similar to: (fast) AGI and AMI synchronization ?

Hi folks, I repeat "as is" the title of a post someone did a few months ago, since I am facing the same problem and did not see one single answer to his post. Maybe I'll be a little bit more lucky. When I'm trying to issue an Atxfer AMI command, in the asterisk 1.8 branch, what happens is that some DTMF's are sent, like this : [Mar 22 15:46:27] DTMF[5910]: channel.c:3900

Hello, I would like to known how to use DUNDi with a Lua dialplan ? In extensions.conf, we should do like these: |[lookupdundi] switch => DUNDi/priv [internal] include => dundiextens include => lookupdundi exten => _XXXX,2,NoOp(calling ${EXTEN}) exten => _XXXX,n,Dial(SIP/${EXTEN}) exten => _XXXX,n,Hangup()| priority 1 is either defined in dundiextens (local registered

Coin, Quoting Quartexx <quartex73 at gmail.com>: > I'm experiencing your same issue. Have you found a fix for this? > Thanks in advance for your reply I downgraded to 3.2.5 to get a working version on the production server, and made a few more tests with 3.4.5. I discovered the ACL bug was linked to the full_audit vfs object, and deactivating it fixed the modification

Hello, Looking the asterisk 1.8 API documentation (http://www.asterisk.org/astdocs/api/index.html), I see a lot of new fields for sip register uris: register => [peer?][transport://]user[@domain][:secret[:authuser]]@host[:port][/extension][~expiry] But the *peer* is not explained anywhere. What it is for ? Regards, Guillaume Bour. -- Guillaume Bour<gbour at proformatique.com>

Hello, After upgrading from Debian Etch with samba 3.0.24-6etch10 to Lenny with a backport of 2:3.4.5~dfsg-1 (with libtalloc2 2.0.1-1), i get a fully working service but with a strange ACL bug : people can create/delete/rename files, but not modify them (error "espace insuffisant pour traiter cette commande" in french, which should translate into "Not enough storage is

Friends, a new dialplan application is now available for testing : http://svn.digium.com/view/asterisk/team/phsultan/jabberreceive/ The corresponding feature request is located here : http://bugs.digium.com/view.php?id=12569 What can you do with it? Well, a direct usage of this application is to make an easy to use GoogleTalk voice gateway out of Asterisk. Here is an example (assuming the

Hi, I'm stucked in situation, and look for a work around if possible in Asterisk. I have a dialplan, [default] exten => 111222,n,Set(fu_callerid=141688xyxzz) exten => _X.,n,NoOp(Callerid ${fu_callerid}) exten => _X.,n,wait(2) exten => _X.,n,Answer() ? When, ?Answer Application is called AMI Event is triggered like this.. ? ? ? ? ? 'Event' => 'Newexten', ? ? ? ?

I used to be able to pass dial strings to IAX2 providers with # characters at the end of the string. This is how we end dial strings for international calls. So, I would like to be able to selectivity chop off any # characters at the end of string, only if they exist. Basically as follows (chopping off the leading '9' with ${EXTEN:1} syntax: EXTEN from Phone EXTEN for Dial String

Hi Everyone, I am writing an open source application that brings desktops widgets to OS X (http://sourceforge.net/projects/astrxtools4osx/), for which I am trying to get my head around the Asterisk Manager Interface. I had been using the Event: NewCallerid to detect a new call which my Asterisk server doesn't seem to send to the socket anymore, because of which I have reverted to using

hi all, Can someone help me with a kind of howto build call center around asterisk with all the necessary features like CTI, call recordings, call spying, real time monitoring etc? I will be glad if it is an open source code. Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi list, I recently purchased an ISDNguard from Junghanns. It came with no software and there is no sign on their website or in any of their documentation where to download it. I have looked in http://www.junghanns.net/downloads/ and there is no sign of it there either. The only thing remotly close ther is isdnguard-asterisk-1.2.13.patch. Their documentation refers to /usr/sbin/ISDNguard. Where

Hi all, what's your opinion about CDR-Event (like Hangup or Ring etc.) on AstManager ? Or, is something like this already implemented ? Regards, Thomas

hi i logged on to manager API from other terminal by telnet IPADDR 5038 now logged in with username mark let's say this connection Window A now i opened another connection with Manager API with same usename lets say this window B now if i give a command like originate,Redirect through window A connection , can i able to see its response:success/failure Originate:failed/succesfully

Hello, all! I'm looking at the wiki page and info on the mailing list and I'm getting conflicting info... I am using the manager API from the telnet CLI and I am testing creating calls with it. I login with events: on and I can originate calls just fine. However, when I set ActionID on an Originate, I cannot see anywhere where that actionid carries into the Event output. But I found

Hi there, I am currently developing the ability to have a unified system/telephone login, with SIP phones paired to a computer. When a user logs into a computer, a notification is sent to an external service program which connects to Asterisk through the manager API. Besides that, the service program tracks user status on the computer, and triggers actions depending on various conditions.

The Asterisk Development Team has announced the release of Asterisk 11.17.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 11.17.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: New

Dear asterisk users, We would like to announce 2 VoIP activites organised by XiVO. For those who've never heard about the project: XiVO is a free/open source telephone system under the GPLv3 that has been based on asterisk since 2005. More info on our website: http://xivo.io [3]. We have an IRC channel, #xivo on freenode and a forum [0]. Feedback and contributions are always welcome :) The

On Thu, Feb 6, 2020 at 12:34 PM sduthil at wazo.io <sduthil at wazo.io> wrote: > On 1/29/20 2:31 PM, George Joseph wrote: > > For those of you who actually process SIP MESSAGE requests... Do you > > use any of the AMI events generated by the "Message/ast_msg_queue" > > channel? We want to change that channel to an "internal" channel that > >

Dne 6.8.2015 v 21:00 Sylvain Boily napsal(a): > Hello, > > Le 2015-08-06 09:24, Marek Cervenka a ?crit : >> hi, >> >> there is updated skills based routing patch for asterisk queue >> please test if you have time >> >> https://issues.asterisk.org/jira/browse/ASTERISK-17366?jql=text%20~%20%22skills%22 >> >> > > You can find the latest

Hello, I did some tests because i'm interesting to transfer a non stasis bridge to a stasis bridge and i found a strange situation. A call B B answer You have a bridge On my asterisk CLI: xivo*CLI> bridge show b1d8fb21-ec6d-469a-9dde-bb6bfd5618cc Id: b1d8fb21-ec6d-469a-9dde-bb6bfd5618cc Type: basic Technology: simple_bridge Num-Channels: 2 Channel: SIP/tcu9tz-00000032 Channel: