similar to: DTMF emulation with SIP INFO and direct media

    Some users have complained that their calls drop after about 30 seconds.  Not all, just some.  After looking at the log files the only difference I can find from the dropped calls is the following line: [2020-09-07 11:29:59] VERBOSE[21666][C-00000055] bridge.c: Bridge 14410400-5e04-4358-af0c-45fd71f6f5cd: switching from simple_bridge technology to native_rtp     Most calls just do:

On 21-11-16 15:17, Matthew Jordan wrote: > > On Mon, Nov 21, 2016 at 7:05 AM, Jonas Kellens > <jonas.kellens at telenet.be <mailto:jonas.kellens at telenet.be>> wrote: > > Hello > > when using Asterisk version 13.12.2 I notice that it takes up to > 30 seconds (sometimes even longer) for a call queue to call its > members. > >

Le 2017-07-05 18:51, Joshua Colp a ?crit : > On Wed, Jul 5, 2017, at 01:45 PM, Jean Aunis wrote: > >> Hello, I am struggling with a problem which I thought would be an easy one : bridging several channels together in a *smart* bridge. I emphasize *smart* : I want my bridge to be a native_rtp one when only two channels are involved, and switch to softmix technology when a third

Hello, I am struggling with a problem which I thought would be an easy one : bridging several channels together in a *smart* bridge. I emphasize *smart* : I want my bridge to be a native_rtp one when only two channels are involved, and switch to softmix technology when a third channel comes in. I thought I could use ConfBridge for that, but it creates a bridge that is not smart (it is of

Hi I'm trying to get the rebuilt parking functionality to work in Asterisk 12.0.0. In Asterisk 11.6.0 I managed to get a call to get parked by adding a dynamic feature in features.conf for the DMTF sequence *# which called a macro in extensions.conf, which then runned the ParkAndAnnounce application, and the call got parked. The syntax for ParkAndAnnounce I used was this (I don't

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

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

Hello when using Asterisk version 13.12.2 I notice that it takes up to 30 seconds (sometimes even longer) for a call queue to call its members. Example 1 : [Nov 21 08:17:57] pbx.c: Executing [queue at pbx-routing:15] Queue("SIP/incoming-00000246", "myqueue1,,,,300,,,") in new stack [Nov 21 08:17:57] res_musiconhold.c: Started music on hold, class 'default', on

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. >

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, 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

Hello, I am searching a way to dial a SIP peer, and if it does not answer within 20 seconds, play an announcement to the caller. This means that the caller would hear a ring tone for 20 seconds, and only then hear the announcement if the callee did not answer. I know it is possible to do this with ARI, but in this particular case I do not want to use ARI. I would like to do this purely with

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

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

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

Hi I'm trying to use Asterisk running as non-root user and selinux enabled. Asterisk is running ok, but astdb not works. When i try to put in astdb, console shows this message: WARNING[1853]: db.c:350 ast_db_put: Couldn't execute statment: SQL logic error or missing database CentOS 7.5.1804 Asterisk certified/13.21-cert3 [root at sv03 asterisk]# ls -lahZ /var/lib/asterisk/astdb.sqlite3

On Mon, Jun 5, 2017, at 04:26 PM, Michael Maier wrote: > On 06/05/2017 at 06:29 PM, Joshua Colp wrote: > > On Mon, Jun 5, 2017, at 01:22 PM, Michael Maier wrote: > >> > >> Do you have any idea where to start to look at? Adding additional output > >> in the source code? Which functions could be interesting? I may add own > >> debug code to see why things

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:

I am trying to send SMS from my grandstream GXV3240 Asterisk receives the message in a NOTIFY block. How can I get asterisk to run dialplan code when receiving these Notify SMS Message Blocks. I can then route them to my SMS provider. Any ideas are appreciated. Below is debug of a message sent from the phone when received no dialplan code is triggered. I am wounding if I need to

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: