AGI Rx << EXEC Dial SIP/MySIPGateway/0119198807xxxxx) -- AGI Script Executing Application: (Dial) Options: (SIP/MySIPGateway/0119198807xxxxx) Audio is at 10.0.0.177 port 15836 Adding codec 0x4 (ulaw) to SDP Reliably Transmitting (no NAT) to 204.aaa.bbb.ccc:5060: INVITE sip:0119198807xxxxx@204.aaa.bbb.ccc SIP/2.0 Via: SIP/2.0/UDP 10.0.0.177:5060;branch=z9hG4bK67a4a901;rport From: "+9198807xxxxx" ;tag=as1fefb70e To: Contact: Call-ID: 3a456e6144e517df688470907ac65dc9@10.0.0.177 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 22 Apr 2008 11:28:04 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 180 v=0 o=root 26994 26994 IN IP4 10.0.0.177 s=session c=IN IP4 10.0.0.177 t=0 0 m=audio 15836 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called MySIPGateway/0119198807xxxxx *CLI> <--- SIP read from 204.aaa.bbb.ccc:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.0.0.177:5060;received=10.0.0.177;branch=z9hG4bK67a4a901;rport=5060 From: "+9198807xxxxx" ;tag=as1fefb70e To: Call-ID: 3a456e6144e517df688470907ac65dc9@10.0.0.177 CSeq: 102 INVITE <-------------> <--- SIP read from 204.aaa.bbb.ccc:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 10.0.0.177:5060;received=10.0.0.177;branch=z9hG4bK67a4a901;rport=5060 From: "+9198807xxxxx" ;tag=as1fefb70e To: ;tag=SDj6a1999-cf20003-0-1611320837 Call-ID: 3a456e6144e517df688470907ac65dc9@10.0.0.177 CSeq: 102 INVITE Server: DC-SIP/1.2 Contact: Content-Type: application/sdp Content-Length: 199 v=0 o=CiscoSystemsSIP-GW-UserAgent 976 8765 IN IP4 204.aaa.bbb.ccc s=SIP Call c=IN IP4 204.aaa.bbb.ccc t=0 0 m=audio 10776 RTP/AVP 0 c=IN IP4 204.aaa.bbb.ccc a=rtpmap:0 PCMU/8000 a=ptime:20 <-------------> --- (10 headers 9 lines) --- Found RTP audio format 0 Peer audio RTP is at port 204.aaa.bbb.ccc:10776 Found description format PCMU for ID 0 Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 204.aaa.bbb.ccc:10776 -- SIP/MySIPGateway-098090a0 is making progress passing it to Local/0119198807xxxxx@c2conf_local-6b86,2 *CLI> <--- SIP read from 204.aaa.bbb.ccc:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.177:5060;received=10.0.0.177;branch=z9hG4bK67a4a901;rport=5060 From: "+9198807xxxxx" ;tag=as1fefb70e To: ;tag=SDj6a1999-cf20003-0-1611320837 Call-ID: 3a456e6144e517df688470907ac65dc9@10.0.0.177 CSeq: 102 INVITE Server: DC-SIP/1.2 Contact: Content-Type: application/sdp Content-Length: 199 v=0 o=CiscoSystemsSIP-GW-UserAgent 976 8765 IN IP4 204.aaa.bbb.ccc s=SIP Call c=IN IP4 204.aaa.bbb.ccc t=0 0 m=audio 10776 RTP/AVP 0 c=IN IP4 204.aaa.bbb.ccc a=rtpmap:0 PCMU/8000 a=ptime:20 <-------------> --- (10 headers 9 lines) --- Found RTP audio format 0 Peer audio RTP is at port 204.aaa.bbb.ccc:10776 Found description format PCMU for ID 0 Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 204.aaa.bbb.ccc:10776 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 204.aaa.bbb.ccc, port 5060 Transmitting (no NAT) to 204.aaa.bbb.ccc:5060: ACK sip:0119198807xxxxx@204.aaa.bbb.ccc:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.0.0.177:5060;branch=z9hG4bK343a317b;rport From: "+9198807xxxxx" ;tag=as1fefb70e To: ;tag=SDj6a1999-cf20003-0-1611320837 Contact: Call-ID: 3a456e6144e517df688470907ac65dc9@10.0.0.177 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- -- SIP/MySIPGateway-098090a0 answered Local/0119198807xxxxx@c2conf_local-6b86,2 *CLI> AGI Rx << SET CALLERID +9198807xxxxx AGI Tx >> 200 result=1 AGI Rx << EXEC Dial SIP/MySIPGateway/31927xxxxx||D(wwww567890#) -- AGI Script Executing Application: (Dial) Options: (SIP/MySIPGateway/31927xxxxx||D(wwww567890#)) Audio is at 10.0.0.177 port 10826 Adding codec 0x4 (ulaw) to SDP Reliably Transmitting (no NAT) to 204.aaa.bbb.ccc:5060: INVITE sip:31927xxxxx@204.aaa.bbb.ccc SIP/2.0 Via: SIP/2.0/UDP 10.0.0.177:5060;branch=z9hG4bK725c9872;rport From: "+9198807xxxxx" ;tag=as0b828a4d To: Contact: Call-ID: 07ce954309a558403b97244f2f78f487@10.0.0.177 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 22 Apr 2008 11:28:17 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 180 v=0 o=root 26994 26994 IN IP4 10.0.0.177 s=session c=IN IP4 10.0.0.177 t=0 0 m=audio 10826 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called MySIPGateway/31927xxxxx <--- SIP read from 204.aaa.bbb.ccc:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.0.0.177:5060;received=10.0.0.177;branch=z9hG4bK725c9872;rport=5060 From: "+9198807xxxxx" ;tag=as0b828a4d To: Call-ID: 07ce954309a558403b97244f2f78f487@10.0.0.177 CSeq: 102 INVITE <-------------> --- (6 headers 0 lines) --- *CLI> <--- SIP read from 204.aaa.bbb.ccc:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.0.0.177:5060;received=10.0.0.177;branch=z9hG4bK725c9872;rport=5060 From: "+9198807xxxxx" ;tag=as0b828a4d To: ;tag=SDtsbb699-3b8b0003-0-722939652 Call-ID: 07ce954309a558403b97244f2f78f487@10.0.0.177 CSeq: 102 INVITE Server: DC-SIP/1.2 Contact: Content-Length: 0 <-------------> --- (9 headers 0 lines) --- -- SIP/MySIPGateway-0980db20 is ringing set_destination: Parsing for address/port to send to set_destination: set destination to 204.aaa.bbb.ccc, port 5060 Audio is at 10.0.0.177 port 15836 Adding codec 0x4 (ulaw) to SDP Reliably Transmitting (no NAT) to 204.aaa.bbb.ccc:5060: INVITE sip:0119198807xxxxx@204.aaa.bbb.ccc:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.0.0.177:5060;branch=z9hG4bK63513d4b;rport From: "+9198807xxxxx" ;tag=as1fefb70e To: ;tag=SDj6a1999-cf20003-0-1611320837 Contact: Call-ID: 3a456e6144e517df688470907ac65dc9@10.0.0.177 CSeq: 103 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 180 v=0 o=root 26994 26995 IN IP4 10.0.0.177 s=session c=IN IP4 10.0.0.177 t=0 0 m=audio 15836 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- *CLI> <--- SIP read from 204.aaa.bbb.ccc:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.177:5060;received=10.0.0.177;branch=z9hG4bK725c9872;rport=5060 From: "+9198807xxxxx" ;tag=as0b828a4d To: ;tag=SDtsbb699-3b8b0003-0-722939652 Call-ID: 07ce954309a558403b97244f2f78f487@10.0.0.177 CSeq: 102 INVITE Server: DC-SIP/1.2 Contact: Content-Type: application/sdp Content-Length: 191 v=0 o=- 3417852927 3417852952 IN IP4 204.aaa.bbb.ccc s=- c=IN IP4 204.aaa.bbb.ccc t=0 0 m=audio 10784 RTP/AVP 0 a=sendrecv a=ptime:20 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - <-------------> --- (10 headers 10 lines) --- Found RTP audio format 0 Peer audio RTP is at port 204.aaa.bbb.ccc:10784 Found description format PCMU for ID 0 Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 204.aaa.bbb.ccc:10784 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 204.aaa.bbb.ccc, port 5060 Transmitting (no NAT) to 204.aaa.bbb.ccc:5060: ACK sip:31927xxxxx@204.aaa.bbb.ccc:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.0.0.177:5060;branch=z9hG4bK79febc8e;rport From: "+9198807xxxxx" ;tag=as0b828a4d To: ;tag=SDtsbb699-3b8b0003-0-722939652 Contact: Call-ID: 07ce954309a558403b97244f2f78f487@10.0.0.177 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- -- SIP/MySIPGateway-0980db20 answered SIP/MySIPGateway-098090a0 -- Sending DTMF 'wwww567890#' to the called party. <--- SIP read from 204.aaa.bbb.ccc:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.177:5060;received=10.0.0.177;branch=z9hG4bK63513d4b;rport=5060 From: "+9198807xxxxx" ;tag=as1fefb70e To: ;tag=SDj6a1999-cf20003-0-1611320837 Call-ID: 3a456e6144e517df688470907ac65dc9@10.0.0.177 CSeq: 103 INVITE Server: DC-SIP/1.2 Contact: Content-Type: application/sdp Content-Length: 226 v=0 o=CiscoSystemsSIP-GW-UserAgent 976 8766 IN IP4 204.aaa.bbb.ccc s=SIP Call c=IN IP4 204.aaa.bbb.ccc t=0 0 m=audio 10776 RTP/AVP 0 c=IN IP4 204.aaa.bbb.ccc a=rtpmap:0 PCMU/8000 a=ptime:20 a=silenceSupp:off - - - - <-------------> --- (10 headers 10 lines) --- Found RTP audio format 0 Peer audio RTP is at port 204.aaa.bbb.ccc:10776 Found description format PCMU for ID 0 Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 204.aaa.bbb.ccc:10776 set_destination: Parsing for address/port to send to set_destination: set destination to 204.aaa.bbb.ccc, port 5060 Transmitting (no NAT) to 204.aaa.bbb.ccc:5060: ACK sip:0119198807xxxxx@204.aaa.bbb.ccc:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.0.0.177:5060;branch=z9hG4bK1b16a56d;rport From: "+9198807xxxxx" ;tag=as1fefb70e To: ;tag=SDj6a1999-cf20003-0-1611320837 Contact: Call-ID: 3a456e6144e517df688470907ac65dc9@10.0.0.177 CSeq: 103 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 204.aaa.bbb.ccc, port 5060 Audio is at 10.0.0.177 port 15836 Adding codec 0x4 (ulaw) to SDP Reliably Transmitting (no NAT) to 204.aaa.bbb.ccc:5060: INVITE sip:0119198807xxxxx@204.aaa.bbb.ccc:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.0.0.177:5060;branch=z9hG4bK0ef76415;rport From: "+9198807xxxxx" ;tag=as1fefb70e To: ;tag=SDj6a1999-cf20003-0-1611320837 Contact: Call-ID: 3a456e6144e517df688470907ac65dc9@10.0.0.177 CSeq: 104 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 190 v=0 o=root 26994 26996 IN IP4 204.aaa.bbb.ccc s=session c=IN IP4 204.aaa.bbb.ccc t=0 0 m=audio 10784 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- <--- SIP read from 204.aaa.bbb.ccc:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.177:5060;received=10.0.0.177;branch=z9hG4bK0ef76415;rport=5060 From: "+9198807xxxxx" ;tag=as1fefb70e To: ;tag=SDj6a1999-cf20003-0-1611320837 Call-ID: 3a456e6144e517df688470907ac65dc9@10.0.0.177 CSeq: 104 INVITE Server: DC-SIP/1.2 Contact: Content-Type: application/sdp Content-Length: 226 v=0 o=CiscoSystemsSIP-GW-UserAgent 976 8767 IN IP4 204.aaa.bbb.ccc s=SIP Call c=IN IP4 204.aaa.bbb.ccc t=0 0 m=audio 10776 RTP/AVP 0 c=IN IP4 204.aaa.bbb.ccc a=rtpmap:0 PCMU/8000 a=ptime:20 a=silenceSupp:off - - - - <-------------> --- (10 headers 10 lines) --- Found RTP audio format 0 Peer audio RTP is at port 204.aaa.bbb.ccc:10776 Found description format PCMU for ID 0 Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 204.aaa.bbb.ccc:10776 set_destination: Parsing for address/port to send to set_destination: set destination to 204.aaa.bbb.ccc, port 5060 Transmitting (no NAT) to 204.aaa.bbb.ccc:5060: ACK sip:0119198807xxxxx@204.aaa.bbb.ccc:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.0.0.177:5060;branch=z9hG4bK4d2027f4;rport From: "+9198807xxxxx" ;tag=as1fefb70e To: ;tag=SDj6a1999-cf20003-0-1611320837 Contact: Call-ID: 3a456e6144e517df688470907ac65dc9@10.0.0.177 CSeq: 104 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- == Parsing '/etc/asterisk/manager.conf': Found == Manager 'ashwin' logged on from 127.0.0.1 == Manager 'ashwin' logged off from 127.0.0.1 -- In function 'ast_senddigit_begin'. -- In function 'ast_senddigit_begin'. -- In function 'ast_senddigit_begin'. -- In function 'ast_senddigit_begin'. -- In function 'ast_senddigit_begin'. -- In function 'ast_senddigit_begin'. -- In function 'ast_senddigit_begin'. -- In function 'ast_senddigit_begin'. -- Native bridging SIP/MySIPGateway-098090a0 and SIP/MySIPGateway-0980db20 set_destination: Parsing for address/port to send to set_destination: set destination to 204.aaa.bbb.ccc, port 5060 Audio is at 10.0.0.177 port 10826 Adding codec 0x4 (ulaw) to SDP Reliably Transmitting (no NAT) to 204.aaa.bbb.ccc:5060: INVITE sip:31927xxxxx@204.aaa.bbb.ccc:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.0.0.177:5060;branch=z9hG4bK28fdc606;rport From: "+9198807xxxxx" ;tag=as0b828a4d To: ;tag=SDtsbb699-3b8b0003-0-722939652 Contact: Call-ID: 07ce954309a558403b97244f2f78f487@10.0.0.177 CSeq: 103 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 190 v=0 o=root 26994 26995 IN IP4 204.aaa.bbb.ccc s=session c=IN IP4 204.aaa.bbb.ccc t=0 0 m=audio 10776 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- <--- SIP read from 204.aaa.bbb.ccc:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.177:5060;received=10.0.0.177;branch=z9hG4bK28fdc606;rport=5060 From: "+9198807xxxxx" ;tag=as0b828a4d To: ;tag=SDtsbb699-3b8b0003-0-722939652 Call-ID: 07ce954309a558403b97244f2f78f487@10.0.0.177 CSeq: 103 INVITE Server: DC-SIP/1.2 Contact: Content-Type: application/sdp Content-Length: 191 v=0 o=- 3417852932 3417853011 IN IP4 204.aaa.bbb.ccc s=- c=IN IP4 204.aaa.bbb.ccc t=0 0 m=audio 10784 RTP/AVP 0 a=sendrecv a=ptime:20 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - <-------------> --- (10 headers 10 lines) --- Found RTP audio format 0 Peer audio RTP is at port 204.aaa.bbb.ccc:10784 Found description format PCMU for ID 0 Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 204.aaa.bbb.ccc:10784 set_destination: Parsing for address/port to send to set_destination: set destination to 204.aaa.bbb.ccc, port 5060 Transmitting (no NAT) to 204.aaa.bbb.ccc:5060: ACK sip:31927xxxxx@204.aaa.bbb.ccc:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.0.0.177:5060;branch=z9hG4bK724c32ec;rport From: "+9198807xxxxx" ;tag=as0b828a4d To: ;tag=SDtsbb699-3b8b0003-0-722939652 Contact: Call-ID: 07ce954309a558403b97244f2f78f487@10.0.0.177 CSeq: 103 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- <--- SIP read from 204.aaa.bbb.ccc:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.177:5060;received=10.0.0.177;branch=z9hG4bK28fdc606;rport=5060 From: "+9198807xxxxx" ;tag=as0b828a4d To: ;tag=SDtsbb699-3b8b0003-0-722939652 Call-ID: 07ce954309a558403b97244f2f78f487@10.0.0.177 CSeq: 103 INVITE Server: DC-SIP/1.2 Contact: Content-Type: application/sdp Content-Length: 191 v=0 o=- 3417852932 3417853011 IN IP4 204.aaa.bbb.ccc s=- c=IN IP4 204.aaa.bbb.ccc t=0 0 m=audio 10784 RTP/AVP 0 a=sendrecv a=ptime:20 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - <-------------> --- (10 headers 10 lines) --- Found RTP audio format 0 Peer audio RTP is at port 204.aaa.bbb.ccc:10784 Found description format PCMU for ID 0 Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 204.aaa.bbb.ccc:10784 set_destination: Parsing for address/port to send to set_destination: set destination to 204.aaa.bbb.ccc, port 5060 Transmitting (no NAT) to 204.aaa.bbb.ccc:5060: ACK sip:31927xxxxx@204.aaa.bbb.ccc:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.0.0.177:5060;branch=z9hG4bK3cfd4bfc;rport From: "+9198807xxxxx" ;tag=as0b828a4d To: ;tag=SDtsbb699-3b8b0003-0-722939652 Contact: Call-ID: 07ce954309a558403b97244f2f78f487@10.0.0.177 CSeq: 103 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 BYE sip:+9198807xxxxx@10.0.0.177 SIP/2.0 Via: SIP/2.0/UDP 204.aaa.bbb.ccc:5060;branch=z9hG4bK0o82jj100gagvo8fu6c1sd0000g00.1 Call-ID: 3a456e6144e517df688470907ac65dc9@10.0.0.177 From: ;tag=SDj6a1999-cf20003-0-1611320837 To: "+9198807xxxxx" ;tag=as1fefb70e CSeq: 1 BYE Content-Length: 0 Max-Forwards: 69 <-------------> --- (8 headers 0 lines) --- Sending to 204.aaa.bbb.ccc : 5060 (no NAT) <--- Transmitting (no NAT) to 204.aaa.bbb.ccc:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 204.aaa.bbb.ccc:5060;branch=z9hG4bK0o82jj100gagvo8fu6c1sd0000g00.1;received=204.aaa.bbb.ccc From: ;tag=SDj6a1999-cf20003-0-1611320837 To: "+9198807xxxxx" ;tag=as1fefb70e Call-ID: 3a456e6144e517df688470907ac65dc9@10.0.0.177 CSeq: 1 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> set_destination: Parsing for address/port to send to set_destination: set destination to 204.aaa.bbb.ccc, port 5060 Audio is at 10.0.0.177 port 10826 Adding codec 0x4 (ulaw) to SDP Reliably Transmitting (no NAT) to 204.aaa.bbb.ccc:5060: INVITE sip:31927xxxxx@204.aaa.bbb.ccc:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.0.0.177:5060;branch=z9hG4bK13db91f8;rport From: "+9198807xxxxx" ;tag=as0b828a4d To: ;tag=SDtsbb699-3b8b0003-0-722939652 Contact: Call-ID: 07ce954309a558403b97244f2f78f487@10.0.0.177 CSeq: 104 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 180 v=0 o=root 26994 26996 IN IP4 10.0.0.177 s=session c=IN IP4 10.0.0.177 t=0 0 m=audio 10826 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- Scheduling destruction of SIP dialog '07ce954309a558403b97244f2f78f487@10.0.0.177' in 32000 ms (Method: INVITE) Really destroying SIP dialog '3a456e6144e517df688470907ac65dc9@10.0.0.177' Method: BYE <--- SIP read from 204.aaa.bbb.ccc:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.177:5060;received=10.0.0.177;branch=z9hG4bK13db91f8;rport=5060 From: "+9198807xxxxx" ;tag=as0b828a4d To: ;tag=SDtsbb699-3b8b0003-0-722939652 Call-ID: 07ce954309a558403b97244f2f78f487@10.0.0.177 CSeq: 104 INVITE Server: DC-SIP/1.2 Contact: Content-Type: application/sdp Content-Length: 191 v=0 o=- 3417852942 3417853126 IN IP4 204.aaa.bbb.ccc s=- c=IN IP4 204.aaa.bbb.ccc t=0 0 m=audio 10784 RTP/AVP 0 a=sendrecv a=ptime:20 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - <-------------> --- (10 headers 10 lines) --- Found RTP audio format 0 Peer audio RTP is at port 204.aaa.bbb.ccc:10784 Found description format PCMU for ID 0 Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 204.aaa.bbb.ccc:10784 set_destination: Parsing for address/port to send to set_destination: set destination to 204.aaa.bbb.ccc, port 5060 Transmitting (no NAT) to 204.aaa.bbb.ccc:5060: ACK sip:31927xxxxx@204.aaa.bbb.ccc:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.0.0.177:5060;branch=z9hG4bK2fda11eb;rport From: "+9198807xxxxx" ;tag=as0b828a4d To: ;tag=SDtsbb699-3b8b0003-0-722939652 Contact: Call-ID: 07ce954309a558403b97244f2f78f487@10.0.0.177 CSeq: 104 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 204.aaa.bbb.ccc, port 5060 Reliably Transmitting (no NAT) to 204.aaa.bbb.ccc:5060: BYE sip:31927xxxxx@204.aaa.bbb.ccc:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.0.0.177:5060;branch=z9hG4bK4594b0e8;rport From: "+9198807xxxxx" ;tag=as0b828a4d To: ;tag=SDtsbb699-3b8b0003-0-722939652 Call-ID: 07ce954309a558403b97244f2f78f487@10.0.0.177 CSeq: 105 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Scheduling destruction of SIP dialog '07ce954309a558403b97244f2f78f487@10.0.0.177' in 32000 ms (Method: INVITE) <--- SIP read from 204.aaa.bbb.ccc:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.177:5060;received=10.0.0.177;branch=z9hG4bK4594b0e8;rport=5060 From: "+9198807xxxxx" ;tag=as0b828a4d To: ;tag=SDtsbb699-3b8b0003-0-722939652 Call-ID: 07ce954309a558403b97244f2f78f487@10.0.0.177 CSeq: 105 BYE Server: DC-SIP/1.2 Content-Length: 0