Compiled cleanly but no joy so far, possibly config my end, I couldn't just add the chan_sip lib to the running server (asterisk now redhat, complained about compile time options being different ;-/ )so I have a very basic test server running at the mo.
I can't get either endpoint to connect: [Apr 17 16:30:50] WARNING[5572]: chan_sip.c:8857 process_sdp_a_audio: Got Siren14 offer at 32000 bps, but only 48000 bps supported; ignoring. [Apr 17 16:30:50] WARNING[5572]: chan_sip.c:8857 process_sdp_a_audio: Got Siren14 offer at 24000 bps, but only 48000 bps supported; ignoring. [Apr 17 16:30:50] WARNING[5572]: chan_sip.c:8314 process_sdp: Unsupported SDP media type in offer: application 60040 RTP/AVP 100 -- Executing [2000@sip:1] Dial("SIP/2001-00000000", "SIP/2000,20,tr") in new stack == Using SIP RTP CoS mark 5 [Apr 17 16:30:50] WARNING[31184]: channel.c:1045 ast_best_codec: Don't know any of 0x0 (nothing) formats [Apr 17 16:30:50] WARNING[31184]: channel.c:1045 ast_best_codec: Don't know any of 0x0 (nothing) formats -- Called 2000 -- Nobody picked up in 20000 ms -- Auto fallthrough, channel 'SIP/2001-00000000' status is 'NOANSWER' [Apr 17 16:31:22] WARNING[5572]: chan_sip.c:3386 retrans_pkt: Retransmission timeout reached on transmission 5f76f24b5c4c98f5239e9f934f49ebd7@192.168.254.56:5060 for seqno 102 (Critical Request) -- See doc/sip-retransmit.txt. Packet timed out after 32000ms with no response == Using SIP RTP CoS mark 5 [Apr 17 16:32:10] WARNING[32084]: chan_sip.c:8857 process_sdp_a_audio: Got Siren14 offer at 32000 bps, but only 48000 bps supported; ignoring. [Apr 17 16:32:10] WARNING[32084]: chan_sip.c:8857 process_sdp_a_audio: Got Siren14 offer at 24000 bps, but only 48000 bps supported; ignoring. [Apr 17 16:32:10] WARNING[32084]: chan_sip.c:8314 process_sdp: Unsupported SDP media type in offer: control 60070 RTP/AVP 96 -- Executing [2001@sip:1] Dial("SIP/2000-00000002", "SIP/2000,20,tr") in new stack == Using SIP RTP CoS mark 5 -- Called 2000 -- Nobody picked up in 20000 ms -- Auto fallthrough, channel 'SIP/2000-00000002' status is 'NOANSWER' [Apr 17 16:32:42] WARNING[5572]: chan_sip.c:3386 retrans_pkt: Retransmission timeout reached on transmission 031126ca272c120d0fcee1002f5a1bc4@192.168.254.56:5060 for seqno 102 (Critical Request) -- See doc/sip-retransmit.txt. Packet timed out after 32000ms with no response == Using SIP RTP CoS mark 5 [Apr 17 16:41:02] WARNING[5572]: chan_sip.c:8857 process_sdp_a_audio: Got Siren14 offer at 32000 bps, but only 48000 bps supported; ignoring. [Apr 17 16:41:02] WARNING[5572]: chan_sip.c:8857 process_sdp_a_audio: Got Siren14 offer at 24000 bps, but only 48000 bps supported; ignoring. [Apr 17 16:41:02] WARNING[5572]: chan_sip.c:8314 process_sdp: Unsupported SDP media type in offer: application 60046 RTP/AVP 100 [Apr 17 16:41:02] NOTICE[5572]: chan_sip.c:21361 handle_request_invite: Call from '2001' to extension '*43' rejected because extension not found in context 'sip'. == Using SIP RTP CoS mark 5 [Apr 17 16:41:04] WARNING[5572]: chan_sip.c:8857 process_sdp_a_audio: Got Siren14 offer at 32000 bps, but only 48000 bps supported; ignoring. [Apr 17 16:41:04] WARNING[5572]: chan_sip.c:8857 process_sdp_a_audio: Got Siren14 offer at 24000 bps, but only 48000 bps supported; ignoring. [Apr 17 16:41:04] WARNING[5572]: chan_sip.c:8314 process_sdp: Unsupported SDP media type in offer: application 60052 RTP/AVP 100 [Apr 17 16:41:04] NOTICE[5572]: chan_sip.c:21361 handle_request_invite: Call from '2001' to extension '*43' rejected because extension not found in context 'sip'. On 16 April 2011 04:52, CM Rahman <cmrah...@yahoo.com> wrote: > Gunnar, > > It compiled fine under both 1.6 and 1.8. Now is the time to test it. > > Thanks > CM > > ________________________________ > From: Gunnar Schaller <li...@nowin.de> > To: Development discussion of video media support in Asterisk > <asterisk-video@lists.digium.com> > Sent: Fri, April 15, 2011 2:05:08 AM > Subject: Re: [Asterisk-video] Lifesize HD X10 endpoints & Video Resolution > and framerate configuration question > > Hello, > > Now I tried to compile it. Works! Attached my patch for 1.8 branch. > For 1.6 you need to add the lines manually around line 10270 in > chan_sip.c. > What about JoelW? Did you try that patch? > > Regards, > Gunnar > > > >> Gunnar, > >> I have tried to compile with the code you provided in 1.6.x > >> Here is what I get.. > > [CC] chan_sip.c ->> chan_sip.o >> chan_sip.c: In function âadd_vcodec_to_sdpâ: >> chan_sip.c:10077: warning: missing terminating " character >> chan_sip.c:10077: error: missing terminating " character >> chan_sip.c:10078: error: âprofileâ undeclared (first use in this function) >> chan_sip.c:10078: error: (Each undeclared identifier is reported only once >> chan_sip.c:10078: error: for each function it appears in.) >> chan_sip.c:10078: error: âlevelâ undeclared (first use in this function) >> chan_sip.c:10078: error: âidâ undeclared (first use in this function) >> chan_sip.c:10078: error: invalid suffix "f" on integer constant >> chan_sip.c:10078: error: expected â)â before â;â token >> chan_sip.c:10079: error: stray â\â in program >> chan_sip.c:10079: error: stray â\â in program >> chan_sip.c:10079: warning: missing terminating " character >> chan_sip.c:10079: error: missing terminating " character >> chan_sip.c:10080: error: expected â;â before â}â token >> make[1]: *** [chan_sip.o] Error 1 >> make: *** [channels] Error 2 > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-video mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-video > -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-video mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-video