similar to: Playing sounds while dialling

Hi all, I want to set up an extension which dials a group of phones while at the same time plays a message ("Press 1 to leave a message") and listens for DTMF. I haven't played around yet but the way I read the docs this isn't possible as Thanks, Peter Spikings This message has been comprehensively scanned for viruses, please visit http://virus.e2e-filter.com/ for details.

Hi all, I want to set up an extension which dials a group of phones while at the same time plays a message ("Press 1 to leave a message") and listens for DTMF. I haven't played around yet but the way I read the docs this isn't possible as the dial command doesn't have appropriate options and takes complete control of the channel. However surely this is a normal thing to want

I am checking on the SetGroup / CheckGroup commands, but I have some troubles to undestand the examples. SetGroup(moh) can be moh anything as I like? Usually moh stands for "music on hold" CheckGroup(1) checks if somebody in in group "moh". Does it mean I can only have one SetGroup(xxx) ?? When I look at example 2 than I see two SetGroup commands and one CheckGroup

should I use mpg123 with asterisk 1.2.7 or should i use the native player asterisk has? the target machine will receive heavy load. also, has anyone succedded in compiling mpg123 in a dual core pentium with centos 4.3 ? -- ------------------------------------------- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama

Hi all i want to know where i can find french sounds for asterisk. I don't have any studio to register good sounds. Bests regards Serge ___________________________________________________________________________ Nouveau : t?l?phonez moins cher avec Yahoo! Messenger ! D?couvez les tarifs exceptionnels pour appeler la France et

Hi I was doing some bandwidth testing, and my incomming usage is 36% more than my outgoing bandwidth. The setup is IAX2 trunking using GSM codec. Is there any obvious reason I am overlooking to figure out why there is such a big difference between the two.? I am using CVS-head September 3rd, maybe there is a version skew? Any suggestions will be appreciated. Thanks Clive

Hi, What port does mpg123 uses to play music on when it starts MoH after asterisk has put called on hold? Zach A

OK. So I did a test last night. All of asterisk's threads where using 0.0% CPU. I made 1 call to our call queue. CPU jumped to average of 9% and stayed around that for the 2 minutes I was in the queue just listening to music on hold. MOH is in MP3 format and I'm using format_mp3. Phone was linksys PAP2-NA using G729. Can I reasonably assume that the 9% was decoding the MP3, then

Can anyone recommend a company that does professional Asterisk recordings for things like IVR, greetings, MOH, announcements, etc? Thanks, Waldo

Guys. Maybe this is asking for a lot :) but is there any web client that can use IAX2 and ilbc? This is for a "call us" web idea.... Any leads?

Hey, I am here at Astricon and about to go down to registration. Is there any interest in pictures if I take my digital camera? I am sure that someone is already doing this. (Probably someone official). I would take pictures of each day and upload them to my website if anyone is interested. Let me know! -- Kristian Kielhofner

This may sound kind of crazy and I maybe missing something. But are you placing the call on hold so you can hear the hold music. This may not be the case but you may have to place the call on hold to here the music. Also you mentioned sound, you do not need a sound card in the asterisk box to use this hold music feature. Hope this helps. -----Original Message----- From:

Hi friends, I am using Real Time Asterisk Architecture where I have put the Sip users/peers and extensions defining the dialplan in tables in a mysql database. Currently, asterisk points to my single database server as configured: ------------------------------------------ /etc/asterisk/res_mysql.conf ------------------------------------------ [general] dbhost = xxx dbname =

We are tracking the following situation: SIP client connects to our Asterisk server, and then connects to another SIP user. Re-invite is OFF, so Asterisk is in the middle of the whole conversation. When one SIP client sends DTMF tones, the SIP client uses RFC2833 to send the tones to the server. (This is correct). The server then sends RFC2833 tones out to the other SIP client. The problem is,

Hi Fran, you could do it using Adobe/Macromedia Flash Media Server 2, but I guess that's not the answer you are looking for. If you manage to do this and release it under GPL I'll kick in $50 for a bounty. Regards, Dean Collins dean@collins.net.pr +1-212-203-4357 +61-2-9016-5642 (Sydney in-dial). -----Original Message----- From: asterisk-users-bounces@lists.digium.com

As the title suggests, I was wondering if there was another source of documentation for Asterisk. Related: If one wanted to contribute to documentation, who would one contact? Thanks! Sean

Hello In my extensions.conf file: [frompstnisdn] exten => s,1,Dial(SIP/200&SIP/202,20) exten => s,2,Voicemail(su200) exten => s,3,Hangup I use the s, start, extension to handle incoming calls. In my zapata.conf: context=frompstnisdn This works ok on another asterisk box I setup. But on incoming calls I get: -- Extension '787367' in context 'frompstnisdn'

Hi There, Thanks for all your suggestions. I have now compiled asterisk from cvs running on FD4. I have performed all the suggested configurations: > busydetect=yes ;changed 17.03.04 from no > busycount=7 ; added as above > for me the distro asterisk package didnt hang up properly on busy > > signal. I > needed to download the source and uncomment BUSYDETECT_MARTIN

Hi, I'm looking for a tutorial or installation guide for SER to be used with asterisk to solve the remote SIP agent problem. All the documents available are for large scale installation. Any help is highly appreciated. Regards. __________________________________ Yahoo! Mail Mobile Take Yahoo! Mail with you! Check email on your mobile phone. http://mobile.yahoo.com/learn/mail

I have an asterisk box and SIP / IAX2 phones. To call out, users have to add 0 (zero) before a real telephone number. That means, that if they want to call someone that has a number 123456, they have to call 0-123456. Simple, right? This has a serious drawback though - when someone calls us from the number 123456, we see the callerID 123456, and we're unable to use the callback/redial