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...;m leaving messages on my * Voicemail that I hear sharp clicks, the same thing happens using the record application... They sound like frame slips... Maybe when timing gets off a certain amount * just hangs up the call? -Chris ----- Original Message ----- From: "Kevin " <Asterisk@gtcus.com> To: <asterisk-users@lists.digium.com> Sent: Friday, August 13, 2004 9:32 AM Subject: RE: [Asterisk-Users] Broadvoice User hung up on voicemail > I place a call through Broadvoice to a phone and put it on mute(no > noise) and it didn't get disconnected. > > > -----...

> -----Original Message----- > From: Asterisk@gtcus.com [mailto:Asterisk@gtcus.com] > Sent: Saturday, September 06, 2003 8:39 AM > To: asterisk-users@lists.digium.com > Subject: RE: [Asterisk-Users] VONAGE or IP Dialtone > > > Thanks for the great feedback on these options. I am fairly > new at this and not familiar with th...

Hello, I cannot accept any inboud calls from any provders in my asterisk which tries to authenticate the provider and at the end rejects the call with tthese message 407 Proxy Authentication Required How do I turn off this message. Thanks. Ehsanul Karim

The Vonage service is offered with a SIP Cisco ATA device for connection to an analog phone. Is it possible to connect the Vonage service directly to the Asterisk PBX bypassing the ATA and FXO card? Are there other services that offer this capability or something similar to IP dialtone? Thanks Kevin -------------- next part -------------- An HTML attachment was scrubbed... URL:

Not I. -----Original Message----- From: Kevin [mailto:Asterisk@gtcus.com] Sent: Tuesday, June 01, 2004 7:44 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Sipura-SPA2000 background noise I have been using Cisco ATA's for analog connections and decided to give a Sipura SPA-2000 a try. I noticed there is a fair amount of background white noise...

> -----Original Message----- > From: Kevin [mailto:Asterisk@gtcus.com] > Sent: Saturday, August 21, 2004 5:16 PM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] Queue Monitor > > > I understand that putting monitor-format in the queues.conf > file will start monitor recording of an active queue call. > Is there a wa...

Has anyone implemented an outcall notification when there is a voice message waiting? I would like to have the system notify me of awaiting voice messages by a telephone call rather than an email notification. I would imagine that a call could be dumped into the asterisk spool directory, but I'm not sure how I would monitor for messages waiting. Has anyone implemented such a feature for

There was a post in the 'wiki' for an application to provide an outcall when there is a voicemail is left on asterisk. I am having a problem that this application will only work if the caller presses the pound sign at the end of recording. As most people just hang up, this application isn't working. Can any offer suggestions to accomplish this out call?

I also experienced this problem and the first thing that really helped out was changing the timing in the span line of the zaptel.conf. Change it to look like this (see below) and see if it helps out. I got the error much less after doing this and eventually got rid of the error completely by removing my raid drives, installing an IDE drive, enabling DMA mode on the Hard Drive, enabling APCI in

When I place a call using Iconnecthere as my sip provider, I hear no ringback when making a call. Does anyone else have this problem or offer any suggestions? Thanks, Kevin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031003/b048f72f/attachment.htm

Hi Folks, I want to setup a follow me routine so that asterisk can call me on the multiple numbers. I tried some of the samples at voip-info but there is a problem with those examples. I dont have coverage in my home area and my cell phone answering machine picks up the phone right away so my home phone never rings. I also want the caller to be able to leave a voicemail and the cell phone

...-family:Arial'>Thanks!!<o:p></o:p></span></font></p> > ></div> > ></body> > ></html> > >------=_NextPart_000_0018_01C431DA.ACE09F10-- > > >--__--__-- > >Message: 9 >From: "Kevin " <Asterisk@gtcus.com> >To: <asterisk-users@lists.digium.com> >Date: Tue, 4 May 2004 13:26:05 -0400 >Subject: [Asterisk-Users] Extension Logic Question >Reply-To: asterisk-users@lists.digium.com > >I have an extension context that performs an assisted ParkandAnnounce >page. I create a t...

...@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ------------------------------ Message: 8 Date: Wed, 22 Dec 2004 22:21:10 -0500 From: Kevin <Asterisk@gtcus.com> Subject: [Asterisk-Users] WARNING Maximum retries exceeded on call for seqno 102 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' <asterisk-users@lists.digium.com> Message-ID: <019901c4e89e$72fc4a30$2c02a8c0@gtcp4> Content-Type: text/plain; charset=US-ASCI...

...gium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users ---------------End of Original Message----------------- --__--__-- Message: 8 From: "Kevin " <Asterisk@gtcus.com> To: <asterisk-users@lists.digium.com> Date: Sat, 10 Apr 2004 09:36:29 -0400 Subject: [Asterisk-Users] Extensions and Include Reply-To: asterisk-users@lists.digium.com This perhaps is a newbie question or I have been up too late working on this. Shouldn't I be able to dial intern...

Is it possible to send an external hookflash command to the Digium FXO card from the asterisk PBX? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031007/fe5be94d/attachment.htm

When I place a call using Iconnecthere or Nikotel as my sip provider, I hear no ringback tone when making a call. I have tried adding the 'r' command to the dial sting with no help. Does anyone else have this problem or offer any suggestions? Thanks, Kevin -------------- next part -------------- An HTML attachment was scrubbed... URL:

I previously reported a problem with receiving inbound calls with Galaxy Voice and the Current CVS Head version. I am currently running version 5/17/04 of the CVS HEAD and my incoming calls work correctly with Galaxy Voice. If I upgrade to the Stable release or the current CVS head I am no longer able to receive incoming calls. I would really like to use the current version of the CVS as it has

I am trying to hook up my Cisco telephones to Asterisk using a Layer 3 switch and am having difficulties it getting it to work. I realize this may not be the proper forum for a discussion on VLAN architecture and configuration so I won't post the question here. I though I had read all the requisite information regarding the configuration for this, but perhaps I am missing something simple.