search for: recvonly

...with meetme a conf with X number of asterisk boxes and "other" devices and phones. I am using the l parameter for all devices being listen only but I'm not sure thats happening as I am getting some feedback (some devices are close to each other like 5 feet). How do I ensure that a=recvonly is being set or sent when bringing a device into the meetme? Can I added that to SIPADDHEADER or something? THere is only "one" device talking and all others should just be listening. I am using 1.4.43 Thanks, Jerry

...inbanddtmf = 1 callerid = 22545062 line => aaln/1 Console output: == Spawn extension (capiring, 9988001133335566, 1) exited non-zero on 'MGCP/aaln/1@iptlf03-1' -- MGCP mgcp_hangup(MGCP/aaln/1@iptlf03-1) on aaln/1@iptlf03 -- Delete connection 4 aaln/1@iptlf03-1 with new mode: recvonly on callid: 5a4b82ad79f47a70 -- MGCP Asked to indicate tone: on aaln/1@iptlf03-1 in cxmode: recvonly -- MGCP mgcp_hangup(MGCP/aaln/1@iptlf03-1) on aaln/1@iptlf03 set vmwi(-) -- MGCP Asked to indicate tone: vmwi(-) on aaln/1@iptlf03-1 in cxmode: inactive NOTICE[49156]: File chan_capi...

...+++++++++++++++++++++++++++++++++ LOG FROM "MGCP DEBUG" ++++++++++++++++++++++++++++++++++++++++ v=0 o=root 2272 2272 IN IP4 192.168.1.30 s=session c=IN IP4 192.168.1.30 t=0 0 m=audio 26040 RTP/AVP 0 a=rtpmap:0 PCMU/8000 to 192.168.1.31:2427 -- Modified aaln/2@atagw-1 with new mode: recvonly on callid: 770afbb9257130a3 Posting Request: MDCX 44 aaln/2@atagw MGCP 1.0 C: 770afbb9257130a3 M: recvonly X: 257130a3 I: 3 R: L/hu(N),L/hf(N),D/[0-9#*](N) to 192.168.1.31:2427 -- MGCP Asked to indicate tone: L/wt,L/ci(10/04/21/07,110,SoftPhone) on aaln/2@atagw-0 in cxmode: recvonly Posting Re...

.../3@Monforte_Euripide_9 MGCP 1.0 C: 05eb2f883bbde177 M: inactive X: 3bbde177 I: 37 R: L/hu(N),L/hf(N),D/[0-9#*](N) to 10.1.1.101:2427 -- Started music on hold, class 'default', on channel 'SIP/024390239-7517' -- Creating connection for aaln/3@Monforte_Euripide_9-1 in cxmode: recvonly callid: 45fd1cb95a923a70 We're at 10.0.0.1 port 12236 Answering with capability 8 Posting Request: CRCX 736 aaln/3@Monforte_Euripide_9 MGCP 1.0 C: 45fd1cb95a923a70 L: p:20, a:PCMA M: recvonly X: 5a923a70 v=0 o=root 2615 2615 IN IP4 10.0.0.1 s=session c=IN IP4 10.0.0.1 t=0 0 m=audio 12236 RTP/A...

Hi. I'm new in Asterisk with MGCP. I set up a MGCP user agent and start asterisk with the next configuration files. '--------------- extensions.conf ---------------------------------------------------- [general] static=yes writeprotect=yes [globals] ap1 => mgcp/aaln/ap200@64.76.148.186 [macro-apl1] exten => s,1,Dial(${ARG1},30,Ttmr) ;exten => s,2,Voicemail(u${MACRO_EXTEN})

Hi All, When asterisk bridges a call between 2 peers and peer-A's user puts the call on hold, then peer-A sends a INVITE with recvonly in the SDP. Asterisk responds to peer-A with sendonly in the SDP and asterisk sends an INVITE to peer-B with recvonly in the SDP. Peer-B then responds with a sendonly in the SDP. I've noticed in the above scenario that peer-B contiutes to send audio to peer-A. What is the point in having audio...

...cting this re-invite or not because it does contain "a=sendrecv". If it should be rejected what error should Asterisk return, and how can we establish two way audio? - After this re-invite Asterisk replies with a "100 Trying" and then a "200 OK" which contains "a=recvonly". - Call is established but called party cannot hear caller. Here's the re-invite message - note that Asterisk is on port 5070: U 2010/05/05 12:47:38.139701 (peer):5060 -> (asterisk):5070 INVITE sip:(called number)@(asterisk):5070 SIP/2.0. Via: SIP/2.0/UDP (peer):5060;branch=z9hG4bK2...

...sting Request: AUEP 3 aaln/1 at 0-13-11-82-bd-a.ssw.dominio.net MGCP 1.0 NCS 1.0 F: A to 10.30.15.254:2427 MGCP read: 200 3 OK A: a:PCMU;PCMA;G728;G729;G729E;G726-16;G726-24;G726-32;G726-40, p:10-30, b:19-100, e:on, t:1, s:off, v:L;fxr;rg;xal;x-xl;fm;lcs;sst;x-jc;x-pol;xrm, m:sendrecv;sendonly;recvonly;inactive;netwloop;netwtest;replcate;confrnce, dq-gi, sc-rtcp: 81/70;81/71;82/70;82/71;80/70;80/71, sc-rtp: 62/51;62/50;64/51;64/50;60/51;60/50 A: a:telephone-event, fmtp:"telephone-event 0-15,144,149,159" A: a:image/t38, p:10-30, b:25-64, dq-gi from 10.30.15.254:2427 Verb: '200'...

...rking too, after a fax call, the connection with T38Modem is established, HylaFAX reports "ANSWER: FAX CONNECTION DEVICE '/dev/ttyT380'", but a fax is never received. In the tcpdump I see the correct handshaking between Asterisk and T38Modem, but finaly T38Modem switches to "recvonly" and Asterisk answers with "SIP/2.0 488 Not acceptable here" and the call ends shortly after that. Probably there is a problem in T38Modem, so I filed a bug at sourceforge for that: https://sourceforge.net/tracker/?func=detail&aid=3337581&group_id=152230&atid=783657 B...

Hi all! I updated my snom200 to 2.02t and now MOH from * don?t works anymore... only the MOH from snom server and if i clear the MOH server field in the phone i have no MOH at all..( with the transfer button, moh plays using a extension). Someone with that problem? I downgrade to 2.01s but nothing changes. Miklos -------------- next part -------------- An HTML attachment was scrubbed... URL:

...oseph) * ASTERISK-25890 - Asterisk 13.8.0 alembic database update fails (Reported by Harley Peters) * ASTERISK-25894 - [patch] webrtc video broken due to missing marker bits in RTP streams (Reported by Jacek Konieczny) * ASTERISK-25854 - No audio after HOLD/RESUME - incorrect a=recvonly in SDP from Asterisk (Reported by Robert McGilvray) * ASTERISK-25873 - res_pjsip: Bundled pjproject: compile error, cannot find -lasteriskpj (Reported by Hans van Eijsden) * ASTERISK-25882 - ARI: Crash can occur due to race condition when attempting to operate on a hung up channel (Pa...

...a=ice-pwd:CFh1JMRfcT5BoH7aoKQuDgDR a=fingerprint:sha-256 6D:5A:B7:7C:8E:1F:2A:F2:8C:DB:49:E4:8D:8F:BE:A0:84:AF:21:E2:A1:D7:DC:29:B1:EE:C3:0C:2C:AF:6C:04 a=setup:actpass a=mid:video a=extmap:2 urn:ietf:params:rtp-hdrext:toffset a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time a=recvonly a=rtcp-mux a=rtpmap:100 VP8/90000 a=rtcp-fb:100 ccm fir a=rtcp-fb:100 nack a=rtcp-fb:100 nack pli a=rtcp-fb:100 goog-remb a=rtpmap:116 red/90000 a=rtpmap:117 ulpfec/90000 a=rtpmap:96 rtx/90000 a=fmtp:96 apt=100 2-BUT when i do channel originate sip/GOROD/XXXXX extension 1065 at office -- Execu...

We are using an MGCP configuration. There seems to be some incompatibilities between our Netergy T2 VOIP chip and Asterisk. This is how our Vendor sees it: Here's what I see. 1. The first call is initiated. (CRCX) The interesting thing here is that the CA (Call Agent) tells us to go directly into sendrecv mode which means that we start streaming audio immediately. All other CAs that

...application/sdp Content-Length: 285 v=0 o=root 2666 2667 IN IP4 192.168.0.20 s=session c=IN IP4 192.168.0.20 t=0 0 m=audio 14254 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/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=recvonly <------------> caideqin*CLI> <--- SIP read from 192.168.0.199:24608 ---> ACK sip:2@192.168.0.20 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.199:24608;branch=z9hG4bK-d87543-da37e161d211bf1f-1--d87543-;rport Max-Forwards: 70 Contact: <sip:1015@192.168.0.199:24608> To: "2"<si...

...t;sip:302 at 192.168.96.5> Content-Type: application/sdp Content-Length: 240 v=0 o=root 1431 1433 IN IP4 192.168.96.16 s=session c=IN IP4 192.168.96.16 t=0 0 m=audio 27002 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=recvonly <------------> set_destination: Parsing <sip:302 at 192.168.96.16:5060;user=phone;transport=udp> for address/port to send to set_destination: set destination to 192.168.96.16, port 5060 Audio is at 192.168.96.5 port 16816 Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitt...

Hey i have an interesting topic to discuss here. The main goal here is to be able to make a video call between two WebRTC endpoints registered on asterisk 13 it is a feature that definitely asterisk 13 should support . the problems that i faced with this is the following and i hope i could get an advise here. asterisk 13 vanilla version has some issues marking the video packets this complain

...> a=fingerprint:sha-256 > 6D:5A:B7:7C:8E:1F:2A:F2:8C:DB:49:E4:8D:8F:BE:A0:84:AF:21:E2:A1:D7:DC:29:B1:EE:C3:0C:2C:AF:6C:04 > a=setup:actpass > a=mid:video > a=extmap:2 urn:ietf:params:rtp-hdrext:toffset > a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time > a=recvonly > a=rtcp-mux > a=rtpmap:100 VP8/90000 > a=rtcp-fb:100 ccm fir > a=rtcp-fb:100 nack > a=rtcp-fb:100 nack pli > a=rtcp-fb:100 goog-remb > a=rtpmap:116 red/90000 > a=rtpmap:117 ulpfec/90000 > a=rtpmap:96 rtx/90000 > a=fmtp:96 apt=100 > > 2-BUT when i do channel ori...

...1:5060> Content-Type: application/sdp Content-Length: 259 v=0 o=root 864806723 864806724 IN IP4 10.7.10.1 s=Asterisk PBX 1.6.0-beta2 c=IN IP4 10.7.10.1 t=0 0 m=audio 19968 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=recvonly <------------> -- Started music on hold, class 'default', on SIP/5253-0823eab0 <--- SIP read from UDP://10.7.10.51:5060 ---> ACK sip:5253 at 10.7.10.1:5060 SIP/2.0 Via: SIP/2.0/UDP 10.7.10.51:5060;branch=z9hG4bKb0a1c610b19e25613 Max-Forwards: 70 From: <sip:5878 at 10.7....

...inari) * ASTERISK-25890 - Asterisk 13.8.0 alembic database update fails (Reported by Harley Peters) * ASTERISK-25894 - [patch] webrtc video broken due to missing marker bits in RTP streams (Reported by Jacek Konieczny) * ASTERISK-25854 - No audio after HOLD/RESUME - incorrect a=recvonly in SDP from Asterisk (Reported by Robert McGilvray) * ASTERISK-25868 - Sorcery "append to category" should allow filters (Reported by Nick Repin) * ASTERISK-25873 - res_pjsip: Bundled pjproject: compile error, cannot find -lasteriskpj (Reported by Hans van Eijsden) * ASTERI...

...sockets exceed pjproject PJ_IOQUEUE_MAX_HANDLES (Reported by George Joseph) * ASTERISK-25707 - Long contact URIs or hostnames can crash pjproject/Asterisk under certain conditions (Reported by George Joseph) * ASTERISK-25854 - No audio after HOLD/RESUME - incorrect a=recvonly in SDP from Asterisk (Reported by Robert McGilvray) * ASTERISK-25882 - ARI: Crash can occur due to race condition when attempting to operate on a hung up channel (Part 2) (Reported by Richard Mudgett) * ASTERISK-25849 - chan_pjsip: transfers with direct media sometimes drops aud...