similar to: Custom UUID in originate and AMI

HiI want to load balance SIP calls between two(or more) Asterisks with only DNS SRV. I used bidirectional sync Unison to synchronize configuration files and internal database file between two Asterisk boxes.The problem is when a calls come to Asterisk1 but SIPendpoint is registered on Asterisk2.How we can check a SIP endpoint is registered or not and what is Contact?information in Dialplan ?

what do you mean by "set" you can use like: Variable: __CUSTOMID=UUID-string\r\n to be able to read back ${CUSTOMID} back in the dialplan ... ? On 8 May 2015 at 19:04, Mehdi Shirazi <mahdi_shirazi at yahoo.com> wrote: > Hi > Could someone please help me how to set Custom generated UUID in Originate > action in AMI ? > > Regards > Babak > > > -- >

Hi I installed Asterisk 1.8.7 with CD ISO(Elastix 2.2) I want to use DAHDISendCallreroutingFacility Application on a PRI link(LIBPRI Already installed). according to https://wiki.asterisk.org/wiki/display/AST/New+in+1.8 Asterisk 1.8 include this application but I cannot see it with "core show applications" Do I need to install mISDN or other modules for using that ? Regards M.Shirazi

Hi If we have a shared RealTime database for sip registration of multiple Asterisk servers, is there a way to realize which Asterisk server registered sip phones ? RegardsM.Shirazi? -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150817/62c5cd3c/attachment.html>

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 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

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 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

I am convinced to use llvm-gcc. As I mentioned in my previous email, I have changed native code generation in llc to generate code with a different stack organization. In order to have working binaries, the libraries must be compiled with the new tool and have the same stack organization. Now that I cannot use llc, it seems that I have to modify llvm-gcc to have it generate code with the

Thank you Chris. Actually, I have changed the stack organization in the "llc" source codes for x86 target. Thus, I need libraries compiled with the new stack organization. If I use llvm-gcc (and apparently also llvmc), llc will not be invoked. I was hoping that llvmc would call llvm-gcc to compile to bc file and then call llc to compile to native assembly, but apparently this

Hi, I am trying to compile glibc on Linux using llvm. I need to do this for a research project. The problem is that llvmc doesn't work, and I guess I have to set CC=llvmc to get the glic compiled. For some reason llvmc complains that it cannot find %llvmcc1%. I have this problem both on my Mac and my Linux system. Other tools that I have tested including llvm-gcc work without any

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:

Reid, Thank you for your quick reply. Yes, I have built and installed llvm-gcc and "llvm-gcc --print-prog- name=cc1" return the correct path for cc1, so I assume its path is set correctly in the config files, but I haven't installed config files separately. Do I need to do that? Is there any documentation about it? Thank you again, Babak On May 8, 2007, at 7:33 PM, Reid