98 Found RTP audio format 101 Found audio description format PCMU for ID 0 Found audio description format G726-32 for ID 2 Found audio description format G723 for ID 4 Found audio description format PCMA for ID 8 Found audio description format G729a for ID 18 Found unknown media description format G726-40 for ID 96 Found unknown media description format G726-24 for ID 97 Found unknown media description format G726-16 for ID 98 Found audio description format telephone-event for ID 101 Capabilities: us - 0x10e (gsm|ulaw|alaw|g729), peer - audio=0x90d (g723|ulaw|alaw|g726|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x10c (ulaw|alaw|g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 192.168.33.24:16404 Looking for 500 in from-test (domain 192.168.33.29) list_route: hop: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.33.24:5060;branch=z9hG4bK-48a8bf6b;received=192.168.33.24 From: "107" ;tag=2199e732c462449fo3 To: Call-ID: 3fdc0cee-ea01fa13@192.168.33.24 CSeq: 102 INVITE Server: FPBX-2.10.1(1.8.15.0) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 -- Executing [500@from-test:1] Answer("SIP/107-00000017", "") in new stack Audio is at 10656 Adding codec 0x8 (alaw) to SDP Adding codec 0x100 (g729) to SDP Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.33.24:5060;branch=z9hG4bK-48a8bf6b;received=192.168.33.24 From: "107" ;tag=2199e732c462449fo3 To: ;tag=as553aa6a0 Call-ID: 3fdc0cee-ea01fa13@192.168.33.24 CSeq: 102 INVITE Server: FPBX-2.10.1(1.8.15.0) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: ontent-Type: application/sdp Content-Length: 337 v=0 o=root 1854165868 1854165868 IN IP4 192.168.33.29 s=Asterisk PBX 1.8.15.0 c=IN IP4 192.168.33.29 t=0 0 m=audio 10656 RTP/AVP 8 18 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv ACK sip:500@192.168.33.29:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.33.24:5060;branch=z9hG4bK-321621be From: "107" ;tag=2199e732c462449fo3 To: ;tag=as553aa6a0 Call-ID: 3fdc0cee-ea01fa13@192.168.33.24 CSeq: 102 ACK Max-Forwards: 70 Authorization: Digest username="107",realm="asterisk",nonce="326bd2e3",uri="sip:500@192.168.33.29",algorithm=MD5,response="e3efcc0cac73e415cc197e619a72de83" Contact: "107" User-Agent: Linksys/SPA942-6.1.5(a) Content-Length: 0 --- (11 headers 0 lines) --- -- Executing [500@from-test:2] Playback("SIP/107-00000017", "agent-pass") in new stack -- Playing 'agent-pass.slin' (language 'ru') [2012-08-20 16:28:04] NOTICE[5087]: channel.c:4176 __ast_read: Dropping incompatible voice frame on SIP/107-00000017 of format alaw since our native format has changed to 0x4 (ulaw) Reliably Transmitting (no NAT) to 192.168.33.24:5060: OPTIONS sip:107@192.168.33.24:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.33.29:5060;branch=z9hG4bK10c27f3d Max-Forwards: 70 From: "Unknown" ;tag=as6611e3c7 To: Contact: Call-ID: 5a7aa61d549896e26a9486934266fabb@192.168.33.29:5060 CSeq: 102 OPTIONS User-Agent: FPBX-2.10.1(1.8.15.0) Date: Mon, 20 Aug 2012 09:28:04 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- SIP/2.0 200 OK To: ;tag=2763bdbbb9049c9ai0 From: "Unknown" ;tag=as6611e3c7 Call-ID: 5a7aa61d549896e26a9486934266fabb@192.168.33.29:5060 CSeq: 102 OPTIONS Via: SIP/2.0/UDP 192.168.33.29:5060;branch=z9hG4bK10c27f3d Server: Linksys/SPA942-6.1.5(a) Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces --- (10 headers 0 lines) --- Really destroying SIP dialog '5a7aa61d549896e26a9486934266fabb@192.168.33.29:5060' Method: OPTIONS -- Executing [500@from-test:3] Wait("SIP/107-00000017", "5") in new stack -- Executing [500@from-test:4] Hangup("SIP/107-00000017", "") in new stack == Spawn extension (from-test, 500, 4) exited non-zero on 'SIP/107-00000017' Scheduling destruction of SIP dialog '3fdc0cee-ea01fa13@192.168.33.24' in 6400 ms (Method: ACK) set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.33.24:5060 Reliably Transmitting (no NAT) to 192.168.33.24:5060: BYE sip:107@192.168.33.24:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.33.29:5060;branch=z9hG4bK79811e8a Max-Forwards: 70 From: ;tag=as553aa6a0 To: "107" ;tag=2199e732c462449fo3 Call-ID: 3fdc0cee-ea01fa13@192.168.33.24 CSeq: 102 BYE User-Agent: FPBX-2.10.1(1.8.15.0) Proxy-Authorization: Digest username="107", realm="asterisk", algorithm=MD5, uri="sip:192.168.33.29", nonce="", response="7d1d465e76545d6e18d6c57c5dca3fae" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- SIP/2.0 481 Call Leg/Transaction Does Not Exist To: "107" ;tag=2199e732c462449fo3 From: ;tag=as553aa6a0 Call-ID: 3fdc0cee-ea01fa13@192.168.33.24 CSeq: 102 BYE Via: SIP/2.0/UDP 192.168.33.29:5060;branch=z9hG4bK79811e8a Server: Linksys/SPA942-6.1.5(a) Content-Length: 0 --- (8 headers 0 lines) --- Really destroying SIP dialog '3fdc0cee-ea01fa13@192.168.33.24' Method: ACK BYE sip:500@192.168.33.29:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.33.24:5060;branch=z9hG4bK-515f7cca From: "107" ;tag=2199e732c462449fo3 To: ;tag=as553aa6a0 Call-ID: 3fdc0cee-ea01fa13@192.168.33.24 CSeq: 103 BYE Max-Forwards: 70 Authorization: Digest username="107",realm="asterisk",nonce="326bd2e3",uri="sip:500@192.168.33.29:5060",algorithm=MD5,response="d8ded9a558c165a8539b71447afecd6a" User-Agent: Linksys/SPA942-6.1.5(a) Content-Length: 0 --- (10 headers 0 lines) --- SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/UDP 192.168.33.24:5060;branch=z9hG4bK-515f7cca;received=192.168.33.24 From: "107" ;tag=2199e732c462449fo3 To: ;tag=as553aa6a0 Call-ID: 3fdc0cee-ea01fa13@192.168.33.24 CSeq: 103 BYE Server: FPBX-2.10.1(1.8.15.0) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 |
Added after 18 minutes:
причем эта часть появляется в логах, после того как я физически кладу трубку
| Код: |
| --- (8 headers 0 lines) --- Really destroying SIP dialog '3fdc0cee-ea01fa13@192.168.33.24' Method: ACK BYE sip:500@192.168.33.29:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.33.24:5060;branch=z9hG4bK-515f7cca From: "107" ;tag=2199e732c462449fo3 To: ;tag=as553aa6a0 Call-ID: 3fdc0cee-ea01fa13@192.168.33.24 CSeq: 103 BYE Max-Forwards: 70 Authorization: Digest username="107",realm="asterisk",nonce="326bd2e3",uri="sip:500@192.168.33.29:5060",algorithm=MD5,response="d8ded9a558c165a8539b71447afecd6a" User-Agent: Linksys/SPA942-6.1.5(a) Content-Length: 0 --- (10 headers 0 lines) --- SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/UDP 192.168.33.24:5060;branch=z9hG4bK-515f7cca;received=192.168.33.24 From: "107" ;tag=2199e732c462449fo3 To: ;tag=as553aa6a0 Call-ID: 3fdc0cee-ea01fa13@192.168.33.24 CSeq: 103 BYE Server: FPBX-2.10.1(1.8.15.0) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 |
Последний раз редактировалось: GS (Чт Авг 30, 2012 12:26)
| Код: |
| Reliably Transmitting (no NAT) to 192.168.33.24:5060: BYE sip:107@192.168.33.24:5060 SIP/2.0 |
на что линксис отвечает
| Код: |
| SIP/2.0 481 Call Leg/Transaction Does Not Exist |
т.е линксис считает, что такого вызова у него нет, в дальнейшем астериск закрывает эту сессию.
Когда ты кладешь трубку на линксисе, то линксис шлет
| Код: |
| BYE sip:500@192.168.33.29:5060 SIP/2.0 |
| Код: |
| SIP/2.0 481 Call leg/transaction does not exist |
Думаю что проблема у тебя с линксисом.
| Alex_asdf @ Пн Авг 20, 2012 16:58 писал(а): |
| Думаю что проблема у тебя с линксисом. |
| Lonely_Ghost @ Вт Авг 21, 2012 13:55 писал(а): |
| Покажи SIP-дебаг "другого астера". |
| Код: |
| SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.33.24:5062;branch=z9hG4bK-a91a9ce;received=192.168.33.24 From: "777" ;tag=f126d35c251f1722o2 To: Call-ID: 9e742b64-29ce5eba@192.168.33.24 CSeq: 102 INVITE Server: FPBX-2.10.1(1.6.2.18) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Length: 0 [2012-08-21 21:12:08] VERBOSE[21795] pbx.c: -- Executing [500@from-test:1] Answer("SIP/777-00000007", "") in new stack [2012-08-21 21:12:08] VERBOSE[21795] chan_sip.c: Audio is at 192.168.33.22 port 17320 [2012-08-21 21:12:08] VERBOSE[21795] chan_sip.c: Adding codec 0x100 (g729) to SDP [2012-08-21 21:12:08] VERBOSE[21795] chan_sip.c: Adding codec 0x8 (alaw) to SDP [2012-08-21 21:12:08] VERBOSE[21795] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [2012-08-21 21:12:08] VERBOSE[21795] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [2012-08-21 21:12:08] VERBOSE[21795] chan_sip.c: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.33.24:5062;branch=z9hG4bK-a91a9ce;received=192.168.33.24 From: "777" ;tag=f126d35c251f1722o2 To: ;tag=as31d43fe5 Call-ID: 9e742b64-29ce5eba@192.168.33.24 CSeq: 102 INVITE Server: FPBX-2.10.1(1.6.2.18) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 335 v=0 o=root 350042229 350042229 IN IP4 192.168.33.22 s=Asterisk PBX 1.6.2.18 c=IN IP4 192.168.33.22 t=0 0 m=audio 17320 RTP/AVP 18 8 0 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv [2012-08-21 21:12:08] VERBOSE[20758] chan_sip.c: ACK sip:500@192.168.33.22 SIP/2.0 Via: SIP/2.0/UDP 192.168.33.24:5062;branch=z9hG4bK-de2be35a From: "777" ;tag=f126d35c251f1722o2 To: ;tag=as31d43fe5 Call-ID: 9e742b64-29ce5eba@192.168.33.24 CSeq: 102 ACK Max-Forwards: 70 Authorization: Digest username="777",realm="asterisk",nonce="332e98a4",uri="sip:500@192.168.33.22",algorithm=MD5,response="afc01e75ca037e6ba6059b29c415c1e1" Contact: "777" User-Agent: Linksys/SPA942-6.1.5(a) Content-Length: 0 [2012-08-21 21:12:08] VERBOSE[20758] chan_sip.c: --- (11 headers 0 lines) --- [2012-08-21 21:12:09] VERBOSE[21795] pbx.c: -- Executing [500@from-test:2] Playback("SIP/777-00000007", "agent-pass") in new stack [2012-08-21 21:12:09] VERBOSE[21795] file.c: -- Playing 'agent-pass.slin' (language 'ru') [2012-08-21 21:12:09] NOTICE[21795] channel.c: Dropping incompatible voice frame on SIP/777-00000007 of format g729 since our native format has changed to 0x4 (ulaw) [2012-08-21 21:12:11] VERBOSE[21795] pbx.c: -- Executing [500@from-test:3] Wait("SIP/777-00000007", "1") in new stack [2012-08-21 21:12:11] VERBOSE[20758] chan_sip.c: SUBSCRIBE sip:777@192.168.33.22 SIP/2.0 Via: SIP/2.0/UDP 192.168.33.24:5062;branch=z9hG4bK-c6061850 From: "777" ;tag=2b2aa83fd55513f4 To: "777" Call-ID: 92b8bb08-56287092@192.168.33.24 CSeq: 1001 SUBSCRIBE Max-Forwards: 70 Contact: "777" Expires: 3600 Event: call-info User-Agent: Linksys/SPA942-6.1.5(a) Content-Length: 0 [2012-08-21 21:12:11] VERBOSE[20758] chan_sip.c: --- (12 headers 0 lines) --- [2012-08-21 21:12:11] VERBOSE[20758] chan_sip.c: Creating new subscription [2012-08-21 21:12:11] VERBOSE[20758] chan_sip.c: Sending to 192.168.33.24 : 5062 (no NAT) [2012-08-21 21:12:11] VERBOSE[20758] chan_sip.c: list_route: hop: [2012-08-21 21:12:11] VERBOSE[20758] chan_sip.c: Found peer '777' for '777' from 192.168.33.24:5062 [2012-08-21 21:12:11] VERBOSE[20758] chan_sip.c: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.33.24:5062;branch=z9hG4bK-c6061850;received=192.168.33.24 From: "777" ;tag=2b2aa83fd55513f4 To: "777" ;tag=as2ce7a4b1 Call-ID: 92b8bb08-56287092@192.168.33.24 CSeq: 1001 SUBSCRIBE Server: FPBX-2.10.1(1.6.2.18) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="73fe3143" Content-Length: 0 [2012-08-21 21:12:11] VERBOSE[20758] chan_sip.c: Scheduling destruction of SIP dialog '92b8bb08-56287092@192.168.33.24' in 6400 ms (Method: SUBSCRIBE) [2012-08-21 21:12:11] VERBOSE[20758] chan_sip.c: SUBSCRIBE sip:777@192.168.33.22 SIP/2.0 Via: SIP/2.0/UDP 192.168.33.24:5062;branch=z9hG4bK-3004662c From: "777" ;tag=2b2aa83fd55513f4 To: "777" Call-ID: 92b8bb08-56287092@192.168.33.24 CSeq: 1002 SUBSCRIBE Max-Forwards: 70 Authorization: Digest username="777",realm="asterisk",nonce="73fe3143",uri="sip:777@192.168.33.22",algorithm=MD5,response="96cf50a27bbc71bd62d9b0b6fcf6536b" Contact: "777" Expires: 3600 Event: call-info User-Agent: Linksys/SPA942-6.1.5(a) Content-Length: 0 [2012-08-21 21:12:11] VERBOSE[20758] chan_sip.c: --- (13 headers 0 lines) --- [2012-08-21 21:12:11] VERBOSE[20758] chan_sip.c: Creating new subscription [2012-08-21 21:12:11] VERBOSE[20758] chan_sip.c: Sending to 192.168.33.24 : 5062 (no NAT) [2012-08-21 21:12:11] VERBOSE[20758] chan_sip.c: Found peer '777' for '777' from 192.168.33.24:5062 [2012-08-21 21:12:11] VERBOSE[20758] chan_sip.c: Looking for 777 in from-test (domain 192.168.33.22) [2012-08-21 21:12:11] VERBOSE[20758] chan_sip.c: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.33.24:5062;branch=z9hG4bK-3004662c;received=192.168.33.24 From: "777" ;tag=2b2aa83fd55513f4 To: "777" ;tag=as2ce7a4b1 Call-ID: 92b8bb08-56287092@192.168.33.24 CSeq: 1002 SUBSCRIBE Server: FPBX-2.10.1(1.6.2.18) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 [2012-08-21 21:12:11] VERBOSE[20758] chan_sip.c: Really destroying SIP dialog '92b8bb08-56287092@192.168.33.24' Method: SUBSCRIBE [2012-08-21 21:12:12] VERBOSE[21795] pbx.c: -- Executing [500@from-test:4] Hangup("SIP/777-00000007", "") in new stack [2012-08-21 21:12:12] VERBOSE[21795] pbx.c: == Spawn extension (from-test, 500, 4) exited non-zero on 'SIP/777-00000007' [2012-08-21 21:12:12] VERBOSE[21795] chan_sip.c: Scheduling destruction of SIP dialog '9e742b64-29ce5eba@192.168.33.24' in 6400 ms (Method: ACK) [2012-08-21 21:12:12] VERBOSE[21795] chan_sip.c: set_destination: Parsing for address/port to send to [2012-08-21 21:12:12] VERBOSE[21795] chan_sip.c: set_destination: set destination to 192.168.33.24, port 5062 [2012-08-21 21:12:12] VERBOSE[21795] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.33.24:5062: BYE sip:777@192.168.33.24:5062 SIP/2.0 Via: SIP/2.0/UDP 192.168.33.22:5060;branch=z9hG4bK7746ae8e;rport Max-Forwards: 70 From: ;tag=as31d43fe5 To: "777" ;tag=f126d35c251f1722o2 Call-ID: 9e742b64-29ce5eba@192.168.33.24 CSeq: 102 BYE User-Agent: FPBX-2.10.1(1.6.2.18) X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- [2012-08-21 21:12:12] VERBOSE[20758] chan_sip.c: SIP/2.0 200 OK To: "777" ;tag=f126d35c251f1722o2 From: ;tag=as31d43fe5 Call-ID: 9e742b64-29ce5eba@192.168.33.24 CSeq: 102 BYE Via: SIP/2.0/UDP 192.168.33.22:5060;branch=z9hG4bK7746ae8e Server: Linksys/SPA942-6.1.5(a) Content-Length: 0 |