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спросил 2011-05-12 10:09:42 +0400

hamel1on Gravatar hamel1on

sip нет звука в обе стороны

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>;tag=as7b3e3b2d To: <sip:zz@192.168.21.1:5060> Contact: <sip:495yy@192.168.21.9:5060> 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>;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 260

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>;tag=as7b3e3b2d To: <sip:zz@192.168.21.1:5060>;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

<-------------> --- (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>;tag=as7b3e3b2d To: <sip:zz@192.168.21.1:5060>;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> Server: CS2000_NGSS/9.0

<-------------> --- (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>;tag=as7b3e3b2d To: <sip:zz@192.168.21.1:5060>;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> Content-Type: application/sdp Server: CS2000_NGSS/9.0

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>;tag=as7b3e3b2d To: <sip:zz@192.168.21.1:5060>;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> Content-Type: application/sdp Server: CS2000_NGSS/9.0

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> setdestination: Parsing <sip:192.168.21.1:5060> 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>;tag=as7b3e3b2d To: <sip:zz@192.168.21.1:5060>;tag=sbc+1+12d90008+a79e4597 Contact: <sip:495yy@192.168.21.9:5060> Call-ID: 30dced837ed7dba2014d57510373dd17@194.67.28.139 CSeq: 102 ACK User-Agent: Asterisk PBX 1.8.3.2 Content-Length: 0

-- 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>;tag=sbc+1+12d90008+a79e4597 To: "495yy" <sip:495yy@194.67.28.139>;tag=as7b3e3b2d CSeq: 14340794 BYE Content-Length: 0 Supported: 100rel Max-Forwards: 69

<-------------> --- (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>;tag=sbc+1+12d90008+a79e4597 To: "495yy" <sip:495yy@194.67.28.139>;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

<------------> == 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>;tag=192.168.20.1+1+2abdd2af+d6a87167 To: <sip:4955439690@192.168.20.9:5060> CSeq: 1 INVITE Expires: 180 Content-Length: 237 Call-Info: <sip:192.168.20.1:5060>;method="NOTIFY;Event=telephone-event;Duration=2000" Contact: <sip:192.168.20.1:5060> Content-Type: application/sdp Call-ID: c30ee825681d114eb9edef0f6ee730d3@192.168.20.1 Max-Forwards: 55 Accept: application/sdp, application/dtmf-relay

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>

<--- 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>;tag=192.168.20.1+1+2abdd2af+d6a87167 To: <sip:4955439690@192.168.20.9:5060> 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> Content-Length: 0

<------------> 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>;tag=192.168.20.1+1+2abdd2af+d6a87167 To: <sip:4955439690@192.168.20.9:5060>;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> Content-Length: 0

<------------> 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>;tag=192.168.20.1+1+2abdd2af+d6a87167 To: <sip:4955439690@192.168.20.9:5060>;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> Content-Type: application/sdp Content-Length: 298

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>;tag=192.168.20.1+1+2abdd2af+d6a87167 To: <sip:4955439690@192.168.20.9:5060>;tag=as4a49900b CSeq: 1 ACK Contact: <sip:192.168.20.1:5060> Content-Length: 0 Supported: 100rel Max-Forwards: 69

<-------------> --- (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> for address/port to send to setdestination: 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>;tag=as4a49900b To: <sip:495yy@192.168.20.1:5060>;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

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>;tag=as4a49900b To: <sip:495yy@192.168.20.1:5060>;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> Server: CS2000_NGSS/9.0

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

sip нет звука в обе стороны

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>;tag=as7b3e3b2d To: <sip:zz@192.168.21.1:5060> Contact: <sip:495yy@192.168.21.9:5060> 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>;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 260

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>;tag=as7b3e3b2d To: <sip:zz@192.168.21.1:5060>;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

<-------------> --- (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>;tag=as7b3e3b2d To: <sip:zz@192.168.21.1:5060>;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> Server: CS2000_NGSS/9.0

<-------------> --- (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>;tag=as7b3e3b2d To: <sip:zz@192.168.21.1:5060>;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> Content-Type: application/sdp Server: CS2000_NGSS/9.0

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>;tag=as7b3e3b2d To: <sip:zz@192.168.21.1:5060>;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> Content-Type: application/sdp Server: CS2000_NGSS/9.0

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> setdestination: Parsing <sip:192.168.21.1:5060> 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>;tag=as7b3e3b2d To: <sip:zz@192.168.21.1:5060>;tag=sbc+1+12d90008+a79e4597 Contact: <sip:495yy@192.168.21.9:5060> Call-ID: 30dced837ed7dba2014d57510373dd17@194.67.28.139 CSeq: 102 ACK User-Agent: Asterisk PBX 1.8.3.2 Content-Length: 0

-- 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>;tag=sbc+1+12d90008+a79e4597 To: "495yy" <sip:495yy@194.67.28.139>;tag=as7b3e3b2d CSeq: 14340794 BYE Content-Length: 0 Supported: 100rel Max-Forwards: 69

<-------------> --- (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>;tag=sbc+1+12d90008+a79e4597 To: "495yy" <sip:495yy@194.67.28.139>;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

<------------> == 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>;tag=192.168.20.1+1+2abdd2af+d6a87167 To: <sip:4955439690@192.168.20.9:5060> CSeq: 1 INVITE Expires: 180 Content-Length: 237 Call-Info: <sip:192.168.20.1:5060>;method="NOTIFY;Event=telephone-event;Duration=2000" Contact: <sip:192.168.20.1:5060> Content-Type: application/sdp Call-ID: c30ee825681d114eb9edef0f6ee730d3@192.168.20.1 Max-Forwards: 55 Accept: application/sdp, application/dtmf-relay

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>

<--- 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>;tag=192.168.20.1+1+2abdd2af+d6a87167 To: <sip:4955439690@192.168.20.9:5060> 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> Content-Length: 0

<------------> 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>;tag=192.168.20.1+1+2abdd2af+d6a87167 To: <sip:4955439690@192.168.20.9:5060>;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> Content-Length: 0

<------------> 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>;tag=192.168.20.1+1+2abdd2af+d6a87167 To: <sip:4955439690@192.168.20.9:5060>;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> Content-Type: application/sdp Content-Length: 298

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>;tag=192.168.20.1+1+2abdd2af+d6a87167 To: <sip:4955439690@192.168.20.9:5060>;tag=as4a49900b CSeq: 1 ACK Contact: <sip:192.168.20.1:5060> Content-Length: 0 Supported: 100rel Max-Forwards: 69

<-------------> --- (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> for address/port to send to setdestination: 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>;tag=as4a49900b To: <sip:495yy@192.168.20.1:5060>;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

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>;tag=as4a49900b To: <sip:495yy@192.168.20.1:5060>;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> Server: CS2000_NGSS/9.0

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

нажмите, чтобы скрыть/показать версии 3
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обновил 2011-05-13 03:29:37 +0400

meral Gravatar meral flag of Ukraine

http://pro-sip.net/

sip нет звука в обе стороны

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>;tag=as7b3e3b2d To: <sip:zz@192.168.21.1:5060> Contact: <sip:495yy@192.168.21.9:5060> 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>;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 260

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>;tag=as7b3e3b2d To: <sip:zz@192.168.21.1:5060>;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

<-------------> --- (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>;tag=as7b3e3b2d To: <sip:zz@192.168.21.1:5060>;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> Server: CS2000_NGSS/9.0

<-------------> --- (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>;tag=as7b3e3b2d To: <sip:zz@192.168.21.1:5060>;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> Content-Type: application/sdp Server: CS2000_NGSS/9.0

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>;tag=as7b3e3b2d To: <sip:zz@192.168.21.1:5060>;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> Content-Type: application/sdp Server: CS2000_NGSS/9.0

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> setdestination: Parsing <sip:192.168.21.1:5060> 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>;tag=as7b3e3b2d To: <sip:zz@192.168.21.1:5060>;tag=sbc+1+12d90008+a79e4597 Contact: <sip:495yy@192.168.21.9:5060> Call-ID: 30dced837ed7dba2014d57510373dd17@194.67.28.139 CSeq: 102 ACK User-Agent: Asterisk PBX 1.8.3.2 Content-Length: 0

-- 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>;tag=sbc+1+12d90008+a79e4597 To: "495yy" <sip:495yy@194.67.28.139>;tag=as7b3e3b2d CSeq: 14340794 BYE Content-Length: 0 Supported: 100rel Max-Forwards: 69

<-------------> --- (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>;tag=sbc+1+12d90008+a79e4597 To: "495yy" <sip:495yy@194.67.28.139>;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

<------------> == 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>;tag=192.168.20.1+1+2abdd2af+d6a87167 To: <sip:4955439690@192.168.20.9:5060> CSeq: 1 INVITE Expires: 180 Content-Length: 237 Call-Info: <sip:192.168.20.1:5060>;method="NOTIFY;Event=telephone-event;Duration=2000" Contact: <sip:192.168.20.1:5060> Content-Type: application/sdp Call-ID: c30ee825681d114eb9edef0f6ee730d3@192.168.20.1 Max-Forwards: 55 Accept: application/sdp, application/dtmf-relay

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>

<--- 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>;tag=192.168.20.1+1+2abdd2af+d6a87167 To: <sip:4955439690@192.168.20.9:5060> 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> Content-Length: 0

<------------> 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>;tag=192.168.20.1+1+2abdd2af+d6a87167 To: <sip:4955439690@192.168.20.9:5060>;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> Content-Length: 0

<------------> 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>;tag=192.168.20.1+1+2abdd2af+d6a87167 To: <sip:4955439690@192.168.20.9:5060>;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> Content-Type: application/sdp Content-Length: 298

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>;tag=192.168.20.1+1+2abdd2af+d6a87167 To: <sip:4955439690@192.168.20.9:5060>;tag=as4a49900b CSeq: 1 ACK Contact: <sip:192.168.20.1:5060> Content-Length: 0 Supported: 100rel Max-Forwards: 69

<-------------> --- (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> for address/port to send to setdestination: 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>;tag=as4a49900b To: <sip:495yy@192.168.20.1:5060>;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

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>;tag=as4a49900b To: <sip:495yy@192.168.20.1:5060>;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> Server: CS2000_NGSS/9.0

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

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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.