similar to: CDR custom variable on second call leg - via originate or .call file

Hi. Im having trouble setting variables in channel dialplan and re-using them in Extension dialplan... Im using the following call file: Channel: Local/210332450 at ZonNew-Outbound CallerID: ZonNew-Outbound:49:210332450: MaxRetries: 5 RetryTime: 10 WaitTime: 60 Account: Outbound210332450 Context: agents Extension: 888210332450 Set: __PARTNER=ZonNew-Outbound Set: NUMBER=210332450 - In

Hello list. We just upgraded to 1.6.1.11. We are using real time information stored on mysql databases. That is all running fine. Now, since we upgraded, some member don't get calls from queues. In CLI: "queue show" shows something like: 611 (Local/611 at agents) with penalty 20 (realtime) (*In use*) has taken no calls yet We use the extension 611 in different computers, in the

Can you explain? Sent from Samsung Mobile <div>-------- Original message --------</div><div>From: Tiago Geada <tiago.geada at gmail.com> </div><div>Date:03/07/2014 9:04 PM (GMT+02:00) </div><div>To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> </div><div>Subject: Re:

Ok, commented out that line. It's still doing it. Reloaded dialplan. Please don't tell me I have to restart asterisk. Thomas M. Peters | Systems Administrator | tpeters at mcts.org Desk: 414.343.1720 | Helpdesk: x3400 or helpdesk at mcts.org >>> Tiago Geada <tiago.geada at gmail.com> 6/26/2015 12:07 PM >>> messages => error states to log error messages to

Hi, i tried to build Samba 2.2.4 under HPUX 10.20, but configure generated a Makfile with systax errors. make Make: line 664: syntax error. Stop. Makefile Line 664 shows: : $(WINBIND_NSS_PICOBJS) there should be something before the : configure said: checking whether to build winbind... no, unsupported on hpux10.20 is there a woraround, or can somebody tell me whats before the : Roland

Hi list! We currently have a PRI gateway composed by a box with two Digium quad-span PRI cards (a TE420 and a ). One of the cards is filled with TELCO1, while the other has first two slots filled with TELCO2, and 3rd slot with TELCO3. I am currently having (timer ?) issues on TELCO3 (span 7) D-Chan (202 as determined by dahdi_genconf ) is constantly failing causing on-going calls to terminate.

> being honest with i have been lost on what to do. > all i want is sent from my email a pdf file and then the server will sent it as fax. > what settings do i have to do regarding emailing to the server? what other settings do i have > to do? > is there a guide on that? > *Sent:* Friday, June 26, 2015 at 7:28 PM > *From:* "Tiago Geada" <tiago.geada at

The line you commented out was writing errors to /var/log/asterisk/messages The problem you are having is the logging to /var/log/messages via syslog. It appears that Asterisk is sending verbose logging out to syslog even though logger.conf does not have syslog configured. I am not sure why Asterisk is doing that, and I do not want to play on a production system. I added a filter to the

Hi All, I?m trying to do ?things? after my Dial application terminates (e.g. play IVR to called party, calling party, etc.). I?m trying to use the local channel for this purpose but so far with no success. I?m using 1.6.1.18 and this is my extensions.conf: [Internal] exten => _22,1,Dial(Local/${EXTEN}@CW/n) ; 22 is test number exten => _22,2,Noop(After Hangup) [CW] exten =>

Hello list, I'm looking for a way to have the estimated hold time on a queue prior to joining it. someone suggested to me to Queue() first for 1 sec, read variable QUEUEHOLDTIME, validade it and Queue() again. But as we're using real time configuration that would mean a event ENTERQUEUE and a LEAVEQUEUE too much in MySQL's queue_log any suggestions?? Thanks in advance

Hi! Since we upgraded to 1.6.1.11, asterisk is only outputing monitored files -in and -out. It is not mixing them in the end. queues.conf has monitor-type=MixMonitor... Would somebody help me debug why it doesn't mix the sounds?? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL:

HiCould someone please help me how to set Custom generated UUID in Originate action in AMI ? RegardsBabak -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150508/528d5ff1/attachment.html>

Hello, ChanSpy can not be used on a Channel that is being recorded with MixMonitor. How can I verify if a channel which I want to spy on, is currently not being recorded ?! Kind regards, Jonas. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120202/7954fe9e/attachment.htm>

I had been using google tts, but it started requiring a captcha for my browser, and via linux I can't access http://translate.google.com/translate_tts?q=test (redirects to captcha) as so, its not reliable On 27 August 2015 at 17:16, Carlos Chavez <cursor at telecomabmex.com> wrote: > On 8/26/15 1:15 PM, Tech Support wrote: > > All; > > I have a customer who is

Switched from Asterisk 1.8 to 13.3.2. Now it logs to /var/log/asterisk/full (good) as well as /var/log/messages (not good). Anyone know why? # grep -v "^;" logger.conf [general] [logfiles] console => notice,warning,error messages => error full => notice,warning,error,debug,verbose,dtmf,fax Thankfully, the .../full logs are rotating properly now (thanks Dale) but we don't

I hope his mother in law doesn't live with him. That's a support issue for sure. From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Kevin Larsen Sent: Thursday, June 25, 2015 2:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] asterisk email to fax > Since the O.P. said

Hello folks. I'm looking into having a web page displaying asterisk callers. We are a call centre, and having operators answering calls at home or whatever, they would need to have a real time application to display how manny callers are queuing, for how long etc. At first, I thought of phpagi. It connects to the manager and does a "core show channels concise". This has most of the

Hello list! I am trying to get hold of ${CDR(duration)} and ${CDR(billsec)} variables in h It seems that these variables always return 0. I am using Asterisk version 1.6.2.11. Can't I get these values other than using CDR reccords ?? -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi folks, We've been having a weird issue... It is happening more often in the last few months... Most inbound calls, we have in our dialplan before Queue(): Set(CALLERID(name)=${PARTNER}:0:${CALLERID(num)}:${UNIQUEID}:${CHANNEL}); So when the call rings a member, softphone will show this string .... The issue is that sometimes the string showing in the softphone is not the same. Its a