similar to: Dialplan - using multiple AND or OR in set is it possible ?

On Tuesday 21 April 2020 at 12:54:49, Administrator wrote: > 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 ? No, logical operators are for comparing True

The mailing list will not receive emails from the forums. What I was referring to is that Discourse does provide the ability to receive emails for posts or threads you're interested in, and you are able to respond over email to them as well. On Mon, Dec 4, 2023 at 8:38 AM John Novack <jhnovack at stromberg-carlson.org> wrote: > > > Frank Vanoni wrote: > > On Mon,

Antony, It is correct. Noone connects to Asterisk box/server from outside.Callcentric SIP trunk configured and Asterisk maintains connection to it itself. No special ports opened, nothing. Connection happens from us to Callcentric and all calls routed in from CallcentricI don't know exactly how it's doing it by it works. Again, keep in mind it is working for many years for most / 90+% of

I've been using Asterisk, including administering and maintaining it, in some aspect since 2003, but this is the first time I have attempted a from-scratch installation and setup on my own. I'm following the instructions in the ePub edition of the book "Asterisk, the Definitive Guide, Fifth Edition," published by O’Reilly Media, Inc. in 2019, for Asterisk version 16 on a

hello list i have an issue related to outbound calls i can contact all the number except on number given by our provider in trunk the issue just when i configure my trunk in our server but when i configure the trunk directly in x-lite i can contact this number without issue below the cli == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Executing [0149xxxxxx at

I just had a user report that they called out to someone on a cell phone this morning, and after a short conversation, the call was dropped/lost. The person on the cell phone says that this is very rare, and would not suspect the dropped/lost call to be on their side. I have looked at the asterisk/full log as thoroughly as I can, and have pasted the lines which seem relevant to that call below.

You would need to give more information really. Your sip.conf file listing the entries for the phones especially which codecs are permitted. A copy of the 'asterisk -rvvv' console output when you make the call. On 27/03/15 17:05, Salaheddine Elharit wrote: > please no body has som with aastra can help me in this issue > > 2015-03-26 11:02 GMT+00:00 Salaheddine Elharit >

Got a strange situation [ext-queues] ... exten => h,2,ExecIf($[${CALLERID(num)} = ' ']?Set(var29=${SHELL(curl -X POST --header "Content-Type: application/json" --header "Accept: application/json" -d "{\"Phone\": ${FROMEXTEN}, \"Source\": \"asterisk\"}" "

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, Is there any way of telling Asteirsk to withhold answer subversion on a call till I call Answer. My DP looks like this: [incoming] Exten => 18005551212,1,Noop() same => n,Answer same => n,Mset(__uid=${SIPCALLID}) same => n,MixMonitor(/tmp/FROM_CALLER_${uid}-${START}.WAV) same => n,Dial(Local/1 at dial_call_center/n&Local/2 at dial_call_center /n&Local/3 at

Hi this message give me when I calling a number than actually not busy: "Dial failed due to trunk reporting BUSY - giving up" max channel is unlimited and sometimes it dial number ok but most of the time it gives me this error. Please inform me how can solve this problem. thanks -------------- next part -------------- An HTML attachment was scrubbed... URL:

I've been working on writing a subroutine to page groups of phones at once and I'm having some difficulty. My goal is to have a user call an extension, I record the page they wish to play, I then page out that recorded file to the phones in groups. [sub-masspage] exten => s,1,NoOP same => n,Answer same => n,Set(filename=$PAGE) same => n,Wait(1) same =>

hi,all when i send a call to other server use SIP trunk, i got this below, <--- SIP read from 222.46.18.52:5060 ---> SIP/2.0 403 Forbidden what's rong with is? > Asterisk 1.4.21.2, Copyright (C) 1999 - 2008 Digium, Inc. and others. > Created by Mark Spencer <markster at digium.com> > Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for

Howdy, I'm looking at enabling autopause on one of my queues where my queue members are bad about leaving their desks without pausing. The problem I see is that when the queue pauses an Member it doesn't jump into the dialplan to do so which means my handy device state and asterisk database driven Light for the Member showing their paused status won't update. My idea for solving

On Thursday 06 April 2023 at 15:48:24, Steve Matzura wrote: > this is the first time I have attempted a > from-scratch installation and setup on my own. ...<so far, so good>... > Then the weeds started to appear, and I was off into them. > > The first was the mention of Alembic. > Reading on, I found this, regarding an SQL database: > SQL? Database? Where ... what

On Monday 04 December 2023 at 13:39:51, Joshua C. Colp wrote: > The mailing list will not receive emails from the forums. What I was > referring to is that Discourse does provide the ability to receive emails > for posts or threads you're interested in, and you are able to respond over > email to them as well. I use this forum via its email interface, and I agree that it works.

i have a prablom here, i want to send a call from A to B use sip trunk , the call can sended B,but not work to PSTN. the message from B server. help pls,what's rong? > > <--- SIP read from 192.168.0.176:5060 ---> > INVITE sip:15921256331 at 192.168.0.151 <sip%3A15921256331 at 192.168.0.151>SIP/2.0 > Via: SIP/2.0/UDP 192.168.0.176:5060;branch=z9hG4bK51a51b96;rport

Hello everyone! I've had this problem for a while and cant figure it out. When an outside caller calls an extension on my asterisk system, they do not hear any sort of ringing. Inside extensions calling other extensions do hear ringing. We have 3 other asterisk systems that are configured the same way, but do not have this problem. We think it has something to do with asterisk 1.6. The other

We're having an issue where we call ReceiveFax in a context that includes a hangup extension and half the time dialplan execution doesn't continue after the fax is received successfully. Am I missing something simple here? Below is a sample call where this happened: The last log line for this channel/call is: [Feb 21 09:10:53] VERBOSE[13730] res_fax_digium.c: -- Channel

Hi All, I am trying to connect my asterisk box with freepbx to PSTN. I have purchased x100p FXO card and installed in my asterisk server. My freepbx detected the x100p FXO card and i can see the card specific details in command line. I have configured the following things. 1. OUTBOUND caller id and Dialing rules in Freepbx. 2. INBOUND route When i call to the PSTN number before