similar to: Application Queue context that calls the extensions

Guys, I have a problem. I have a queue on asterisk 1.8 that members are added dynamically via the AMI QueueAdd. When you run the CLI a "reload app_queue.so" all members who were in the queue disappear. This is a bug or some parameter that I do not know? Would have another way to do the reload queue without any risk to members who are already in it? tks Ed -------------- next part

Hi All I have been looking at this problem for a few days/weeks now and after some advice please. I currently have a customer on 11.25.3 and I am in the process of upgrading versions and OS (Debian) and all things that involves mysql -> PDO etc The problem I have is the customer want a simple call distribution like this Extn 1001, 1002, 1003 to be called on an incoming call - if they

Hi John This works fine providing extensions 1001,1002 and 1003 are "Incall" or "Paused" - the problem appears to be that is a handset say 1002 is "ringing" then the 2xxx then the penalty is not honoured. This is well described in the History section of the following link https://wiki.freepbx.org/display/PPS/lazymembers+patch+to+app_queue As I say this seems to

people I have a running Asterisk 1.8.28 in great Dell server with two xeon processors and 16gb of ram and HD SAS 15k (Raid 1). This server is recording all calls (placed to record the audio in a ram disk), the entire CDR goes straight to MySQL by cdr_mysql.so. Each call runs some validation and AGI's have an auto dialer system that generates calls over the manager. Calls originate and

My settings for the queue.log are in the [general] section of logger.conf I'm running 13, I didn't see what version you said you were running. If I wanted to add a LOCAL channel to my queue I'd do it as member => LOCAL/7124 at kiniston-intern,0,John,hint:7124 at kiniston-intern On Thu, Nov 15, 2018 at 2:38 PM Ivan Demkovitch <idemkovitch at yahoo.com> wrote: > John,

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

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

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

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. >

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

John, >From looking at the wiki won't STRFIME just give me what I need based on the unix time that I put in? What I am actually looking to do is convert over from 12 hour format to 24 (unless strftime does just that and I don't kow what am I am doing?). On Thu, Feb 13, 2020 at 12:03 PM John Kiniston <johnkiniston at gmail.com> wrote: > Try using the STRFIME function

Thanks for the link. Are there no free alternatives for speech recognition?

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

Hello list members, after programing of dialplan I have some messy Custom:hints which I can see in 'devstate list'. I didn't find any possibility how to remove this hints from Asterisk and I want remove them.? Can you help me with that, please? I tried search about that something in documentation or on Google, but I didn't find anything.? asterisk*CLI> devstate list ?

thanks John yeah, your approach is much siple, i've tried it but i'm not able do detect DTMF tones. it seems that on calls that i receive DTMF tones are handled correctly, but on calls generated from Asterisk to the world when the called side sends some DTMF digits they are not detected: -- Executing [s at macro-myconnector:1] NoOp("SIP/pbx2-000004b2", "") in new

Hello, Is there a way to let someone access to Asterisk CLI and type whatever command (s)he likes but the shell command (the ones started by !) ? Ideally, it could be an argument to rasterisk: rasterisk --no-shell When done, a session could be like this: > pjsip show endpoints ... > core reload ... > !rm /etc/foobar Forbidden Suggestions ? Best regards -------------- next part

Hello All, I will like to configure Asterisk to use PIN Code for all outgoing international calls. Also, any suggestions as to when should I prompt users for code prior to dialing the number or after dialing the number? can someone provide with a example on how to accomplish this goal? I am a bit confuse by this :

Hi Is there any way to set the presence state of a peer to in-use in asterisk 1.8? The idea is to integrate DND buttons on phones to BLF. Regards -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)161 660 2350 f: +44 (0)161 660 9825 e: ish at pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street