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sip нет звука в обе стороны

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звонок идет через сип провайдера с одного asterisk на второй. если звонить на городские телефоны через этого же провайдера голос есть. Обращался к провайдеру, они отвечают у нас все в порядке, в чем я сильно сомневаюсь. лог астериск 1 == Using UDPTL CoS mark 5 == Using SIP RTP CoS mark 5 -- Executing [9zz@from-internal:1] Set("SIP/7926xx-0000003b", "CALLERID(all)="<495yy>"") in new stack -- Executing [9zz@from-internal:2] Dial("SIP/7926xx-0000003b", "SIP/prov/zz,40") in new stack == Using UDPTL CoS mark 5 == Using SIP RTP CoS mark 5 Audio is at 5060 Adding codec 0x8 (alaw) to SDP Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.21.1:5060: INVITE sip:zz@192.168.21.1:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.21.9:5060;branch=z9hG4bK04039ef6 Max-Forwards: 70 From: "495yy" <sip:495yy@194.67.28.139&gt;;tag=as7b3e3b2d to:="" <sip:zz@192.168.21.1:5060&gt;="" contact:="" <sip:495yy@192.168.21.9:5060&gt;="" call-id:="" 30dced837ed7dba2014d57510373dd17@194.67.28.139="" cseq:="" 102="" invite="" user-agent:="" asterisk="" pbx="" 1.8.3.2="" date:="" tue,="" 10="" may="" 2011="" 13:52:06="" gmt="" allow:="" invite,="" ack,="" cancel,="" options,="" bye,="" refer,="" subscribe,="" notify,="" info,="" publish="" supported:="" replaces,="" timer="" remote-party-id:="" "495yy"="" <sip:495yy@194.67.28.139&gt;;party="calling;privacy=off;screen=no" content-type:="" application="" sdp="" content-length:="" 260<="" p="">

v=0 o=root 1145872005 1145872005 IN IP4 192.168.21.9 s=Asterisk PBX 1.8.3.2 c=IN IP4 192.168.21.9 t=0 0 m=audio 12110 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv


-- Called prov/zz

<--- SIP read from UDP:192.168.21.1:5060 ---> SIP/2.0 100 Trying Call-ID: 30dced837ed7dba2014d57510373dd17@194.67.28.139 CSeq: 102 INVITE From: "495yy" <sip:495yy@194.67.28.139&gt;;tag=as7b3e3b2d to:="" <sip:zz@192.168.21.1:5060&gt;;tag="sbc+1+12d90008+a79e4597" via:="" sip="" 2.0="" udp="" 192.168.21.9:5060;received="192.168.21.9;branch=z9hG4bK04039ef6" server:="" cisco-sbc="" 2.x="" content-length:="" 0<="" p="">

<-------------> --- (8 headers 0 lines) ---

<--- SIP read from UDP:192.168.21.1:5060 ---> SIP/2.0 180 Ringing Call-ID: 30dced837ed7dba2014d57510373dd17@194.67.28.139 CSeq: 102 INVITE From: "495yy" <sip:495yy@194.67.28.139&gt;;tag=as7b3e3b2d to:="" <sip:zz@192.168.21.1:5060&gt;;tag="sbc+1+12d90008+a79e4597" via:="" sip="" 2.0="" udp="" 192.168.21.9:5060;received="192.168.21.9;branch=z9hG4bK04039ef6" content-length:="" 0="" contact:="" <sip:192.168.21.1:5060&gt;="" server:="" cs2000_ngss="" 9.0<="" p="">

<-------------> --- (9 headers 0 lines) --- -- SIP/prov-0000003c is ringing

<--- SIP read from UDP:192.168.21.1:5060 ---> SIP/2.0 183 Session Progress Call-ID: 30dced837ed7dba2014d57510373dd17@194.67.28.139 CSeq: 102 INVITE From: "495yy" <sip:495yy@194.67.28.139&gt;;tag=as7b3e3b2d to:="" <sip:zz@192.168.21.1:5060&gt;;tag="sbc+1+12d90008+a79e4597" via:="" sip="" 2.0="" udp="" 192.168.21.9:5060;received="192.168.21.9;branch=z9hG4bK04039ef6" content-length:="" 266="" contact:="" <sip:192.168.21.1:5060&gt;="" content-type:="" application="" sdp="" server:="" cs2000_ngss="" 9.0<="" p="">

v=0 o=root 127952923454121 127952923454121 IN IP4 192.168.21.1 s=- c=IN IP4 192.168.21.1 t=0 0 m=audio 21696 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 <-------------> --- (10 headers 12 lines) --- Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 101 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found audio description format telephone-event for ID 101 Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 192.168.21.1:21696 -- SIP/prov-0000003c is making progress passing it to SIP/7926xx-0000003b

<--- SIP read from UDP:192.168.21.1:5060 ---> SIP/2.0 200 OK Call-ID: 30dced837ed7dba2014d57510373dd17@194.67.28.139 CSeq: 102 INVITE From: "495yy" <sip:495yy@194.67.28.139&gt;;tag=as7b3e3b2d to:="" <sip:zz@192.168.21.1:5060&gt;;tag="sbc+1+12d90008+a79e4597" via:="" sip="" 2.0="" udp="" 192.168.21.9:5060;received="192.168.21.9;branch=z9hG4bK04039ef6" content-length:="" 266="" contact:="" <sip:192.168.21.1:5060&gt;="" content-type:="" application="" sdp="" server:="" cs2000_ngss="" 9.0<="" p="">

v=0 o=root 127952923454121 127952923454121 IN IP4 192.168.21.1 s=- c=IN IP4 192.168.21.1 t=0 0 m=audio 21696 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 <-------------> --- (10 headers 12 lines) --- listroute: hop: <sip:192.168.21.1:5060&gt; set<="" em="">destination: Parsing <sip:192.168.21.1:5060&gt; for="" address="" port="" to="" send="" to="" set_destination:="" set="" destination="" to="" 192.168.21.1:5060="" transmitting="" (no="" nat)="" to="" 192.168.21.1:5060:="" ack="" sip:192.168.21.1:5060="" sip="" 2.0="" via:="" sip="" 2.0="" udp="" 192.168.21.9:5060;branch="z9hG4bK422688d2" max-forwards:="" 70="" from:="" "495yy"="" <sip:495yy@194.67.28.139&gt;;tag="as7b3e3b2d" to:="" <sip:zz@192.168.21.1:5060&gt;;tag="sbc+1+12d90008+a79e4597" contact:="" <sip:495yy@192.168.21.9:5060&gt;="" call-id:="" 30dced837ed7dba2014d57510373dd17@194.67.28.139="" cseq:="" 102="" ack="" user-agent:="" asterisk="" pbx="" 1.8.3.2="" content-length:="" 0<="" p="">


-- SIP/prov-0000003c answered SIP/7926xx-0000003b
-- Locally bridging SIP/7926xx-0000003b and SIP/prov-0000003c

<--- SIP read from UDP:192.168.21.1:5060 ---> BYE sip:495yy@192.168.21.9:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.21.1:5060;branch=z9hG4bK+040c457c794316aaf876c95864d4f2bc+sbc+1 Call-ID: 30dced837ed7dba2014d57510373dd17@194.67.28.139 From: <sip:zz@192.168.21.1:5060&gt;;tag=sbc+1+12d90008+a79e4597 to:="" "495yy"="" <sip:495yy@194.67.28.139&gt;;tag="as7b3e3b2d" cseq:="" 14340794="" bye="" content-length:="" 0="" supported:="" 100rel="" max-forwards:="" 69<="" p="">

<-------------> --- (9 headers 0 lines) --- Sending to 192.168.21.1:5060 (no NAT) Scheduling destruction of SIP dialog '30dced837ed7dba2014d57510373dd17@194.67.28.139' in 6400 ms (Method: BYE)

<--- Transmitting (no NAT) to 192.168.21.1:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.21.1:5060;branch=z9hG4bK+040c457c794316aaf876c95864d4f2bc+sbc+1;received=192.168.21.1 From: <sip:zz@192.168.21.1:5060&gt;;tag=sbc+1+12d90008+a79e4597 to:="" "495yy"="" <sip:495yy@194.67.28.139&gt;;tag="as7b3e3b2d" call-id:="" 30dced837ed7dba2014d57510373dd17@194.67.28.139="" cseq:="" 14340794="" bye="" server:="" asterisk="" pbx="" 1.8.3.2="" allow:="" invite,="" ack,="" cancel,="" options,="" bye,="" refer,="" subscribe,="" notify,="" info,="" publish="" supported:="" replaces,="" timer="" content-length:="" 0<="" p="">

<------------> == Spawn extension (from-internal, 9zz, 2) exited non-zero on 'SIP/7926xx-0000003b'

лог астериск 2 <--- SIP read from UDP:192.168.20.1:5060 ---> INVITE sip:4955439690@192.168.20.9:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.20.1:5060;branch=z9hG4bK+c4103e3f6eafcc9b0d5f35c30eab3af9+sbc+1 Supported: 100rel From: <sip:495yy@192.168.20.1:5060&gt;;tag=192.168.20.1+1+2abdd2af+d6a87167 to:="" <sip:4955439690@192.168.20.9:5060&gt;="" cseq:="" 1="" invite="" expires:="" 180="" content-length:="" 237="" call-info:="" <sip:192.168.20.1:5060&gt;;method="NOTIFY;Event=telephone-event;Duration=2000" contact:="" <sip:192.168.20.1:5060&gt;="" content-type:="" application="" sdp="" call-id:="" c30ee825681d114eb9edef0f6ee730d3@192.168.20.1="" max-forwards:="" 55="" accept:="" application="" sdp,="" application="" dtmf-relay<="" p="">

v=0 o=root 29937474789401 29937474789401 IN IP4 192.168.20.1 s=- c=IN IP4 192.168.20.1 t=0 0 m=audio 26738 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20

<-------------> --- (14 headers 11 lines) --- Sending to 192.168.20.1 : 5060 (no NAT) Using INVITE request as basis request - c30ee825681d114eb9edef0f6ee730d3@192.168.20.1 Found peer 'prov' for '495yy' from 192.168.20.1:5060 Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 101 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.20.1:26738 Looking for 4955439690 in from-internal (domain 192.168.20.9) list_route: hop: <sip:192.168.20.1:5060&gt;< p="">

<--- Transmitting (no NAT) to 192.168.20.1:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.20.1:5060;branch=z9hG4bK+c4103e3f6eafcc9b0d5f35c30eab3af9+sbc+1;received=192.168.20.1 From: <sip:495yy@192.168.20.1:5060&gt;;tag=192.168.20.1+1+2abdd2af+d6a87167 to:="" <sip:4955439690@192.168.20.9:5060&gt;="" call-id:="" c30ee825681d114eb9edef0f6ee730d3@192.168.20.1="" cseq:="" 1="" invite="" server:="" asterisk="" pbx="" 1.6.2.5-0ubuntu1.1="" allow:="" invite,="" ack,="" cancel,="" options,="" bye,="" refer,="" subscribe,="" notify,="" info="" supported:="" replaces,="" timer="" contact:="" <sip:4955439690@192.168.20.9&gt;="" content-length:="" 0<="" p="">

<------------> asterisk1*CLI> <--- Transmitting (no NAT) to 192.168.20.1:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.20.1:5060;branch=z9hG4bK+c4103e3f6eafcc9b0d5f35c30eab3af9+sbc+1;received=192.168.20.1 From: <sip:495yy@192.168.20.1:5060&gt;;tag=192.168.20.1+1+2abdd2af+d6a87167 to:="" <sip:4955439690@192.168.20.9:5060&gt;;tag="as4a49900b" call-id:="" c30ee825681d114eb9edef0f6ee730d3@192.168.20.1="" cseq:="" 1="" invite="" server:="" asterisk="" pbx="" 1.6.2.5-0ubuntu1.1="" allow:="" invite,="" ack,="" cancel,="" options,="" bye,="" refer,="" subscribe,="" notify,="" info="" supported:="" replaces,="" timer="" contact:="" <sip:4955439690@192.168.20.9&gt;="" content-length:="" 0<="" p="">

<------------> Audio is at 192.168.20.9 port 17644 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 192.168.20.1:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.20.1:5060;branch=z9hG4bK+c4103e3f6eafcc9b0d5f35c30eab3af9+sbc+1;received=192.168.20.1 From: <sip:495yy@192.168.20.1:5060&gt;;tag=192.168.20.1+1+2abdd2af+d6a87167 to:="" <sip:4955439690@192.168.20.9:5060&gt;;tag="as4a49900b" call-id:="" c30ee825681d114eb9edef0f6ee730d3@192.168.20.1="" cseq:="" 1="" invite="" server:="" asterisk="" pbx="" 1.6.2.5-0ubuntu1.1="" allow:="" invite,="" ack,="" cancel,="" options,="" bye,="" refer,="" subscribe,="" notify,="" info="" supported:="" replaces,="" timer="" contact:="" <sip:4955439690@192.168.20.9&gt;="" content-type:="" application="" sdp="" content-length:="" 298<="" p="">

v=0 o=root 1393445746 1393445746 IN IP4 192.168.20.9 s=Asterisk PBX 1.6.2.5-0ubuntu1.1 c=IN IP4 192.168.20.9 t=0 0 m=audio 17644 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv

<------------> asterisk1*CLI> <--- SIP read from UDP:192.168.20.1:5060 ---> ACK sip:4955439690@192.168.20.9 SIP/2.0 Via: SIP/2.0/UDP 192.168.20.1:5060;branch=z9hG4bK+2efd085d76ca2c37bb3152129559b0aa+sbc+1 Call-ID: c30ee825681d114eb9edef0f6ee730d3@192.168.20.1 From: <sip:495yy@192.168.20.1:5060&gt;;tag=192.168.20.1+1+2abdd2af+d6a87167 to:="" <sip:4955439690@192.168.20.9:5060&gt;;tag="as4a49900b" cseq:="" 1="" ack="" contact:="" <sip:192.168.20.1:5060&gt;="" content-length:="" 0="" supported:="" 100rel="" max-forwards:="" 69<="" p="">

<-------------> --- (10 headers 0 lines) --- Scheduling destruction of SIP dialog 'c30ee825681d114eb9edef0f6ee730d3@192.168.20.1' in 6400 ms (Method: ACK) setdestination: Parsing <sip:192.168.20.1:5060&gt; for="" address="" port="" to="" send="" to="" set<="" em="">destination: set destination to 192.168.20.1, port 5060 Reliably Transmitting (no NAT) to 192.168.20.1:5060: BYE sip:192.168.20.1:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.20.9:5060;branch=z9hG4bK20f61525;rport Max-Forwards: 70 From: <sip:4955439690@192.168.20.9:5060&gt;;tag=as4a49900b to:="" <sip:495yy@192.168.20.1:5060&gt;;tag="192.168.20.1+1+2abdd2af+d6a87167" call-id:="" c30ee825681d114eb9edef0f6ee730d3@192.168.20.1="" cseq:="" 102="" bye="" user-agent:="" asterisk="" pbx="" 1.6.2.5-0ubuntu1.1="" x-asterisk-hangupcause:="" normal="" clearing="" x-asterisk-hangupcausecode:="" 16="" content-length:="" 0<="" p="">


asterisk1*CLI> <--- SIP read from UDP:192.168.20.1:5060 ---> SIP/2.0 200 OK Call-ID: c30ee825681d114eb9edef0f6ee730d3@192.168.20.1 CSeq: 102 BYE From: <sip:4955439690@192.168.20.9:5060&gt;;tag=as4a49900b to:="" <sip:495yy@192.168.20.1:5060&gt;;tag="192.168.20.1+1+2abdd2af+d6a87167" via:="" sip="" 2.0="" udp="" 192.168.20.9:5060;received="192.168.20.9;rport=5060;branch=z9hG4bK20f61525" content-length:="" 0="" supported:="" 100rel="" contact:="" <sip:192.168.20.1:5060&gt;="" server:="" cs2000_ngss="" 9.0<="" p="">

<-------------> --- (10 headers 0 lines) --- SIP Response message for INCOMING dialog BYE arrived Really destroying SIP dialog 'c30ee825681d114eb9edef0f6ee730d3@192.168.20.1' Method: ACK

удалить закрыть спам изменить тег редактировать

спросил 2011-05-12 10:09:42 +0400

hamel1on Gravatar hamel1on
25 1 3

обновил 2011-05-13 03:29:37 +0400

meral Gravatar meral flag of Ukraine
21228 24 19 169
http://pro-sip.net/

Comments

етить.. редактируйте, там есть кнопочка "код" с нулями и единицами meral ( 2011-05-13 03:29:57 +0400 )редактировать

1 Ответ

0

там судя по всему не поставлен nat=yes и externip=194.67.28.xxx в sip.conf. Хотя этот топик и с бородой =)

ссылка удалить спам редактировать

ответил 2011-12-19 03:08:15 +0400

Zavr2008 Gravatar Zavr2008 flag of Russian Federation
2685 11 9 37
http://mh.otx.ru/

Comments

не некромансь и живым будешь.

meral ( 2011-12-19 06:57:26 +0400 )редактировать

я люблю чтобы закрыты были вопросы :)

Zavr2008 ( 2011-12-19 11:28:02 +0400 )редактировать

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Задан: 2011-05-12 10:09:42 +0400

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Обновлен: Dec 19 '11

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