<-- SIP read from my.home.public.ip:2051: INVITE sip:14169671111@my.server.com;user=phone SIP/2.0 Via: SIP/2.0/UDP my.home.public.ip:2051;branch=z9hG4bK-fsodav1arm2p;rport From: "Nabeel" ;tag=echuketgez To: Call-ID: 3c27a32f975e-trhwxc53m21t@snom320-000413240230 CSeq: 1 INVITE Max-Forwards: 70 Contact: ;flow-id=1 P-Key-Flags: keys="3" User-Agent: snom320/5.2 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces, callerid Session-Expires: 3600;refresher=uas Content-Type: application/sdp Content-Length: 574 v=0 o=root 586024269 586024269 IN IP4 my.home.public.ip s=call c=IN IP4 my.home.public.ip t=0 0 m=audio 64266 RTP/AVP 0 8 9 2 3 18 4 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:FHo83d5cYdrIXXJUnkfFjJ5Zk1cpVYHTgx4jKCXt a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:18 g729/8000 a=rtpmap:4 g723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=encryption:optional a=alt:1 1.0 : user 9kksj== my.home.public.ip 64266 a=alt:2 0.9 : user 9kksj== my.device.private.ip 64266 a=sendrecv --- (17 headers 21 lines)--- Using INVITE request as basis request - 3c27a32f975e-trhwxc53m21t@snom320-000413240230 Sending to my.home.public.ip : 2051 (non-NAT) Reliably Transmitting (NAT) to my.home.public.ip:2051: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP my.home.public.ip:2051;branch=z9hG4bK-fsodav1arm2p;received=my.home.public.ip;rport=2051 From: "Nabeel" ;tag=echuketgez To: ;tag=as3f0d1652 Call-ID: 3c27a32f975e-trhwxc53m21t@snom320-000413240230 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70 Contact: Proxy-Authenticate: Digest realm="asterisk", nonce="14301f04" Content-Length: 0 --- Scheduling destruction of call '3c27a32f975e-trhwxc53m21t@snom320-000413240230' in 15000 ms Found user 'test' phone1*CLI> <-- SIP read from my.home.public.ip:2051: ACK sip:14169671111@my.server.com;user=phone SIP/2.0 Via: SIP/2.0/UDP my.home.public.ip:2051;branch=z9hG4bK-fsodav1arm2p;rport From: "Nabeel" ;tag=echuketgez To: ;tag=as3f0d1652 Call-ID: 3c27a32f975e-trhwxc53m21t@snom320-000413240230 CSeq: 1 ACK Max-Forwards: 70 Contact: ;flow-id=1 Content-Length: 0 --- (9 headers 0 lines)--- phone1*CLI> <-- SIP read from my.home.public.ip:2051: INVITE sip:14169671111@my.server.com;user=phone SIP/2.0 Via: SIP/2.0/UDP my.home.public.ip:2051;branch=z9hG4bK-7k1e70a3oh3p;rport From: "Nabeel" ;tag=echuketgez To: Call-ID: 3c27a32f975e-trhwxc53m21t@snom320-000413240230 CSeq: 2 INVITE Max-Forwards: 70 Contact: ;flow-id=1 P-Key-Flags: keys="3" User-Agent: snom320/5.2 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces, callerid Session-Expires: 3600;refresher=uas Proxy-Authorization: Digest username="test",realm="asterisk",nonce="14301f04",uri="sip:14169671111@my.server.com;user=phone",response="a ea7372e6876e278629b1dd837bc9c3b",algorithm=md5 Content-Type: application/sdp Content-Length: 574 v=0 o=root 586024269 586024269 IN IP4 my.home.public.ip s=call c=IN IP4 my.home.public.ip t=0 0 m=audio 64266 RTP/AVP 0 8 9 2 3 18 4 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:FHo83d5cYdrIXXJUnkfFjJ5Zk1cpVYHTgx4jKCXt a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:18 g729/8000 a=rtpmap:4 g723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=encryption:optional a=alt:1 1.0 : user 9kksj== my.home.public.ip 64266 a=alt:2 0.9 : user 9kksj== my.device.private.ip 64266 a=sendrecv> --- (18 headers 21 lines)--- Using INVITE request as basis request - 3c27a32f975e-trhwxc53m21t@snom320-000413240230 Sending to my.home.public.ip : 2051 (NAT) Found user 'test' Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 9 Found RTP audio format 2 Found RTP audio format 3 Found RTP audio format 18 Found RTP audio format 4 Found RTP audio format 101 Peer audio RTP is at port my.home.public.ip:64266 Found description format pcmu Found description format pcma Found description format g722 Found description format g726-32 Found description format gsm Found description format g729 Found description format g723 Found description format telephone-event Capabilities: us - 0x4 (ulaw), peer - audio=0x11f (g723|gsm|ulaw|alaw|g726|g729)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for 14169671111 in test (domain my.server.com) list_route: hop: Transmitting (NAT) to my.home.public.ip:2051: SIP/2.0 100 Trying Via: SIP/2.0/UDP my.home.public.ip:2051;branch=z9hG4bK-7k1e70a3oh3p;received=my.home.public.ip;rport=2051 From: "Nabeel" ;tag=echuketgez To: Call-ID: 3c27a32f975e-trhwxc53m21t@snom320-000413240230 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70 Contact: Content-Length: 0 --- -- Executing Goto("SIP/test-34fa", "pri-out|4169671111|1") in new stack -- Goto (pri-out,4169671111,1) -- Executing Set("SIP/test-34fa", "GROUP=1000") in new stack -- Executing GotoIf("SIP/test-34fa", "0?BLOCK") in new stack -- Executing Dial("SIP/test-34fa", "ZAP/g1/4169671111") in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called g1/4169671111 We're at 204.14.18.145 port 14186 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Transmitting (NAT) to my.home.public.ip:2051: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP my.home.public.ip:2051;branch=z9hG4bK-7k1e70a3oh3p;received=my.home.public.ip;rport=2051 From: "Nabeel" ;tag=echuketgez To: ;tag=as6e899773 Call-ID: 3c27a32f975e-trhwxc53m21t@snom320-000413240230 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70 Contact: Content-Type: application/sdp Content-Length: 218 v=0 o=root 30438 30438 IN IP4 204.14.18.145 s=session c=IN IP4 204.14.18.145 t=0 0 m=audio 14186 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Zap/1-1 is proceeding passing it to SIP/test-34fa -- Zap/1-1 is making progress passing it to SIP/test-34fa -- Zap/1-1 answered SIP/test-34fa We're at 204.14.18.145 port 14186 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to my.home.public.ip:2051: SIP/2.0 200 OK Via: SIP/2.0/UDP my.home.public.ip:2051;branch=z9hG4bK-7k1e70a3oh3p;received=my.home.public.ip;rport=2051 From: "Nabeel" ;tag=echuketgez To: ;tag=as6e899773 Call-ID: 3c27a32f975e-trhwxc53m21t@snom320-000413240230 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70 Contact: Content-Type: application/sdp Content-Length: 218 v=0 o=root 30438 30439 IN IP4 204.14.18.145 s=session c=IN IP4 204.14.18.145 t=0 0 m=audio 14186 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- phone1*CLI> <-- SIP read from my.home.public.ip:2051: ACK sip:14169671111@204.14.18.145 SIP/2.0 Via: SIP/2.0/UDP my.home.public.ip:2051;branch=z9hG4bK-7vogu0vmo8g3;rport From: "Nabeel" ;tag=echuketgez To: ;tag=as6e899773 Call-ID: 3c27a32f975e-trhwxc53m21t@snom320-000413240230 CSeq: 2 ACK Max-Forwards: 70 Contact: ;flow-id=1 Content-Length: 0 --- (9 headers 0 lines)--- phone1*CLI> <-- SIP read from my.home.public.ip:2051: INVITE sip:14169671111@204.14.18.145 SIP/2.0 Via: SIP/2.0/UDP my.home.public.ip:2051;branch=z9hG4bK-trefd36ho5p8;rport From: "Nabeel" ;tag=echuketgez To: ;tag=as6e899773 Call-ID: 3c27a32f975e-trhwxc53m21t@snom320-000413240230 CSeq: 3 INVITE Max-Forwards: 70 Contact: ;flow-id=1 P-Key-Flags: keys="3" User-Agent: snom320/5.2 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces, callerid Session-Expires: 3600;refresher=uas Content-Type: application/sdp Content-Length: 567 v=0 o=root 586024269 586024270 IN IP4 my.home.public.ip s=call c=IN IP4 0.0.0.0 t=0 0 m=audio 64266 RTP/AVP 0 8 9 2 3 18 4 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:FHo83d5cYdrIXXJUnkfFjJ5Zk1cpVYHTgx4jKCXt a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:18 g729/8000 a=rtpmap:4 g723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=encryption:optional a=alt:1 1.0 : user 9kksj== my.home.public.ip 64266 a=alt:2 0.9 : user 9kksj== my.device.private.ip 64266 a=sendonly --- (17 headers 21 lines)--- Using INVITE request as basis request - 3c27a32f975e-trhwxc53m21t@snom320-000413240230 Sending to my.home.public.ip : 2051 (NAT) Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 9 Found RTP audio format 2 Found RTP audio format 3 Found RTP audio format 18 Found RTP audio format 4 Found RTP audio format 101 Peer audio RTP is at port 0.0.0.0:64266 Found description format pcmu Found description format pcma Found description format g722 Found description format g726-32 Found description format gsm Found description format g729 Found description format g723 Found description format telephone-event Capabilities: us - 0x4 (ulaw), peer - audio=0x11f (g723|gsm|ulaw|alaw|g726|g729)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) -- Started music on hold, class 'default', on channel 'Zap/1-1' We're at 204.14.18.145 port 14186 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to my.home.public.ip:2051: SIP/2.0 200 OK Via: SIP/2.0/UDP my.home.public.ip:2051;branch=z9hG4bK-trefd36ho5p8;received=my.home.public.ip;rport=2051 From: "Nabeel" ;tag=echuketgez To: ;tag=as6e899773 Call-ID: 3c27a32f975e-trhwxc53m21t@snom320-000413240230 CSeq: 3 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70 Contact: Content-Type: application/sdp Content-Length: 218 v=0 o=root 30438 30440 IN IP4 204.14.18.145 s=session c=IN IP4 204.14.18.145 t=0 0 m=audio 14186 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- phone1*CLI> <-- SIP read from my.home.public.ip:2051: ACK sip:14169671111@204.14.18.145 SIP/2.0 Via: SIP/2.0/UDP my.home.public.ip:2051;branch=z9hG4bK-zcwgf5s71gnf;rport From: "Nabeel" ;tag=echuketgez To: ;tag=as6e899773 Call-ID: 3c27a32f975e-trhwxc53m21t@snom320-000413240230 CSeq: 3 ACK Max-Forwards: 70 Contact: ;flow-id=1 Content-Length: 0 --- (9 headers 0 lines)--- -- Stopped music on hold on Zap/1-1 phone1*CLI> <-- SIP read from my.home.public.ip:2051: REFER sip:14169671111@204.14.18.145 SIP/2.0 Via: SIP/2.0/UDP my.home.public.ip:2051;branch=z9hG4bK-02l8u1b7c96q;rport From: "Nabeel" ;tag=echuketgez To: ;tag=as6e899773 Call-ID: 3c27a32f975e-trhwxc53m21t@snom320-000413240230 CSeq: 4 REFER Max-Forwards: 70 Contact: ;flow-id=1 Refer-To: sip:14169671111@my.server.com;user=phone Referred-By: sip:test@my.server.com User-Agent: snom320/5.2 Content-Length: 0 --- (12 headers 0 lines)--- Transfer to 14169671111 in test Transfer from test in test Transmitting (NAT) to my.home.public.ip:2051: SIP/2.0 202 Accepted Via: SIP/2.0/UDP my.home.public.ip:2051;branch=z9hG4bK-02l8u1b7c96q;received=my.home.public.ip;rport=2051 From: "Nabeel" ;tag=echuketgez To: ;tag=as6e899773 Call-ID: 3c27a32f975e-trhwxc53m21t@snom320-000413240230 CSeq: 4 REFER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70 Contact: Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing --- set_destination: Parsing for address/port to send to set_destination: set destination to my.home.public.ip, port 2051 Reliably Transmitting (NAT) to my.home.public.ip:2051: NOTIFY sip:test@my.home.public.ip:2051 SIP/2.0 Via: SIP/2.0/UDP 204.14.18.145:5060;branch=z9hG4bK15b13cb4;rport From: ;tag=as6e899773 To: "Nabeel" ;tag=echuketgez Contact: Call-ID: 3c27a32f975e-trhwxc53m21t@snom320-000413240230 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: refer;id=4 Subscription-state: terminated;reason=noresource Content-Type: message/sipfrag;version=2.0 Content-Length: 14 SIP/2.0 200 OK --- -- Executing Goto("Zap/1-1", "pri-out|4169671111|1") in new stack -- Goto (pri-out,4169671111,1) set_destination: Parsing for address/port to send to -- Executing Set("Zap/1-1", "GROUP=") in new stack set_destination: set destination to my.home.public.ip, port 2051 Reliably Transmitting (NAT) to my.home.public.ip:2051: BYE sip:test@my.home.public.ip:2051 SIP/2.0 Via: SIP/2.0/UDP 204.14.18.145:5060;branch=z9hG4bK715c55e2;rport From: ;tag=as6e899773 To: "Nabeel" ;tag=echuketgez Contact: Call-ID: 3c27a32f975e-trhwxc53m21t@snom320-000413240230 CSeq: 103 BYE User-Agent: Asterisk PBX Max-Forwards: 70 X-Asterisk-HangupCause: Normal Clearing Content-Length: 0 --- Feb 5 13:03:52 WARNING[30449]: ast_expr2.fl:176 ast_yyerror: ast_yyerror(): syntax error: syntax error, unexpected $end, expecting TOK_MINUS or TOK_COMPL or TOK_LP or TOKEN; Input: 0 > ^ Feb 5 13:03:52 WARNING[30449]: ast_expr2.fl:180 ast_yyerror: If you have questions, please refer to doc/README.variables in the asterisk sour ce. -- Executing GotoIf("Zap/1-1", "0?BLOCK") in new stack -- Executing Dial("Zap/1-1", "ZAP/g1/4169671111") in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called g1/4169671111 phone1*CLI> <-- SIP read from my.home.public.ip:2051: SIP/2.0 200 Ok Via: SIP/2.0/UDP 204.14.18.145:5060;branch=z9hG4bK15b13cb4;rport=5060 From: ;tag=as6e899773 To: "Nabeel" ;tag=echuketgez Call-ID: 3c27a32f975e-trhwxc53m21t@snom320-000413240230 CSeq: 102 NOTIFY Contact: ;flow-id=1 Content-Length: 0 --- (8 headers 0 lines)--- phone1*CLI> <-- SIP read from my.home.public.ip:2051: SIP/2.0 200 OK Via: SIP/2.0/UDP 204.14.18.145:5060;branch=z9hG4bK715c55e2;rport=5060 From: ;tag=as6e899773 To: "Nabeel" ;tag=echuketgez Call-ID: 3c27a32f975e-trhwxc53m21t@snom320-000413240230 CSeq: 103 BYE Contact: ;flow-id=1 User-Agent: snom320/5.2 Content-Length: 0 --- (9 headers 0 lines)--- Destroying call '3c27a32f975e-trhwxc53m21t@snom320-000413240230' -- Zap/2-1 is proceeding passing it to Zap/1-1 -- Zap/2-1 is making progress passing it to Zap/1-1 -- Zap/2-1 answered Zap/1-1 -- Attempting native bridge of Zap/1-1 and Zap/2-1 -- Channel 0/1, span 1 got hangup request -- Hungup 'Zap/2-1'ls -- Hungup 'Zap/1-1'ls