vagrant@asterisk13:~$ sudo asterisk -rvvvvvv Asterisk 13.3.2, Copyright (C) 1999 - 2014, Digium, Inc. and others. Created by Mark Spencer Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. ========================================================================= Connected to Asterisk 13.3.2 currently running on asterisk13 (pid = 8747) asterisk13*CLI> sip set debug on SIP Debugging enabled <--- SIP read from UDP:10.203.175.1:5060 ---> REGISTER sip:10.203.175.22 SIP/2.0 Via: SIP/2.0/UDP 10.203.175.1:5060;rport;branch=z9hG4bKPjxIsC7bLyLORCndntMCtbW9IvdKkXCNqp Max-Forwards: 70 From: ;tag=eowc9rshI31UpXWN4CR0Fd8rza1.aq4m To: Call-ID: v4IVJ1CI2Hj39OAhF9kK9NtObdJYoVE1 CSeq: 36288 REGISTER User-Agent: PJSUA v2.3 Darwin-14.3/x86_64 Contact: Expires: 300 Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Sending to 10.203.175.1:5060 (no NAT) Sending to 10.203.175.1:5060 (no NAT) <--- Transmitting (no NAT) to 10.203.175.1:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.203.175.1:5060;branch=z9hG4bKPjxIsC7bLyLORCndntMCtbW9IvdKkXCNqp;received=10.203.175.1;rport=5060 From: ;tag=eowc9rshI31UpXWN4CR0Fd8rza1.aq4m To: ;tag=as3cb77650 Call-ID: v4IVJ1CI2Hj39OAhF9kK9NtObdJYoVE1 CSeq: 36288 REGISTER Server: Asterisk PBX 13.3.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="64dfd815" Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'v4IVJ1CI2Hj39OAhF9kK9NtObdJYoVE1' in 32000 ms (Method: REGISTER) <--- SIP read from UDP:10.203.175.1:5060 ---> REGISTER sip:10.203.175.22 SIP/2.0 Via: SIP/2.0/UDP 10.203.175.1:5060;rport;branch=z9hG4bKPjnddaINTE8n8tD3UfFIckDRSqsG8PBAnm Max-Forwards: 70 From: ;tag=eowc9rshI31UpXWN4CR0Fd8rza1.aq4m To: Call-ID: v4IVJ1CI2Hj39OAhF9kK9NtObdJYoVE1 CSeq: 36289 REGISTER User-Agent: PJSUA v2.3 Darwin-14.3/x86_64 Contact: Expires: 300 Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Authorization: Digest username="userb", realm="asterisk", nonce="64dfd815", uri="sip:10.203.175.22", response="5427afad5ca24cda2a8b7993f3b4b5c9", algorithm=MD5 Content-Length: 0 <-------------> --- (13 headers 0 lines) --- Sending to 10.203.175.1:5060 (no NAT) -- Registered SIP 'userb' at 10.203.175.1:5060 > Saved useragent "PJSUA v2.3 Darwin-14.3/x86_64" for peer userb <--- Transmitting (no NAT) to 10.203.175.1:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.203.175.1:5060;branch=z9hG4bKPjnddaINTE8n8tD3UfFIckDRSqsG8PBAnm;received=10.203.175.1;rport=5060 From: ;tag=eowc9rshI31UpXWN4CR0Fd8rza1.aq4m To: ;tag=as3cb77650 Call-ID: v4IVJ1CI2Hj39OAhF9kK9NtObdJYoVE1 CSeq: 36289 REGISTER Server: Asterisk PBX 13.3.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Expires: 300 Contact: ;expires=300 Date: Tue, 26 May 2015 12:57:39 GMT Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'v4IVJ1CI2Hj39OAhF9kK9NtObdJYoVE1' in 32000 ms (Method: REGISTER) <--- SIP read from UDP:10.203.175.1:54191 ---> INVITE sip:123@10.203.175.22 SIP/2.0 Via: SIP/2.0/UDP 192.168.11.215:54191;rport;branch=z9hG4bKPjwnL2oWdLUYcPaTkEwYKtkXbtbGFgPUzg Max-Forwards: 70 From: "Luca Pradovera" ;tag=moOzaAjm6CrEUO8QBoP2AQNoZjZeXz3y To: Contact: Call-ID: IqoQ0P0y1J3vf7mPyDmRJA7uwlcaWuHH CSeq: 13233 INVITE Allow: SUBSCRIBE, NOTIFY, PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE, REFER Supported: 100rel, replaces, norefersub, gruu User-Agent: Blink Pro 4.2.1 (MacOSX) Content-Type: application/sdp Content-Length: 346 v=0 o=- 3641633873 3641633873 IN IP4 10.203.175.1 s=Blink Pro 4.2.1 (MacOSX) t=0 0 m=audio 50056 RTP/AVP 8 0 101 c=IN IP4 10.203.175.1 a=rtcp:50057 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=zrtp-hash:1.10 5bb8c0e2d97c3f0185e1b645af54d3ca9e8c4082e3bcba5c2844b692644d1de4 a=sendrecv <-------------> --- (13 headers 13 lines) --- Sending to 10.203.175.1:54191 (NAT) Sending to 10.203.175.1:54191 (NAT) Using INVITE request as basis request - IqoQ0P0y1J3vf7mPyDmRJA7uwlcaWuHH Found peer 'usera' for 'usera' from 10.203.175.1:54191 <--- Reliably Transmitting (NAT) to 10.203.175.1:54191 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.11.215:54191;branch=z9hG4bKPjwnL2oWdLUYcPaTkEwYKtkXbtbGFgPUzg;received=10.203.175.1;rport=54191 From: "Luca Pradovera" ;tag=moOzaAjm6CrEUO8QBoP2AQNoZjZeXz3y To: ;tag=as4f22895a Call-ID: IqoQ0P0y1J3vf7mPyDmRJA7uwlcaWuHH CSeq: 13233 INVITE Server: Asterisk PBX 13.3.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="009e3230" Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'IqoQ0P0y1J3vf7mPyDmRJA7uwlcaWuHH' in 32000 ms (Method: INVITE) <--- SIP read from UDP:10.203.175.1:54191 ---> ACK sip:123@10.203.175.22 SIP/2.0 Via: SIP/2.0/UDP 192.168.11.215:54191;rport;branch=z9hG4bKPjwnL2oWdLUYcPaTkEwYKtkXbtbGFgPUzg Max-Forwards: 70 From: "Luca Pradovera" ;tag=moOzaAjm6CrEUO8QBoP2AQNoZjZeXz3y To: ;tag=as4f22895a Call-ID: IqoQ0P0y1J3vf7mPyDmRJA7uwlcaWuHH CSeq: 13233 ACK User-Agent: Blink Pro 4.2.1 (MacOSX) Content-Length: 0 <-------------> --- (9 headers 0 lines) --- <--- SIP read from UDP:10.203.175.1:54191 ---> INVITE sip:123@10.203.175.22 SIP/2.0 Via: SIP/2.0/UDP 192.168.11.215:54191;rport;branch=z9hG4bKPjoDTYY-GwjrOHB3K9jt1z0-O.XJDBGteX Max-Forwards: 70 From: "Luca Pradovera" ;tag=moOzaAjm6CrEUO8QBoP2AQNoZjZeXz3y To: Contact: Call-ID: IqoQ0P0y1J3vf7mPyDmRJA7uwlcaWuHH CSeq: 13234 INVITE Allow: SUBSCRIBE, NOTIFY, PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE, REFER Supported: 100rel, replaces, norefersub, gruu User-Agent: Blink Pro 4.2.1 (MacOSX) Authorization: Digest username="usera", realm="asterisk", nonce="009e3230", uri="sip:123@10.203.175.22", response="250e969040dd98a32587c5a6311becc7", algorithm=MD5 Content-Type: application/sdp Content-Length: 346 v=0 o=- 3641633873 3641633873 IN IP4 10.203.175.1 s=Blink Pro 4.2.1 (MacOSX) t=0 0 m=audio 50056 RTP/AVP 8 0 101 c=IN IP4 10.203.175.1 a=rtcp:50057 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=zrtp-hash:1.10 5bb8c0e2d97c3f0185e1b645af54d3ca9e8c4082e3bcba5c2844b692644d1de4 a=sendrecv <-------------> --- (14 headers 13 lines) --- Sending to 10.203.175.1:54191 (NAT) Using INVITE request as basis request - IqoQ0P0y1J3vf7mPyDmRJA7uwlcaWuHH Found peer 'usera' for 'usera' from 10.203.175.1:54191 == Using SIP RTP CoS mark 5 Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 101 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Capabilities: us - (ulaw|alaw|gsm|h263), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 10.203.175.1:50056 Looking for 123 in outbound-monitor (domain 10.203.175.22) sip_route_dump: route/path hop: <--- Transmitting (NAT) to 10.203.175.1:54191 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.11.215:54191;branch=z9hG4bKPjoDTYY-GwjrOHB3K9jt1z0-O.XJDBGteX;received=10.203.175.1;rport=54191 From: "Luca Pradovera" ;tag=moOzaAjm6CrEUO8QBoP2AQNoZjZeXz3y To: Call-ID: IqoQ0P0y1J3vf7mPyDmRJA7uwlcaWuHH CSeq: 13234 INVITE Server: Asterisk PBX 13.3.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Length: 0 <------------> -- Executing [123@outbound-monitor:1] Answer("SIP/usera-00000000", "") in new stack Audio is at 15196 Adding codec ulaw to SDP Adding codec alaw to SDP Adding codec gsm to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (NAT) to 10.203.175.1:54191 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.11.215:54191;branch=z9hG4bKPjoDTYY-GwjrOHB3K9jt1z0-O.XJDBGteX;received=10.203.175.1;rport=54191 From: "Luca Pradovera" ;tag=moOzaAjm6CrEUO8QBoP2AQNoZjZeXz3y To: ;tag=as190f62dc Call-ID: IqoQ0P0y1J3vf7mPyDmRJA7uwlcaWuHH CSeq: 13234 INVITE Server: Asterisk PBX 13.3.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 288 v=0 o=root 1706769702 1706769702 IN IP4 10.203.175.22 s=Asterisk PBX 13.3.2 c=IN IP4 10.203.175.22 t=0 0 m=audio 15196 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv <------------> <--- SIP read from UDP:10.203.175.1:54191 ---> PUBLISH sip:usera@10.203.175.22 SIP/2.0 Via: SIP/2.0/UDP 192.168.11.215:54191;rport;branch=z9hG4bKPj2aCtFA-VnpyXh4.JaoiurphzIv0ytGJM Max-Forwards: 70 From: "Luca Pradovera" ;tag=8d1kSqu9UrdzpIdRQqIiuW8DSiyg43QI To: "Luca Pradovera" Call-ID: BJJ87s2B-DBJc182LlZ5WHu0IqF2nvP9 CSeq: 1 PUBLISH Event: presence Expires: 600 User-Agent: Blink Pro 4.2.1 (MacOSX) Content-Type: application/pidf+xml Content-Length: 1965 openbusyLuca PradoveraItaly/Borgaro%20TorineseLucas-MacBook-ProBlink Pro 4.2.1 (MacOSX)120truetruetruetruetruetruetrueactive6c954adc-02d4-4a0a-a617-47ff61cab4c3sip%3Ausera%4010.203.175.22On the Phone2015-05-26T14:57:53.308686+02:001202015-05-26T14:57:53.308686+02:006c954adc-02d4-4a0a-a617-47ff61cab4c3Blink Pro 4.2.1 (MacOSX) at Lucas-MacBook-Pro2015-05-26T14:57:53.308686+02:00 <-------------> --- (12 headers 2 lines) --- Sending to 10.203.175.1:54191 (NAT) <--- Transmitting (NAT) to 10.203.175.1:54191 ---> SIP/2.0 489 Bad Event Via: SIP/2.0/UDP 192.168.11.215:54191;branch=z9hG4bKPj2aCtFA-VnpyXh4.JaoiurphzIv0ytGJM;received=10.203.175.1;rport=54191 From: "Luca Pradovera" ;tag=8d1kSqu9UrdzpIdRQqIiuW8DSiyg43QI To: "Luca Pradovera" ;tag=as2ac7479c Call-ID: BJJ87s2B-DBJc182LlZ5WHu0IqF2nvP9 CSeq: 1 PUBLISH Server: Asterisk PBX 13.3.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <------------> Really destroying SIP dialog 'BJJ87s2B-DBJc182LlZ5WHu0IqF2nvP9' Method: PUBLISH <--- SIP read from UDP:10.203.175.1:54191 ---> ACK sip:123@10.203.175.22:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.11.215:54191;rport;branch=z9hG4bKPjUmJv17q8rW5Ul14.77l3L.GXtg2C24Ke Max-Forwards: 70 From: "Luca Pradovera" ;tag=moOzaAjm6CrEUO8QBoP2AQNoZjZeXz3y To: ;tag=as190f62dc Call-ID: IqoQ0P0y1J3vf7mPyDmRJA7uwlcaWuHH CSeq: 13234 ACK User-Agent: Blink Pro 4.2.1 (MacOSX) Content-Length: 0 <-------------> --- (9 headers 0 lines) --- > 0x7f0e5000d5d0 -- Probation passed - setting RTP source address to 10.203.175.1:50056 -- Executing [123@outbound-monitor:2] Monitor("SIP/usera-00000000", "wav,12345678,m") in new stack -- Executing [123@outbound-monitor:3] Dial("SIP/usera-00000000", "SIP/userb, 10") in new stack == Using SIP RTP CoS mark 5 Audio is at 17296 Adding codec ulaw to SDP Adding codec alaw to SDP Adding codec gsm to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 10.203.175.1:5060: INVITE sip:userb@10.203.175.1:5060;ob SIP/2.0 Via: SIP/2.0/UDP 10.203.175.22:5060;branch=z9hG4bK4df816bb Max-Forwards: 70 From: "User A" ;tag=as711a417a To: Contact: Call-ID: 5a457c525415ef9c70c1ac4470edda59@10.203.175.22:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 13.3.2 Date: Tue, 26 May 2015 12:57:53 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 286 v=0 o=root 674367894 674367894 IN IP4 10.203.175.22 s=Asterisk PBX 13.3.2 c=IN IP4 10.203.175.22 t=0 0 m=audio 17296 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- -- Called SIP/userb [May 26 12:57:53] WARNING[8946][C-00000000]: translate.c:392 framein: no samples for ulawtolin <--- SIP read from UDP:10.203.175.1:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.203.175.22:5060;received=10.203.175.22;branch=z9hG4bK4df816bb Call-ID: 5a457c525415ef9c70c1ac4470edda59@10.203.175.22:5060 From: "User A" ;tag=as711a417a To: CSeq: 102 INVITE Content-Length: 0 <-------------> --- (7 headers 0 lines) --- [May 26 12:57:53] WARNING[8946][C-00000000]: translate.c:392 framein: no samples for ulawtolin <--- SIP read from UDP:10.203.175.1:5060 ---> <-------------> <--- SIP read from UDP:10.203.175.1:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.203.175.22:5060;received=10.203.175.22;branch=z9hG4bK4df816bb Call-ID: 5a457c525415ef9c70c1ac4470edda59@10.203.175.22:5060 From: "User A" ;tag=as711a417a To: ;tag=FHHpxyueDtR90Y.KOlIipmoYCclCkwI1 CSeq: 102 INVITE Contact: Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Content-Type: application/sdp Content-Length: 271 v=0 o=- 3641633873 3641633874 IN IP4 192.168.2.7 s=pjmedia b=AS:84 t=0 0 a=X-nat:0 m=audio 4000 RTP/AVP 0 101 c=IN IP4 192.168.2.7 b=TIAS:64000 a=rtcp:4001 IN IP4 192.168.2.7 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 <-------------> --- (11 headers 14 lines) --- Found RTP audio format 0 Found RTP audio format 101 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Capabilities: us - (ulaw|alaw|gsm|h263), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 192.168.2.7:4000 sip_route_dump: route/path hop: set_destination: Parsing for address/port to send to set_destination: set destination to 10.203.175.1:5060 Transmitting (no NAT) to 10.203.175.1:5060: ACK sip:userb@10.203.175.1:5060;ob SIP/2.0 Via: SIP/2.0/UDP 10.203.175.22:5060;branch=z9hG4bK4420ce83 Max-Forwards: 70 From: "User A" ;tag=as711a417a To: ;tag=FHHpxyueDtR90Y.KOlIipmoYCclCkwI1 Contact: Call-ID: 5a457c525415ef9c70c1ac4470edda59@10.203.175.22:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 13.3.2 Content-Length: 0 --- -- SIP/userb-00000001 answered SIP/usera-00000000 -- Channel SIP/usera-00000000 joined 'simple_bridge' basic-bridge <8d94acfe-4da4-4d7b-8792-7974bf7a9671> -- Channel SIP/userb-00000001 joined 'simple_bridge' basic-bridge <8d94acfe-4da4-4d7b-8792-7974bf7a9671> > 0x7f0e5002ac90 -- Probation passed - setting RTP source address to 10.203.175.1:4000 > 0x7f0e5002ac90 -- Switching RTP source address to 192.168.2.7:4000 <--- SIP read from UDP:10.203.175.1:5060 ---> INVITE sip:usera@10.203.175.22:5060 SIP/2.0 Via: SIP/2.0/UDP 10.203.175.1:5060;rport;branch=z9hG4bKPjfVUyNCExX37QLzhnkA2nATDo7OcEHtTa Max-Forwards: 70 From: ;tag=FHHpxyueDtR90Y.KOlIipmoYCclCkwI1 To: "User A" ;tag=as711a417a Contact: Call-ID: 5a457c525415ef9c70c1ac4470edda59@10.203.175.22:5060 CSeq: 2680 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Session-Expires: 1800 Min-SE: 90 User-Agent: PJSUA v2.3 Darwin-14.3/x86_64 Content-Type: application/sdp Content-Length: 470 v=0 o=- 3641633873 3641633875 IN IP4 192.168.2.7 s=pjmedia b=AS:84 t=0 0 a=X-nat:0 m=audio 4000 RTP/AVP 98 97 99 104 3 0 8 9 96 c=IN IP4 192.168.2.7 b=TIAS:64000 a=rtcp:4001 IN IP4 192.168.2.7 a=sendrecv a=rtpmap:98 speex/16000 a=rtpmap:97 speex/8000 a=rtpmap:99 speex/32000 a=rtpmap:104 iLBC/8000 a=fmtp:104 mode=30 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 <-------------> --- (15 headers 22 lines) --- Sending to 10.203.175.1:5060 (no NAT) Found RTP audio format 98 Found RTP audio format 97 Found RTP audio format 99 Found RTP audio format 104 Found RTP audio format 3 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 9 Found RTP audio format 96 Found audio description format speex for ID 98 Found audio description format speex for ID 97 Found audio description format speex for ID 99 Found audio description format iLBC for ID 104 Found audio description format GSM for ID 3 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found audio description format G722 for ID 9 Found audio description format telephone-event for ID 96 Capabilities: us - (ulaw|alaw|gsm|h263), peer - audio=(ulaw|gsm|alaw|g722|speex|speex16|speex32|ilbc)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|gsm) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 192.168.2.7:4000 <--- Transmitting (no NAT) to 10.203.175.1:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.203.175.1:5060;branch=z9hG4bKPjfVUyNCExX37QLzhnkA2nATDo7OcEHtTa;received=10.203.175.1;rport=5060 From: ;tag=FHHpxyueDtR90Y.KOlIipmoYCclCkwI1 To: "User A" ;tag=as711a417a Call-ID: 5a457c525415ef9c70c1ac4470edda59@10.203.175.22:5060 CSeq: 2680 INVITE Server: Asterisk PBX 13.3.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Length: 0 <------------> Audio is at 17296 Adding codec ulaw to SDP Adding codec alaw to SDP Adding codec gsm to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 10.203.175.1:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.203.175.1:5060;branch=z9hG4bKPjfVUyNCExX37QLzhnkA2nATDo7OcEHtTa;received=10.203.175.1;rport=5060 From: ;tag=FHHpxyueDtR90Y.KOlIipmoYCclCkwI1 To: "User A" ;tag=as711a417a Call-ID: 5a457c525415ef9c70c1ac4470edda59@10.203.175.22:5060 CSeq: 2680 INVITE Server: Asterisk PBX 13.3.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 283 v=0 o=root 674367894 674367895 IN IP4 10.203.175.22 s=Asterisk PBX 13.3.2 c=IN IP4 10.203.175.22 t=0 0 m=audio 17296 RTP/AVP 0 8 3 96 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=maxptime:150 a=sendrecv <------------> <--- SIP read from UDP:10.203.175.1:5060 ---> ACK sip:usera@10.203.175.22:5060 SIP/2.0 Via: SIP/2.0/UDP 10.203.175.1:5060;rport;branch=z9hG4bKPjuWBA.R0.al9.Zc7RN75S1MjfWvjvtrxY Max-Forwards: 70 From: ;tag=FHHpxyueDtR90Y.KOlIipmoYCclCkwI1 To: "User A" ;tag=as711a417a Call-ID: 5a457c525415ef9c70c1ac4470edda59@10.203.175.22:5060 CSeq: 2680 ACK Content-Length: 0 <-------------> --- (8 headers 0 lines) --- <--- SIP read from UDP:10.203.175.1:5060 ---> INVITE sip:usera@10.203.175.22:5060 SIP/2.0 Via: SIP/2.0/UDP 10.203.175.1:5060;rport;branch=z9hG4bKPjtA153PF43OvW7zy5fZ8WHPVkmhtp.eS8 Max-Forwards: 70 From: ;tag=FHHpxyueDtR90Y.KOlIipmoYCclCkwI1 To: "User A" ;tag=as711a417a Contact: Call-ID: 5a457c525415ef9c70c1ac4470edda59@10.203.175.22:5060 CSeq: 2681 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Session-Expires: 1800 Min-SE: 90 Content-Type: application/sdp Content-Length: 268 v=0 o=- 3641633873 3641633876 IN IP4 192.168.2.7 s=pjmedia b=AS:84 t=0 0 a=X-nat:0 m=audio 4000 RTP/AVP 0 96 c=IN IP4 192.168.2.7 b=TIAS:64000 a=rtcp:4001 IN IP4 192.168.2.7 a=rtpmap:0 PCMU/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=sendrecv <-------------> --- (14 headers 14 lines) --- Sending to 10.203.175.1:5060 (no NAT) Found RTP audio format 0 Found RTP audio format 96 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 96 Capabilities: us - (ulaw|alaw|gsm|h263), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 192.168.2.7:4000 <--- Transmitting (no NAT) to 10.203.175.1:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.203.175.1:5060;branch=z9hG4bKPjtA153PF43OvW7zy5fZ8WHPVkmhtp.eS8;received=10.203.175.1;rport=5060 From: ;tag=FHHpxyueDtR90Y.KOlIipmoYCclCkwI1 To: "User A" ;tag=as711a417a Call-ID: 5a457c525415ef9c70c1ac4470edda59@10.203.175.22:5060 CSeq: 2681 INVITE Server: Asterisk PBX 13.3.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Length: 0 <------------> Audio is at 17296 Adding codec ulaw to SDP Adding codec alaw to SDP Adding codec gsm to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 10.203.175.1:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.203.175.1:5060;branch=z9hG4bKPjtA153PF43OvW7zy5fZ8WHPVkmhtp.eS8;received=10.203.175.1;rport=5060 From: ;tag=FHHpxyueDtR90Y.KOlIipmoYCclCkwI1 To: "User A" ;tag=as711a417a Call-ID: 5a457c525415ef9c70c1ac4470edda59@10.203.175.22:5060 CSeq: 2681 INVITE Server: Asterisk PBX 13.3.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 283 v=0 o=root 674367894 674367896 IN IP4 10.203.175.22 s=Asterisk PBX 13.3.2 c=IN IP4 10.203.175.22 t=0 0 m=audio 17296 RTP/AVP 0 8 3 96 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=maxptime:150 a=sendrecv <------------> <--- SIP read from UDP:10.203.175.1:5060 ---> ACK sip:usera@10.203.175.22:5060 SIP/2.0 Via: SIP/2.0/UDP 10.203.175.1:5060;rport;branch=z9hG4bKPjithTi8Kpqv-WCQ4FdRh4.FObq.ekV1YB Max-Forwards: 70 From: ;tag=FHHpxyueDtR90Y.KOlIipmoYCclCkwI1 To: "User A" ;tag=as711a417a Call-ID: 5a457c525415ef9c70c1ac4470edda59@10.203.175.22:5060 CSeq: 2681 ACK Content-Length: 0 <-------------> --- (8 headers 0 lines) --- > 0x7f0e5002ac90 -- Probation passed - setting RTP source address to 10.203.175.1:4000 > 0x7f0e5002ac90 -- Switching RTP source address to 192.168.2.7:4000 <--- SIP read from UDP:10.203.175.1:5060 ---> <-------------> Really destroying SIP dialog 'v4IVJ1CI2Hj39OAhF9kK9NtObdJYoVE1' Method: REGISTER <--- SIP read from UDP:10.203.175.1:5060 ---> INVITE sip:usera@10.203.175.22:5060 SIP/2.0 Via: SIP/2.0/UDP 10.203.175.1:5060;rport;branch=z9hG4bKPjkH0DQ94pEnH1QoJ89UoTaxT0uXgK4jFF Max-Forwards: 70 From: ;tag=FHHpxyueDtR90Y.KOlIipmoYCclCkwI1 To: "User A" ;tag=as711a417a Contact: Call-ID: 5a457c525415ef9c70c1ac4470edda59@10.203.175.22:5060 CSeq: 2682 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Session-Expires: 1800 Min-SE: 90 User-Agent: PJSUA v2.3 Darwin-14.3/x86_64 Content-Type: application/sdp Content-Length: 470 v=0 o=- 3641633873 3641633877 IN IP4 192.168.2.7 s=pjmedia b=AS:84 t=0 0 a=X-nat:0 m=audio 4000 RTP/AVP 98 97 99 104 3 0 8 9 96 c=IN IP4 192.168.2.7 b=TIAS:64000 a=rtcp:4001 IN IP4 192.168.2.7 a=sendrecv a=rtpmap:98 speex/16000 a=rtpmap:97 speex/8000 a=rtpmap:99 speex/32000 a=rtpmap:104 iLBC/8000 a=fmtp:104 mode=30 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 <-------------> --- (15 headers 22 lines) --- Sending to 10.203.175.1:5060 (no NAT) Found RTP audio format 98 Found RTP audio format 97 Found RTP audio format 99 Found RTP audio format 104 Found RTP audio format 3 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 9 Found RTP audio format 96 Found audio description format speex for ID 98 Found audio description format speex for ID 97 Found audio description format speex for ID 99 Found audio description format iLBC for ID 104 Found audio description format GSM for ID 3 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found audio description format G722 for ID 9 Found audio description format telephone-event for ID 96 Capabilities: us - (ulaw|alaw|gsm|h263), peer - audio=(ulaw|gsm|alaw|g722|speex|speex16|speex32|ilbc)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|gsm) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 192.168.2.7:4000 <--- Transmitting (no NAT) to 10.203.175.1:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.203.175.1:5060;branch=z9hG4bKPjkH0DQ94pEnH1QoJ89UoTaxT0uXgK4jFF;received=10.203.175.1;rport=5060 From: ;tag=FHHpxyueDtR90Y.KOlIipmoYCclCkwI1 To: "User A" ;tag=as711a417a Call-ID: 5a457c525415ef9c70c1ac4470edda59@10.203.175.22:5060 CSeq: 2682 INVITE Server: Asterisk PBX 13.3.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Length: 0 <------------> Audio is at 17296 Adding codec ulaw to SDP Adding codec alaw to SDP Adding codec gsm to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 10.203.175.1:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.203.175.1:5060;branch=z9hG4bKPjkH0DQ94pEnH1QoJ89UoTaxT0uXgK4jFF;received=10.203.175.1;rport=5060 From: ;tag=FHHpxyueDtR90Y.KOlIipmoYCclCkwI1 To: "User A" ;tag=as711a417a Call-ID: 5a457c525415ef9c70c1ac4470edda59@10.203.175.22:5060 CSeq: 2682 INVITE Server: Asterisk PBX 13.3.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 283 v=0 o=root 674367894 674367897 IN IP4 10.203.175.22 s=Asterisk PBX 13.3.2 c=IN IP4 10.203.175.22 t=0 0 m=audio 17296 RTP/AVP 0 8 3 96 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=maxptime:150 a=sendrecv <------------> <--- SIP read from UDP:10.203.175.1:5060 ---> ACK sip:usera@10.203.175.22:5060 SIP/2.0 Via: SIP/2.0/UDP 10.203.175.1:5060;rport;branch=z9hG4bKPjY2wGxSqbH0w6KrEBnmEfP74CE6gjlmXP Max-Forwards: 70 From: ;tag=FHHpxyueDtR90Y.KOlIipmoYCclCkwI1 To: "User A" ;tag=as711a417a Call-ID: 5a457c525415ef9c70c1ac4470edda59@10.203.175.22:5060 CSeq: 2682 ACK Content-Length: 0 <-------------> --- (8 headers 0 lines) --- <--- SIP read from UDP:10.203.175.1:5060 ---> INVITE sip:usera@10.203.175.22:5060 SIP/2.0 Via: SIP/2.0/UDP 10.203.175.1:5060;rport;branch=z9hG4bKPjOmPUt8f5LpffV2OvpkU0vOA0T4Fr9n36 Max-Forwards: 70 From: ;tag=FHHpxyueDtR90Y.KOlIipmoYCclCkwI1 To: "User A" ;tag=as711a417a Contact: Call-ID: 5a457c525415ef9c70c1ac4470edda59@10.203.175.22:5060 CSeq: 2683 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Session-Expires: 1800 Min-SE: 90 Content-Type: application/sdp Content-Length: 268 v=0 o=- 3641633873 3641633878 IN IP4 192.168.2.7 s=pjmedia b=AS:84 t=0 0 a=X-nat:0 m=audio 4000 RTP/AVP 0 96 c=IN IP4 192.168.2.7 b=TIAS:64000 a=rtcp:4001 IN IP4 192.168.2.7 a=rtpmap:0 PCMU/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=sendrecv <-------------> --- (14 headers 14 lines) --- Sending to 10.203.175.1:5060 (no NAT) Found RTP audio format 0 Found RTP audio format 96 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 96 Capabilities: us - (ulaw|alaw|gsm|h263), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 192.168.2.7:4000 <--- Transmitting (no NAT) to 10.203.175.1:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.203.175.1:5060;branch=z9hG4bKPjOmPUt8f5LpffV2OvpkU0vOA0T4Fr9n36;received=10.203.175.1;rport=5060 From: ;tag=FHHpxyueDtR90Y.KOlIipmoYCclCkwI1 To: "User A" ;tag=as711a417a Call-ID: 5a457c525415ef9c70c1ac4470edda59@10.203.175.22:5060 CSeq: 2683 INVITE Server: Asterisk PBX 13.3.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Length: 0 <------------> Audio is at 17296 Adding codec ulaw to SDP Adding codec alaw to SDP Adding codec gsm to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 10.203.175.1:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.203.175.1:5060;branch=z9hG4bKPjOmPUt8f5LpffV2OvpkU0vOA0T4Fr9n36;received=10.203.175.1;rport=5060 From: ;tag=FHHpxyueDtR90Y.KOlIipmoYCclCkwI1 To: "User A" ;tag=as711a417a Call-ID: 5a457c525415ef9c70c1ac4470edda59@10.203.175.22:5060 CSeq: 2683 INVITE Server: Asterisk PBX 13.3.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 283 v=0 o=root 674367894 674367898 IN IP4 10.203.175.22 s=Asterisk PBX 13.3.2 c=IN IP4 10.203.175.22 t=0 0 m=audio 17296 RTP/AVP 0 8 3 96 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=maxptime:150 a=sendrecv <------------> <--- SIP read from UDP:10.203.175.1:5060 ---> ACK sip:usera@10.203.175.22:5060 SIP/2.0 Via: SIP/2.0/UDP 10.203.175.1:5060;rport;branch=z9hG4bKPj4X7eC9fmXZPK-GBJhw4OK0V.oZpVsxyS Max-Forwards: 70 From: ;tag=FHHpxyueDtR90Y.KOlIipmoYCclCkwI1 To: "User A" ;tag=as711a417a Call-ID: 5a457c525415ef9c70c1ac4470edda59@10.203.175.22:5060 CSeq: 2683 ACK Content-Length: 0 <-------------> --- (8 headers 0 lines) --- > 0x7f0e5002ac90 -- Probation passed - setting RTP source address to 192.168.2.7:4000 <--- SIP read from UDP:10.203.175.1:5060 ---> <-------------> <--- SIP read from UDP:10.203.175.1:54191 ---> BYE sip:123@10.203.175.22:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.11.215:54191;rport;branch=z9hG4bKPjc1tIE9tZipkRRnQgS6kx5KVrdh4sKY5E Max-Forwards: 70 From: "Luca Pradovera" ;tag=moOzaAjm6CrEUO8QBoP2AQNoZjZeXz3y To: ;tag=as190f62dc Call-ID: IqoQ0P0y1J3vf7mPyDmRJA7uwlcaWuHH CSeq: 13235 BYE User-Agent: Blink Pro 4.2.1 (MacOSX) Content-Length: 0 <-------------> --- (9 headers 0 lines) --- Sending to 10.203.175.1:54191 (NAT) Scheduling destruction of SIP dialog 'IqoQ0P0y1J3vf7mPyDmRJA7uwlcaWuHH' in 32000 ms (Method: BYE) <--- Transmitting (NAT) to 10.203.175.1:54191 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.11.215:54191;branch=z9hG4bKPjc1tIE9tZipkRRnQgS6kx5KVrdh4sKY5E;received=10.203.175.1;rport=54191 From: "Luca Pradovera" ;tag=moOzaAjm6CrEUO8QBoP2AQNoZjZeXz3y To: ;tag=as190f62dc Call-ID: IqoQ0P0y1J3vf7mPyDmRJA7uwlcaWuHH CSeq: 13235 BYE Server: Asterisk PBX 13.3.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <------------> -- Channel SIP/usera-00000000 left 'simple_bridge' basic-bridge <8d94acfe-4da4-4d7b-8792-7974bf7a9671> == Spawn extension (outbound-monitor, 123, 3) exited non-zero on 'SIP/usera-00000000' -- Executing [h@outbound-monitor:1] Answer("SIP/usera-00000000", "") in new stack == Spawn extension (outbound-monitor, h, 1) exited non-zero on 'SIP/usera-00000000' -- Channel SIP/userb-00000001 left 'simple_bridge' basic-bridge <8d94acfe-4da4-4d7b-8792-7974bf7a9671> Scheduling destruction of SIP dialog '5a457c525415ef9c70c1ac4470edda59@10.203.175.22:5060' in 32000 ms (Method: ACK) set_destination: Parsing for address/port to send to set_destination: set destination to 10.203.175.1:5060 Reliably Transmitting (no NAT) to 10.203.175.1:5060: BYE sip:userb@10.203.175.1:5060;ob SIP/2.0 Via: SIP/2.0/UDP 10.203.175.22:5060;branch=z9hG4bK7cfb27b6;rport Max-Forwards: 70 From: "User A" ;tag=as711a417a To: ;tag=FHHpxyueDtR90Y.KOlIipmoYCclCkwI1 Call-ID: 5a457c525415ef9c70c1ac4470edda59@10.203.175.22:5060 CSeq: 103 BYE User-Agent: Asterisk PBX 13.3.2 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- <--- SIP read from UDP:10.203.175.1:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.203.175.22:5060;rport=5060;received=10.203.175.22;branch=z9hG4bK7cfb27b6 Call-ID: 5a457c525415ef9c70c1ac4470edda59@10.203.175.22:5060 From: "User A" ;tag=as711a417a To: ;tag=FHHpxyueDtR90Y.KOlIipmoYCclCkwI1 CSeq: 103 BYE Content-Length: 0 <-------------> --- (7 headers 0 lines) --- SIP Response message for INCOMING dialog BYE arrived Really destroying SIP dialog '5a457c525415ef9c70c1ac4470edda59@10.203.175.22:5060' Method: ACK asterisk13*CLI> Disconnected from Asterisk server Asterisk cleanly ending (0). Executing last minute cleanups vagrant@asterisk13:~$