similar to: quoting arguments to System command in dialplan

hi. currently i am running the phonebook in astdb with *database put cidname 0123456789 "name_surname"* and i retrive it with *exten =>9876543210,1,Set(CALLERID(name)=${DB(cidname/${CALLERID(num)})})* Now, my system has mysql and i got all my contacts in there in a database is called *asterisk *and a table called *addressbook**. *password of the mysql is *whateverpasswd* how do i

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

I am using Sipura 841 phones and Asterisk CVS-v1-0-06/14/05. Whenever a call comes in with blank CIDName or CIDNum the phone reports the respective variable as "asterisk". I can manually set the variables to whatever I want: CIDName (alpha-numeric) & CIDNum (Numeric). But if I try to make them blank, or null, or maybe throw some alpha characters into CIDNum, they get reported

Hello, Sometimes, when a call comes in from the PSTN through our VoIP gateway, the information that is sent to our web page that logs calls includes the original CID name instead of the one that is we expect to be rewritten on the fly using Asterisk's LookupCIDName: ================= ;extensions.conf [internal] exten => group,1,LookupCIDName exten =>

Seems I can write to an existing file, but is there really no way of creating a new file to log some data to, without reverting to AGI? (will be different for each caller ID)

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

Hello everybody. Is there a way how to setup asterisk to notify caller's phone? Example: I have some numbers and names in asterisk database ( cidname, cidnum), and I want to display the name of person on my phone ( which has no addressbook, but can display chars ) which I am calling to be sure that I have dialed the right number. Thank you for any answer. Andrej

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. >

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? --

Hello, we want to use something like same = n,ExecIf($["A" = "B"]?Set(C=1) & Set(D=2) & ...) Problem is that result gives C=1) & Set(D=2) & ... Is there a possibility to use multiple AND or OR in such a way ? -- Daniel

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

Hello, I am running Asterisk 11 on CentOS 6.x. I have configured several queues as follows in extensions.conf: exten => s,1,Queue(myqueue,rtnC,18) same => n,Background(user_unavail) same => n,WaitExten(10) exten => 1,1,Voicemail(1111 at my-vm,s) This rings the phones in the queue for 18 seconds. If no queue members answer, the caller is then prompted to press 1 and leave a

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 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

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

On Tuesday 22 September 2020 at 13:27:27, Joshua C. Colp wrote: > On Tue, Sep 22, 2020 at 7:37 AM Antony Stone wrote: > > Hi. > > > > (Asterisk 16.2.1) > > > > I'm using AMI Originate to initiate calls, and I'm passing some > > additional data in to the dialplan context using the Variable: > > parameter. Works fine. > > > >

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 ?