ok then... sorry it took so long, but I already posted it yesterday but it was rejected by the list's moderator as it exceeded 40k in size :( so i'll resend it now: part #1 is this, part #2 will only contain the attachment with working "direct"-Video (not early media) for comparison...
--- 1) i tried with asterisk 11.6... no change. i.e. same behaviour as with asterisk 12.2 --> no preview video (i verified with wireshark that - as with asterisk 12.2 - the h264 data is sent from the caller to asterisk, but isn't forwarded to the callee... only the audio (g722) is passed on) 2) setup: the caller is 301 (linphone on windows) calling 306 (Grandstream 3175v2 with preview enabled) 2.1) sip.conf: [general] context=unauthenticated allowguest=yes srvlookup=no udpbindaddr=0.0.0.0 tcpenable=yes tcpbindaddr=0.0.0.0 callcounter=yes textsupport=yes accept_outofcall_messages=yes outofcall_message_context=messages-sip auth_message_requests=no insecure=invite directmedia=no prematuremedia=yes progressinband=never [office-phone](!) type=friend context=LocalSets host=dynamic nat=force_rport,comedia dtmfmode=auto disallow=all allow=g722 allow=ulaw allow=alaw videosupport=yes allow=h264 # define some devices [301](office-phone) secret=pwd301 [306](office-phone) secret=pwd306 2.2) extensions.conf ... [LocalSets] ... exten => 306,hint,SIP/306 exten => 306,1,NoOp(dial 306) same => n,Dial(SIP/306) ... 3) debug-logs: I used "core set debug 9" and "core set verbose 9" and "sip set debug on" i attached two files: 3.1) "asterisk-debug-earlymedia.txt" in which I dial 306 from 301 and after some (5?) seconds i press "preview" on the grandstream. after that, the grandstream only allows me to choose beween "Accept Audio" and "Reject" - no "Accept Video" as I would expect... another 5 seconds later i hang up. 3.2) "asterisk-debug-videodirect.txt" here I push the "Accept Video" button instead of the "preview" button; here the video works! thanks a lot for your support!
<--- SIP read from UDP:10.10.1.144:5060 ---> INVITE sip:306@10.10.1.201 SIP/2.0 Via: SIP/2.0/UDP 10.10.1.144:5060;branch=z9hG4bK.~izIMUo0u;rport From: <sip:301@10.10.1.201>;tag=WSFA2GLK9 To: sip:306@10.10.1.201 CSeq: 20 INVITE Call-ID: 64TsxOvORf Max-Forwards: 70 Supported: replaces, outbound Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Content-Length: 562 Contact: <sip:301@10.10.1.144>;+sip.instance="<urn:uuid:c03a376f-325c-4764-a9c0-827fb2a23a79>" User-Agent: Linphone/3.7.0 (belle-sip/1.3.0) v=0 o=301 4066 3784 IN IP4 192.168.220.16 s=Talk c=IN IP4 192.168.220.16 t=0 0 m=audio 7078 RTP/AVP 9 124 111 110 0 8 101 a=rtpmap:124 opus/48000 a=fmtp:124 useinbandfec=1; usedtx=1 a=rtpmap:111 speex/16000 a=fmtp:111 vbr=on a=rtpmap:110 speex/8000 a=fmtp:110 vbr=on a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 m=video 9078 RTP/AVP 102 98 103 99 a=rtpmap:102 H264/90000 a=fmtp:102 profile-level-id=42801F a=rtpmap:98 H263-1998/90000 a=fmtp:98 CIF=1;QCIF=1 a=rtpmap:103 VP8/90000 a=rtpmap:99 MP4V-ES/90000 a=fmtp:99 profile-level-id=3 <-------------> --- (13 headers 22 lines) --- Sending to 10.10.1.144:5060 (no NAT) Sending to 10.10.1.144:5060 (no NAT) Using INVITE request as basis request - 64TsxOvORf Found peer '301' for '301' from 10.10.1.144:5060 == Using SIP VIDEO CoS mark 6 == Using SIP RTP CoS mark 5 Found RTP audio format 9 Found RTP audio format 124 Found RTP audio format 111 Found RTP audio format 110 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 101 Found audio description format opus for ID 124 Found audio description format speex for ID 111 Found audio description format speex for ID 110 Found audio description format telephone-event for ID 101 Found RTP video format 102 Found RTP video format 98 Found RTP video format 103 Found RTP video format 99 Found video description format H264 for ID 102 Found video description format H263-1998 for ID 98 Found video description format VP8 for ID 103 Found video description format MP4V-ES for ID 99 Capabilities: us - (ulaw|alaw|g722|h264), peer - audio=(ulaw|alaw|speex|speex16|g722|opus)/video=(h263p|h264|mpeg4|vp8)/text=(nothing), combined - (ulaw|alaw|g722|h264) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 192.168.220.16:7078 Peer video RTP is at port 192.168.220.16:9078 Peer doesn't provide T.140 Looking for 306 in LocalSets (domain 10.10.1.201) list_route: route/path hop: <sip:301@10.10.1.144> <--- Transmitting (NAT) to 10.10.1.144:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.1.144:5060;branch=z9hG4bK.~izIMUo0u;received=10.10.1.144;rport=5060 From: <sip:301@10.10.1.201>;tag=WSFA2GLK9 To: sip:306@10.10.1.201 Call-ID: 64TsxOvORf CSeq: 20 INVITE Server: Asterisk PBX 12.2.0-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: <sip:306@10.10.1.201:5060> Content-Length: 0 <------------> -- Executing [306@LocalSets:1] NoOp("SIP/301-00000008", "dial 306") in new stack -- Executing [306@LocalSets:3] Dial("SIP/301-00000008", "SIP/306") in new stack == Using SIP VIDEO CoS mark 6 == Using SIP RTP CoS mark 5 Audio is at 13998 Video is at 10.10.1.201:19236 Adding codec 100012 (g722) to SDP Adding codec 100003 (ulaw) to SDP Adding codec 100004 (alaw) to SDP Adding video codec 200004 (h264) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 10.10.1.145:5062: INVITE sip:306@10.10.1.145:5062 SIP/2.0 Via: SIP/2.0/UDP 10.10.1.201:5060;branch=z9hG4bK0adf25b7;rport Max-Forwards: 70 From: <sip:301@10.10.1.201>;tag=as2e0a7bb2 To: <sip:306@10.10.1.145:5062> Contact: <sip:301@10.10.1.201:5060> Call-ID: 3c7241696275b34c664111f736759783@10.10.1.201:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 12.2.0-rc1 Date: Wed, 07 May 2014 12:01:20 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 433 v=0 o=root 763823955 763823955 IN IP4 10.10.1.201 s=Asterisk PBX 12.2.0-rc1 c=IN IP4 10.10.1.201 b=CT:384 t=0 0 m=audio 13998 RTP/AVP 9 0 8 101 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=maxptime:150 a=sendrecv m=video 19236 RTP/AVP 99 a=rtpmap:99 H264/90000 a=fmtp:99 profile-level-id=42801F a=sendrecv --- -- Called SIP/306 <--- SIP read from UDP:10.10.1.145:5062 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.1.201:5060;branch=z9hG4bK0adf25b7;rport=5060 From: <sip:301@10.10.1.201>;tag=as2e0a7bb2 To: <sip:306@10.10.1.145:5062> Call-ID: 3c7241696275b34c664111f736759783@10.10.1.201:5060 CSeq: 102 INVITE Supported: replaces, path, timer, eventlist User-Agent: Grandstream GXV3175v2 1.0.1.55 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from UDP:10.10.1.145:5062 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.10.1.201:5060;branch=z9hG4bK0adf25b7;rport=5060 From: <sip:301@10.10.1.201>;tag=as2e0a7bb2 To: <sip:306@10.10.1.145:5062>;tag=177996708 Call-ID: 3c7241696275b34c664111f736759783@10.10.1.201:5060 CSeq: 102 INVITE Contact: <sip:306@10.10.1.145:5062> Supported: replaces, path, timer, eventlist User-Agent: Grandstream GXV3175v2 1.0.1.55 Allow-Events: talk, hold Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 <-------------> --- (12 headers 0 lines) --- list_route: route/path hop: <sip:306@10.10.1.145:5062> -- SIP/306-00000009 is ringing <--- Transmitting (NAT) to 10.10.1.144:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.10.1.144:5060;branch=z9hG4bK.~izIMUo0u;received=10.10.1.144;rport=5060 From: <sip:301@10.10.1.201>;tag=WSFA2GLK9 To: sip:306@10.10.1.201;tag=as0ae1acfc Call-ID: 64TsxOvORf CSeq: 20 INVITE Server: Asterisk PBX 12.2.0-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: <sip:306@10.10.1.201:5060> Content-Length: 0 <------------> <--- SIP read from UDP:10.10.1.144:5060 ---> <-------------> Reliably Transmitting (NAT) to 10.10.1.145:5062: OPTIONS sip:306@10.10.1.145:5062 SIP/2.0 Via: SIP/2.0/UDP 10.10.1.201:5060;branch=z9hG4bK4b77810b;rport Max-Forwards: 70 From: "asterisk" <sip:asterisk@10.10.1.201>;tag=as7bef7d1c To: <sip:306@10.10.1.145:5062> Contact: <sip:asterisk@10.10.1.201:5060> Call-ID: 28e06368307769e702a73e524062623a@10.10.1.201:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 12.2.0-rc1 Date: Wed, 07 May 2014 12:01:25 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- <--- SIP read from UDP:10.10.1.145:5062 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.1.201:5060;branch=z9hG4bK4b77810b;rport=5060 From: "asterisk" <sip:asterisk@10.10.1.201>;tag=as7bef7d1c To: <sip:306@10.10.1.145:5062>;tag=637705810 Call-ID: 28e06368307769e702a73e524062623a@10.10.1.201:5060 CSeq: 102 OPTIONS Supported: replaces, path, timer, eventlist User-Agent: Grandstream GXV3175v2 1.0.1.55 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Really destroying SIP dialog '28e06368307769e702a73e524062623a@10.10.1.201:5060' Method: OPTIONS <--- SIP read from UDP:10.10.1.145:5062 ---> <-------------> <--- SIP read from UDP:10.10.1.145:5062 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 10.10.1.201:5060;branch=z9hG4bK0adf25b7;rport=5060 From: <sip:301@10.10.1.201>;tag=as2e0a7bb2 To: <sip:306@10.10.1.145:5062>;tag=177996708 Call-ID: 3c7241696275b34c664111f736759783@10.10.1.201:5060 CSeq: 102 INVITE Contact: <sip:306@10.10.1.145:5062> Supported: replaces, path, timer, eventlist User-Agent: Grandstream GXV3175v2 1.0.1.55 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Type: application/sdp Content-Length: 447 v=0 o=306 8000 8000 IN IP4 10.10.1.145 s=SIP Call c=IN IP4 10.10.1.145 t=0 0 m=audio 43870 RTP/AVP 9 0 8 101 a=sendrecv a=rtpmap:9 G722/8000 a=ptime:20 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=silenceSupp:off - - - - m=video 31652 RTP/AVP 99 b=AS:320 a=sendrecv a=rtpmap:99 H264/90000 a=fmtp:99 profile-level-id=42801F; sprop-parameter-sets=Z0KADJWgUH5A,aM4Ecg==; max-br=320 <-------------> --- (12 headers 19 lines) --- list_route: route/path hop: <sip:306@10.10.1.145:5062> Found RTP audio format 9 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 101 Found audio description format G722 for ID 9 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found audio description format telephone-event for ID 101 Found RTP video format 99 Found video description format H264 for ID 99 Capabilities: us - (ulaw|alaw|g722|h264), peer - audio=(ulaw|alaw|g722)/video=(h264)/text=(nothing), combined - (ulaw|alaw|g722|h264) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 10.10.1.145:43870 Peer video RTP is at port 10.10.1.145:31652 Peer doesn't provide T.140 -- SIP/306-00000009 is making progress passing it to SIP/301-00000008 Audio is at 16968 Video is at 10.10.1.201:16576 Adding codec 100012 (g722) to SDP Adding codec 100003 (ulaw) to SDP Adding codec 100004 (alaw) to SDP Adding video codec 200004 (h264) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Transmitting (NAT) to 10.10.1.144:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 10.10.1.144:5060;branch=z9hG4bK.~izIMUo0u;received=10.10.1.144;rport=5060 From: <sip:301@10.10.1.201>;tag=WSFA2GLK9 To: sip:306@10.10.1.201;tag=as0ae1acfc Call-ID: 64TsxOvORf CSeq: 20 INVITE Server: Asterisk PBX 12.2.0-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: <sip:306@10.10.1.201:5060> Content-Type: application/sdp Content-Length: 436 v=0 o=root 120783996 120783996 IN IP4 10.10.1.201 s=Asterisk PBX 12.2.0-rc1 c=IN IP4 10.10.1.201 b=CT:384 t=0 0 m=audio 16968 RTP/AVP 9 0 8 101 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=maxptime:150 a=sendrecv m=video 16576 RTP/AVP 102 a=rtpmap:102 H264/90000 a=fmtp:102 profile-level-id=42801F a=sendrecv <------------> > 0x7f84dc026930 -- Probation passed - setting RTP source address to 10.10.1.144:9078 > 0x7f84dc026930 -- Probation passed - setting RTP source address to 10.10.1.144:9078 > 0x7f84dc023590 -- Probation passed - setting RTP source address to 10.10.1.144:7078 > 0x7f84dc023590 -- Probation passed - setting RTP source address to 10.10.1.144:7078 <--- SIP read from UDP:10.10.1.144:5060 ---> <-------------> Reliably Transmitting (NAT) to 10.10.1.144:5060: OPTIONS sip:301@10.10.1.144 SIP/2.0 Via: SIP/2.0/UDP 10.10.1.201:5060;branch=z9hG4bK6c349143;rport Max-Forwards: 70 From: "asterisk" <sip:asterisk@10.10.1.201>;tag=as0db0840e To: <sip:301@10.10.1.144> Contact: <sip:asterisk@10.10.1.201:5060> Call-ID: 588ac3e618f7bcc458801e4f111b21e9@10.10.1.201:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 12.2.0-rc1 Date: Wed, 07 May 2014 12:01:34 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- <--- SIP read from UDP:10.10.1.144:5060 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 10.10.1.201:5060;branch=z9hG4bK6c349143;rport From: "asterisk" <sip:asterisk@10.10.1.201>;tag=as0db0840e To: <sip:301@10.10.1.144>;tag=O6KBR Call-ID: 588ac3e618f7bcc458801e4f111b21e9@10.10.1.201:5060 CSeq: 102 OPTIONS <-------------> --- (6 headers 0 lines) --- Really destroying SIP dialog '588ac3e618f7bcc458801e4f111b21e9@10.10.1.201:5060' Method: OPTIONS <--- SIP read from UDP:10.10.1.145:5062 ---> SIP/2.0 486 Busy Here Via: SIP/2.0/UDP 10.10.1.201:5060;branch=z9hG4bK0adf25b7;rport=5060 From: <sip:301@10.10.1.201>;tag=as2e0a7bb2 To: <sip:306@10.10.1.145:5062>;tag=177996708 Call-ID: 3c7241696275b34c664111f736759783@10.10.1.201:5060 CSeq: 102 INVITE Supported: replaces, path, timer, eventlist User-Agent: Grandstream GXV3175v2 1.0.1.55 Warning: 399 GS "The call is rejected" Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 <-------------> --- (11 headers 0 lines) --- -- Got SIP response 486 "Busy Here" back from 10.10.1.145:5062 Transmitting (NAT) to 10.10.1.145:5062: ACK sip:306@10.10.1.145:5062 SIP/2.0 Via: SIP/2.0/UDP 10.10.1.201:5060;branch=z9hG4bK0adf25b7;rport Max-Forwards: 70 From: <sip:301@10.10.1.201>;tag=as2e0a7bb2 To: <sip:306@10.10.1.145:5062>;tag=177996708 Contact: <sip:301@10.10.1.201:5060> Call-ID: 3c7241696275b34c664111f736759783@10.10.1.201:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 12.2.0-rc1 Content-Length: 0 --- -- SIP/306-00000009 is busy == Everyone is busy/congested at this time (1:1/0/0) -- Auto fallthrough, channel 'SIP/301-00000008' status is 'BUSY' <--- Reliably Transmitting (NAT) to 10.10.1.144:5060 ---> SIP/2.0 486 Busy Here Via: SIP/2.0/UDP 10.10.1.144:5060;branch=z9hG4bK.~izIMUo0u;received=10.10.1.144;rport=5060 From: <sip:301@10.10.1.201>;tag=WSFA2GLK9 To: sip:306@10.10.1.201;tag=as0ae1acfc Call-ID: 64TsxOvORf CSeq: 20 INVITE Server: Asterisk PBX 12.2.0-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer X-Asterisk-HangupCause: User busy X-Asterisk-HangupCauseCode: 17 Content-Length: 0 <------------> Really destroying SIP dialog '3c7241696275b34c664111f736759783@10.10.1.201:5060' Method: INVITE <--- SIP read from UDP:10.10.1.144:5060 ---> ACK sip:306@10.10.1.201 SIP/2.0 Via: SIP/2.0/UDP 10.10.1.144:5060;branch=z9hG4bK.~izIMUo0u;rport Call-ID: 64TsxOvORf From: <sip:301@10.10.1.201>;tag=WSFA2GLK9 To: <sip:306@10.10.1.201>;tag=as0ae1acfc Contact: <sip:301@10.10.1.144>;+sip.instance="<urn:uuid:c03a376f-325c-4764-a9c0-827fb2a23a79>" Max-Forwards: 70 CSeq: 20 ACK <-------------> --- (8 headers 0 lines) --- Really destroying SIP dialog '64TsxOvORf' Method: ACK <--- SIP read from UDP:10.10.1.144:5060 ---> PUBLISH sip:301@10.10.1.201 SIP/2.0 Via: SIP/2.0/UDP 10.10.1.144:5060;branch=z9hG4bK.sq11cyw3l;rport From: <sip:301@10.10.1.201>;tag=TOdTkgRTc To: sip:301@10.10.1.201 CSeq: 26 PUBLISH Call-ID: LErqEXzaBt Max-Forwards: 70 Supported: replaces, outbound Event: presence Expires: 3600 User-Agent: Linphone/3.7.0 (belle-sip/1.3.0) Content-Type: application/pidf+xml Content-Length: 381 <?xml version="1.0" encoding="UTF-8"?> <presence xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpid="urn:ietf:params:xml:ns:pidf:rpid" entity="sip:301@10.10.1.201" xmlns="urn:ietf:params:xml:ns:pidf"><tuple id="nqcpwx"><status><basic>open</basic></status><contact priority="0.8">sip:301@10.10.1.201</contact><timestamp>2014-05-07T11:55:33Z</timestamp></tuple></presence> <-------------> --- (13 headers 2 lines) --- Sending to 10.10.1.144:5060 (no NAT) <--- Transmitting (no NAT) to 10.10.1.144:5060 ---> SIP/2.0 489 Bad Event Via: SIP/2.0/UDP 10.10.1.144:5060;branch=z9hG4bK.sq11cyw3l;received=10.10.1.144;rport=5060 From: <sip:301@10.10.1.201>;tag=TOdTkgRTc To: sip:301@10.10.1.201;tag=as22c12a03 Call-ID: LErqEXzaBt CSeq: 26 PUBLISH Server: Asterisk PBX 12.2.0-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <------------> Really destroying SIP dialog 'LErqEXzaBt' Method: PUBLISH <--- SIP read from UDP:10.10.1.144:5060 ---> <-------------> <--- SIP read from UDP:10.10.1.145:5062 ---> <-------------> Reliably Transmitting (NAT) to 10.10.1.113:42055: OPTIONS sip:402@10.10.1.113:42055;ob SIP/2.0 Via: SIP/2.0/UDP 10.10.1.201:5060;branch=z9hG4bK20f2fbcb;rport Max-Forwards: 70 From: "asterisk" <sip:asterisk@10.10.1.201>;tag=as5a859160 To: <sip:402@10.10.1.113:42055;ob> Contact: <sip:asterisk@10.10.1.201:5060> Call-ID: 01e706c701493e44456abe8551c383a3@10.10.1.201:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 12.2.0-rc1 Date: Wed, 07 May 2014 12:01:47 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- <--- SIP read from UDP:10.10.1.113:42055 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.1.201:5060;rport=5060;received=10.10.1.201;branch=z9hG4bK20f2fbcb Call-ID: 01e706c701493e44456abe8551c383a3@10.10.1.201:5060 From: "asterisk" <sip:asterisk@10.10.1.201>;tag=as5a859160 To: <sip:402@10.10.1.113;ob>;tag=z9hG4bK20f2fbcb CSeq: 102 OPTIONS Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Accept: application/sdp, application/pidf+xml, application/xpidf+xml, application/simple-message-summary, message/sipfrag;version=2.0, application/im-iscomposing+xml, text/plain Supported: replaces, 100rel, timer, norefersub Allow-Events: presence, message-summary, refer User-Agent: CSipSimple_hammerhead-19/r2398 Content-Type: application/sdp Content-Length: 253 v=0 o=- 3608452942 3608452942 IN IP4 10.10.1.113 s=pjmedia t=0 0 m=audio 4000 RTP/AVP 9 0 8 101 c=IN IP4 10.10.1.113 a=sendrecv a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 <-------------> --- (13 headers 12 lines) --- Really destroying SIP dialog '01e706c701493e44456abe8551c383a3@10.10.1.201:5060' Method: OPTIONS [May 7 14:01:51] WARNING[1820]: res_musiconhold.c:744 monmp3thread: poll() failed: Interrupted system call <--- SIP read from UDP:10.10.1.144:5060 ---> <------------->
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