звонок с 214058 на (9)214020 через кошку Вывод с астериска sip set debug bee*CLI> <--- SIP read from 192.168.11.123:5060 ---> INVITE sip:9214020@192.168.11.124 SIP/2.0 Via: SIP/2.0/UDP 192.168.11.123;branch=z9hG4bK25FE1777 CSeq: 1604 INVITE To: Content-Type: application/sdp From: ;tag=59297E9A Call-ID: 767094243@192.168.11.123 Subject: sip:214058@192.168.11.124 Content-Length: 262 User-Agent: kphone/4.2 Contact: v=0 o=username 0 0 IN IP4 192.168.11.123 s=The Funky Flow c=IN IP4 192.168.11.123 t=0 0 m=audio 32916 RTP/AVP 8 97 3 0 m=audio 32916 RTP/AVP 97 3 0 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 <-------------> --- (11 headers 12 lines) --- Sending to 192.168.11.123 : 5060 (no NAT) Using INVITE request as basis request - 767094243@192.168.11.123 <--- Reliably Transmitting (no NAT) to 192.168.11.123:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.11.123;branch=z9hG4bK25FE1777;received=192.168.11.123 From: ;tag=59297E9A To: ;tag=as6dc44a40 Call-ID: 767094243@192.168.11.123 CSeq: 1604 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="15f9d22c" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '767094243@192.168.11.123' in 32000 ms (Method: INVITE) Found user '214058' bee*CLI> <--- SIP read from 192.168.11.123:5060 ---> ACK sip:9214020@192.168.11.124 SIP/2.0 Via: SIP/2.0/UDP 192.168.11.123;branch=z9hG4bK25FE1777 CSeq: 1604 ACK To: ;tag=as6dc44a40 From: ;tag=59297E9A Call-ID: 767094243@192.168.11.123 Content-Length: 0 User-Agent: kphone/4.2 Contact: <-------------> --- (9 headers 0 lines) --- bee*CLI> <--- SIP read from 192.168.11.123:5060 ---> INVITE sip:9214020@192.168.11.124 SIP/2.0 Via: SIP/2.0/UDP 192.168.11.123;branch=z9hG4bK5624B9F CSeq: 1605 INVITE To: Proxy-Authorization: Digest username="214058", realm="asterisk", nonce="15f9d22c", uri="sip:9214020@192.168.11.124", cnonce="abcdefghi", nc=00000001, response="0b64017e9c7cb3b3275bd7f39f14d339", opaque="", algorithm="MD5" Content-Type: application/sdp From: ;tag=59297E9A Call-ID: 767094243@192.168.11.123 Subject: sip:214058@192.168.11.124 Content-Length: 262 User-Agent: kphone/4.2 Contact: v=0 o=username 0 0 IN IP4 192.168.11.123 s=The Funky Flow c=IN IP4 192.168.11.123 t=0 0 m=audio 32916 RTP/AVP 8 97 3 0 m=audio 32916 RTP/AVP 97 3 0 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 <-------------> --- (12 headers 12 lines) --- Sending to 192.168.11.123 : 5060 (no NAT) Using INVITE request as basis request - 767094243@192.168.11.123 Found user '214058' Found RTP audio format 8 Found RTP audio format 97 Found RTP audio format 3 Found RTP audio format 0 Found RTP audio format 97 Found RTP audio format 3 Found RTP audio format 0 Peer audio RTP is at port 192.168.11.123:32916 Found description format PCMU for ID 0 Found description format GSM for ID 3 Found description format PCMA for ID 8 Found description format iLBC for ID 97 Capabilities: us - 0x90c (ulaw|alaw|g726|g729), peer - audio=0x40e (gsm|ulaw|alaw|ilbc)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 192.168.11.123:32916 Looking for 9214020 in local-phones (domain 192.168.11.124) list_route: hop: <--- Transmitting (no NAT) to 192.168.11.123:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.11.123;branch=z9hG4bK5624B9F;received=192.168.11.123 From: ;tag=59297E9A To: Call-ID: 767094243@192.168.11.123 CSeq: 1605 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> Audio is at 192.168.11.124 port 14596 Adding codec 0x8 (alaw) to SDP Adding codec 0x800 (g726) to SDP Adding codec 0x10 (g726aal2) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.11.120:5060: INVITE sip:214020@192.168.11.120 SIP/2.0 Via: SIP/2.0/UDP 192.168.11.124:5060;branch=z9hG4bK460d39aa;rport From: "214058" ;tag=as5ca83e36 To: Contact: Call-ID: 2b4510122f2f403b052800453841a143@192.168.11.124 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 10 Jul 2007 17:09:28 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 356 v=0 o=root 8154 8154 IN IP4 192.168.11.124 s=session c=IN IP4 192.168.11.124 t=0 0 m=audio 14596 RTP/AVP 8 111 112 0 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:112 AAL2-G726-32/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Jul 10 21:09:28] WARNING[9033]: channel.c:3147 ast_channel_make_compatible: No path to translate from SIP/cisco-081e8ea0(256) to SIP/214058-081e3600(8) bee*CLI> <--- SIP read from 192.168.11.120:5060 ---> SIP/2.0 488 Not Acceptable Media Via: SIP/2.0/UDP 192.168.11.124:5060;branch=z9hG4bK460d39aa;rport From: "214058" ;tag=as5ca83e36 To: ;tag=A070E75-1A98 Date: Tue, 10 Jul 2007 17:09:26 GMT Call-ID: 2b4510122f2f403b052800453841a143@192.168.11.124 Server: Cisco-SIPGateway/IOS-12.x CSeq: 102 INVITE Allow-Events: telephone-event Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Transmitting (no NAT) to 192.168.11.120:5060: ACK sip:214020@192.168.11.120 SIP/2.0 Via: SIP/2.0/UDP 192.168.11.124:5060;branch=z9hG4bK460d39aa;rport From: "214058" ;tag=as5ca83e36 To: ;tag=A070E75-1A98 Contact: Call-ID: 2b4510122f2f403b052800453841a143@192.168.11.124 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Scheduling destruction of SIP dialog '2b4510122f2f403b052800453841a143@192.168.11.124' in 32000 ms (Method: INVITE) Reliably Transmitting (no NAT) to 192.168.11.120:5060: BYE sip:214020@192.168.11.120 SIP/2.0 Via: SIP/2.0/UDP 192.168.11.124:5060;branch=z9hG4bK5b83d3ec;rport From: "214058" ;tag=as5ca83e36 To: ;tag=A070E75-1A98 Call-ID: 2b4510122f2f403b052800453841a143@192.168.11.124 CSeq: 103 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- <--- Transmitting (no NAT) to 192.168.11.123:5060 ---> SIP/2.0 503 Service Unavailable Via: SIP/2.0/UDP 192.168.11.123;branch=z9hG4bK5624B9F;received=192.168.11.123 From: ;tag=59297E9A To: ;tag=as576dd9e2 Call-ID: 767094243@192.168.11.123 CSeq: 1605 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 X-Asterisk-HangupCause: Bearer capability not available X-Asterisk-HangupCauseCode: 58 <------------> bee*CLI> <--- SIP read from 192.168.11.123:5060 ---> ACK sip:9214020@192.168.11.124 SIP/2.0 Via: SIP/2.0/UDP 192.168.11.123;branch=z9hG4bK5624B9F CSeq: 1605 ACK To: ;tag=as576dd9e2 From: ;tag=59297E9A Call-ID: 767094243@192.168.11.123 Content-Length: 0 User-Agent: kphone/4.2 Contact: <-------------> --- (9 headers 0 lines) --- Really destroying SIP dialog '767094243@192.168.11.123' Method: ACK bee*CLI> <--- SIP read from 192.168.11.120:5060 ---> SIP/2.0 481 Call Leg/Transaction Does Not Exist Via: SIP/2.0/UDP 192.168.11.124:5060;branch=z9hG4bK5b83d3ec;rport From: "214058" ;tag=as5ca83e36 To: ;tag=A070E75-1A98 Call-ID: 2b4510122f2f403b052800453841a143@192.168.11.124 CSeq: 103 BYE Content-Length: 0 <-------------> --- (7 headers 0 lines) --- [Jul 10 21:09:28] WARNING[8183]: chan_sip.c:12311 handle_response: Remote host can't match request BYE to call '2b4510122f2f403b052800453841a143@192.168.11.124'. Giving up. Really destroying SIP dialog '2b4510122f2f403b052800453841a143@192.168.11.124' Method: INVITE >>>>>>>>>>>>>>>>>>>>>>>>>>>>> sh run voice class codec 1 codec preference 1 g729r8 codec preference 2 g726r16 codec preference 3 g726r24 codec preference 4 g726r32 codec preference 5 g711alaw codec preference 6 g711ulaw dial-peer voice 1 pots tone ringback alert-no-PI incoming called-number .T destination-pattern . direct-inward-dial port 1:15 forward-digits all ! dial-peer voice 2 voip incoming called-number . destination-pattern 2140[5-9][5-9] session protocol sipv2 session target ipv4:192.168.11.124 (как только добавляем сюда ) (codec g711alaw - все прекрасно работает) debug c кошки пустой :( sip.conf [cisco] context=pstn-incoming type=friend host=192.168.11.120 insecure=very dtmfmode=rfc2833 allow=g729 allow=g726 allow=g726aal2 allow=alaw allow=ulaw [214058] context=local-phones type=friend username=214058 secret=214058 callerid=214058 host=dynamic disallow=all allow=g729 allow=g726 allow=alaw allow=ulaw вот.