similar to: Call engaged

Hi guys, Can someone tell me if there is a way to increase the volume of a conversation that occurs between two SIP channels or between a SIP and an IAX channel ? My headsets are set to the maximum volume but the voice is still low ... I know there is a configuration in zapata.conf for the digium cards, but is there a place I can set this up for RTP conversations ? Thanks,

When phone A registers, I want phone B to ring, when picked up, it should call phone A and connect the phones. Translated: When GF in Mexico powers up laptop where soft iax-phone registers automatically, I want to talk to her asap :-) How to? 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

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, 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

Hi all, I would like to ask please if there is a way to play a ring back tone from asterisk when the customer try to make a call...I already added the ringing function to the context in extensions .conf and it work perfectly...But the issue that the asterisk server is stoping playing back his own ring back tone as soon as it detect a ring back tone coming from the carrier side... Is there a way

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

Does asterisk have a way in the dialplan to generate tones? Say I want to play a tone 300Hz for 3 seconds. Can I do that? If not, can I use some system command to generate the wav file then just have asterisk play it? Jerry

Hi All I would like to be able to have an announcement played to an operator advising them of the queue the call came from before the call is pasted over to them, so they know how to greet the customer. Does anyone have any ideas or can point me to some resource which details this? Many Thanks in Advance. SP

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?

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

Hello, is anyone on the list using a normal cell/mobile phone which is able to act as a SIP client over WLAN? Or has anyone heard of a SIP client for cell/mobile phones running windows mobile 6.x? The phone should use SIP, when the asterisk server is reachable and should automatically switch to a German telco if it is not reachable. Thanks for any hints, Stefan --

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

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, 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! 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

Wow. My message made it to the list after more than 3 hours. 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 14998

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