similar to: camp on off-line phone

Mostly to debug/test BLF, is there a softphone or another app. which can subscribe to hints on Asterisk? Heck, it doesn't even need to be able to do calls :-) Leif

Hi All; Which is better (to have more stable or release versions) of zaptel, libpri and asterisk: to use cvs or svn? In case of using cvs, why I need to type: export CVSROOT=:pserver:anoncvs:anoncvs at cvs.digium.com:/usr/cvsroot In other words: what is the use of pserver, anoncvs, ... with cvs checkout? Note: How can I know all the variables needed for cvs checkout so I might need to do

Hi, this is to inform everybody that the translation of my new book (unstable version) is online at http://www.the-asterisk-book.com The book is a GNU FDL project. So everybody who wants to participate is welcome to do so. Also, everybody who needs material for his own work, feel free to take it as long as the new material will become GNU FDL too. I am glad that Stephen Bosch (who you

I'm sure this has been discussed many times, but I have a question about hoteling. My understanding would be this: A phone sitting on a desk. A user hits 9000 and it asks what extension you'd like to become. You type "1001" and then it asks for your password. You type 1234, and it says you're "logged in". You now are accepting calls at your phone and you're

Hi all! I am trying to figure out how DEVICE_STATE is working, no luck so far. sip.conf [0317998975] type=friend regexten=0317998975 secret=???? username=0317998975 callerid="Magnus Benngard" mailbox=0317998975 at inputinterior.se host=dynamic canreinvite=yes dtmfmode=rfc2833 nat=yes disallow=all allow=alaw extensions.conf exten => 0317998975,hint,SIP/0317998975 exten =>

I have a large dial plan here with over 3000 lines, and several dozen macros. As it grew, it became apparent that there was some problems. 1. When you pass arguments to a macro in the form of $ARG1, $ARG2 etc, if that macro calls another macro, and passes arguments like this as well, you lose the original values. 2. When the macro's 'return' some value, it has to set a channel

What Asterisk command I can use to send a SIP INFO command? Thanks for pointers. Yuan Liu

Hi List; Can someone advise me why in the below context, it does not run the Background step? Once I dial 1000, then it hangup and give congestion signal? If I comment the ResponseTimeOut, then it run the Background but it does not wait till caller enter the digits, once the sound file finish, then it hangup (congestion signal), also in all the situation, it does not go for the t extension, why?

Hi, I have remove the comment defor res_odbc.so and res_config_odbc.so in my modules.conf, but the module is still not loading when I do: module show like odbc I have o module returned anybody knows why? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090127/0963b5a4/attachment.htm

Hi, is it possible to have the information stored in /var/log/asterisk/queue_log realtime in mySQL? thanks

I want to use JabberSend in my dialplan, but I saw that my Asterisk does not support Jabber. Also I have nowhere a module res_jabber.so... So I thought I'd rebuild my Asterisk. In menuselect I saw that res_jabber was dependent of 'iksemel' and 'gnutls'. In my yum repositories I can find a gnutls.i386, but what is this iksemel-beast ??? There is info to find via google on

Hello everybody, I was wondering what is postponing the 1.4.26 release? I thought it was scheculed for last week. Is there something we can do to help to release this version? There is no more issue reported on https://issues.asterisk.org/ for the time being. Best Regards, -- -- Marc LEURENT lftsy at leurent.eu -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi! http://www.voip-info.org/tiki-index.php?page=Asterisk+sound+files+international#German lists a plenty of sound files for German. Can someone recommend one for Asterisk 1.4 (any maybe 1.6 soon). thanks klaus

I am trying to setup an extension of *7XXX that will allow me to dial *7 and then any extension and use the Pickup application to pickup a ringing phone. Ideally it will also check if the phone is ringing somehow and then either dial the extension or pick it up if it is ringing. But I can't get that far. If I use *7268 specially it works fine, but as soon as I introduce any wild

Hi! I have the following questions/problems with * 1.4. We have several Snom phones (320 and 360). Hints are configured in extensions.conf (core show hints shows the correct values). My Snom phone is registered to some numbers (validated by using sip show subscriptions). I see the lights blinking if someone calls the subscribed number and steady lights if the call is established. So far, so

Is the "Whats New at Digium the Asterisk Company" message I got from digium at en25.com really from Digium? If so I suggest to send it from digium.com and not to use those shady Eloqua redirect URLs. Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk?

Could someone with inside knowledge comment on that? If the source code says "deprecated" but the CLI help does not mention that - whom do I trust? -------- Original message -------- Subject: Re: [asterisk-users] Ex-Girlfriend syntax and RealTime Extensions From: Philipp Kempgen <philipp.kempgen@amooma.de> Thomas Kenyon wrote: > Philipp Kempgen wrote: >> You might use

Dear, I added new field to cdr table , named "service" and type varchar(20), but in extensions.conf with the following command, nothing to be saved. exten => _X.,1,Set(CDR(service)=OUT) does asterisk support this ability ? is any setting must be changed, before that ? best Mani

Hi, Does anyone know how to include a file in AEL using the #include "filename" syntax in .conf files? Regards, Philipp -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? -> http://www.das-asterisk-buch.de Gesch?ftsf?hrer: Stefan Wintermeyer Handelsregister: Neuwied B

Hi, is it possible to hangup a channel from another channel? I want to finish a call from another channel, but if I put exten => h,n,HangUp(channelname) it doesn't hangup... Is that correct? Thanks, _________________________________________________________________ -------------- next part -------------- An HTML attachment was scrubbed... URL: