similar to: Cell Phone dialing

Hello, I sent a wav-audio to Asterisk though SIP and ISDN channels and recorded it in wav-audio at the Asterisk server. I found the loudness level of the recorded audio was too high comparing with the orginal audio. How can I ajust it, so that there will be no amplifier used for recording. Thanks a lot. best regards Felix -------------- next part -------------- An HTML attachment was

Hi, please forgive me for this (hopefully) simple question. I cannot seem to find an answer or solution while searching around. I want to be able to call in to my server using my cell phone and be able to set call forwarding for my extension and enter a phone number and also be able to call in to that extension and disable the call forwarding. I see I can do this through the ARI web interface

Hi gang, We are moving our 1.4 asterisk with DAHDI over to 10.0 with SIP. Everything is going nicely except that I can't get NV_FAXDETECT to compile properly into 10.0. Because of this, I will have to have my receptionist manually transfer incoming faxes. Any suggestions? Thanks in Advance Danny Nicholas -------------- next part -------------- An HTML attachment

Hi, I was trying to register a VoIP trunk in Asterisk , where its keep on sending Register message to the server, where I am not getting any response from server. But whereas if i register in Xlite softphone the account is getting registered. I suspect it could be network related issue, but since in softphone it is getting registered from the same network. Any ideas to isolate things would be

Hello guys, i would like to implement authentication for my sip extension with an openldap server. Following this guide http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/ExternalServices_id291590.html i see a template named [sip] to map the information of sip peers into ldap. But i'm not interested to create a template, i would only authenticate sip extensions using username

> Date: Wed, 28 Jan 2009 13:11:19 -0600 > From: "Danny Nicholas" <danny at debsinc.com> > Subject: Re: [asterisk-users] SIP Registrations broken on 1.4.22.1? > To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" > <asterisk-users at lists.digium.com> > Message-ID: <D32AD473FC574B41AE6A842E46549174 at db0005> >

Hi, I am using asterisk via AGI and want to be able to record a call. The scenario is: 1. A call comes in 2. The call is redirected to a mobile number via a local extension and ChannelRedirect 3. The local extension looks like something this: exten => _X.,1,Dial(SIP/${EXTEN},60,?) exten => _X.,n,Agi(agi://localhost/aj.agi?action=??..) I have looked through all arguments of Dial

Dear, I need to create a custom device extension in order to dial a local queue. Suppose my queue number is 8888, how can fill the Dial field from the custom extension ??? Because if I put just 8888 or Local/8888, I don't succeed. Thanks a lot

Hi, I have used basic Asterisk as a PBX controlling few extensions. My question is simple, at work there is an existing Call Manager/PBX and all which manages all the extensions for SCCP VOIP phones. Can Asterisk be used to manage just 1 VOIP phone and still can make internal calls to the other extensions? Thanks, Jorge -------------- next part -------------- An

Hi, Is there a way to execute next priority in the dialplan if you have called agi:async? I want to play warning message if adhearsion is down. Currently I wasn't able to make it work. The dialplan execution ends after the first priority. [incomming] exten => _X.,1,AGI(agi:async) exten => _X.,2,Answer exten => _X.,3,Playback(some-message) exten => _X.,4,Hangup Regards, Pavel

Hi all, I'm using ConfBridge within Asterisk 1.6.20 and want to record the conference, so I'd like to start the recording when the second user joins, so in the example below, for example, how can I get the current user count in ConfBridge 3000? [conferences] ;authenticated conference (ext C-O-N-F = 2663) exten => 2663,1,Answer same => n,Wait(1) same => n,Authenticate(143382)

Hi all, I am using asterisk-1.6.2.10. I changed say.conf script for customized number reading. In the extension.conf: -------------------------- [number-to-voice] exten => 8765,1,playback(num:344345,say) exten => 8765,n,hangup It executes corresponding say.conf script and produces good results for me. but when I write it in agi does not working. Here is agi debug output from asterisk.

Hi, I've tested 1.8.6.0, 1.8.4.0 and 1.8.0 I can get proper start and answer time but not the end time of call <SIP/11-00000000>AGI Rx << GET VARIABLE CDR(start) <SIP/11-00000000>AGI Tx >> 200 result=1 (2011-12-16 18:34:48) <SIP/11-00000000>AGI Rx << GET VARIABLE CDR(end) <SIP/11-00000000>AGI Tx >> 200 result=1 (2011 12-16 18:34:48)

Hi everybody, I' ve been following this list for a while now. Is there a way to detect the individual and cumulative s/n ratio values for the incoming calls in Asterisk or any other Call Center solution?... -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20111130/d0d53c1f/attachment.htm>

Hi I am currently using 1.2.x and 1.4.x behind OpenSER. One of the things I do on INVITES is to re-authenticate the user from OpenSER. Then when the INVITE gets passed to Asterisk I capture the AUTH to a variable in the dialplan and pass to an AGI script. I am now trying to set the same thing up in 1.6 However because the argument delimter in 1.6 has changed from pipe to comma this breaks as the

I have this [TRONCAL-SIP] exten=>225/91,1,Answer exten=>225/91,2,Echo exten=>225/91,3,Hangup exten=>225/92,1,Answer exten=>225/92,2,Playback(conf-invalid) exten=>225/92,3,Hangup When I make a call CLI> -- Recv IAM CIC=8 ANI=91 DNI=225 RNI= redirect=no/0 complete=1 Dont work If I add this rule exten=>225,1,Answer Works ok -------------- next part --------------

Somebody knows if I can save files in mp3 with the Record command on Asterisk? I try to recompile sox to suport mp3 but Asterisk return the folowing message when I use the Record command: - Executing [*40 at liberado15:15] Record("SIP/1201-083453c8", "/var/spool/asterisk/alarme/alarme-1201-200905121212:mp3") in new stack ??? -- <SIP/1201-083453c8> Playing 'beep'

Hi all, I'm trying to run an AGI in PERL which uses the module DBD-Oracle. Currently my AGI is working fine in my two servers but not in my other four servers. When I tried execute an AGI (as a user asterisk) in command line it works fine (even I also declare environmental variables in user profile and in my AGI), but when I tried to call my AGI (perl) in dial plan, it don't get

Hello I need to write a script that will dial a list of customers and play a message. I couldn't find a way to tell Asterisk/Zaptel to wait until the callee has actually picked up the phone before proceeding with Playback(): ============ ;call made through Dial(): Doesn't proceed after off-hook/hangup [internal] exten => 8888,1,Dial(Zap/1/${IPPI}) exten => 8888,n,NoOp(We never

Dear all, Anyone of you knows how to play an audio file at the beginning of a call for both Caller and Callee? A(x) of Dial application only plays audio for callee. I don't want to use MeetMe because I want to use Monitor and MixMonitor. Thank you! ________________________________ Este mensaje se dirige exclusivamente a su destinatario. Puede consultar nuestra pol?tica de env?o y recepci?n