similar to: do not start MoH when caller pres HOLD on mobile

Hi all If Asterisk is bridging a call between two SIP peers and one peer puts the other on hold by means of a re-INVITE with SDP containing a=sendonly, Asterisk will play locally generated MOH instead of relaying the media streamed by the SIP peer which took the hold action. Any ideas how to change that? (This is understandable if the peer is a handset but can be a problem if it is a PBX with

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

Hello, I am having problems with music on hold on grandstream phones. When I press Hold button on grandstream phone this is the debug of sip. But nothing happens, no music. Is it problem of asterisk or grandstream budget phone? Sip read: INVITE sip:1105@192.168.1.1 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.5;branch=z9hG4bK7fcd3a44e7721b41 From:

hi, i'm searching solution for playing media(moh,prompts,voicemail,recordings - wav format) from adobe flash player (web browser) flash cannot play wav directly (imho) i must convert files to any other format on-the-fly - i cannot use mp3 because of royalties - next option is swf (with ffmpeg), supported free audio codecs (http://en.wikipedia.org/wiki/Flash_Video#Format_details) *

Hi All, I have a problem with asterisk and call hold. In the re-invite package when I take the call to the hold, the SDP value "a=sendrecv" is present, according to the rfc3264 the sdp value a must be mark with "sendonly". I've already tried with Asterisk 1.8 and Asterisk 11, but there is the same problem. I've already read all the information about canreinvite and

Hello, I think I met a case similar to the one solved by [1] . Quoting this case : * res_pjsip: Handle deferred SDP hold/unhold properly. Some SIP devices indicate hold/unhold using deferred SDP reinvites. In other words, they provide no SDP in the reinvite. A typical transaction that starts hold might look something like this: * Device sends reinvite with no SDP * Asterisk

hi, i'm evaluating performance of centos7 i did tests on centos6 x86_64/distro kernel 2.6.32, asterisk 11.16.0 with 500calls (7sec alaw, simple dialplan, pass trough - sipp generators/asterisk receiver with answer/playback) scenario - sipp generators - asterisk - asterisk receiver (i wrote ansible scenario for this if you are interested) then i reinstalled system to centos7 x86_64/distro

Hi list, I have a strange comportment of the MOH system on my asterisk. When i respond to a call and after fews second i set this call in hold mode the correspondent listen the music fine. When i re-take my correspondent at T0 instant the music is paused. And when i re-hold him at T60 (60 second later) the sound is always at T0 when he was stopped at T0. So the music is stopped and don't

Hi all, I've asterisk 1.2.5 , and what is happening is this: Sip user agent "A" calls a pstn "phone B" Sip User agent Activates MOH. "B" starts listening. "A" doesn't hangup and just Disconnect Sipoftphone XLITE (exit) "B" stills listenning Music on Hold and "A" has left Asterisk, who hangs the call? only when B hangs...

Hi, when I put a call on hold from my Nokia E51 (SIP client), the other side does NOT hear music on hold although sip debug / wireshark shows that the E51 tells the asterisk that it now holds the call. Canreinvite is set to "no". Also, when parking a call (features.conf), the parked caller does not hear music on hold. In queues, when using "#" and when using the hold

I get MOH when the caller hangs up. Is there any way I can just get Busy tone. Regards Michael Knill -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060522/4905a84b/attachment.htm

The setup is this: 2 sip handsets (a Cisco 7960 and a 7961) exten 401/402 1 fxs/dahdi cordless phone, exten 201 rhino fxo/fxs analog card asterisk 1.4.31 This is running on a Soekris 5501 with Astlinux 0.7.2 While I do have FXO capabilities, no POTS lines are connected. When a call comes in (VoIP, either SIP or IAX) it is usually answered on one of the SIP Cisco phones(x 401 or 402). If it is

Hello, any plans for astricon videos? Thanks Marek

Hello all, I have a problem where problem with one way audio, and I think it's related to "a=sendonly" and a re-invite. Can anyone please assist? The scenario is as follows.... - We send an INVITE to a peer, and it replies with a "100 Trying", and then a "183 Session Progress" message containing "a=sendonly". - Asterisk plays the caller music on hold,

Can anyone shed some light on what happened? Asterisk 1.2.1 with Zaptel 1.2.1 Here is what I know happened: A call came into our main number and was answered Asterisk set the monitor CALLFILENAME and then started monitor. The call was directed to a context called "open" where all calls go during business hours. The dial plan has a Answer() again, and then played a message (custom/1)

Ok... I know I have asked this question before, but have never gotten an answer... When I press the hold button on my phone, should the caller hear music just like when I park the caller or transfer them to another extension? Please assist... -gcc

hi, i have following topology PSTN - Asterisk ---- internet -----  router - jssip client (wss) Asterisk 13.29.1 on public IP, chan_pjsip for wss, chan_sip/udp for SIP connection to PSTN router - public IP/private IP (NAT) jssip client - private IP - sip over websocket to Asterisk PJSIP ~30% of calls has problem with no audio. reason is that Asterisk is sending RTP to private IP of jssip

thank you very much. this is exactly whats needed for debug example output for your info [Dec 12 15:39:19] DEBUG[2182][C-00000000]: pjproject: <?>:         icess0x7f5d44081e88 .Added new remote candidate from the request: 2.2.2.2:57536 [Dec 12 15:39:19] DEBUG[2182][C-00000000]: pjproject: <?>:         icess0x7f5d44081e88 .New triggered check added: 1 [Dec 12 15:39:19]

I'm working through my list of features people will expect, and Hold Pickup is at the top at the moment -- has anyone done any work on this? We've had some unpleasant experiences with call parking, and everyone seems to like the Hold Pickup model. If you don't know what I mean by Hold Pickup, it's sort of a reverse transfer; pick up the nearest phone and dial

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: