<------------> -- Executing [345678@default:1] Macro("SIP/102-082195c8", "dialout-omskelecom|345678") in new stack -- Executing [s@macro-dialout-omskelecom:1] Set("SIP/102-082195c8", "GROUP()=OMSKELECOM") in new stack -- Executing [s@macro-dialout-omskelecom:2] GotoIf("SIP/102-082195c8", "0 ?limit") in new stack -- Executing [s@macro-dialout-omskelecom:3] Set("SIP/102-082195c8", "CALLERID(name)=""") in new stack -- Executing [s@macro-dialout-omskelecom:4] Dial("SIP/102-082195c8", "SIP/omskelecom/345678|50|T") in new stack [May 8 10:09:55] WARNING[5515]: rtp.c:1946 ast_rtp_settos: Unable to set TOS to 184 [May 8 10:09:55] WARNING[5515]: udptl.c:846 ast_udptl_settos: UDPTL unable to set TOS to 184 Audio is at 192.168.1.12 port 16770 Adding codec 0x8 (alaw) to SDP Adding codec 0x4 (ulaw) to SDP Reliably Transmitting (no NAT) to 192.168.2.10:5060: INVITE sip:345678@omskelecom.ru SIP/2.0 Via: SIP/2.0/UDP 192.168.1.12:5060;branch=z9hG4bK4bd4a6f1;rport From: "102" ;tag=as334c19ea To: Contact: Call-ID: 10527ca46177abe0290b2a3a7086f6c1@omskelecom.ru CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 08 May 2007 03:09:55 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 183 v=0 o=root 2123 2123 IN IP4 192.168.1.12 s=session c=IN IP4 192.168.1.12 t=0 0 m=audio 16770 RTP/AVP 8 0 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=ptime:20 a=sendrecv --- -- Called omskelecom/345678 linux*CLI> <--- SIP read from 192.168.2.10:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.12:5060;branch=z9hG4bK4bd4a6f1;rport=5060 From: "102" ;tag=as334c19ea To: Call-ID: 10527ca46177abe0290b2a3a7086f6c1@omskelecom.ru CSeq: 102 INVITE <-------------> --- (6 headers 0 lines) --- linux*CLI> <--- SIP read from 192.168.2.10:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.12:5060;branch=z9hG4bK4bd4a6f1;rport=5060 From: "102" ;tag=as334c19ea To: ;tag=test_tag_0008365820 Call-ID: 10527ca46177abe0290b2a3a7086f6c1@omskelecom.ru CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Tue, 08 May 2007 03:09:55 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 WWW-Authenticate: Digest realm="omskelecom.ru", nonce="3196078125-f0317751d112cd02267f3d49f670a9e4", algorithm=MD5, qop="auth" <-------------> --- (12 headers 0 lines) --- Transmitting (no NAT) to 192.168.2.10:5060: ACK sip:345678@omskelecom.ru SIP/2.0 Via: SIP/2.0/UDP 192.168.1.12:5060;branch=z9hG4bK4bd4a6f1;rport From: "102" ;tag=as334c19ea To: ;tag=test_tag_0008365820 Contact: Call-ID: 10527ca46177abe0290b2a3a7086f6c1@omskelecom.ru CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Audio is at 192.168.1.12 port 16770 Adding codec 0x8 (alaw) to SDP Adding codec 0x4 (ulaw) ещ SDP Reliably Transmitting (no NAT) to 192.168.2.10:5060: INVITE sip:345678@omskelecom.ru SIP/2.0 Via: SIP/2.0/UDP 192.168.1.12:5060;branch=z9hG4bK2227280e;rport From: "102" ;tag=as334c19ea To: Contact: Call-ID: 10527ca46177abe0290b2a3a7086f6c1@omskelecom.ru CSeq: 103 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Authorization: Digest username="73812777005", realm="omskelecom.ru", algorithm=MD5, uri="sip:345678@omskelecom.ru", nonce="3196078125-f0317751d112cd02267f3d49f670a9e4", response="bd3f40e91c16233b8c24fdfe56c45920", opaque="", qop=auth, cnonce="0ae78df9", nc=00000001 Date: Tue, 08 May 2007 03:09:55 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 183 v=0 o=root 2123 2124 IN IP4 192.168.1.12 s=session c=IN IP4 192.168.1.12 t=0 0 m=audio 16770 RTP/AVP 8 0 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=ptime:20 a=sendrecv --- linux*CLI> <--- SIP read from 192.168.2.10:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.12:5060;branch=z9hG4bK2227280e;rport=5060 From: "102" ;tag=as334c19ea To: Call-ID: 10527ca46177abe0290b2a3a7086f6c1@omskelecom.ru CSeq: 103 INVITE <-------------> --- (6 headers 0 lines) --- linux*CLI> <--- SIP read from 192.168.2.10:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.1.12:5060;branch=z9hG4bK2227280e;rport=5060 From: "102" ;tag=as334c19ea To: ;tag=test_tag_0008365821 Call-ID: 10527ca46177abe0290b2a3a7086f6c1@omskelecom.ru CSeq: 103 INVITE Contact: Date: Tue,08 May 2007 05:14:32 GMT Content-Type: application/sdp Content-Length: 306 v=0 o=hiQ9200/CN0/071/006/017 470620070407231432 1191575569 IN IP4 192.168.2.10 s=Phone Call via hiQ9200 SIPCA c=IN IP4 192.168.2.10 t=0 0 m=audio 49162 RTP/AVP 8 a=rtpmap:8 PCMA/8000 a=sqn: 0 a=cdsc: 1 audio RTP/AVP 8 0 a=cpar: a=rtpmap:8 PCMA/8000 a=cpar: a=rtpmap:0 PCMU/8000 a=sendrecv <-------------> --- (10 headers 12 lines) --- Found RTP audio format 8 Peer audio RTP is at port 192.168.2.10:49162 Found description format PCMA for ID 8 Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 192.168.2.10:49162 -- Call on SIP/omskelecom-0821ab48 left from hold -- SIP/omskelecom-0821ab48 is making progress passing it to SIP/102-082195c8 Audio is at 192.168.32.1 port 16490 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Transmitting (no NAT) to 192.168.32.99:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.32.99:5060;branch=z9hG4bK-a31f55ee;received=192.168.32.99 From: "IgorG" ;tag=9a88fec6122519beo0 To: "345678" ;tag=as417ea90d Call-ID: 87e60966-3b9a8f9e@192.168.32.99 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 211 v=0 o=root 2123 2123 IN IP4 192.168.32.1 s=session c=IN IP4 192.168.32.1 t=0 0 m=audio 16490 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <------------> linux*CLI> <--- SIP read from 192.168.2.10:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.1.12:5060;branch=z9hG4bK2227280e;rport=5060 From: "102" ;tag=as334c19ea To: ;tag=test_tag_0008365821 Call-ID: 10527ca46177abe0290b2a3a7086f6c1@omskelecom.ru CSeq: 103 INVITE Contact: Date: Tue,08 May 2007 05:14:32 GMT Content-Type: application/sdp Content-Length: 306 v=0 o=hiQ9200/CN0/071/006/017 470620070407231432 1191575569 IN IP4 192.168.2.10 s=Phone Call via hiQ9200 SIPCA c=IN IP4 192.168.2.10 t=0 0 m=audio 49162 RTP/AVP 8 a=rtpmap:8 PCMA/8000 a=sqn: 0 a=cdsc: 1 audio RTP/AVP 8 0 a=cpar: a=rtpmap:8 PCMA/8000 a=cpar: a=rtpmap:0 PCMU/8000 a=sendrecv <-------------> --- (10 headers 12 lines) --- Found RTP audio format 8 Peer audio RTP is at port 192.168.2.10:49162 Found description format PCMA for ID 8 Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 192.168.2.10:49162 -- SIP/omskelecom-0821ab48 is ringing -- Call on SIP/omskelecom-0821ab48 left from hold -- SIP/omskelecom-0821ab48 is making progress passing it to SIP/102-082195c8 linux*CLI> <--- SIP read from 192.168.2.10:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.12:5060;branch=z9hG4bK2227280e;rport=5060 From: "102" ;tag=as334c19ea To: ;tag=test_tag_0008365821 Call-ID: 10527ca46177abe0290b2a3a7086f6c1@omskelecom.ru CSeq: 103 INVITE Contact: Accept-Language: en;q=0.0 Allow: REGISTER, INVITE, ACK, BYE, CANCEL, NOTIFY, REFER Date: Tue,08 May 2007 05:14:35 GMT Content-Type: application/sdp Content-Length: 306 v=0 o=hiQ9200/CN0/071/006/017 470620070407231432 1191575569 IN IP4 192.168.2.10 s=Phone Call via hiQ9200 SIPCA c=IN IP4 192.168.2.10 t=0 0 m=audio 49162 RTP/AVP 8 a=rtpmap:8 PCMA/8000 a=sqn: 0 a=cdsc: 1 audio RTP/AVP 8 0 a=cpar: a=rtpmap:8 PCMA/8000 a=cpar: a=rtpmap:0 PCMU/8000 a=sendrecv <-------------> --- (12 headers 12 lines) --- Found RTP audio format 8 Peer audio RTP is at port 192.168.2.10:49162 Found description format PCMA for ID 8 Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 192.168.2.10:49162 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.2.10, port 5060 Transmitting (no NAT) to 192.168.2.10:5060: ACK sip:345678@192.168.2.10:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.1.12:5060;branch=z9hG4bK1f3d927f;rport From: "102" ;tag=as334c19ea To: ;tag=test_tag_0008365821 Contact: Call-ID: 10527ca46177abe0290b2a3a7086f6c1@omskelecom.ru CSeq: 103 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- -- Call on SIP/omskelecom-0821ab48 left from hold -- SIP/omskelecom-0821ab48 answered SIP/102-082195c8 Audio is at 192.168.32.1 port 16490 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 192.168.32.99:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.32.99:5060;branch=z9hG4bK-a31f55ee;received=192.168.32.99 From: "IgorG" ;tag=9a88fec6122519beo0 To: "345678" ;tag=as417ea90d Call-ID: 87e60966-3b9a8f9e@192.168.32.99 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 211 v=0 o=root 2123 2124 IN IP4 192.168.32.1 s=session c=IN IP4 192.168.32.1 t=0 0 m=audio 16490 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <------------> linux*CLI> <--- SIP read from 192.168.32.99:5060 ---> ACK sip:345678@192.168.32.1 SIP/2.0 Via: SIP/2.0/UDP 192.168.32.99:5060;branch=z9hG4bK-95fa6e0c From: "IgorG" ;tag=9a88fec6122519beo0 To: "345678" ;tag=as417ea90d Call-ID: 87e60966-3b9a8f9e@192.168.32.99 CSeq: 102 ACK Max-Forwards: 70 Proxy-Authorization: Digest username="102",realm="asterisk.local",nonce="740fc6c5",uri="sip:345678@192.168.32.1",algorithm=MD5,response="e4a62c9069a05781ebb5090acc164311" Contact: "IgorG" User-Agent: Sipura/SPA941-4.1.8 Content-Length: 0 <-------------> --- (11 headers 0 lines) --- linux*CLI> <--- SIP read from 192.168.32.99:5060 ---> BYE sip:345678@192.168.32.1 SIP/2.0 Via: SIP/2.0/UDP 192.168.32.99:5060;branch=z9hG4bK-cd3605c0 From: "IgorG" ;tag=9a88fec6122519beo0 To: "345678" ;tag=as417ea90d Call-ID: 87e60966-3b9a8f9e@192.168.32.99 CSeq: 103 BYE Max-Forwards: 70 Proxy-Authorization: Digest username="102",realm="asterisk.local",nonce="740fc6c5",uri="sip:345678@192.168.32.1",algorithm=MD5,response="f0eb42130a3ecbc6bdeb543e77e7ae0c" User-Agent: Sipura/SPA941-4.1.8 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 192.168.32.99 : 5060 (no NAT) <--- Transmitting (no NAT) to 192.168.32.99:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.32.99:5060;branch=z9hG4bK-cd3605c0;received=192.168.32.99 From: "IgorG" ;tag=9a88fec6122519beo0 To: "345678" ;tag=as417ea90d Call-ID: 87e60966-3b9a8f9e@192.168.32.99 Cseq: 103 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> Scheduling destruction of SIP dialog '10527ca46177abe0290b2a3a7086f6c1@omskelecom.ru' in 32000 ms (Method: INVITE) set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.2.10, port 5060 Reliably Transmitting (no NAT) to 192.168.2.10:5060: BYE sip:345678@192.168.2.10:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.1.12:5060;branch=z9hG4bK124f13c6;rport From: "102" ;tag=as334c19ea To: ;tag=test_tag_0008365821 Call-ID: 10527ca46177abe0290b2a3a7086f6c1@omskelecom.ru CSeq: 104 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Authorization: Digest username="73812777005", realm="omskelecom.ru", algorithm=MD5, uri="sip:345678@192.168.2.10:5060", nonce="3196078125-f0317751d112cd02267f3d49f670a9e4", response="65d5adfded1885b9c5ef4d4d4508f90c", opaque="", qop=auth, cnonce="1efe93c1", nc=00000002 Content-Length: 0 --- == Spawn extension (macro-dialout-omskelecom, s, 4) exited non-zero on 'SIP/102-082195c8' linux*CLI> <--- SIP read from 192.168.2.10:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.12:5060;branch=z9hG4bK124f13c6;rport=5060 From: "102" ;tag=as334c19ea To: ;tag=test_tag_0008365821 Call-ID: 10527ca46177abe0290b2a3a7086f6c1@omskelecom.ru CSeq: 104 BYE User-Agent: Asterisk PBX Content-Length: 0 WWW-Authenticate: Digest realm="omskelecom.ru", nonce="3196078190-1d310f901532fc6b254cc69cb0e84186", stale=true, algorithm=MD5, qop="auth" <-------------> --- (9 headers 0 lines) --- [May 8 10:10:01] WARNING[2949]: chan_sip.c:12278 handle_response: Got authentication request (401) on unknown BYE to ';tag=test_tag_0008365821' Really destroying SIP dialog '10527ca46177abe0290b2a3a7086f6c1@omskelecom.ru' Method: INVITE Really destroying SIP dialog '87e60966-3b9a8f9e@192.168.32.99' Method: BYE linux*CLI> exit