similar to: How to retrieve voicemail

Hi i have a agi script that gets called when a user wants to dialout externally. it gets passed in the exten number and the number dialled and looks up in a db to see if they are allowed to dial the number. the problem is if someone forwards their phone to a external number the CALLERIDNUM is the CLID of the calling party not the extension forwarded thus the call is denied. Can anyone think of a

Hi I have a client using a Avaya IP Office PBX that is taking a SIP trunk from me terminating on a * box. It all works perfectly apart from DTMF. Although you can hear the tones they don't seem to get recognised. I have tried DTMF mode auto, inband, out of band and rfc2833 but no luck. Any ideas? Regards Jon -- Jon Farmer Telford, Shropshire, UK

Been working fine for us so far. -----Original Message----- From: Andrew Joakimsen [mailto:joakimsen@gmail.com] Sent: Wednesday, December 06, 2006 6:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] MWI across multiple servers How well would NFS work in this situation? On 12/6/06, Porier, Jeremy M. < jporier@ccu.edu> wrote: We are about to

Hi all I think this could be en old question. I would like to do a realtime billing prepaid system, mainly using asterisk. I have found few things; I can not get CDR function into agi because asterisk set them once the call is absolutely finish (at least main values for the main porpouse, billsec,duration, etc..) There is a patch that allow you to use CDR

Hi all, i have an * version: Asterisk SVN-branch-1.2-r45691, I need to register a linksys 922 phone thru internet and when I make sip debug command i see this debug information: -- SIP read from x.x.x.x:1024: REGISTER sip:mysipserver.com SIP/2.0 Via: SIP/2.0/UDP x.x.x.x:1025;branch=z9hG4bK-839856dc From: "SPA922" <sip:5403@mysipserver.com>;tag=685bbad1fae3325do0 To:

I have a fairly complicated setup. Extensions (1,2 and 3). In 3 - I execute AGI in java which plays few wav files depending on external parameters. Can I have a dial plan inside my AGI? If not, how do I accomodate user who needs to reach extension 2 from my agi? I have tried stream file and get data but the two commands did not work at all.

Hi! I did manage to load phone with SIP image : SIP70.8-0-3S, made SEP-MAC.cnf.xml, but phone never read the configuration from it. On the screen it's written "Unprovisioned", and phone is not trying to register with asterisk. Please help!! MihaelaMJ -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, I need to connect two asterisk server in same network and i'm using sip user as my clients...... plz anyone suggest me.... Regards, Thiru -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061227/aa4e409c/attachment.htm

Hi All Does anyone know how to use the MySQL cmd in Asterisk with LIKE and % in the query? I have: exten => s,5,Set(query=SELECT name from contacts where tel like %${number}) exten => s,6,MySQL(Connect connid hostname username password dbname) exten => s,7,MySQL(Query resultid ${connid} ${query}) But there seems to be a problem with the % sign and I don't know how to

Hi, I have a basic dial plan question; Here is the scenario. Call comes through IAX and my * authenticate, then collect the digits and dials out, simple :). Here is the dial plan; [did-in] ;for did callers exten => 866219xxxx,1,Ringing exten => 866219xxxx,2,Wait,4 exten => 866219xxxx,3,Answer exten => 866219xxxx,4,Authenticate(/etc/asterisk/authenticate.txt|a) exten =>

Hello, I have setup an Asterisk 1.2.7.1 system, with a working voicemail box: /etc/asterisk/extensions.conf exten => 83086921,1,Answer exten => 83086921,2,Dial(SIP/stefan,5,r) exten => 83086921,3,VoiceMail,u111 exten => 83086921,4,Hangup exten => 83086921,103,VoiceMail,b111 exten => 83086921,104,Hangup /etc/asterisk/voicemail.conf [default] language=de 111 =>

Hi I can't for the life of me work out why this is not working. When in the campon contect if you hit a DTMF key 2 you get moved to the exten => 2 defined in the mainmenu context not the exten => 2 defined in the campon context. What is wrong? The same happens if you hit key 1. [campon] exten => _*1XXX,1,Answer exten => _*1XXX,2,SetCallerID(${CALLERIDNUM}) exten =>

Hello, Upgraded from 1.6.1.0 to 1.6.1.1 and my voicemail setup does not work anymore. I use ODBC storage for voicemail. Comes out that the "voicemessages" table schema should have changed, because the log says Asterisk needed to store data to an additional field called "flag". Any new message cannot be saved. The thing is that I'd like to know where I can find an updated

Hi list, yesterday I converted my voicemail.conf to realtime voicemail and also configured to store the voicemessages in a database using odbc as described here <http://www.voip-info.org/wiki/view/Asterisk+RealTime+Voicemail> and here <http://www.voip-info.org/wiki/view/Asterisk+Voicemail+ODBC+storage>. I am using asterisk 1.4.2 with mysql. I also installed the proper odbc driver for

Hi How do i disable dialling out from voicemail? -- Jon Farmer Telford, Shropshire, UK

Hi All, I have an issue on voice mail with odbc in asterisk 11.7 box. Voice message can be received through Google mail but it doesn't show in phone. The error messages is as follow and let me get your kind advice. -- <SIP/0015-00000007> Playing 'auth-thankyou.g722' (language 'en') [2014-05-28 14:55:13] DEBUG[12260][C-00000006]: app_voicemail.c:3824 last_message_index:

Hello, I wish to use asterisk as a voicemail server with ser . I want to use asterisk external configuration toHello, I wish to use asterisk as a voicemail server with ser . I want to use asterisk external configuration to manage users and storing voicemail messages according to ser database. Where can i find the schema of the SQL DB for voicemail accounts . for example in extconfig ;

Hi - We've got a number of offices, and they're all using ODBC message storage using MySQL. I've been trying to get MySQL replication set up so messages left in a voicemail box at one office will get copied to the corresponding voicemail box at all the offices. We're also using MySQL replication for the voicemail user info, and that part works just fine. I'd like to

Hi, I am having a strange problem with attempting to get voicemail-to-mysql to work. The biggest problem is that I am not able to store voicemail into database. So, I followed the instructor found on the web: Updated the /usr/src/asterisk/apps/Makefile to have USE_MYSQL_VM_INTERFACE=1 and recompiled asterisk, with make clean; make; make install (By the way, is it necessary to update the Makefile

Hello! I've got a little problem with Asterisk (11.14.1), the voicemessages are kinda limited to 40 seconds (average) aproximately; because when a message reach this long I got a cut in the file (*.wav) after I got this message: WARNING[15035][C-000021ef]: format_wav_gsm.c:418 wav_read: Short read (20) (Resource temporarily unavailable)! Does anyone got this problem, any idea of what is