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...mono comes...from where it is originating.?? As there is nothing written in .conf file. Console output is below: I am using mpg123 version 0.59r. Although I am able to play music with mpg123 but why it is on No-cooperation movement against asterisk ? Need help..any ideas any direction... Thanks Gurminder Console output ***************************Snip*********************************** -- Executing MusicOnHold("Zap/1-1", "default") in new stack -- Started music on hold, class 'default', on channel 'Zap/1-1' -- Stopped music on hold on Zap/1-1 Unknown opt...

...fication: some email servers rejects the email from asterisk. My engineer added Asterisk IP into the DNS to slove the email rejection issue. Thanks for those who replied this email. Best regards, Larry ------------------------------ Message: 21 Date: Thu, 11 Aug 2005 20:46:39 +0530 From: Gurminder Arora <gurmi.linux@gmail.com> Subject: Re: [Asterisk-Users] re: how to set the voice message as email attachment ? To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID: <e5d0d9d705081108163334a4d7@mail.gmail.com> Content-Type:...

...Asterisk on Red hat linux 9 for first time. After compiling and installing: 1. Zaptel 2. lippri 3. mpg123 4. asterisk and configuring files I started asterisk but it exited by printing error message Can't write audio data: Broken pipe Please tell me what is possible reason for this. Regards Gurminder -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050715/7625ad46/attachment.htm

Hi All, I am trying to connect to cvs.digium.com but connection gets timed out. Even pinging to cvs.digium.com is not working. I m using cvs login password - anoncvs Regards Gurminder

...*************** I am seaching Why loop exits before reaching limit of 8867 or what makes zt_handle_event to control the flow. Please help me with any idea you have. Also tell if I am on wrong path for right problem PS: I have tried best to explain it but if ny doubt prevails pls tell me. Regards Gurminder

...incoming context is ************************************************************************ [incoming] exten => s,1,Answer exten => s,2,Dial(Zap/1,5,tr) exten => s,3,Voicemail(100@default) exten => s,4,Hangup ************************************************************************* Gurminder

Hello I have couple issues with AAH. 1.5 1. Flash panel doesn't show proper status. Sometime accessing with IP seems to work and it shows current line status etc. Sometimes accessing with hostname of the asterisk server seems to show lines, but it doesn't show off hook etc when we pickup a extension and talk. In /var/www/html/panel/op_server.cfg I have tried setting manager_host to all

Hello I have created an iax exten in my iax.conf file: [300] type=friend username=300 secret=*** context=default host=dynamic callerid="some name" <300> auth=md5 Then in my extensions.conf I have: exten => 300,1,Dial(IAX/${EXTEN},20) exten => 300,2,Hangup I can dial from iaxComm (a soft IAX client) and that works fine. But when I try to dial 300 get: WARNING[22077]:

Hi there, I am using redhat 9.0 with asterisk 1.0.7. I created an user and was be able to leave voice messages to that user and retrieve the voice message. I looked the wiki and setup the voice message as the email attachment. However, I have never received email with the voice attachment. Here is the setting for voicemail.conf: ; ; Voicemail Configuration ; [general] ; Default formats

Hello, is it possible to change how are caller id on incoming call from isdn, capi lines displayed od sip phones ? ( e.g. SNOM ) standard is 1234567@domain.net. I just want only 1234567 to be displayed. is it possible ? regards Marian -- SUNTEQ s. r. o. Hviezdoslavova 9 # Prievidza # 971 04 # Slovak republic Tel: +421-46-5430 754 # Fax: +421-46-5439 144 http://www.sunteq.sk/