similar to: Weird issue with Set(CALLERID(name)=string);

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

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>

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

Hi We have the following test .call file and test dialplan: I can't set a custom CDR var to a value on one channel leg, and another value on the connected channel leg? Is there a way I can woraround this issue? ## test call file Channel: Local/queue at TiagoGeada CallerID: teste-geada:0:210332450: MaxRetries: 0 RetryTime: 1 WaitTime: 8640 Account: teste-geada Context: TiagoGeada

Hello guys, We're currently working on asterisk trying to create our own SIP phone, because we need special features. But dunno maybe there's other people who already done that before. Basically, we are a inbound call center. We have serveral customers with different phone numbers, which are redirected to us. When we receive a call coming on a specific phone number, the call gets

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.

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:

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

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

Guys. Anybody knows why this is happening? Seems every time I make an internal call, the manager shows this and I don't get the callerid on my identapop but rather the calledid.. Event: Dial Privilege: call,all Source: SIP/intruder1-85f0 Destination: SIP/test-f037 CallerID: 201 CallerIDName: Anton Krall SrcUniqueID: 1117038116.7 DestUniqueID: 1117038116.8 Event: Newchannel Privilege:

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 all! Has anyone expirienced the following:? With an IAXclient softphone (like diax/iaxcomm/etc) Dialing to the PSTN (zap) or a SIP device has no problems .. but when I make calls between 2 softphones I have weird problems.... in about 4 out of 10 IAX-2-IAX softphone calls I get a big delay .. in the beginning of the call it's all okay... (delay < 0.5 sec) but the longer the call

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: