similar to: How to put someone on hold with Astersik Manager

Hello, I am writing a program based on Astersik Manager which needs to put calls on hold and to redirect them to others extensions. I haven't funded any action able to do this. Is there a way to put calls on hold using Asterisk Manager Actions? Amaury ?

Hi all, I am building an Asterisk PBX with voicemail and music on hold functions. An ISDN BRI line will also be available and G.729 IP-phones will be used. Are there patents rights applicable to France? Which licence could I use and how many ones are required (only one per phone or also for voicemail and MOH)? Regards Amaury -------------- next part -------------- An HTML

Hi, I have tried to use UserEvent() command to send data to Asterisk Manager from my dialplan. It works fine if the body only contains 1 line but I don't know how to send multiple arguments in the body. I have tested: UserEvent(eventname|body1|body2) UserEvent(eventname|body1\r\nbody2) But no one seems to work. Is it possible to do that and what is the correct syntax? Amaury

HI; Thanks for your reply. The reason for why I am going through asterisk in such case is just "using asterisk voicemail service" I mean: ATA1 calls ATA2, suppose ATA2 is unreachable or he is not at the office, then the call reroute (my GK is able to reroute calls if the first route is not valid) to atersik for voicemail service. Do you think I can handle it with asterisk native

Ok, this is probably a stupid question, but I can't seem to find anywhere where it tells how to put someone on hold on a zap channel. Flash gives me a dialtone and # tells me to enter a new extension, how can i just put the caller on hold. Pressing # then hanging up drops the call. Is there a simple way of doing this without transfering the user to a parking lot? Thanks, Jon.

On 01/04/2016 07:32 AM, Amaury SECHET wrote: > After a bit more investigation, it turns out that because %0 is stored > into %1 (after bitcast) and so %3 may have access to it and clobber it. Can you give a bit more context? I'm not sure which of the examples you're talking about. > > After a bit of thought, it is correct in the general case, but > definitively something

> On Jan 4, 2016, at 9:55 AM, Amaury SECHET via llvm-dev <llvm-dev at lists.llvm.org> wrote: > > > > 2016-01-04 18:21 GMT+01:00 Philip Reames <listmail at philipreames.com <mailto:listmail at philipreames.com>>: > On 01/04/2016 07:32 AM, Amaury SECHET wrote: >> After a bit more investigation, it turns out that because %0 is stored into %1 (after

Hi all, I don't find where you can setup the date (${VM_DATE}) in french for the mail. Is anybody can help me? Amaury BOSS?

Slash-dot is pointing to this article on Asterisk and Pingtel. http://www.theregister.co.uk/2005/05/22/pingtel_voip/ Paul Paul Mahler www.signate.com

Hello All: Does the Asterisk support to insert an off the board transcoder for a call? Thanks, Charles ____________________________________________________________________________________ Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping -------------- next part -------------- An

Hi to all!! I'm successful to connect Asterisk to PostgreSQL database... If it's possible, can anyone learn me how to store sip user in PostgreSQL database and how to configure voicemail?? Thanks for all!!!

Dear Sirs, I had compiled PWlib and OpenH323 correctly in my Fedora Core 2. But when I try to compile asterisk-oh323 I get the following error: `__use_AST_MUTEX_DEFINE_STATIC_rather_than_AST_MUTEX_INITIALIZER__' How can I solve it? Thank you for your help. Juanjo

Hi, I am new to the Asterisk world. I don't know much about the architecture, but I am involved in installing and configuring the VoIP system. My requirement is to build a VoIP system using the 4 input lines (ISDN up0 telephone lines), it must be possible to receive calls from outside through the 4 ISDN up0 input lines, and also possible for outgoing calls, conferencing .etc. I

We are planning to install a voip system based on asterisk for 2000-3000 retail locations and up to 6000-8000 sip accounts/users. Instead of setting up a new, centralized PSTN gateway, we are intend to use a CISCO gateway/router of an existing CISCO voip solution in the headquarter and we must able to call all CISCO based voip phones in the headquarter running together with a CCM. SIP-Phones

Hi everyone, I want to call from one Asterisk to another Asterisk via SIP, but i dn't know how. I have found out something in these links: http://www.voip-info.org/wiki/index.php?page=Asterisk+config+sip.conf http://www.voip-info.org/tiki-index.php?page=Asterisk%20SIP%20Channels but I don't understand them very well. At first, I tried simply doing this: In SIP Client:

Hi all, I'm building a karaoke service. Asterisk will play a music file, people can detect the point when they want to sing and record by press * key during the music is playing, and press # key to stop recording. I use 2 functions: ast_streamfile and ast_seekstream to play audio file, and function ast_waitstream_fr to detect whenever people press DTMF key. The problems is that, Asterisk

In my Asterisk server following things have been done to detect answering machines before the answered call connects to the agents in queue. In extension_additional.conf ============================== [ext-queues] include => ext-queues-custom exten => 5000,20,Macro(user-callerid,) ; changed the priority to 20 ............... ============================== In extension_custom.conf

Hello, I've got very annoying behaviour from our asterisk PBX. We have 12 channels T1 e&m wink start for TDM and using iax softphones internally (iaxcomm, but tried firefly-thirdparty and discarded for bad sound quality). Slackware 9.1 w/ kernel 2.4.26+ digium TE110P card. In some cases when call is placed from softphone to TDM, system does not detect call answered on Zap channel and

Right, later in the same file we have: // Reserve entries in the vector for each of the SUnits we are creating. This // ensure that reallocation of the vector won't happen, so SUnit*'s won't get // invalidated. // FIXME: Multiply by 2 because we may clone nodes during scheduling. // This is a temporary workaround. SUnits.reserve(NumNodes * 2); So for some reason *2 is

2015-12-26 18:32 GMT+01:00 Philip Reames <listmail at philipreames.com>: > On 12/26/2015 02:17 AM, Amaury SECHET via llvm-dev wrote: > > I'm trying to fix that bug: https://llvm.org/bugs/show_bug.cgi?id=20049 > > It turns out this is the kind of optimization that I really need, as when > it isn't done, all kind of other optimizations opportunities down the road