similar to: Dial and start music on hold after timeout

Sorry, I forgot to write that the SIP peer must keep ringing while the announcement is being played. Le 22/08/2016 ? 17:42, John Kiniston a ?crit : > This seems like the obvious answer but maybe I'm misunderstanding the > question. > > exten => s,1,Dial(SIP/alice,20) > same => n,Playback(myannouncement) > same => n,NoOP(Whatever else you want to do goes

Thank you for the idea. The problem with RetryDial, is that it will cancel the first call, play the announce and then dial the SIP peer once again, so the telephone will display a missed call. I would prefer to do everything in a single call. Le 22/08/2016 ? 17:57, John Kiniston a ?crit : > You could try using RetryDial() instead of Dial, It supports playing > an announcement. >

How about: exten => s,1,Dial(SIP/alice&LOCAL/555 at delayed-announce,40) [delayed-announce] exten => 555,1,Wait(20) same => n,Playback(myannouncement,noanswer) same => n,NoOP(Whatever else you want to do goes here) The 'noanswer' option on the Playback means that SIP/alice should continue to ring for the remaining 20 of the 40 seconds, as the Playback will not answer

Maybe try progress() instead of answer () ?????? 23 ????? 2016 7:19 PM,? "Jean Aunis" <jean.aunis at prescom.fr> ???: > Thank you, I just tried your suggestion. Strangely, the announcement is > played only if I try to dial a SIP peer which is not available (not > registered to be more precise). If the SIP peer is available, I only get > the ring tone, and never hear

Hello, Has anyone already implemented some sort of call preemption in Asterisk ? I am trying to achieve something like this : - I want to limit the number of calls on a given SIP peer to 10 - on the other hand, some calls have higher priority than others - when the ceiling of 10 calls is reached and a call with a high priority is attempted, I would like to drop a call with a lower priority

How can we set the music on hold class using the Dynamic Parking lots? The variables set the PARKINGLOT, PARKINGDYNAMIC, PARKINGDYNPOS,PARKINGEXT,PARKINGDYNCONTEXT I can't find a PARKINGMOH variable. This is becoming a big issue. We are using the current release 11. version We have to be able to set the MOH dynamically I just can't find the mechanism. Any ideas? Thanks

Hiya, When I hang up on a call to my stasis app I’m getting multiple channelDestroyed events for the same channel: app.js:985:13) Channel was destroyed: 1547220509.77 app.js:1029:17) This was a customer app.js:1030:17) Checking if this was a customer talking to an agent app.js:1043:21) Customer was not talking to anyone app.js:1126:13) 2019-01-11 10:28:29 app.js:985:13) Channel was destroyed:

On Thu, Jan 30, 2020 at 12:18 AM Jean Aunis <jean.aunis at prescom.fr> wrote: > Hello, > > I use UserEvents generated by the Message/ast_message_queue channel with > the UserEvent application. > > Regards > > Jean > Thanks Jean. We're looking at alternatives. > Le 29/01/2020 à 20:31, George Joseph a écrit : > > For those of you who actually

Hello, I'm familiar with Pickup/PickupChan for taking a ringing call, but does anyone know how a phone can "steal" an already answered call from another phone? Our users have decided that call parking is too long-winded and don't want to use that. For example: phone A calls phone B, phone B answers the call, phone C dials something to "steal" the call from B, and

hi all, can anyone give me a guide on any asterisk admin solution / interface for config management, and monitoring? No database use is intended and I prefer open source. Thanks for support. Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160516/98f6e448/attachment.html>

Hello, I think there is an issue when DTMF are handled with SIP INFO and direct media is enabled. When I receive a SIP INFO, the logs tell me that a "DTMF begin" is generated, but no related "DTMF end" is generated, unless the call is ended. Here is an excerpt of the logs : *--- SIP INFO received **on **SIP/xxx-00000004:* [Dec 13 11:56:16] DTMF[18193][C-00000005]

Hi Is it a normal behavior of Asterisk put a call on hold when receive a Session Progress with media address 0.0.0.0 in SDP? I believe the call on hold should be initiate with a re-invite. Thanks -- Att, Rafael Saraiva -------------- next part -------------- An HTML attachment was scrubbed... URL:

On Thu, Jan 30, 2020 at 3:18 AM Jean Aunis <jean.aunis at prescom.fr> wrote: > Hello, > > I use UserEvents generated by the Message/ast_message_queue channel with > the UserEvent application. > Do you use any aspects of the channel itself in the user events, or merely the contents of the user event and what you've placed in it? -- Joshua C. Colp Asterisk Technical Lead

That's just what I'm using, John. But I'm getting (eg) [Nov 4 21:46:16] ERROR[1676][C-00000003]: func_env.c:449 file2format: Cannot open '/home/logs/anonymous.txt': No such file or directory [Nov 4 21:46:16] ERROR[1676][C-00000003]: func_env.c:949 file_write: File '/home/logs/anonymous.txt' not in line format Asterisk is running as root (yeah, I know!), and has

For those of you who actually process SIP MESSAGE requests... Do you use any of the AMI events generated by the "Message/ast_msg_queue" channel? We want to change that channel to an "internal" channel that doesn't generate AMI events (for performance reasons) but we need to know if anyone's using them first. Thanks! -- George Joseph Asterisk Software Developer

On Wed, 2019-02-20 at 11:46 -0700, John Kiniston wrote: > Use the IF function to evaluate and change the dial command directly. Thanks for taking the time, but that doesn't actually answer the question I asked. It in fact answers the caveat I specifically mentioned: > Granted the particular above example could probably be better > written to simply modify $ARG2 based on ${SIP}

Hi, 1. How do you then, synced then unread message presence with custom device status ? From an external program ? When a user leaves VoiceMailMan application ? Using externnotify ? 2. What is MWI:101 at default expression for (see [2] ? Cheers [2] https://wiki.freepbx.org/display/FPG/Subscribe+a+BLF+button+to+Monitor+a+Voicemail+Box 2017-11-21 17:58 GMT+01:00 John Kiniston <johnkiniston at

I'm implementing a SBC with my Asterisk PBX but the keeps disabling the trunk group I've configured and I think it may be because Asterisk is returning a 4r04 to the OPTIONS. I've created a test context and have put in a wildcard pattern match to try and catch those options but it doesn't seem to work. Is there a way to have asterisk respond with an 200 OK instead of a 404? --

More testing. Test test test. :-) -- Matthew Fredrickson Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

That is the general idea. But how do i make it work? is there somewhere ready? On Wed, Jan 10, 2018 at 6:39 PM, John Kiniston <johnkiniston at gmail.com> wrote: > Define your *72 and *73 extensions in your internal context, Have them set > a value in the ASTDB that you then check when dialing your handsets. > > The same can be done for call forwarding, store a number in the