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спросил 2015-08-14 11:37:07 +0400

Vlady_32 Gravatar Vlady_32

Пропадание звука в звонках с астериск?

Ситуация следующая. Есть сервер с Asterisk 11.18.0. К нему подключен gsm шлюз Openvox. Также создано 3 внутренних абонента, к каждому из которых привязана симкарта на шлюзе. Ситуация следующая, менеджер звонит (либо менеджеру звонят), разговор начинается. Все друг друга хорошо слышат. Но в последнее время стал всё чаще пропадать звук. ТО есть разговор не прерывается, а просто абонент и менеджер не слышит друг друга. И это происходит не всегда. Скажите, как можно отследить, почему пропадает звук о обеих сторон? Может у вас есть какие-либо предположения, что это может быть?

<--- SIP read from UDP:192.168.0.150:5060 --->

INVITE sip:+375296022323@192.168.0.104:4524 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.150:5060;branch=z9hG4bK233836d7;rport Max-Forwards: 70 From: "+375293204872" <sip:+375293204872@192.168.0.150>;tag=as71d12b59 To: <sip:+375296022323@192.168.0.104:4524> Contact: <sip:+375293204872@192.168.0.150:5060> Call-ID: 76c8f9af1a3a4cc960a1d9d23ed84964@192.168.0.150:5060 CSeq: 102 INVITE User-Agent: VoxStack Wireless Gateway Date: Wed, 25 Feb 1970 00:45:35 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-Best-Codec: alaw Content-Type: application/sdp Content-Length: 434

v=0 o=root 615765964 615765964 IN IP4 192.168.0.150 s=VoxStack Wireless Gateway c=IN IP4 192.168.0.150 t=0 0 m=audio 18710 RTP/AVP 8 0 3 9 4 111 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:9 G722/8000 a=rtpmap:4 G723/8000 a=fmtp:4 annexa=no a=rtpmap:111 G726-32/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <-------------> [Aug 13 14:40:46] VERBOSE[31702] chansip.c: --- (15 headers 19 lines) --- [Aug 13 14:40:46] VERBOSE[31702] chansip.c: Sending to 192.168.0.150:5060 (NAT) [Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chansip.c: Sending to 192.168.0.150:5060 (NAT) [Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chansip.c: Using INVITE request as basis request - 76c8f9af1a3a4cc960a1d9d23ed84964@192.168.0.150:5060 [Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chansip.c: Found peer 'openvox' for '+375293204872' from 192.168.0.150:5060 [Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chansip.c: <--- Reliably Transmitting (no NAT) to 192.168.0.150:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.0.150:5060;branch=z9hG4bK233836d7;received=192.168.0.150;rport=5060 From: "+375293204872" <sip:+375293204872@192.168.0.150>;tag=as71d12b59 To: <sip:+375296022323@192.168.0.104:4524>;tag=as378bb9e4 Call-ID: 76c8f9af1a3a4cc960a1d9d23ed84964@192.168.0.150:5060 CSeq: 102 INVITE Server: Asterisk PBX 11.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1edea583" Content-Length: 0

<------------> [Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chansip.c: Scheduling destruction of SIP dialog '76c8f9af1a3a4cc960a1d9d23ed84964@192.168.0.150:5060' in 6400 ms (Method:$ [Aug 13 14:40:46] VERBOSE[31702] chansip.c: <--- SIP read from UDP:192.168.0.150:5060 ---> ACK sip:+375296022323@192.168.0.104:4524 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.150:5060;branch=z9hG4bK233836d7;rport Max-Forwards: 70 From: "+375293204872" <sip:+375293204872@192.168.0.150>;tag=as71d12b59 To: <sip:+375296022323@192.168.0.104:4524>;tag=as378bb9e4 Contact: <sip:+375293204872@192.168.0.150:5060> Call-ID: 76c8f9af1a3a4cc960a1d9d23ed84964@192.168.0.150:5060 CSeq: 102 ACK User-Agent: VoxStack Wireless Gateway Content-Length: 0 <-------------> [Aug 13 14:40:46] VERBOSE[31702] chansip.c: --- (10 headers 0 lines) --- [Aug 13 14:40:46] VERBOSE[31702] chansip.c: <--- SIP read from UDP:192.168.0.150:5060 ---> INVITE sip:+375296022323@192.168.0.104:4524 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.150:5060;branch=z9hG4bK49ade539;rport Max-Forwards: 70 From: "+375293204872" <sip:+375293204872@192.168.0.150>;tag=as71d12b59 To: <sip:+375296022323@192.168.0.104:4524> Contact: <sip:+375293204872@192.168.0.150:5060> Call-ID: 76c8f9af1a3a4cc960a1d9d23ed84964@192.168.0.150:5060 CSeq: 103 INVITE User-Agent: VoxStack Wireless Gateway Authorization: Digest username="openvox", realm="asterisk", algorithm=MD5, uri="sip:+375296022323@192.168.0.104:4524", nonce="1edea583", response="a9b79f406c3e3acd7088$ Date: Wed, 25 Feb 1970 00:45:35 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-Best-Codec: alaw Content-Type: application/sdp Content-Length: 434

v=0 o=root 615765964 615765965 IN IP4 192.168.0.150 s=VoxStack Wireless Gateway c=IN IP4 192.168.0.150 t=0 0 m=audio 18710 RTP/AVP 8 0 3 9 4 111 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:9 G722/8000 a=rtpmap:4 G723/8000 a=fmtp:4 annexa=no a=rtpmap:111 G726-32/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <-------------> [Aug 13 14:40:46] VERBOSE[31702] chansip.c: --- (16 headers 19 lines) --- [Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chansip.c: Sending to 192.168.0.150:5060 (no NAT) [Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chansip.c: Using INVITE request as basis request - 76c8f9af1a3a4cc960a1d9d23ed84964@192.168.0.150:5060 [Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chansip.c: Found peer 'openvox' for '+375293204872' from 192.168.0.150:5060 [Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] netsock2.c: == Using SIP RTP CoS mark 5 [Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chansip.c: Found RTP audio format 8 [Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chansip.c: Found RTP audio format 0 [Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chansip.c: Found RTP audio format 3 [Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chansip.c: Found RTP audio format 9 [Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chansip.c: Found RTP audio format 4 [Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chansip.c: Found RTP audio format 111 [Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chansip.c: Found RTP audio format 18 [Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chansip.c: Found RTP audio format 101 [Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chansip.c: Found audio description format PCMA for ID 8 [Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chansip.c: Found audio description format PCMU for ID 0 [Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chansip.c: Found audio description format GSM for ID 3 [Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chansip.c: Found audio description format G722 for ID 9 [Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chansip.c: Found audio description format G723 for ID 4 [Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chansip.c: Found audio description format G726-32 for ID 111 [Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chansip.c: Found audio description format G729 for ID 18 [Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chansip.c: Found audio description format telephone-event for ID 101 [Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chansip.c: Capabilities: us - (gsm|ulaw|alaw), peer - audio=(g723|gsm|ulaw|alaw|g726|g729|g722)/video=(nothing)/text=(not$ [Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chansip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (tel$ [Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chansip.c: Peer audio RTP is at port 192.168.0.150:18710 [Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chansip.c: Looking for +375296022323 in from-gsm (domain 192.168.0.104) [Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chansip.c: listroute: hop: <sip:+375293204872@192.168.0.150:5060> [Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chansip.c: <--- Transmitting (no NAT) to 192.168.0.150:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.150:5060;branch=z9hG4bK49ade539;received=192.168.0.150;rport=5060 From: "+375293204872" <sip:+375293204872@192.168.0.150>;tag=as71d12b59 To: <sip:+375296022323@192.168.0.104:4524> Call-ID: 76c8f9af1a3a4cc960a1d9d23ed84964@192.168.0.150:5060 CSeq: 103 INVITE Server: Asterisk PBX 11.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: <sip:+375296022323@192.168.0.104:4524> Content-Length: 0 <------------> [Aug 13 14:40:46] VERBOSE[20710][C-00000e7f] pbx.c: -- Executing [+375296022323@from-gsm:1] GotoIfTime("SIP/openvox-0000192d", "00:00-23:59|sat-sun|*|*?workinghour$ [Aug 13 14:40:46] VERBOSE[20710][C-00000e7f] pbx.c: -- Executing [+375296022323@from-gsm:2] GotoIfTime("SIP/openvox-0000192d", "18:00-09:00|mon-fri|*|*?workinghour$ [Aug 13 14:40:46] VERBOSE[20710][C-00000e7f] pbx.c: -- Executing [+375296022323@from-gsm:3] Goto("SIP/openvox-0000192d", "otvet3,s,1") in new stack [Aug 13 14:40:46] VERBOSE[20710][C-00000e7f] pbx.c: -- Goto (otvet3,s,1) [Aug 13 14:40:46] VERBOSE[20710][C-00000e7f] pbx.c: -- Executing [s@otvet3:1] Set("SIP/openvox-0000192d", "fname=201508131440-+375293204872-s") in new stack [Aug 13 14:40:46] VERBOSE[20710][C-00000e7f] pbx.c: -- Executing [s@otvet3:2] MixMonitor("SIP/openvox-0000192d", "/var/www/html/callrecords/201508131440-+375293204$ [Aug 13 14:40:46] VERBOSE[20710][C-00000e7f] pbx.c: -- Executing [s@otvet3:3] BackGround("SIP/openvox-0000192d", "/var/lib/asterisk/moh/privetstvie/privetstvie") i$ [Aug 13 14:40:46] VERBOSE[20711][C-00000e7f] appmixmonitor.c: == Begin MixMonitor Recording SIP/openvox-0000192d [Aug 13 14:40:46] VERBOSE[20710][C-00000e7f] chansip.c: Audio is at 10704 [Aug 13 14:40:46] VERBOSE[20710][C-00000e7f] chansip.c: Adding codec 100004 (alaw) to SDP [Aug 13 14:40:46] VERBOSE[20710][C-00000e7f] chansip.c: Adding codec 100003 (ulaw) to SDP [Aug 13 14:40:46] VERBOSE[20710][C-00000e7f] chansip.c: Adding codec 100002 (gsm) to SDP [Aug 13 14:40:46] VERBOSE[20710][C-00000e7f] chansip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 13 14:40:46] VERBOSE[20710][C-00000e7f] chansip.c: <--- Reliably Transmitting (no NAT) to 192.168.0.150:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.150:5060;branch=z9hG4bK49ade539;received=192.168.0.150;rport=5060 From: "+375293204872" <sip:+375293204872@192.168.0.150>;tag=as71d12b59 To: <sip:+375296022323@192.168.0.104:4524>;tag=as5f9b50a2 Call-ID: 76c8f9af1a3a4cc960a1d9d23ed84964@192.168.0.150:5060 CSeq: 103 INVITE Server: Asterisk PBX 11.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: <sip:+375296022323@192.168.0.104:4524> Content-Type: application/sdp Require: timer Content-Length: 277

v=0 o=root 392634 392634 IN IP4 192.168.0.104 s=Asterisk PBX 11.18.0 c=IN IP4 192.168.0.104 t=0 0 m=audio 10704 RTP/AVP 8 0 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv

<------------> [Aug 13 14:40:46] VERBOSE[31702] chansip.c: <--- SIP read from UDP:192.168.0.150:5060 ---> ACK sip:+375296022323@192.168.0.104:4524 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.150:5060;branch=z9hG4bK5bf0957e;rport Max-Forwards: 70 From: "+375293204872" <sip:+375293204872@192.168.0.150>;tag=as71d12b59 To: <sip:+375296022323@192.168.0.104:4524>;tag=as5f9b50a2 Contact: <sip:+375293204872@192.168.0.150:5060> Call-ID: 76c8f9af1a3a4cc960a1d9d23ed84964@192.168.0.150:5060 CSeq: 103 ACK User-Agent: VoxStack Wireless Gateway Content-Length: 0 <-------------> [Aug 13 14:40:46] VERBOSE[31702] chansip.c: --- (10 headers 0 lines) --- [Aug 13 14:40:46] VERBOSE[20710][C-00000e7f] file.c: -- <sip openvox-0000192d=""> Playing '/var/lib/asterisk/moh/privetstvie/privetstvie.slin' (language 'ru') [Aug 13 14:40:46] VERBOSE[31676] manager.c: == HTTP Manager 'amocrm' timed out from 127.0.0.1 [Aug 13 14:40:46] VERBOSE[31702] chan_sip.c: <--- SIP read from UDP:192.168.0.100:54206 --->

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обновил 2015-08-17 12:41:56 +0400

Пропадание звука в звонках с астериск?

Ситуация следующая. Есть сервер с Asterisk 11.18.0. К нему подключен gsm шлюз Openvox. Также создано 3 внутренних абонента, к каждому из которых привязана симкарта на шлюзе. Ситуация следующая, менеджер звонит (либо менеджеру звонят), разговор начинается. Все друг друга хорошо слышат. Но в последнее время стал всё чаще пропадать звук. ТО есть разговор не прерывается, а просто абонент и менеджер не слышит друг друга. И это происходит не всегда. Скажите, как можно отследить, почему пропадает звук о обеих сторон? Может у вас есть какие-либо предположения, что это может быть?

 <--- SIP read from UDP:192.168.0.150:5060 --->

INVITE sip:+375296022323@192.168.0.104:4524 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.150:5060;branch=z9hG4bK233836d7;rport Max-Forwards: 70 From: "+375293204872" <sip:+375293204872@192.168.0.150>;tag=as71d12b59 To: <sip:+375296022323@192.168.0.104:4524> Contact: <sip:+375293204872@192.168.0.150:5060> Call-ID: 76c8f9af1a3a4cc960a1d9d23ed84964@192.168.0.150:5060 CSeq: 102 INVITE User-Agent: VoxStack Wireless Gateway Date: Wed, 25 Feb 1970 00:45:35 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-Best-Codec: alaw Content-Type: application/sdp Content-Length: 434

434

v=0 o=root 615765964 615765964 IN IP4 192.168.0.150 s=VoxStack Wireless Gateway c=IN IP4 192.168.0.150 t=0 0 m=audio 18710 RTP/AVP 8 0 3 9 4 111 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:9 G722/8000 a=rtpmap:4 G723/8000 a=fmtp:4 annexa=no a=rtpmap:111 G726-32/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <-------------> [Aug 13 14:40:46] VERBOSE[31702] chansip.c: chan_sip.c: --- (15 headers 19 lines) --- [Aug 13 14:40:46] VERBOSE[31702] chansip.c: chan_sip.c: Sending to 192.168.0.150:5060 (NAT) [Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chansip.c: chan_sip.c: Sending to 192.168.0.150:5060 (NAT) [Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chansip.c: chan_sip.c: Using INVITE request as basis request - 76c8f9af1a3a4cc960a1d9d23ed84964@192.168.0.150:5060 [Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chansip.c: chan_sip.c: Found peer 'openvox' for '+375293204872' from 192.168.0.150:5060 [Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chansip.c: chan_sip.c: <--- Reliably Transmitting (no NAT) to 192.168.0.150:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.0.150:5060;branch=z9hG4bK233836d7;received=192.168.0.150;rport=5060 From: "+375293204872" <sip:+375293204872@192.168.0.150>;tag=as71d12b59 To: <sip:+375296022323@192.168.0.104:4524>;tag=as378bb9e4 Call-ID: 76c8f9af1a3a4cc960a1d9d23ed84964@192.168.0.150:5060 CSeq: 102 INVITE Server: Asterisk PBX 11.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1edea583" Content-Length: 0

0

<------------> [Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chansip.c: chan_sip.c: Scheduling destruction of SIP dialog '76c8f9af1a3a4cc960a1d9d23ed84964@192.168.0.150:5060' in 6400 ms (Method:$ [Aug 13 14:40:46] VERBOSE[31702] chansip.c: chan_sip.c: <--- SIP read from UDP:192.168.0.150:5060 ---> ACK sip:+375296022323@192.168.0.104:4524 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.150:5060;branch=z9hG4bK233836d7;rport Max-Forwards: 70 From: "+375293204872" <sip:+375293204872@192.168.0.150>;tag=as71d12b59 To: <sip:+375296022323@192.168.0.104:4524>;tag=as378bb9e4 Contact: <sip:+375293204872@192.168.0.150:5060> Call-ID: 76c8f9af1a3a4cc960a1d9d23ed84964@192.168.0.150:5060 CSeq: 102 ACK User-Agent: VoxStack Wireless Gateway Content-Length: 0 <-------------> [Aug 13 14:40:46] VERBOSE[31702] chansip.c: chan_sip.c: --- (10 headers 0 lines) --- [Aug 13 14:40:46] VERBOSE[31702] chansip.c: chan_sip.c: <--- SIP read from UDP:192.168.0.150:5060 ---> INVITE sip:+375296022323@192.168.0.104:4524 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.150:5060;branch=z9hG4bK49ade539;rport Max-Forwards: 70 From: "+375293204872" <sip:+375293204872@192.168.0.150>;tag=as71d12b59 To: <sip:+375296022323@192.168.0.104:4524> Contact: <sip:+375293204872@192.168.0.150:5060> Call-ID: 76c8f9af1a3a4cc960a1d9d23ed84964@192.168.0.150:5060 CSeq: 103 INVITE User-Agent: VoxStack Wireless Gateway Authorization: Digest username="openvox", realm="asterisk", algorithm=MD5, uri="sip:+375296022323@192.168.0.104:4524", nonce="1edea583", response="a9b79f406c3e3acd7088$ Date: Wed, 25 Feb 1970 00:45:35 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-Best-Codec: alaw Content-Type: application/sdp Content-Length: 434

434

v=0 o=root 615765964 615765965 IN IP4 192.168.0.150 s=VoxStack Wireless Gateway c=IN IP4 192.168.0.150 t=0 0 m=audio 18710 RTP/AVP 8 0 3 9 4 111 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:9 G722/8000 a=rtpmap:4 G723/8000 a=fmtp:4 annexa=no a=rtpmap:111 G726-32/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <-------------> [Aug 13 14:40:46] VERBOSE[31702] chansip.c: chan_sip.c: --- (16 headers 19 lines) --- [Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chansip.c: chan_sip.c: Sending to 192.168.0.150:5060 (no NAT) [Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chansip.c: chan_sip.c: Using INVITE request as basis request - 76c8f9af1a3a4cc960a1d9d23ed84964@192.168.0.150:5060 [Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chansip.c: chan_sip.c: Found peer 'openvox' for '+375293204872' from 192.168.0.150:5060 [Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] netsock2.c: == Using SIP RTP CoS mark 5 [Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chansip.c: chan_sip.c: Found RTP audio format 8 [Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chansip.c: chan_sip.c: Found RTP audio format 0 [Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chansip.c: chan_sip.c: Found RTP audio format 3 [Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chansip.c: chan_sip.c: Found RTP audio format 9 [Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chansip.c: chan_sip.c: Found RTP audio format 4 [Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chansip.c: chan_sip.c: Found RTP audio format 111 [Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chansip.c: chan_sip.c: Found RTP audio format 18 [Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chansip.c: chan_sip.c: Found RTP audio format 101 [Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chansip.c: chan_sip.c: Found audio description format PCMA for ID 8 [Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chansip.c: chan_sip.c: Found audio description format PCMU for ID 0 [Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chansip.c: chan_sip.c: Found audio description format GSM for ID 3 [Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chansip.c: chan_sip.c: Found audio description format G722 for ID 9 [Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chansip.c: chan_sip.c: Found audio description format G723 for ID 4 [Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chansip.c: chan_sip.c: Found audio description format G726-32 for ID 111 [Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chansip.c: chan_sip.c: Found audio description format G729 for ID 18 [Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chansip.c: chan_sip.c: Found audio description format telephone-event for ID 101 [Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chansip.c: chan_sip.c: Capabilities: us - (gsm|ulaw|alaw), peer - audio=(g723|gsm|ulaw|alaw|g726|g729|g722)/video=(nothing)/text=(not$ [Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chansip.c: chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (tel$ [Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chansip.c: chan_sip.c: Peer audio RTP is at port 192.168.0.150:18710 [Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chansip.c: chan_sip.c: Looking for +375296022323 in from-gsm (domain 192.168.0.104) [Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chansip.c: listroute: chan_sip.c: list_route: hop: <sip:+375293204872@192.168.0.150:5060> [Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chansip.c: chan_sip.c: <--- Transmitting (no NAT) to 192.168.0.150:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.150:5060;branch=z9hG4bK49ade539;received=192.168.0.150;rport=5060 From: "+375293204872" <sip:+375293204872@192.168.0.150>;tag=as71d12b59 To: <sip:+375296022323@192.168.0.104:4524> Call-ID: 76c8f9af1a3a4cc960a1d9d23ed84964@192.168.0.150:5060 CSeq: 103 INVITE Server: Asterisk PBX 11.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: <sip:+375296022323@192.168.0.104:4524> Content-Length: 0 <------------> [Aug 13 14:40:46] VERBOSE[20710][C-00000e7f] pbx.c: -- Executing [+375296022323@from-gsm:1] GotoIfTime("SIP/openvox-0000192d", "00:00-23:59|sat-sun|*|*?workinghour$ [Aug 13 14:40:46] VERBOSE[20710][C-00000e7f] pbx.c: -- Executing [+375296022323@from-gsm:2] GotoIfTime("SIP/openvox-0000192d", "18:00-09:00|mon-fri|*|*?workinghour$ [Aug 13 14:40:46] VERBOSE[20710][C-00000e7f] pbx.c: -- Executing [+375296022323@from-gsm:3] Goto("SIP/openvox-0000192d", "otvet3,s,1") in new stack [Aug 13 14:40:46] VERBOSE[20710][C-00000e7f] pbx.c: -- Goto (otvet3,s,1) [Aug 13 14:40:46] VERBOSE[20710][C-00000e7f] pbx.c: -- Executing [s@otvet3:1] Set("SIP/openvox-0000192d", "fname=201508131440-+375293204872-s") in new stack [Aug 13 14:40:46] VERBOSE[20710][C-00000e7f] pbx.c: -- Executing [s@otvet3:2] MixMonitor("SIP/openvox-0000192d", "/var/www/html/callrecords/201508131440-+375293204$ [Aug 13 14:40:46] VERBOSE[20710][C-00000e7f] pbx.c: -- Executing [s@otvet3:3] BackGround("SIP/openvox-0000192d", "/var/lib/asterisk/moh/privetstvie/privetstvie") i$ [Aug 13 14:40:46] VERBOSE[20711][C-00000e7f] appmixmonitor.c: app_mixmonitor.c: == Begin MixMonitor Recording SIP/openvox-0000192d [Aug 13 14:40:46] VERBOSE[20710][C-00000e7f] chansip.c: chan_sip.c: Audio is at 10704 [Aug 13 14:40:46] VERBOSE[20710][C-00000e7f] chansip.c: chan_sip.c: Adding codec 100004 (alaw) to SDP [Aug 13 14:40:46] VERBOSE[20710][C-00000e7f] chansip.c: chan_sip.c: Adding codec 100003 (ulaw) to SDP [Aug 13 14:40:46] VERBOSE[20710][C-00000e7f] chansip.c: chan_sip.c: Adding codec 100002 (gsm) to SDP [Aug 13 14:40:46] VERBOSE[20710][C-00000e7f] chansip.c: chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 13 14:40:46] VERBOSE[20710][C-00000e7f] chansip.c: chan_sip.c: <--- Reliably Transmitting (no NAT) to 192.168.0.150:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.150:5060;branch=z9hG4bK49ade539;received=192.168.0.150;rport=5060 From: "+375293204872" <sip:+375293204872@192.168.0.150>;tag=as71d12b59 To: <sip:+375296022323@192.168.0.104:4524>;tag=as5f9b50a2 Call-ID: 76c8f9af1a3a4cc960a1d9d23ed84964@192.168.0.150:5060 CSeq: 103 INVITE Server: Asterisk PBX 11.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: <sip:+375296022323@192.168.0.104:4524> Content-Type: application/sdp Require: timer Content-Length: 277

277

v=0 o=root 392634 392634 IN IP4 192.168.0.104 s=Asterisk PBX 11.18.0 c=IN IP4 192.168.0.104 t=0 0 m=audio 10704 RTP/AVP 8 0 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv

a=sendrecv

<------------> [Aug 13 14:40:46] VERBOSE[31702] chansip.c: chan_sip.c: <--- SIP read from UDP:192.168.0.150:5060 ---> ACK sip:+375296022323@192.168.0.104:4524 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.150:5060;branch=z9hG4bK5bf0957e;rport Max-Forwards: 70 From: "+375293204872" <sip:+375293204872@192.168.0.150>;tag=as71d12b59 To: <sip:+375296022323@192.168.0.104:4524>;tag=as5f9b50a2 Contact: <sip:+375293204872@192.168.0.150:5060> Call-ID: 76c8f9af1a3a4cc960a1d9d23ed84964@192.168.0.150:5060 CSeq: 103 ACK User-Agent: VoxStack Wireless Gateway Content-Length: 0 <-------------> [Aug 13 14:40:46] VERBOSE[31702] chansip.c: chan_sip.c: --- (10 headers 0 lines) --- [Aug 13 14:40:46] VERBOSE[20710][C-00000e7f] file.c: -- <sip openvox-0000192d=""> <SIP/openvox-0000192d> Playing '/var/lib/asterisk/moh/privetstvie/privetstvie.slin' (language 'ru') [Aug 13 14:40:46] VERBOSE[31676] manager.c: == HTTP Manager 'amocrm' timed out from 127.0.0.1 [Aug 13 14:40:46] VERBOSE[31702] chan_sip.c: <--- SIP read from UDP:192.168.0.100:54206 --->

--->
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обновил 2015-08-17 12:41:56 +0400

Пропадание звука в звонках с астериск?

Ситуация следующая. Есть сервер с Asterisk 11.18.0. К нему подключен gsm шлюз Openvox. Также создано 3 внутренних абонента, к каждому из которых привязана симкарта на шлюзе. Ситуация следующая, менеджер звонит (либо менеджеру звонят), разговор начинается. Все друг друга хорошо слышат. Но в последнее время стал всё чаще пропадать звук. ТО есть разговор не прерывается, а просто абонент и менеджер не слышит друг друга. И это происходит не всегда. Скажите, как можно отследить, почему пропадает звук о обеих сторон? Может у вас есть какие-либо предположения, что это может быть?

    <--- SIP read from UDP:192.168.0.150:5060 --->
INVITE sip:+375296022323@192.168.0.104:4524 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.150:5060;branch=z9hG4bK233836d7;rport
Max-Forwards: 70
From: "+375293204872" <sip:+375293204872@192.168.0.150>;tag=as71d12b59
To: <sip:+375296022323@192.168.0.104:4524>
Contact: <sip:+375293204872@192.168.0.150:5060>
Call-ID: 76c8f9af1a3a4cc960a1d9d23ed84964@192.168.0.150:5060
CSeq: 102 INVITE
User-Agent: VoxStack Wireless Gateway
Date: Wed, 25 Feb 1970 00:45:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-Best-Codec: alaw
Content-Type: application/sdp
Content-Length: 434

v=0
o=root 615765964 615765964 IN IP4 192.168.0.150
s=VoxStack Wireless Gateway
c=IN IP4 192.168.0.150
t=0 0
m=audio 18710 RTP/AVP 8 0 3 9 4 111 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:9 G722/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:111 G726-32/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
[Aug 13 14:40:46] VERBOSE[31702] chan_sip.c: --- (15 headers 19 lines) ---
[Aug 13 14:40:46] VERBOSE[31702] chan_sip.c: Sending to 192.168.0.150:5060 (NAT)
[Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chan_sip.c: Sending to 192.168.0.150:5060 (NAT)
[Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chan_sip.c: Using INVITE request as basis request - 76c8f9af1a3a4cc960a1d9d23ed84964@192.168.0.150:5060
[Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chan_sip.c: Found peer 'openvox' for '+375293204872' from 192.168.0.150:5060
[Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chan_sip.c:
<--- Reliably Transmitting (no NAT) to 192.168.0.150:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.150:5060;branch=z9hG4bK233836d7;received=192.168.0.150;rport=5060
From: "+375293204872" <sip:+375293204872@192.168.0.150>;tag=as71d12b59
To: <sip:+375296022323@192.168.0.104:4524>;tag=as378bb9e4
Call-ID: 76c8f9af1a3a4cc960a1d9d23ed84964@192.168.0.150:5060
CSeq: 102 INVITE
Server: Asterisk PBX 11.18.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1edea583"
Content-Length: 0


<------------>
[Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chan_sip.c: Scheduling destruction of SIP dialog '76c8f9af1a3a4cc960a1d9d23ed84964@192.168.0.150:5060' in 6400 ms (Method:$
[Aug 13 14:40:46] VERBOSE[31702] chan_sip.c:
<--- SIP read from UDP:192.168.0.150:5060 --->
ACK sip:+375296022323@192.168.0.104:4524 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.150:5060;branch=z9hG4bK233836d7;rport
Max-Forwards: 70
From: "+375293204872" <sip:+375293204872@192.168.0.150>;tag=as71d12b59
To: <sip:+375296022323@192.168.0.104:4524>;tag=as378bb9e4
Contact: <sip:+375293204872@192.168.0.150:5060>
Call-ID: 76c8f9af1a3a4cc960a1d9d23ed84964@192.168.0.150:5060
CSeq: 102 ACK
User-Agent: VoxStack Wireless Gateway
Content-Length: 0
<------------->
[Aug 13 14:40:46] VERBOSE[31702] chan_sip.c: --- (10 headers 0 lines) ---
[Aug 13 14:40:46] VERBOSE[31702] chan_sip.c:
<--- SIP read from UDP:192.168.0.150:5060 --->
INVITE sip:+375296022323@192.168.0.104:4524 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.150:5060;branch=z9hG4bK49ade539;rport
Max-Forwards: 70
From: "+375293204872" <sip:+375293204872@192.168.0.150>;tag=as71d12b59
To: <sip:+375296022323@192.168.0.104:4524>
Contact: <sip:+375293204872@192.168.0.150:5060>
Call-ID: 76c8f9af1a3a4cc960a1d9d23ed84964@192.168.0.150:5060
CSeq: 103 INVITE
User-Agent: VoxStack Wireless Gateway
Authorization: Digest username="openvox", realm="asterisk", algorithm=MD5, uri="sip:+375296022323@192.168.0.104:4524", nonce="1edea583", response="a9b79f406c3e3acd7088$
Date: Wed, 25 Feb 1970 00:45:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-Best-Codec: alaw
Content-Type: application/sdp
Content-Length: 434

v=0
o=root 615765964 615765965 IN IP4 192.168.0.150
s=VoxStack Wireless Gateway
c=IN IP4 192.168.0.150
t=0 0
m=audio 18710 RTP/AVP 8 0 3 9 4 111 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:9 G722/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:111 G726-32/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
[Aug 13 14:40:46] VERBOSE[31702] chan_sip.c: --- (16 headers 19 lines) ---
[Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chan_sip.c: Sending to 192.168.0.150:5060 (no NAT)
[Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chan_sip.c: Using INVITE request as basis request - 76c8f9af1a3a4cc960a1d9d23ed84964@192.168.0.150:5060
[Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chan_sip.c: Found peer 'openvox' for '+375293204872' from 192.168.0.150:5060
[Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] netsock2.c:   == Using SIP RTP CoS mark 5
[Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chan_sip.c: Found RTP audio format 8
[Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chan_sip.c: Found RTP audio format 0
[Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chan_sip.c: Found RTP audio format 3
[Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chan_sip.c: Found RTP audio format 9
[Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chan_sip.c: Found RTP audio format 4
[Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chan_sip.c: Found RTP audio format 111
[Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chan_sip.c: Found RTP audio format 18
[Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chan_sip.c: Found RTP audio format 101
[Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chan_sip.c: Found audio description format PCMA for ID 8
[Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chan_sip.c: Found audio description format PCMU for ID 0
[Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chan_sip.c: Found audio description format GSM for ID 3
[Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chan_sip.c: Found audio description format G722 for ID 9
[Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chan_sip.c: Found audio description format G723 for ID 4
[Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chan_sip.c: Found audio description format G726-32 for ID 111
[Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chan_sip.c: Found audio description format G729 for ID 18
[Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chan_sip.c: Found audio description format telephone-event for ID 101
[Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chan_sip.c: Capabilities: us - (gsm|ulaw|alaw), peer - audio=(g723|gsm|ulaw|alaw|g726|g729|g722)/video=(nothing)/text=(not$
[Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (tel$
[Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chan_sip.c: Peer audio RTP is at port 192.168.0.150:18710
[Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chan_sip.c: Looking for +375296022323 in from-gsm (domain 192.168.0.104)
[Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chan_sip.c: list_route: hop: <sip:+375293204872@192.168.0.150:5060>
[Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chan_sip.c:
<--- Transmitting (no NAT) to 192.168.0.150:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.150:5060;branch=z9hG4bK49ade539;received=192.168.0.150;rport=5060
From: "+375293204872" <sip:+375293204872@192.168.0.150>;tag=as71d12b59
To: <sip:+375296022323@192.168.0.104:4524>
Call-ID: 76c8f9af1a3a4cc960a1d9d23ed84964@192.168.0.150:5060
CSeq: 103 INVITE
Server: Asterisk PBX 11.18.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:+375296022323@192.168.0.104:4524>
Content-Length: 0
<------------>
[Aug 13 14:40:46] VERBOSE[20710][C-00000e7f] pbx.c:     -- Executing [+375296022323@from-gsm:1] GotoIfTime("SIP/openvox-0000192d", "00:00-23:59|sat-sun|*|*?workinghour$
[Aug 13 14:40:46] VERBOSE[20710][C-00000e7f] pbx.c:     -- Executing [+375296022323@from-gsm:2] GotoIfTime("SIP/openvox-0000192d", "18:00-09:00|mon-fri|*|*?workinghour$
[Aug 13 14:40:46] VERBOSE[20710][C-00000e7f] pbx.c:     -- Executing [+375296022323@from-gsm:3] Goto("SIP/openvox-0000192d", "otvet3,s,1") in new stack
[Aug 13 14:40:46] VERBOSE[20710][C-00000e7f] pbx.c:     -- Goto (otvet3,s,1)
[Aug 13 14:40:46] VERBOSE[20710][C-00000e7f] pbx.c:     -- Executing [s@otvet3:1] Set("SIP/openvox-0000192d", "fname=201508131440-+375293204872-s") in new stack
[Aug 13 14:40:46] VERBOSE[20710][C-00000e7f] pbx.c:     -- Executing [s@otvet3:2] MixMonitor("SIP/openvox-0000192d", "/var/www/html/callrecords/201508131440-+375293204$
[Aug 13 14:40:46] VERBOSE[20710][C-00000e7f] pbx.c:     -- Executing [s@otvet3:3] BackGround("SIP/openvox-0000192d", "/var/lib/asterisk/moh/privetstvie/privetstvie") i$
[Aug 13 14:40:46] VERBOSE[20711][C-00000e7f] app_mixmonitor.c:   == Begin MixMonitor Recording SIP/openvox-0000192d
[Aug 13 14:40:46] VERBOSE[20710][C-00000e7f] chan_sip.c: Audio is at 10704
[Aug 13 14:40:46] VERBOSE[20710][C-00000e7f] chan_sip.c: Adding codec 100004 (alaw) to SDP
[Aug 13 14:40:46] VERBOSE[20710][C-00000e7f] chan_sip.c: Adding codec 100003 (ulaw) to SDP
[Aug 13 14:40:46] VERBOSE[20710][C-00000e7f] chan_sip.c: Adding codec 100002 (gsm) to SDP
[Aug 13 14:40:46] VERBOSE[20710][C-00000e7f] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[Aug 13 14:40:46] VERBOSE[20710][C-00000e7f] chan_sip.c:
<--- Reliably Transmitting (no NAT) to 192.168.0.150:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.150:5060;branch=z9hG4bK49ade539;received=192.168.0.150;rport=5060
From: "+375293204872" <sip:+375293204872@192.168.0.150>;tag=as71d12b59
To: <sip:+375296022323@192.168.0.104:4524>;tag=as5f9b50a2
Call-ID: 76c8f9af1a3a4cc960a1d9d23ed84964@192.168.0.150:5060
CSeq: 103 INVITE
Server: Asterisk PBX 11.18.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:+375296022323@192.168.0.104:4524>
Content-Type: application/sdp
Require: timer
Content-Length: 277

v=0
o=root 392634 392634 IN IP4 192.168.0.104
s=Asterisk PBX 11.18.0
c=IN IP4 192.168.0.104
t=0 0
m=audio 10704 RTP/AVP 8 0 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
[Aug 13 14:40:46] VERBOSE[31702] chan_sip.c:
<--- SIP read from UDP:192.168.0.150:5060 --->
ACK sip:+375296022323@192.168.0.104:4524 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.150:5060;branch=z9hG4bK5bf0957e;rport
Max-Forwards: 70
From: "+375293204872" <sip:+375293204872@192.168.0.150>;tag=as71d12b59
To: <sip:+375296022323@192.168.0.104:4524>;tag=as5f9b50a2
Contact: <sip:+375293204872@192.168.0.150:5060>
Call-ID: 76c8f9af1a3a4cc960a1d9d23ed84964@192.168.0.150:5060
CSeq: 103 ACK
User-Agent: VoxStack Wireless Gateway
Content-Length: 0
<------------->
[Aug 13 14:40:46] VERBOSE[31702] chan_sip.c: --- (10 headers 0 lines) ---
[Aug 13 14:40:46] VERBOSE[20710][C-00000e7f] file.c:     -- <SIP/openvox-0000192d> Playing '/var/lib/asterisk/moh/privetstvie/privetstvie.slin' (language 'ru')
[Aug 13 14:40:46] VERBOSE[31676] manager.c:   == HTTP Manager 'amocrm' timed out from 127.0.0.1
[Aug 13 14:40:46] VERBOSE[31702] chan_sip.c:
<--- SIP read from UDP:192.168.0.100:54206 --->

Пропадание звука в звонках с астериск?

Ситуация следующая. Есть сервер с Asterisk 11.18.0. К нему подключен gsm шлюз Openvox. Также создано 3 внутренних абонента, к каждому из которых привязана симкарта на шлюзе. Ситуация следующая, менеджер звонит (либо менеджеру звонят), разговор начинается. Все друг друга хорошо слышат. Но в последнее время стал всё чаще пропадать звук. ТО есть разговор не прерывается, а просто абонент и менеджер не слышит друг друга. И это происходит не всегда. Скажите, как можно отследить, почему пропадает звук о обеих сторон? Может у вас есть какие-либо предположения, что это может быть?

    <--- SIP read from UDP:192.168.0.150:5060 --->
INVITE sip:+375296022323@192.168.0.104:4524 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.150:5060;branch=z9hG4bK233836d7;rport
Max-Forwards: 70
From: "+375293204872" <sip:+375293204872@192.168.0.150>;tag=as71d12b59
To: <sip:+375296022323@192.168.0.104:4524>
Contact: <sip:+375293204872@192.168.0.150:5060>
Call-ID: 76c8f9af1a3a4cc960a1d9d23ed84964@192.168.0.150:5060
CSeq: 102 INVITE
User-Agent: VoxStack Wireless Gateway
Date: Wed, 25 Feb 1970 00:45:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-Best-Codec: alaw
Content-Type: application/sdp
Content-Length: 434

v=0
o=root 615765964 615765964 IN IP4 192.168.0.150
s=VoxStack Wireless Gateway
c=IN IP4 192.168.0.150
t=0 0
m=audio 18710 RTP/AVP 8 0 3 9 4 111 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:9 G722/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:111 G726-32/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
[Aug 13 14:40:46] VERBOSE[31702] chan_sip.c: --- (15 headers 19 lines) ---
[Aug 13 14:40:46] VERBOSE[31702] chan_sip.c: Sending to 192.168.0.150:5060 (NAT)
[Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chan_sip.c: Sending to 192.168.0.150:5060 (NAT)
[Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chan_sip.c: Using INVITE request as basis request - 76c8f9af1a3a4cc960a1d9d23ed84964@192.168.0.150:5060
[Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chan_sip.c: Found peer 'openvox' for '+375293204872' from 192.168.0.150:5060
[Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chan_sip.c:
<--- Reliably Transmitting (no NAT) to 192.168.0.150:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.150:5060;branch=z9hG4bK233836d7;received=192.168.0.150;rport=5060
From: "+375293204872" <sip:+375293204872@192.168.0.150>;tag=as71d12b59
To: <sip:+375296022323@192.168.0.104:4524>;tag=as378bb9e4
Call-ID: 76c8f9af1a3a4cc960a1d9d23ed84964@192.168.0.150:5060
CSeq: 102 INVITE
Server: Asterisk PBX 11.18.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1edea583"
Content-Length: 0


<------------>
[Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chan_sip.c: Scheduling destruction of SIP dialog '76c8f9af1a3a4cc960a1d9d23ed84964@192.168.0.150:5060' in 6400 ms (Method:$
[Aug 13 14:40:46] VERBOSE[31702] chan_sip.c:
<--- SIP read from UDP:192.168.0.150:5060 --->
ACK sip:+375296022323@192.168.0.104:4524 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.150:5060;branch=z9hG4bK233836d7;rport
Max-Forwards: 70
From: "+375293204872" <sip:+375293204872@192.168.0.150>;tag=as71d12b59
To: <sip:+375296022323@192.168.0.104:4524>;tag=as378bb9e4
Contact: <sip:+375293204872@192.168.0.150:5060>
Call-ID: 76c8f9af1a3a4cc960a1d9d23ed84964@192.168.0.150:5060
CSeq: 102 ACK
User-Agent: VoxStack Wireless Gateway
Content-Length: 0
<------------->
[Aug 13 14:40:46] VERBOSE[31702] chan_sip.c: --- (10 headers 0 lines) ---
[Aug 13 14:40:46] VERBOSE[31702] chan_sip.c:
<--- SIP read from UDP:192.168.0.150:5060 --->
INVITE sip:+375296022323@192.168.0.104:4524 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.150:5060;branch=z9hG4bK49ade539;rport
Max-Forwards: 70
From: "+375293204872" <sip:+375293204872@192.168.0.150>;tag=as71d12b59
To: <sip:+375296022323@192.168.0.104:4524>
Contact: <sip:+375293204872@192.168.0.150:5060>
Call-ID: 76c8f9af1a3a4cc960a1d9d23ed84964@192.168.0.150:5060
CSeq: 103 INVITE
User-Agent: VoxStack Wireless Gateway
Authorization: Digest username="openvox", realm="asterisk", algorithm=MD5, uri="sip:+375296022323@192.168.0.104:4524", nonce="1edea583", response="a9b79f406c3e3acd7088$
Date: Wed, 25 Feb 1970 00:45:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-Best-Codec: alaw
Content-Type: application/sdp
Content-Length: 434

v=0
o=root 615765964 615765965 IN IP4 192.168.0.150
s=VoxStack Wireless Gateway
c=IN IP4 192.168.0.150
t=0 0
m=audio 18710 RTP/AVP 8 0 3 9 4 111 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:9 G722/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:111 G726-32/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
[Aug 13 14:40:46] VERBOSE[31702] chan_sip.c: --- (16 headers 19 lines) ---
[Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chan_sip.c: Sending to 192.168.0.150:5060 (no NAT)
[Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chan_sip.c: Using INVITE request as basis request - 76c8f9af1a3a4cc960a1d9d23ed84964@192.168.0.150:5060
[Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chan_sip.c: Found peer 'openvox' for '+375293204872' from 192.168.0.150:5060
[Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] netsock2.c:   == Using SIP RTP CoS mark 5
[Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chan_sip.c: Found RTP audio format 8
[Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chan_sip.c: Found RTP audio format 0
[Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chan_sip.c: Found RTP audio format 3
[Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chan_sip.c: Found RTP audio format 9
[Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chan_sip.c: Found RTP audio format 4
[Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chan_sip.c: Found RTP audio format 111
[Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chan_sip.c: Found RTP audio format 18
[Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chan_sip.c: Found RTP audio format 101
[Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chan_sip.c: Found audio description format PCMA for ID 8
[Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chan_sip.c: Found audio description format PCMU for ID 0
[Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chan_sip.c: Found audio description format GSM for ID 3
[Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chan_sip.c: Found audio description format G722 for ID 9
[Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chan_sip.c: Found audio description format G723 for ID 4
[Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chan_sip.c: Found audio description format G726-32 for ID 111
[Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chan_sip.c: Found audio description format G729 for ID 18
[Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chan_sip.c: Found audio description format telephone-event for ID 101
[Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chan_sip.c: Capabilities: us - (gsm|ulaw|alaw), peer - audio=(g723|gsm|ulaw|alaw|g726|g729|g722)/video=(nothing)/text=(not$
[Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (tel$
[Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chan_sip.c: Peer audio RTP is at port 192.168.0.150:18710
[Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chan_sip.c: Looking for +375296022323 in from-gsm (domain 192.168.0.104)
[Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chan_sip.c: list_route: hop: <sip:+375293204872@192.168.0.150:5060>
[Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chan_sip.c:
<--- Transmitting (no NAT) to 192.168.0.150:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.150:5060;branch=z9hG4bK49ade539;received=192.168.0.150;rport=5060
From: "+375293204872" <sip:+375293204872@192.168.0.150>;tag=as71d12b59
To: <sip:+375296022323@192.168.0.104:4524>
Call-ID: 76c8f9af1a3a4cc960a1d9d23ed84964@192.168.0.150:5060
CSeq: 103 INVITE
Server: Asterisk PBX 11.18.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:+375296022323@192.168.0.104:4524>
Content-Length: 0
<------------>
[Aug 13 14:40:46] VERBOSE[20710][C-00000e7f] pbx.c:     -- Executing [+375296022323@from-gsm:1] GotoIfTime("SIP/openvox-0000192d", "00:00-23:59|sat-sun|*|*?workinghour$
[Aug 13 14:40:46] VERBOSE[20710][C-00000e7f] pbx.c:     -- Executing [+375296022323@from-gsm:2] GotoIfTime("SIP/openvox-0000192d", "18:00-09:00|mon-fri|*|*?workinghour$
[Aug 13 14:40:46] VERBOSE[20710][C-00000e7f] pbx.c:     -- Executing [+375296022323@from-gsm:3] Goto("SIP/openvox-0000192d", "otvet3,s,1") in new stack
[Aug 13 14:40:46] VERBOSE[20710][C-00000e7f] pbx.c:     -- Goto (otvet3,s,1)
[Aug 13 14:40:46] VERBOSE[20710][C-00000e7f] pbx.c:     -- Executing [s@otvet3:1] Set("SIP/openvox-0000192d", "fname=201508131440-+375293204872-s") in new stack
[Aug 13 14:40:46] VERBOSE[20710][C-00000e7f] pbx.c:     -- Executing [s@otvet3:2] MixMonitor("SIP/openvox-0000192d", "/var/www/html/callrecords/201508131440-+375293204$
[Aug 13 14:40:46] VERBOSE[20710][C-00000e7f] pbx.c:     -- Executing [s@otvet3:3] BackGround("SIP/openvox-0000192d", "/var/lib/asterisk/moh/privetstvie/privetstvie") i$
[Aug 13 14:40:46] VERBOSE[20711][C-00000e7f] app_mixmonitor.c:   == Begin MixMonitor Recording SIP/openvox-0000192d
[Aug 13 14:40:46] VERBOSE[20710][C-00000e7f] chan_sip.c: Audio is at 10704
[Aug 13 14:40:46] VERBOSE[20710][C-00000e7f] chan_sip.c: Adding codec 100004 (alaw) to SDP
[Aug 13 14:40:46] VERBOSE[20710][C-00000e7f] chan_sip.c: Adding codec 100003 (ulaw) to SDP
[Aug 13 14:40:46] VERBOSE[20710][C-00000e7f] chan_sip.c: Adding codec 100002 (gsm) to SDP
[Aug 13 14:40:46] VERBOSE[20710][C-00000e7f] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[Aug 13 14:40:46] VERBOSE[20710][C-00000e7f] chan_sip.c:
<--- Reliably Transmitting (no NAT) to 192.168.0.150:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.150:5060;branch=z9hG4bK49ade539;received=192.168.0.150;rport=5060
From: "+375293204872" <sip:+375293204872@192.168.0.150>;tag=as71d12b59
To: <sip:+375296022323@192.168.0.104:4524>;tag=as5f9b50a2
Call-ID: 76c8f9af1a3a4cc960a1d9d23ed84964@192.168.0.150:5060
CSeq: 103 INVITE
Server: Asterisk PBX 11.18.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:+375296022323@192.168.0.104:4524>
Content-Type: application/sdp
Require: timer
Content-Length: 277

v=0
o=root 392634 392634 IN IP4 192.168.0.104
s=Asterisk PBX 11.18.0
c=IN IP4 192.168.0.104
t=0 0
m=audio 10704 RTP/AVP 8 0 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
[Aug 13 14:40:46] VERBOSE[31702] chan_sip.c:
<--- SIP read from UDP:192.168.0.150:5060 --->
ACK sip:+375296022323@192.168.0.104:4524 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.150:5060;branch=z9hG4bK5bf0957e;rport
Max-Forwards: 70
From: "+375293204872" <sip:+375293204872@192.168.0.150>;tag=as71d12b59
To: <sip:+375296022323@192.168.0.104:4524>;tag=as5f9b50a2
Contact: <sip:+375293204872@192.168.0.150:5060>
Call-ID: 76c8f9af1a3a4cc960a1d9d23ed84964@192.168.0.150:5060
CSeq: 103 ACK
User-Agent: VoxStack Wireless Gateway
Content-Length: 0
<------------->
[Aug 13 14:40:46] VERBOSE[31702] chan_sip.c: --- (10 headers 0 lines) ---
[Aug 13 14:40:46] VERBOSE[20710][C-00000e7f] file.c:     -- <SIP/openvox-0000192d> Playing '/var/lib/asterisk/moh/privetstvie/privetstvie.slin' (language 'ru')
[Aug 13 14:40:46] VERBOSE[31676] manager.c:   == HTTP Manager 'amocrm' timed out from 127.0.0.1
[Aug 13 14:40:46] VERBOSE[31702] chan_sip.c:
<--- SIP read from UDP:192.168.0.100:54206 --->

Пропадание звука в звонках с астериск?

Ситуация следующая. Есть сервер с Asterisk 11.18.0. К нему подключен gsm шлюз Openvox. Также создано 3 внутренних абонента, к каждому из которых привязана симкарта на шлюзе. Ситуация следующая, менеджер звонит (либо менеджеру звонят), разговор начинается. Все друг друга хорошо слышат. Но в последнее время стал всё чаще пропадать звук. ТО есть разговор не прерывается, а просто абонент и менеджер не слышит друг друга. И это происходит не всегда. Скажите, как можно отследить, почему пропадает звук о обеих сторон? Может у вас есть какие-либо предположения, что это может быть?

192.168.0.103 - это ip адрес компьютера с софтфоном 192.168.0.104 - это ip адрес сервера с астериск 192.168.0.150 - это ip адрес gsm шлюза openvox Проанализировал, вот что мне вывело: 1 рисунок - обведен звонок, который сорвался image description 2 картинка - диаграмма первого выбранного файла image description 3 картинка - диаграмма второго выбранного файла image description

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Проект компании "АТС Дизайн"
Asterisk® и Digium® являются зарегистрированными торговыми марками компании Digium, Inc., США.
IP АТС Asterisk распространяется под лицензией GNU GPL.