<--- SIP read from UDP://192.168.3.197:5061 ---> INVITE sip:92931575@asterisk.norma.perm.ru SIP/2.0 Via: SIP/2.0/UDP 192.168.3.197:5061;branch=z9hG4bK-b837bfb From: rybalko81-voip3 ;tag=d1f818b6208cf3o1 To: Remote-Party-ID: rybalko81-voip3 ;screen=yes;party=calling Call-ID: 98b4cebc-fa99069f@192.168.3.197 CSeq: 101 INVITE Max-Forwards: 70 Contact: rybalko81-voip3 Expires: 240 User-Agent: Linksys/SPA2102-5.2.10 Content-Length: 448 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura, replaces Content-Type: application/sdp v=0 o=- 32420780 32420780 IN IP4 192.168.3.197 s=- c=IN IP4 192.168.3.197 t=0 0 m=audio 16448 RTP/AVP 18 4 0 2 8 96 97 98 100 101 a=rtpmap:18 G729a/8000 a=rtpmap:4 G723/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:100 NSE/8000 a=fmtp:100 192-193 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv <-------------> --- (15 headers 20 lines) --- Sending to 192.168.3.197 : 5061 (no NAT) Using INVITE request as basis request - 98b4cebc-fa99069f@192.168.3.197 Found user 'rybalko81-voip3' for 'rybalko81-voip3' <--- Reliably Transmitting (no NAT) to 192.168.3.197:5061 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.3.197:5061;branch=z9hG4bK-b837bfb;received=192.168.3.197 From: rybalko81-voip3 ;tag=d1f818b6208cf3o1 To: ;tag=as4d600cb0 Call-ID: 98b4cebc-fa99069f@192.168.3.197 CSeq: 101 INVITE User-Agent: Asterisk PBX 1.6.0.25 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk.norma.perm.ru", nonce="6acb16ba" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '98b4cebc-fa99069f@192.168.3.197' in 32000 ms (Method: INVITE) asterisk-alpha*CLI> <--- SIP read from UDP://192.168.3.197:5061 ---> ACK sip:92931575@asterisk.norma.perm.ru SIP/2.0 Via: SIP/2.0/UDP 192.168.3.197:5061;branch=z9hG4bK-b837bfb From: rybalko81-voip3 ;tag=d1f818b6208cf3o1 To: ;tag=as4d600cb0 Call-ID: 98b4cebc-fa99069f@192.168.3.197 CSeq: 101 ACK Max-Forwards: 70 Contact: rybalko81-voip3 User-Agent: Linksys/SPA2102-5.2.10 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- asterisk-alpha*CLI> <--- SIP read from UDP://192.168.3.197:5061 ---> INVITE sip:92931575@asterisk.norma.perm.ru SIP/2.0 Via: SIP/2.0/UDP 192.168.3.197:5061;branch=z9hG4bK-240f5ea3 From: rybalko81-voip3 ;tag=d1f818b6208cf3o1 To: Remote-Party-ID: rybalko81-voip3 ;screen=yes;party=calling Call-ID: 98b4cebc-fa99069f@192.168.3.197 CSeq: 102 INVITE Max-Forwards: 70 Authorization: Digest username="rybalko81-voip3",realm="asterisk.norma.perm.ru",nonce="6acb16ba",uri="sip:92931575@asterisk.norma.perm.ru",algorithm=MD5,response="c430e1c52bf85f70e7b6e477c4a61be5" Contact: rybalko81-voip3 Expires: 240 User-Agent: Linksys/SPA2102-5.2.10 Content-Length: 448 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura, replaces Content-Type: application/sdp v=0 o=- 32420780 32420780 IN IP4 192.168.3.197 s=- c=IN IP4 192.168.3.197 t=0 0 m=audio 16448 RTP/AVP 18 4 0 2 8 96 97 98 100 101 a=rtpmap:18 G729a/8000 a=rtpmap:4 G723/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:100 NSE/8000 a=fmtp:100 192-193 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv <-------------> --- (16 headers 20 lines) --- Sending to 192.168.3.197 : 5061 (no NAT) Using INVITE request as basis request - 98b4cebc-fa99069f@192.168.3.197 Found user 'rybalko81-voip3' for 'rybalko81-voip3' Found RTP audio format 18 Found RTP audio format 4 Found RTP audio format 0 Found RTP audio format 2 Found RTP audio format 8 Found RTP audio format 96 Found RTP audio format 97 Found RTP audio format 98 Found RTP audio format 100 Found RTP audio format 101 Found audio description format G729a for ID 18 Found audio description format G723 for ID 4 Found audio description format PCMU for ID 0 Found audio description format G726-32 for ID 2 Found audio description format PCMA for ID 8 Found unknown media description format G726-40 for ID 96 Found unknown media description format G726-24 for ID 97 Found unknown media description format G726-16 for ID 98 Found unknown media description format NSE for ID 100 Found audio description format telephone-event for ID 101 Capabilities: us - 0x101 (g723|g729), peer - audio=0x90d (g723|ulaw|alaw|g726|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x101 (g723|g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.3.197:16448 Looking for 92931575 in ordinary (domain asterisk.norma.perm.ru) [Mar 3 11:00:15] WARNING[46612]: res_config_pgsql.c:353 realtime_multi_pgsql: PostgreSQL RealTime: Could not find any rows in table extensions. list_route: hop: asterisk-alpha*CLI> <--- Transmitting (no NAT) to 192.168.3.197:5061 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.3.197:5061;branch=z9hG4bK-240f5ea3;received=192.168.3.197 From: rybalko81-voip3 ;tag=d1f818b6208cf3o1 To: Call-ID: 98b4cebc-fa99069f@192.168.3.197 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.0.25 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Length: 0 <------------> Audio is at 192.168.3.20 port 10600 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.3.40:5060: INVITE sip:92931575@192.168.3.40 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.20:5060;branch=z9hG4bK1fc99a3c;rport Max-Forwards: 70 From: "rybalko81-voip3" ;tag=as1b45cf27 To: Contact: Call-ID: 58d990816fee861423e500f87f21ee9a@asterisk.norma.perm.ru CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.0.25 Date: Wed, 03 Mar 2010 06:00:15 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 1400162537 1400162537 IN IP4 192.168.3.20 s=Asterisk PBX 1.6.0.25 c=IN IP4 192.168.3.20 t=0 0 m=audio 10600 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Mar 3 11:00:15] WARNING[46612]: res_config_pgsql.c:353 realtime_multi_pgsql: PostgreSQL RealTime: Could not find any rows in table extensions. asterisk-alpha*CLI> <--- SIP read from UDP://192.168.3.40:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.3.20:5060;branch=z9hG4bK1fc99a3c;rport From: "rybalko81-voip3" ;tag=as1b45cf27 To: ;tag=E4E50828-1DCC Date: Wed, 03 Mar 2010 06:00:15 GMT Call-ID: 58d990816fee861423e500f87f21ee9a@asterisk.norma.perm.ru Server: Cisco-SIPGateway/IOS-12.x CSeq: 102 INVITE Allow-Events: telephone-event Content-Length: 0 <-------------> --- (11 headers 0 lines) --- asterisk-alpha*CLI> <--- SIP read from UDP://192.168.3.40:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.3.20:5060;branch=z9hG4bK1fc99a3c;rport From: "rybalko81-voip3" ;tag=as1b45cf27 To: ;tag=E4E50828-1DCC Date: Wed, 03 Mar 2010 06:00:15 GMT Call-ID: 58d990816fee861423e500f87f21ee9a@asterisk.norma.perm.ru Server: Cisco-SIPGateway/IOS-12.x CSeq: 102 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER Allow-Events: telephone-event Contact: Content-Length: 0 <-------------> --- (12 headers 0 lines) --- asterisk-alpha*CLI> <--- Transmitting (no NAT) to 192.168.3.197:5061 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.3.197:5061;branch=z9hG4bK-240f5ea3;received=192.168.3.197 From: rybalko81-voip3 ;tag=d1f818b6208cf3o1 To: ;tag=as58e3b592 Call-ID: 98b4cebc-fa99069f@192.168.3.197 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.0.25 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Length: 0 <------------> Scheduling destruction of SIP dialog '27f8514-ca6b50d8@192.168.7.117' in 32000 ms (Method: REGISTER) asterisk-alpha*CLI> <--- SIP read from UDP://192.168.3.40:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.3.20:5060;branch=z9hG4bK1fc99a3c;rport From: "rybalko81-voip3" ;tag=as1b45cf27 To: ;tag=E4E50828-1DCC Date: Wed, 03 Mar 2010 06:00:15 GMT Call-ID: 58d990816fee861423e500f87f21ee9a@asterisk.norma.perm.ru Server: Cisco-SIPGateway/IOS-12.x CSeq: 102 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER Supported: replaces Allow-Events: telephone-event Contact: Content-Type: application/sdp Content-Length: 247 v=0 o=CiscoSystemsSIP-GW-UserAgent 3571 4077 IN IP4 192.168.3.40 s=SIP Call c=IN IP4 192.168.3.40 t=0 0 m=audio 17602 RTP/AVP 8 101 c=IN IP4 192.168.3.40 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 <-------------> --- (14 headers 11 lines) --- Found RTP audio format 8 Found RTP audio format 101 Found audio description format PCMA for ID 8 Found audio description format telephone-event for ID 101 Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.3.40:17602 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.3.40, port 5060 Transmitting (no NAT) to 192.168.3.40:5060: ACK sip:92931575@192.168.3.40:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.20:5060;branch=z9hG4bK39de28a2;rport Max-Forwards: 70 From: "rybalko81-voip3" ;tag=as1b45cf27 To: ;tag=E4E50828-1DCC Contact: Call-ID: 58d990816fee861423e500f87f21ee9a@asterisk.norma.perm.ru CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.0.25 Content-Length: 0 --- Audio is at 192.168.3.20 port 14624 Adding codec 0x100 (g729) to SDP Adding codec 0x1 (g723) to SDP Adding non-codec 0x1 (telephone-event) to SDP asterisk-alpha*CLI> <--- Reliably Transmitting (no NAT) to 192.168.3.197:5061 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.3.197:5061;branch=z9hG4bK-240f5ea3;received=192.168.3.197 From: rybalko81-voip3 ;tag=d1f818b6208cf3o1 To: ;tag=as58e3b592 Call-ID: 98b4cebc-fa99069f@192.168.3.197 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.0.25 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 329 v=0 o=root 757568896 757568896 IN IP4 192.168.3.20 s=Asterisk PBX 1.6.0.25 c=IN IP4 192.168.3.20 t=0 0 m=audio 14624 RTP/AVP 18 4 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:4 G723/8000 a=fmtp:4 annexa=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.3.40, port 5060 Audio is at 192.168.3.20 port 10600 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.3.40:5060: INVITE sip:92931575@192.168.3.40:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.20:5060;branch=z9hG4bK6fa77e8e;rport Max-Forwards: 70 From: "rybalko81-voip3" ;tag=as1b45cf27 To: ;tag=E4E50828-1DCC Contact: Call-ID: 58d990816fee861423e500f87f21ee9a@asterisk.norma.perm.ru CSeq: 103 INVITE User-Agent: Asterisk PBX 1.6.0.25 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 266 v=0 o=root 1400162537 1400162538 IN IP4 192.168.3.197 s=Asterisk PBX 1.6.0.25 c=IN IP4 192.168.3.197 t=0 0 m=audio 16448 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- asterisk-alpha*CLI> <--- SIP read from UDP://192.168.3.40:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.3.20:5060;branch=z9hG4bK6fa77e8e;rport From: "rybalko81-voip3" ;tag=as1b45cf27 To: ;tag=E4E50828-1DCC Date: Wed, 03 Mar 2010 06:00:24 GMT Call-ID: 58d990816fee861423e500f87f21ee9a@asterisk.norma.perm.ru Server: Cisco-SIPGateway/IOS-12.x CSeq: 103 INVITE Allow-Events: telephone-event Content-Length: 0 <-------------> --- (10 headers 0 lines) --- asterisk-alpha*CLI> <--- SIP read from UDP://192.168.3.40:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.3.20:5060;branch=z9hG4bK6fa77e8e;rport From: "rybalko81-voip3" ;tag=as1b45cf27 To: ;tag=E4E50828-1DCC Date: Wed, 03 Mar 2010 06:00:24 GMT Call-ID: 58d990816fee861423e500f87f21ee9a@asterisk.norma.perm.ru Server: Cisco-SIPGateway/IOS-12.x CSeq: 103 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER Supported: replaces Allow-Events: telephone-event Contact: Content-Type: application/sdp Content-Length: 247 v=0 o=CiscoSystemsSIP-GW-UserAgent 3571 4078 IN IP4 192.168.3.40 s=SIP Call c=IN IP4 192.168.3.40 t=0 0 m=audio 17602 RTP/AVP 8 101 c=IN IP4 192.168.3.40 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 <-------------> --- (14 headers 11 lines) --- Found RTP audio format 8 Found RTP audio format 101 Found audio description format PCMA for ID 8 Found audio description format telephone-event for ID 101 Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.3.40:17602 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.3.40, port 5060 Transmitting (no NAT) to 192.168.3.40:5060: ACK sip:92931575@192.168.3.40:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.20:5060;branch=z9hG4bK64ebc032;rport Max-Forwards: 70 From: "rybalko81-voip3" ;tag=as1b45cf27 To: ;tag=E4E50828-1DCC Contact: Call-ID: 58d990816fee861423e500f87f21ee9a@asterisk.norma.perm.ru CSeq: 103 ACK User-Agent: Asterisk PBX 1.6.0.25 Content-Length: 0 --- asterisk-alpha*CLI> <--- SIP read from UDP://192.168.3.197:5061 ---> ACK sip:92931575@192.168.3.20 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.197:5061;branch=z9hG4bK-bde39a From: rybalko81-voip3 ;tag=d1f818b6208cf3o1 To: ;tag=as58e3b592 Call-ID: 98b4cebc-fa99069f@192.168.3.197 CSeq: 102 ACK Max-Forwards: 70 Authorization: Digest username="rybalko81-voip3",realm="asterisk.norma.perm.ru",nonce="6acb16ba",uri="sip:92931575@asterisk.norma.perm.ru",algorithm=MD5,response="c430e1c52bf85f70e7b6e477c4a61be5" Contact: rybalko81-voip3 User-Agent: Linksys/SPA2102-5.2.10 Content-Length: 0 <-------------> --- (11 headers 0 lines) --- set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.3.197, port 5061 Audio is at 192.168.3.20 port 14624 Adding codec 0x100 (g729) to SDP Adding codec 0x1 (g723) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.3.197:5061: INVITE sip:rybalko81-voip3@192.168.3.197:5061 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.20:5060;branch=z9hG4bK3cd6b397;rport Max-Forwards: 70 From: ;tag=as58e3b592 To: rybalko81-voip3 ;tag=d1f818b6208cf3o1 Contact: Call-ID: 98b4cebc-fa99069f@192.168.3.197 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.0.25 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 329 v=0 o=root 757568896 757568897 IN IP4 192.168.3.40 s=Asterisk PBX 1.6.0.25 c=IN IP4 192.168.3.40 t=0 0 m=audio 17602 RTP/AVP 18 4 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:4 G723/8000 a=fmtp:4 annexa=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- asterisk-alpha*CLI> <--- SIP read from UDP://192.168.3.197:5061 ---> SIP/2.0 200 OK To: rybalko81-voip3 ;tag=d1f818b6208cf3o1 From: ;tag=as58e3b592 Call-ID: 98b4cebc-fa99069f@192.168.3.197 CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.3.20:5060;branch=z9hG4bK3cd6b397 Contact: rybalko81-voip3 Server: Linksys/SPA2102-5.2.10 Content-Length: 310 Content-Type: application/sdp v=0 o=- 32421731 32421731 IN IP4 192.168.3.197 s=- c=IN IP4 192.168.3.197 t=0 0 m=audio 16448 RTP/AVP 18 100 101 a=rtpmap:18 G729a/8000 a=fmtp:18 annexb=no a=rtpmap:100 NSE/8000 a=fmtp:100 192-193 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv a=silenceSupp:off - - - - <-------------> --- (10 headers 15 lines) --- Found RTP audio format 18 Found RTP audio format 100 Found RTP audio format 101 Found audio description format G729a for ID 18 Found unknown media description format NSE for ID 100 Found audio description format telephone-event for ID 101 Capabilities: us - 0x101 (g723|g729), peer - audio=0x100 (g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.3.197:16448 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.3.197, port 5061 Transmitting (no NAT) to 192.168.3.197:5061: ACK sip:rybalko81-voip3@192.168.3.197:5061 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.20:5060;branch=z9hG4bK5284b921;rport Max-Forwards: 70 From: ;tag=as58e3b592 To: rybalko81-voip3 ;tag=d1f818b6208cf3o1 Contact: Call-ID: 98b4cebc-fa99069f@192.168.3.197 CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.0.25 Content-Length: 0 --- set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.3.40, port 5060 Audio is at 192.168.3.20 port 10600 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.3.40:5060: INVITE sip:92931575@192.168.3.40:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.20:5060;branch=z9hG4bK0e159653;rport Max-Forwards: 70 From: "rybalko81-voip3" ;tag=as1b45cf27 To: ;tag=E4E50828-1DCC Contact: Call-ID: 58d990816fee861423e500f87f21ee9a@asterisk.norma.perm.ru CSeq: 104 INVITE User-Agent: Asterisk PBX 1.6.0.25 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 266 v=0 o=root 1400162537 1400162539 IN IP4 192.168.3.197 s=Asterisk PBX 1.6.0.25 c=IN IP4 192.168.3.197 t=0 0 m=audio 16448 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- asterisk-alpha*CLI> <--- SIP read from UDP://192.168.3.40:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.3.20:5060;branch=z9hG4bK0e159653;rport From: "rybalko81-voip3" ;tag=as1b45cf27 To: ;tag=E4E50828-1DCC Date: Wed, 03 Mar 2010 06:00:24 GMT Call-ID: 58d990816fee861423e500f87f21ee9a@asterisk.norma.perm.ru Server: Cisco-SIPGateway/IOS-12.x CSeq: 104 INVITE Allow-Events: telephone-event Content-Length: 0 <-------------> --- (10 headers 0 lines) --- asterisk-alpha*CLI> <--- SIP read from UDP://192.168.3.40:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.3.20:5060;branch=z9hG4bK0e159653;rport From: "rybalko81-voip3" ;tag=as1b45cf27 To: ;tag=E4E50828-1DCC Date: Wed, 03 Mar 2010 06:00:24 GMT Call-ID: 58d990816fee861423e500f87f21ee9a@asterisk.norma.perm.ru Server: Cisco-SIPGateway/IOS-12.x CSeq: 104 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER Supported: replaces Allow-Events: telephone-event Contact: Content-Type: application/sdp Content-Length: 247 v=0 o=CiscoSystemsSIP-GW-UserAgent 3571 4078 IN IP4 192.168.3.40 s=SIP Call c=IN IP4 192.168.3.40 t=0 0 m=audio 17602 RTP/AVP 8 101 c=IN IP4 192.168.3.40 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 <-------------> --- (14 headers 11 lines) --- set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.3.40, port 5060 Transmitting (no NAT) to 192.168.3.40:5060: ACK sip:92931575@192.168.3.40:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.20:5060;branch=z9hG4bK6de719f0;rport Max-Forwards: 70 From: "rybalko81-voip3" ;tag=as1b45cf27 To: ;tag=E4E50828-1DCC Contact: Call-ID: 58d990816fee861423e500f87f21ee9a@asterisk.norma.perm.ru CSeq: 104 ACK User-Agent: Asterisk PBX 1.6.0.25 Content-Length: 0 --- asterisk-alpha*CLI> <--- SIP read from UDP://192.168.3.197:5061 ---> BYE sip:92931575@192.168.3.20 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.197:5061;branch=z9hG4bK-a0d0cc3e From: rybalko81-voip3 ;tag=d1f818b6208cf3o1 To: ;tag=as58e3b592 Call-ID: 98b4cebc-fa99069f@192.168.3.197 CSeq: 103 BYE Max-Forwards: 70 Authorization: Digest username="rybalko81-voip3",realm="asterisk.norma.perm.ru",nonce="6acb16ba",uri="sip:92931575@192.168.3.20",algorithm=MD5,response="db2b2796a296665799847bbefc2812a0" User-Agent: Linksys/SPA2102-5.2.10 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 192.168.3.197 : 5061 (no NAT) asterisk-alpha*CLI> <--- Transmitting (no NAT) to 192.168.3.197:5061 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.3.197:5061;branch=z9hG4bK-a0d0cc3e;received=192.168.3.197 From: rybalko81-voip3 ;tag=d1f818b6208cf3o1 To: ;tag=as58e3b592 Call-ID: 98b4cebc-fa99069f@192.168.3.197 CSeq: 103 BYE User-Agent: Asterisk PBX 1.6.0.25 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.3.40, port 5060 Audio is at 192.168.3.20 port 10600 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.3.40:5060: INVITE sip:92931575@192.168.3.40:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.20:5060;branch=z9hG4bK004ca9d7;rport Max-Forwards: 70 From: "rybalko81-voip3" ;tag=as1b45cf27 To: ;tag=E4E50828-1DCC Contact: Call-ID: 58d990816fee861423e500f87f21ee9a@asterisk.norma.perm.ru CSeq: 105 INVITE User-Agent: Asterisk PBX 1.6.0.25 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 264 v=0 o=root 1400162537 1400162540 IN IP4 192.168.3.20 s=Asterisk PBX 1.6.0.25 c=IN IP4 192.168.3.20 t=0 0 m=audio 10600 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Mar 3 11:00:27] WARNING[46612]: res_config_pgsql.c:353 realtime_multi_pgsql: PostgreSQL RealTime: Could not find any rows in table extensions. Scheduling destruction of SIP dialog '58d990816fee861423e500f87f21ee9a@asterisk.norma.perm.ru' in 32000 ms (Method: INVITE) [Mar 3 11:00:27] WARNING[46612]: res_config_pgsql.c:353 realtime_multi_pgsql: PostgreSQL RealTime: Could not find any rows in table extensions. asterisk-alpha*CLI> <--- SIP read from UDP://192.168.3.40:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.3.20:5060;branch=z9hG4bK004ca9d7;rport From: "rybalko81-voip3" ;tag=as1b45cf27 To: ;tag=E4E50828-1DCC Date: Wed, 03 Mar 2010 06:00:27 GMT Call-ID: 58d990816fee861423e500f87f21ee9a@asterisk.norma.perm.ru Server: Cisco-SIPGateway/IOS-12.x CSeq: 105 INVITE Allow-Events: telephone-event Content-Length: 0 <-------------> --- (10 headers 0 lines) --- asterisk-alpha*CLI> <--- SIP read from UDP://192.168.3.40:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.3.20:5060;branch=z9hG4bK004ca9d7;rport From: "rybalko81-voip3" ;tag=as1b45cf27 To: ;tag=E4E50828-1DCC Date: Wed, 03 Mar 2010 06:00:27 GMT Call-ID: 58d990816fee861423e500f87f21ee9a@asterisk.norma.perm.ru Server: Cisco-SIPGateway/IOS-12.x CSeq: 105 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER Supported: replaces Allow-Events: telephone-event Contact: Content-Type: application/sdp Content-Length: 247 v=0 o=CiscoSystemsSIP-GW-UserAgent 3571 4079 IN IP4 192.168.3.40 s=SIP Call c=IN IP4 192.168.3.40 t=0 0 m=audio 17602 RTP/AVP 8 101 c=IN IP4 192.168.3.40 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 <-------------> --- (14 headers 11 lines) --- Found RTP audio format 8 Found RTP audio format 101 Found audio description format PCMA for ID 8 Found audio description format telephone-event for ID 101 Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.3.40:17602 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.3.40, port 5060 Transmitting (no NAT) to 192.168.3.40:5060: ACK sip:92931575@192.168.3.40:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.20:5060;branch=z9hG4bK5fa73e7c;rport Max-Forwards: 70 From: "rybalko81-voip3" ;tag=as1b45cf27 To: ;tag=E4E50828-1DCC Contact: Call-ID: 58d990816fee861423e500f87f21ee9a@asterisk.norma.perm.ru CSeq: 105 ACK User-Agent: Asterisk PBX 1.6.0.25 Content-Length: 0 --- set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.3.40, port 5060 Reliably Transmitting (no NAT) to 192.168.3.40:5060: BYE sip:92931575@192.168.3.40:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.20:5060;branch=z9hG4bK03c237e9;rport Max-Forwards: 70 From: "rybalko81-voip3" ;tag=as1b45cf27 To: ;tag=E4E50828-1DCC all-ID: 58d990816fee861423e500f87f21ee9a@asterisk.norma.perm.ru CSeq: 106 BYE User-Agent: Asterisk PBX 1.6.0.25 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- Scheduling destruction of SIP dialog '58d990816fee861423e500f87f21ee9a@asterisk.norma.perm.ru' in 32000 ms (Method: INVITE) asterisk-alpha*CLI> <--- SIP read from UDP://192.168.3.40:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.3.20:5060;branch=z9hG4bK03c237e9;rport From: "rybalko81-voip3" ;tag=as1b45cf27 To: ;tag=E4E50828-1DCC Date: Wed, 03 Mar 2010 06:00:27 GMT Call-ID: 58d990816fee861423e500f87f21ee9a@asterisk.norma.perm.ru Server: Cisco-SIPGateway/IOS-12.x Content-Length: 0 CSeq: 106 BYE