Connected to Asterisk 1.2.16 currently running on kc1 (pid = 31139) Verbosity is at least 3 kc1*CLI> <-- SIP read from phone_ip:52236: INVITE sip:dest_number@asterisk_server SIP/2.0 Via: SIP/2.0/UDP phone_ip:52236;branch=z9hG4bK-7a95dcd7 From: "Gaspar Zoltan" ;tag=b90af9deb3a01c3fo0 To: Call-ID: b969158a-f2ad483@192.168.0.3 CSeq: 101 INVITE Max-Forwards: 70 Contact: "Gaspar Zoltan" Expires: 240 User-Agent: Sipura/SPA841-3.1.4(a) Content-Length: 393 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Content-Type: application/sdp v=0 o=- 248326 248326 IN IP4 phone_ip s=- c=IN IP4 phone_ip t=0 0 m=audio 60432 RTP/AVP 0 2 4 8 18 96 97 98 101 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729a/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv --- (13 headers 18 lines) --- Using INVITE request as basis request - b969158a-f2ad483@192.168.0.3 Sending to phone_ip : 52236 (NAT) Reliably Transmitting (NAT) to phone_ip:52236: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP phone_ip:52236;branch=z9hG4bK-7a95dcd7;received=phone_ip From: "Gaspar Zoltan" ;tag=b90af9deb3a01c3fo0 To: ;tag=as0a0f5d75 Call-ID: b969158a-f2ad483@192.168.0.3 CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="68614334" Content-Length: 0 --- Scheduling destruction of call 'b969158a-f2ad483@192.168.0.3' in 15000 ms Found user '45e2b8014cddb47a9cfd3cf34d3d6db4' kc1*CLI> <-- SIP read from phone_ip:52236: ACK sip:dest_number@asterisk_server SIP/2.0 Via: SIP/2.0/UDP phone_ip:52236;branch=z9hG4bK-7a95dcd7 From: "Gaspar Zoltan" ;tag=b90af9deb3a01c3fo0 To: ;tag=as0a0f5d75 Call-ID: b969158a-f2ad483@192.168.0.3 CSeq: 101 ACK Max-Forwards: 70 Contact: "Gaspar Zoltan" User-Agent: Sipura/SPA841-3.1.4(a) Content-Length: 0 --- (10 headers 0 lines) --- kc1*CLI> <-- SIP read from phone_ip:52236: INVITE sip:dest_number@asterisk_server SIP/2.0 Via: SIP/2.0/UDP phone_ip:52236;branch=z9hG4bK-5cfbaa4c From: "Gaspar Zoltan" ;tag=b90af9deb3a01c3fo0 To: Call-ID: b969158a-f2ad483@192.168.0.3 CSeq: 102 INVITE Max-Forwards: 70 Proxy-Authorization: Digest username="digest_username",realm="asterisk",nonce="68614334",uri="sip:dest_number@asterisk_server",algorithm=MD5,response="response" Contact: "Gaspar Zoltan" Expires: 240 User-Agent: Sipura/SPA841-3.1.4(a) Content-Length: 393 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Content-Type: application/sdp v=0 o=- 248326 248326 IN IP4 phone_ip s=- c=IN IP4 phone_ip t=0 0 m=audio 60434 RTP/AVP 0 2 4 8 18 96 97 98 101 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729a/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv --- (14 headers 18 lines) --- Using INVITE request as basis request - b969158a-f2ad483@192.168.0.3 Sending to phone_ip : 52236 (NAT) Found user '45e2b8014cddb47a9cfd3cf34d3d6db4' Found RTP audio format 0 Found RTP audio format 2 Found RTP audio format 4 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 96 Found RTP audio format 97 Found RTP audio format 98 Found RTP audio format 101 Peer audio RTP is at port phone_ip:60434 Found description format PCMU Found description format G726-32 Found description format G723 Found description format PCMA Found description format G729a Found description format G726-40 Found description format G726-24 Found description format G726-16 Found description format telephone-event Capabilities: us - 0x104 (ulaw|g729), peer - audio=0x51d (g723|ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing), combined - 0x104 (ulaw|g729) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for dest_number in default (domain asterisk_server) list_route: hop: Transmitting (NAT) to phone_ip:52236: SIP/2.0 100 Trying Via: SIP/2.0/UDP phone_ip:52236;branch=z9hG4bK-5cfbaa4c;received=phone_ip From: "Gaspar Zoltan" ;tag=b90af9deb3a01c3fo0 To: Call-ID: b969158a-f2ad483@192.168.0.3 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- -- Executing DeadAGI("SIP/45e2b8014cddb47a9cfd3cf34d3d6db4-08577918", "exec_main.php") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/exec_main.php exec_main.php: agi_request = exec_main.php exec_main.php: agi_channel = SIP/45e2b8014cddb47a9cfd3cf34d3d6db4-08577918 exec_main.php: agi_language = en exec_main.php: agi_type = SIP exec_main.php: agi_uniqueid = 1181204785.18 exec_main.php: agi_callerid = 45e2b8014cddb47a9cfd3cf34d3d6db4 exec_main.php: agi_calleridname = Gaspar Zoltan exec_main.php: agi_callingpres = 0 exec_main.php: agi_callingani2 = 0 exec_main.php: agi_callington = 0 exec_main.php: agi_callingtns = 0 exec_main.php: agi_dnid = dest_number exec_main.php: agi_rdnis = unknown exec_main.php: agi_context = default exec_main.php: agi_extension = dest_number exec_main.php: agi_priority = 1 exec_main.php: agi_enhanced = 0.0 exec_main.php: agi_accountcode = 45e2b8014cddb47a9cf exec_main.php: X-- exec_main.php: END of header_agi.inc.php exec_main.php: *** connection_count_inc start *** exec_main.php: case_nr = 10 We're at asterisk_ip port 18930 Adding codec 0x4 (ulaw) to SDP Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to phone_ip:52236: SIP/2.0 200 OK Via: SIP/2.0/UDP phone_ip:52236;branch=z9hG4bK-5cfbaa4c;received=phone_ip From: "Gaspar Zoltan" ;tag=b90af9deb3a01c3fo0 To: ;tag=as00fa36c8 Call-ID: b969158a-f2ad483@192.168.0.3 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 265 v=0 o=root 31139 31139 IN IP4 asterisk_ip s=session c=IN IP4 asterisk_ip t=0 0 m=audio 18930 RTP/AVP 0 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Playing 'digits/1' (language 'en') kc1*CLI> <-- SIP read from phone_ip:52236: ACK sip:dest_number@asterisk_ip SIP/2.0 Via: SIP/2.0/UDP phone_ip:52236;branch=z9hG4bK-6ca276f2 From: "Gaspar Zoltan" ;tag=b90af9deb3a01c3fo0 To: ;tag=as00fa36c8 Call-ID: b969158a-f2ad483@192.168.0.3 CSeq: 102 ACK Max-Forwards: 70 Proxy-Authorization: Digest username="45e2b8014cddb47a9cfd3cf34d3d6db4",realm="asterisk",nonce="68614334",uri="sip:dest_number@asterisk_ip",algorithm=MD5,response="ee8cf0942199a4724cff944c015e4fcf" Contact: "Gaspar Zoltan" User-Agent: Sipura/SPA841-3.1.4(a) Content-Length: 0 --- (11 headers 0 lines) --- -- Playing 'digits/thousand' (language 'en') -- Playing 'digits/2' (language 'en') -- Playing 'digits/hundred' (language 'en') -- Playing 'digits/30' (language 'en') -- Playing 'digits/4' (language 'en') -- AGI Script Executing Application: (dial) Options: (SIP/tech_prefix_dest_number@provider_ip||S(20)) -- Setting call duration limit to 20 seconds. We're at asterisk_ip port 19356 Adding codec 0x4 (ulaw) to SDP Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP 13 headers, 12 lines Reliably Transmitting (NAT) to provider_ip:5060: INVITE sip:tech_prefix_dest_number@provider_ip SIP/2.0 Via: SIP/2.0/UDP asterisk_ip:5060;branch=z9hG4bK4645ccbc;rport From: "Gaspar Zoltan" ;tag=as2f9caf85 To: Contact: Call-ID: 20a155c3054ed7a974bf482719e640ba@asterisk_ip CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 07 Jun 2007 08:26:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 265 v=0 o=root 31139 31139 IN IP4 asterisk_ip s=session c=IN IP4 asterisk_ip t=0 0 m=audio 19356 RTP/AVP 0 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Called tech_prefix_dest_number@provider_ip kc1*CLI> <-- SIP read from provider_ip:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP asterisk_ip:5060;branch=z9hG4bK4645ccbc;rport From: "Gaspar Zoltan" ;tag=as2f9caf85 To: ;tag=73936DA0-1ACD Date: Thu, 07 Jun 2007 08:26:29 GMT Call-ID: 20a155c3054ed7a974bf482719e640ba@asterisk_ip Server: Cisco-SIPGateway/IOS-12.x CSeq: 102 INVITE Allow-Events: telephone-event Content-Length: 0 --- kc1*CLI> <-- SIP read from provider_ip:5060: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP asterisk_ip:5060;branch=z9hG4bK4645ccbc;rport From: "Gaspar Zoltan" ;tag=as2f9caf85 To: ;tag=73936DA0-1ACD Date: Thu, 07 Jun 2007 08:26:29 GMT Call-ID: 20a155c3054ed7a974bf482719e640ba@asterisk_ip 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-Disposition: session;handling=required Content-Type: application/sdp Content-Length: 235 v=0 o=CiscoSystemsSIP-GW-UserAgent 3288 9303 IN IP4 provider_ip s=SIP Call c=IN IP4 provider_ip t=0 0 m=audio 21440 RTP/AVP 0 101 c=IN IP4 provider_ip a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 --- (14 headers 10 lines) --- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port provider_ip:21440 Found description format PCMU Found description format telephone-event Capabilities: us - 0x104 (ulaw|g729), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) -- SIP/provider_ip-08581060 is making progress passing it to SIP/45e2b8014cddb47a9cfd3cf34d3d6db4-08577918 kc1*CLI> <-- SIP read from provider_ip:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP asterisk_ip:5060;branch=z9hG4bK4645ccbc;rport From: "Gaspar Zoltan" ;tag=as2f9caf85 To: ;tag=73936DA0-1ACD Date: Thu, 07 Jun 2007 08:26:29 GMT Call-ID: 20a155c3054ed7a974bf482719e640ba@asterisk_ip 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-Type: application/sdp Content-Length: 235 v=0 o=CiscoSystemsSIP-GW-UserAgent 3288 9303 IN IP4 provider_ip s=SIP Call c=IN IP4 provider_ip t=0 0 m=audio 21440 RTP/AVP 0 101 c=IN IP4 provider_ip a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 --- (13 headers 10 lines) --- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port provider_ip:21440 Found description format PCMU Found description format telephone-event Capabilities: us - 0x104 (ulaw|g729), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to provider_ip, port 5060 Transmitting (NAT) to provider_ip:5060: ACK sip:tech_prefix_dest_number@provider_ip:5060 SIP/2.0 Via: SIP/2.0/UDP asterisk_ip:5060;branch=z9hG4bK1af50f7f;rport From: "Gaspar Zoltan" ;tag=as2f9caf85 To: ;tag=73936DA0-1ACD Contact: Call-ID: 20a155c3054ed7a974bf482719e640ba@asterisk_ip CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 kc1*CLI> --- -- SIP/provider_ip-08581060 answered SIP/45e2b8014cddb47a9cfd3cf34d3d6db4-08577918 -- Attempting native bridge of SIP/45e2b8014cddb47a9cfd3cf34d3d6db4-08577918 and SIP/provider_ip-08581060 set_destination: Parsing for address/port to send to set_destination: set destination to phone_ip, port 52236 We're at asterisk_ip port 18930 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP 13 headers, 10 lines Reliably Transmitting (NAT) to phone_ip:52236: INVITE sip:45e2b8014cddb47a9cfd3cf34d3d6db4@phone_ip:52236 SIP/2.0 Via: SIP/2.0/UDP asterisk_ip:5060;branch=z9hG4bK61f68ef9;rport From: ;tag=as00fa36c8 To: "Gaspar Zoltan" ;tag=b90af9deb3a01c3fo0 Contact: Call-ID: b969158a-f2ad483@192.168.0.3 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY X-asterisk-info: SIP re-invite (RTP bridge) Content-Type: application/sdp Content-Length: 216 v=0 o=root 31139 31140 IN IP4 provider_ip s=session c=IN IP4 provider_ip t=0 0 m=audio 21440 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- set_destination: Parsing for address/port to send to set_destination: set destination to provider_ip, port 5060 We're at asterisk_ip port 19356 Adding codec 0x4 (ulaw) to SDP Adding codec 0x100 (g729) to SDP Adding codec 0x1 (g723) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x10 (g726) to SDP Adding codec 0x400 (ilbc) to SDP Adding non-codec 0x1 (telephone-event) to SDP 13 headers, 16 lines Reliably Transmitting (NAT) to provider_ip:5060: INVITE sip:tech_prefix_dest_number@provider_ip:5060 SIP/2.0 Via: SIP/2.0/UDP asterisk_ip:5060;branch=z9hG4bK2a90ede4;rport From: "Gaspar Zoltan" ;tag=as2f9caf85 To: ;tag=73936DA0-1ACD Contact: Call-ID: 20a155c3054ed7a974bf482719e640ba@asterisk_ip CSeq: 103 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY X-asterisk-info: SIP re-invite (RTP bridge) Content-Type: application/sdp Content-Length: 366 v=0 o=root 31139 31140 IN IP4 phone_ip s=session c=IN IP4 phone_ip t=0 0 m=audio 60434 RTP/AVP 0 18 4 8 111 97 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- kc1*CLI> <-- SIP read from provider_ip:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP asterisk_ip:5060;branch=z9hG4bK2a90ede4;rport From: "Gaspar Zoltan" ;tag=as2f9caf85 To: ;tag=73936DA0-1ACD Date: Thu, 07 Jun 2007 08:26:35 GMT Call-ID: 20a155c3054ed7a974bf482719e640ba@asterisk_ip Server: Cisco-SIPGateway/IOS-12.x CSeq: 103 INVITE Allow-Events: telephone-event Content-Length: 0 --- (10 headers 0 lines) --- kc1*CLI> <-- SIP read from provider_ip:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP asterisk_ip:5060;branch=z9hG4bK2a90ede4;rport From: "Gaspar Zoltan" ;tag=as2f9caf85 To: ;tag=73936DA0-1ACD Date: Thu, 07 Jun 2007 08:26:35 GMT Call-ID: 20a155c3054ed7a974bf482719e640ba@asterisk_ip Server: Cisco-SIPGateway/IOS-12.x CSeq: 103 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER Allow-Events: telephone-event Contact: Content-Type: application/sdp Content-Length: 206 v=0 o=CiscoSystemsSIP-GW-UserAgent 3288 9304 IN IP4 provider_ip s=SIP Call c=IN IP4 provider_ip t=0 0 m=audio 21440 RTP/AVP 0 c=IN IP4 provider_ip a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - --- (13 headers 9 lines) --- Found RTP audio format 0 Peer audio RTP is at port provider_ip:21440 Found description format PCMU Capabilities: us - 0x104 (ulaw|g729), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) set_destination: Parsing for address/port to send to set_destination: set destination to provider_ip, port 5060 Transmitting (NAT) to provider_ip:5060: ACK sip:tech_prefix_dest_number@provider_ip:5060 SIP/2.0 Via: SIP/2.0/UDP asterisk_ip:5060;branch=z9hG4bK0caa74e7;rport From: "Gaspar Zoltan" ;tag=as2f9caf85 To: ;tag=73936DA0-1ACD Contact: Call-ID: 20a155c3054ed7a974bf482719e640ba@asterisk_ip CSeq: 103 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- kc1*CLI> <-- SIP read from phone_ip:52236: SIP/2.0 200 OK To: "Gaspar Zoltan" ;tag=b90af9deb3a01c3fo0 From: ;tag=as00fa36c8 Call-ID: b969158a-f2ad483@192.168.0.3 CSeq: 102 INVITE Via: SIP/2.0/UDP asterisk_ip:5060;branch=z9hG4bK61f68ef9 Contact: "Gaspar Zoltan" Server: Sipura/SPA841-3.1.4(a) Content-Length: 204 Content-Type: application/sdp v=0 o=- 249397 249397 IN IP4 phone_ip s=- c=IN IP4 phone_ip t=0 0 m=audio 60440 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv --- (10 headers 11 lines) --- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port phone_ip:60440 Found description format PCMU Found description format telephone-event Capabilities: us - 0x104 (ulaw|g729), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to phone_ip, port 52236 Transmitting (NAT) to phone_ip:52236: ACK sip:45e2b8014cddb47a9cfd3cf34d3d6db4@phone_ip:52236 SIP/2.0 Via: SIP/2.0/UDP asterisk_ip:5060;branch=z9hG4bK00329433;rport From: ;tag=as00fa36c8 To: "Gaspar Zoltan" ;tag=b90af9deb3a01c3fo0 Contact: Call-ID: b969158a-f2ad483@192.168.0.3 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- set_destination: Parsing for address/port to send to set_destination: set destination to provider_ip, port 5060 We're at asterisk_ip port 19356 Adding codec 0x4 (ulaw) to SDP 13 headers, 8 lines Reliably Transmitting (NAT) to provider_ip:5060: INVITE sip:tech_prefix_dest_number@provider_ip:5060 SIP/2.0 Via: SIP/2.0/UDP asterisk_ip:5060;branch=z9hG4bK79b639d0;rport From: "Gaspar Zoltan" ;tag=as2f9caf85 To: ;tag=73936DA0-1ACD Contact: Call-ID: 20a155c3054ed7a974bf482719e640ba@asterisk_ip CSeq: 104 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY X-asterisk-info: SIP re-invite (RTP bridge) Content-Type: application/sdp Content-Length: 158 v=0 o=root 31139 31141 IN IP4 phone_ip s=session c=IN IP4 phone_ip t=0 0 m=audio 60440 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - --- kc1*CLI> <-- SIP read from provider_ip:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP asterisk_ip:5060;branch=z9hG4bK79b639d0;rport From: "Gaspar Zoltan" ;tag=as2f9caf85 To: ;tag=73936DA0-1ACD Date: Thu, 07 Jun 2007 08:26:36 GMT Call-ID: 20a155c3054ed7a974bf482719e640ba@asterisk_ip Server: Cisco-SIPGateway/IOS-12.x CSeq: 104 INVITE Allow-Events: telephone-event Content-Length: 0 --- (10 headers 0 lines) --- kc1*CLI> <-- SIP read from provider_ip:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP asterisk_ip:5060;branch=z9hG4bK79b639d0;rport From: "Gaspar Zoltan" ;tag=as2f9caf85 To: ;tag=73936DA0-1ACD Date: Thu, 07 Jun 2007 08:26:36 GMT Call-ID: 20a155c3054ed7a974bf482719e640ba@asterisk_ip Server: Cisco-SIPGateway/IOS-12.x CSeq: 104 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER Allow-Events: telephone-event Contact: Content-Type: application/sdp Content-Length: 206 v=0 o=CiscoSystemsSIP-GW-UserAgent 3288 9305 IN IP4 provider_ip s=SIP Call c=IN IP4 provider_ip t=0 0 m=audio 21440 RTP/AVP 0 c=IN IP4 provider_ip a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - --- (13 headers 9 lines) --- Found RTP audio format 0 Peer audio RTP is at port provider_ip:21440 Found description format PCMU Capabilities: us - 0x104 (ulaw|g729), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) set_destination: Parsing for address/port to send to set_destination: set destination to provider_ip, port 5060 Transmitting (NAT) to provider_ip:5060: ACK sip:tech_prefix_dest_number@provider_ip:5060 SIP/2.0 Via: SIP/2.0/UDP asterisk_ip:5060;branch=z9hG4bK57bd476f;rport From: "Gaspar Zoltan" ;tag=as2f9caf85 To: ;tag=73936DA0-1ACD Contact: Call-ID: 20a155c3054ed7a974bf482719e640ba@asterisk_ip CSeq: 104 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- kc1*CLI> <-- SIP read from phone_ip:59568: --- (0 headers 0 lines) Nat keepalive --- set_destination: Parsing for address/port to send to set_destination: set destination to phone_ip, port 52236 We're at asterisk_ip port 18930 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP 13 headers, 10 lines Reliably Transmitting (NAT) to phone_ip:52236: INVITE sip:45e2b8014cddb47a9cfd3cf34d3d6db4@phone_ip:52236 SIP/2.0 Via: SIP/2.0/UDP asterisk_ip:5060;branch=z9hG4bK2b28bf3b;rport From: ;tag=as00fa36c8 To: "Gaspar Zoltan" ;tag=b90af9deb3a01c3fo0 Contact: Call-ID: b969158a-f2ad483@192.168.0.3 CSeq: 103 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY X-asterisk-info: SIP re-invite (RTP bridge) Content-Type: application/sdp Content-Length: 218 v=0 o=root 31139 31141 IN IP4 asterisk_ip s=session c=IN IP4 asterisk_ip t=0 0 m=audio 18930 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- set_destination: Parsing for address/port to send to set_destination: set destination to provider_ip, port 5060 We're at asterisk_ip port 19356 Adding codec 0x4 (ulaw) to SDP 13 headers, 8 lines Reliably Transmitting (NAT) to provider_ip:5060: INVITE sip:tech_prefix_dest_number@provider_ip:5060 SIP/2.0 Via: SIP/2.0/UDP asterisk_ip:5060;branch=z9hG4bK2dff5d13;rport From: "Gaspar Zoltan" ;tag=as2f9caf85 To: ;tag=73936DA0-1ACD Contact: Call-ID: 20a155c3054ed7a974bf482719e640ba@asterisk_ip CSeq: 105 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY X-asterisk-info: SIP re-invite (RTP bridge) Content-Type: application/sdp Content-Length: 162 v=0 o=root 31139 31142 IN IP4 asterisk_ip s=session c=IN IP4 asterisk_ip t=0 0 m=audio 19356 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - --- Scheduling destruction of call '20a155c3054ed7a974bf482719e640ba@asterisk_ip' in 32000 ms kc1*CLI> <-- SIP read from provider_ip:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP asterisk_ip:5060;branch=z9hG4bK2dff5d13;rport From: "Gaspar Zoltan" ;tag=as2f9caf85 To: ;tag=73936DA0-1ACD Date: Thu, 07 Jun 2007 08:26:55 GMT Call-ID: 20a155c3054ed7a974bf482719e640ba@asterisk_ip Server: Cisco-SIPGateway/IOS-12.x CSeq: 105 INVITE Allow-Events: telephone-event Content-Length: 0 --- (10 headers 0 lines) --- set_destination: Parsing for address/port to send to set_destination: set destination to provider_ip, port 5060 Reliably Transmitting (NAT) to provider_ip:5060: BYE sip:tech_prefix_dest_number@provider_ip:5060 SIP/2.0 Via: SIP/2.0/UDP asterisk_ip:5060;branch=z9hG4bK018d33ef;rport From: "Gaspar Zoltan" ;tag=as2f9caf85 To: ;tag=73936DA0-1ACD Call-ID: 20a155c3054ed7a974bf482719e640ba@asterisk_ip CSeq: 106 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Scheduling destruction of call '20a155c3054ed7a974bf482719e640ba@asterisk_ip' in 32000 ms <-- SIP read from provider_ip:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP asterisk_ip:5060;branch=z9hG4bK2dff5d13;rport From: "Gaspar Zoltan" ;tag=as2f9caf85 To: ;tag=73936DA0-1ACD Date: Thu, 07 Jun 2007 08:26:55 GMT Call-ID: 20a155c3054ed7a974bf482719e640ba@asterisk_ip Server: Cisco-SIPGateway/IOS-12.x CSeq: 105 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER Allow-Events: telephone-event Contact: Content-Type: application/sdp Content-Length: 206 v=0 o=CiscoSystemsSIP-GW-UserAgent 3288 9306 IN IP4 provider_ip s=SIP Call c=IN IP4 provider_ip t=0 0 m=audio 21440 RTP/AVP 0 c=IN IP4 provider_ip a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - --- (13 headers 9 lines) --- Found RTP audio format 0 Peer audio RTP is at port provider_ip:21440 Found description format PCMU Capabilities: us - 0x104 (ulaw|g729), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) set_destination: Parsing for address/port to send to set_destination: set destination to provider_ip, port 5060 Transmitting (NAT) to provider_ip:5060: ACK sip:tech_prefix_dest_number@provider_ip:5060 SIP/2.0 Via: SIP/2.0/UDP asterisk_ip:5060;branch=z9hG4bK76702de1;rport From: "Gaspar Zoltan" ;tag=as2f9caf85 To: ;tag=73936DA0-1ACD Contact: Call-ID: 20a155c3054ed7a974bf482719e640ba@asterisk_ip CSeq: 105 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- kc1*CLI> <-- SIP read from provider_ip:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP asterisk_ip:5060;branch=z9hG4bK018d33ef;rport From: "Gaspar Zoltan" ;tag=as2f9caf85 To: ;tag=73936DA0-1ACD Date: Thu, 07 Jun 2007 08:26:55 GMT Call-ID: 20a155c3054ed7a974bf482719e640ba@asterisk_ip Server: Cisco-SIPGateway/IOS-12.x Content-Length: 0 CSeq: 106 BYE --- (9 headers 0 lines) --- Destroying call '20a155c3054ed7a974bf482719e640ba@asterisk_ip' kc1*CLI> <-- SIP read from phone_ip:52236: SIP/2.0 200 OK To: "Gaspar Zoltan" ;tag=b90af9deb3a01c3fo0 From: ;tag=as00fa36c8 Call-ID: b969158a-f2ad483@192.168.0.3 CSeq: 103 INVITE Via: SIP/2.0/UDP asterisk_ip:5060;branch=z9hG4bK2b28bf3b Contact: "Gaspar Zoltan" Server: Sipura/SPA841-3.1.4(a) Content-Length: 204 Content-Type: application/sdp v=0 o=- 251349 251349 IN IP4 phone_ip s=- c=IN IP4 phone_ip t=0 0 m=audio 60440 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv --- (10 headers 11 lines) --- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port phone_ip:60440 Found description format PCMU Found description format telephone-event Capabilities: us - 0x104 (ulaw|g729), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) set_destination: Parsing for address/port to send to set_destination: set destination to phone_ip, port 52236 Transmitting (NAT) to phone_ip:52236: ACK sip:45e2b8014cddb47a9cfd3cf34d3d6db4@phone_ip:52236 SIP/2.0 Via: SIP/2.0/UDP asterisk_ip:5060;branch=z9hG4bK6bee9513;rport From: ;tag=as00fa36c8 To: "Gaspar Zoltan" ;tag=b90af9deb3a01c3fo0 Contact: Call-ID: b969158a-f2ad483@192.168.0.3 CSeq: 103 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Jun 7 04:27:00 WARNING[31665]: file.c:587 ast_readaudio_callback: Failed to write frame -- Playing 'digits/4' (language 'en') -- AGI Script exec_hangup.php completed, returning 0 -- Executing Hangup("SIP/45e2b8014cddb47a9cfd3cf34d3d6db4-08577918", "") in new stack == Spawn extension (default, h, 2) exited non-zero on 'SIP/45e2b8014cddb47a9cfd3cf34d3d6db4-08577918' Scheduling destruction of call 'b969158a-f2ad483@192.168.0.3' in 32000 ms set_destination: Parsing for address/port to send to set_destination: set destination to phone_ip, port 52236 Reliably Transmitting (NAT) to phone_ip:52236: BYE sip:45e2b8014cddb47a9cfd3cf34d3d6db4@phone_ip:52236 SIP/2.0 Via: SIP/2.0/UDP asterisk_ip:5060;branch=z9hG4bK124300f4;rport From: ;tag=as00fa36c8 To: "Gaspar Zoltan" ;tag=b90af9deb3a01c3fo0 Call-ID: b969158a-f2ad483@192.168.0.3 CSeq: 104 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- kc1*CLI> <-- SIP read from phone_ip:52236: SIP/2.0 200 OK To: "Gaspar Zoltan" ;tag=b90af9deb3a01c3fo0 From: ;tag=as00fa36c8 Call-ID: b969158a-f2ad483@192.168.0.3 CSeq: 104 BYE Via: SIP/2.0/UDP asterisk_ip:5060;branch=z9hG4bK124300f4 Server: Sipura/SPA841-3.1.4(a) Content-Length: 0 --- (8 headers 0 lines) --- Destroying call 'b969158a-f2ad483@192.168.0.3' kc1*CLI>