[asterisk-users] Autoreply: Re: Queue stats

On Thu, 2007-07-26 at 09:37 -0500, Jay Moore wrote: > My boss would like some statistics on how many calls are answered out of > specific queues during a given time period, and I'm not sure how exactly > to obtain those stats. It sounds like you've got quite the queue setup (although I don't

[asterisk-users] Autoreply: Re: Newbie Advice on Asterisk and Linux

Mark Burrows wrote: > HI All, > > > > I’m new to Asterisk and also to Linux. I have a large IVR project that > I’m about to embark on. I’m new to programming; new to Linux and new to > Asterisk. I think I’m about to climb a steep learning curve. I have an > existing IVR which is getting

[asterisk-users] Autoreply: Queue stats

Greetings, list! My boss would like some statistics on how many calls are answered out of specific queues during a given time period, and I'm not sure how exactly to obtain those stats. Here's how our queue system works. 1) Call comes in and enters our 'ring' queue where the phones ring for

[asterisk-users] Autoreply: Autoreply: Re: Display IE

Hi! Thank you all for the info! But I think I haven't explained my scenario well enough. I am not relaying the calls to SIP. What happens is the following (the scenario is: a call started from an ISDN E1 PBX, through the asterisk, to the PSTN, and was answered by the remote party): 1. A CONNEC

[asterisk-users] Autoreply: Autoreply: Is it possible transcode (ilbc <-> g.729) in software ?

Hi guys, I don't know if this is the question ... But I have my softphones set up to use ilbc (because I found that it is better for me) and I'm trying to connect them to my provider that provides me termination through g.729. I don't have any T1 or whatever card in my server. But everytime I tr

[asterisk-users] Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Re: Dialtone when automatically picking up.

On 2007 Jul 26 (Thu) at 09:32:00 -0400 (-0400), Jared Smith wrote: :On Wed, 2007-07-25 at 15:06 -0700, Peter Hessler wrote: :> ;; dialtone in the background isn't there any more :> ;; dialed '305' :> ;; everything from here is exactly as expected. : :OK, I missed this in the first email you s

[asterisk-users] Autoreply: Re: Attaching VoiceMails on E-Mails

Hello Marco, On 7/27/07, Marco Mouta <[EMAIL PROTECTED]> wrote: > > hi, > > The > VoiceMailapplication > uses > */usr/sbin/sendmail* to mail voicemail messages to users. This can be any > sendmail-compatible MTA. In practice y

[asterisk-users] Autoreply: Autoreply: Autoreply: Re: Display IE

Hi! Thank you all for the info! But I think I haven't explained my scenario well enough. I am not relaying the calls to SIP. What happens is the following (the scenario is: a call started from an ISDN E1 PBX, through the asterisk, to the PSTN, and was answered by the remote party): 1. A CONNEC

[asterisk-users] Autoreply: Autoreply: Autoreply: Is it possible transcode (ilbc <-> g.729) in software ?

Hi guys, I don't know if this is the question ... But I have my softphones set up to use ilbc (because I found that it is better for me) and I'm trying to connect them to my provider that provides me termination through g.729. I don't have any T1 or whatever card in my server. But everytime I tr

[asterisk-users] Autoreply: Autoreply: Re: Attaching VoiceMails on E-Mails

Hello Marco, On 7/27/07, Marco Mouta <[EMAIL PROTECTED]> wrote: > > hi, > > The > VoiceMailapplication > uses > */usr/sbin/sendmail* to mail voicemail messages to users. This can be any > sendmail-compatible MTA. In practice y

[asterisk-users] Autoreply: Is it possible transcode (ilbc <-> g.729) in software ?

Hi guys, I don't know if this is the question ... But I have my softphones set up to use ilbc (because I found that it is better for me) and I'm trying to connect them to my provider that provides me termination through g.729. I don't have any T1 or whatever card in my server. But everytime I tr

[asterisk-users] Autoreply: Autoreply: Autoreply: Autoreply: Re: Dialtone when automatically picking up.

On 2007 Jul 26 (Thu) at 09:32:00 -0400 (-0400), Jared Smith wrote: :On Wed, 2007-07-25 at 15:06 -0700, Peter Hessler wrote: :> ;; dialtone in the background isn't there any more :> ;; dialed '305' :> ;; everything from here is exactly as expected. : :OK, I missed this in the first email you se

[asterisk-users] Autoreply: Re: Dialtone when automatically picking up.

On 2007 Jul 26 (Thu) at 09:32:00 -0400 (-0400), Jared Smith wrote: :On Wed, 2007-07-25 at 15:06 -0700, Peter Hessler wrote: :> ;; dialtone in the background isn't there any more :> ;; dialed '305' :> ;; everything from here is exactly as expected. : :OK, I missed this in the first email you sent.

[asterisk-users] Autoreply: Autoreply: Autoreply: Re: Dialtone when automatically picking up.

On 2007 Jul 26 (Thu) at 09:32:00 -0400 (-0400), Jared Smith wrote: :On Wed, 2007-07-25 at 15:06 -0700, Peter Hessler wrote: :> ;; dialtone in the background isn't there any more :> ;; dialed '305' :> ;; everything from here is exactly as expected. : :OK, I missed this in the first email you sen

[asterisk-users] Autoreply: Autoreply: Re: Dialtone when automatically picking up.

On 2007 Jul 26 (Thu) at 09:32:00 -0400 (-0400), Jared Smith wrote: :On Wed, 2007-07-25 at 15:06 -0700, Peter Hessler wrote: :> ;; dialtone in the background isn't there any more :> ;; dialed '305' :> ;; everything from here is exactly as expected. : :OK, I missed this in the first email you sent

[asterisk-users] Autoreply: Re: Display IE

Hi! Thank you all for the info! But I think I haven't explained my scenario well enough. I am not relaying the calls to SIP. What happens is the following (the scenario is: a call started from an ISDN E1 PBX, through the asterisk, to the PSTN, and was answered by the remote party): 1. A CONNEC

[asterisk-users] Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Re: Dialtone when automatically picking up.

On 2007 Jul 26 (Thu) at 09:32:00 -0400 (-0400), Jared Smith wrote: :On Wed, 2007-07-25 at 15:06 -0700, Peter Hessler wrote: :> ;; dialtone in the background isn't there any more :> ;; dialed '305' :> ;; everything from here is exactly as expected. : :OK, I missed this in the first email you

[asterisk-users] Autoreply: Autoreply: Autoreply: Autoreply: Re: Attaching VoiceMails on E-Mails

Hello Marco, On 7/27/07, Marco Mouta <[EMAIL PROTECTED]> wrote: > > hi, > > The > VoiceMailapplication > uses > */usr/sbin/sendmail* to mail voicemail messages to users. This can be any > sendmail-compatible MTA. In practice y

[asterisk-users] Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Re: Attaching VoiceMails on E-Mails

Hello Marco, On 7/27/07, Marco Mouta <[EMAIL PROTECTED]> wrote: > > hi, > > The > VoiceMailapplication > uses > */usr/sbin/sendmail* to mail voicemail messages to users. This can be any > sendmail-compatible MTA. In practice y

[asterisk-users] Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Re: Display IE

Hi! Thank you all for the info! But I think I haven't explained my scenario well enough. I am not relaying the calls to SIP. What happens is the following (the scenario is: a call started from an ISDN E1 PBX, through the asterisk, to the PSTN, and was answered by the remote party): 1. A CONNEC

[asterisk-users] Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Re: Dialtone when automatically picking up.

On 2007 Jul 26 (Thu) at 09:32:00 -0400 (-0400), Jared Smith wrote: :On Wed, 2007-07-25 at 15:06 -0700, Peter Hessler wrote: :> ;; dialtone in the background isn't there any more :> ;; dialed '305' :> ;; everything from here is exactly as expected. : :OK, I missed this in the first email y

[asterisk-users] Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Re: Attaching VoiceMails on E-Mails

Hello Marco, On 7/27/07, Marco Mouta <[EMAIL PROTECTED]> wrote: > > hi, > > The > VoiceMailapplication > uses > */usr/sbin/sendmail* to mail voicemail messages to users. This can be any > sendmail-compatible MTA. In practice y

[asterisk-users] Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Re: Dialtone when automatically picking up.

On 2007 Jul 26 (Thu) at 09:32:00 -0400 (-0400), Jared Smith wrote: :On Wed, 2007-07-25 at 15:06 -0700, Peter Hessler wrote: :> ;; dialtone in the background isn't there any more :> ;; dialed '305' :> ;; everything from here is exactly as expected. : :OK, I missed this in the first email

[asterisk-users] Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Re: Dialtone when automatically picking up.

On 2007 Jul 26 (Thu) at 09:32:00 -0400 (-0400), Jared Smith wrote: :On Wed, 2007-07-25 at 15:06 -0700, Peter Hessler wrote: :> ;; dialtone in the background isn't there any more :> ;; dialed '305' :> ;; everything from here is exactly as expected. : :OK, I missed this in the first email

[asterisk-users] Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Re: Attaching VoiceMails on E-Mails

Hello Marco, On 7/27/07, Marco Mouta <[EMAIL PROTECTED]> wrote: > > hi, > > The > VoiceMailapplication > uses > */usr/sbin/sendmail* to mail voicemail messages to users. This can be any > sendmail-compatible MTA. In practice y

[asterisk-users] Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Is it possible transcode (ilbc <-> g.729) in software ?

Hi guys, I don't know if this is the question ... But I have my softphones set up to use ilbc (because I found that it is better for me) and I'm trying to connect them to my provider that provides me termination through g.729. I don't have any T1 or whatever card in my server. But everytime I tr

[asterisk-users] Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Is it possible transcode (ilbc <-> g.729) in software ?

Hi guys, I don't know if this is the question ... But I have my softphones set up to use ilbc (because I found that it is better for me) and I'm trying to connect them to my provider that provides me termination through g.729. I don't have any T1 or whatever card in my server. But everytime I tr

[asterisk-users] Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Re: Dialtone when automatically picking up.

On 2007 Jul 26 (Thu) at 09:32:00 -0400 (-0400), Jared Smith wrote: :On Wed, 2007-07-25 at 15:06 -0700, Peter Hessler wrote: :> ;; dialtone in the background isn't there any more :> ;; dialed '305' :> ;; everything from here is exactly as expected. : :OK, I missed this in the first email yo

[asterisk-users] Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Re: Display IE

Hi! Thank you all for the info! But I think I haven't explained my scenario well enough. I am not relaying the calls to SIP. What happens is the following (the scenario is: a call started from an ISDN E1 PBX, through the asterisk, to the PSTN, and was answered by the remote party): 1. A CONNEC

[asterisk-users] Autoreply: Autoreply: Autoreply: Re: Attaching VoiceMails on E-Mails

Hello Marco, On 7/27/07, Marco Mouta <[EMAIL PROTECTED]> wrote: > > hi, > > The > VoiceMailapplication > uses > */usr/sbin/sendmail* to mail voicemail messages to users. This can be any > sendmail-compatible MTA. In practice y

[asterisk-users] Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Re: Dialtone when automatically picking up.

On 2007 Jul 26 (Thu) at 09:32:00 -0400 (-0400), Jared Smith wrote: :On Wed, 2007-07-25 at 15:06 -0700, Peter Hessler wrote: :> ;; dialtone in the background isn't there any more :> ;; dialed '305' :> ;; everything from here is exactly as expected. : :OK, I missed this in the first email you

[asterisk-users] Autoreply: Autoreply: Autoreply: Autoreply: Is it possible transcode (ilbc <-> g.729) in software ?

Hi guys, I don't know if this is the question ... But I have my softphones set up to use ilbc (because I found that it is better for me) and I'm trying to connect them to my provider that provides me termination through g.729. I don't have any T1 or whatever card in my server. But everytime I tr

[asterisk-users] Autoreply: Autoreply: Autoreply: Autoreply: Re: Display IE

Hi! Thank you all for the info! But I think I haven't explained my scenario well enough. I am not relaying the calls to SIP. What happens is the following (the scenario is: a call started from an ISDN E1 PBX, through the asterisk, to the PSTN, and was answered by the remote party): 1. A CONNEC

[asterisk-users] Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Is it possible transcode (ilbc <-> g.729) in software ?

Hi guys, I don't know if this is the question ... But I have my softphones set up to use ilbc (because I found that it is better for me) and I'm trying to connect them to my provider that provides me termination through g.729. I don't have any T1 or whatever card in my server. But everytime I tr

[asterisk-users] Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Re: Display IE

Hi! Thank you all for the info! But I think I haven't explained my scenario well enough. I am not relaying the calls to SIP. What happens is the following (the scenario is: a call started from an ISDN E1 PBX, through the asterisk, to the PSTN, and was answered by the remote party): 1. A CONNEC

[asterisk-users] Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Re: Attaching VoiceMails on E-Mails

Hello Marco, On 7/27/07, Marco Mouta <[EMAIL PROTECTED]> wrote: > > hi, > > The > VoiceMailapplication > uses > */usr/sbin/sendmail* to mail voicemail messages to users. This can be any > sendmail-compatible MTA. In practice y

[asterisk-users] Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Is it possible transcode (ilbc <-> g.729) in software ?

Hi guys, I don't know if this is the question ... But I have my softphones set up to use ilbc (because I found that it is better for me) and I'm trying to connect them to my provider that provides me termination through g.729. I don't have any T1 or whatever card in my server. But everytime I tr

[asterisk-users] Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Re: Display IE

Hi! Thank you all for the info! But I think I haven't explained my scenario well enough. I am not relaying the calls to SIP. What happens is the following (the scenario is: a call started from an ISDN E1 PBX, through the asterisk, to the PSTN, and was answered by the remote party): 1. A CONNEC

[asterisk-users] Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Is it possible transcode (ilbc <-> g.729) in software ?

Hi guys, I don't know if this is the question ... But I have my softphones set up to use ilbc (because I found that it is better for me) and I'm trying to connect them to my provider that provides me termination through g.729. I don't have any T1 or whatever card in my server. But everytime I tr

[asterisk-users] Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Re: Dialtone when automatically picking up.

On 2007 Jul 26 (Thu) at 09:32:00 -0400 (-0400), Jared Smith wrote: :On Wed, 2007-07-25 at 15:06 -0700, Peter Hessler wrote: :> ;; dialtone in the background isn't there any more :> ;; dialed '305' :> ;; everything from here is exactly as expected. : :OK, I missed this in the first emai

[asterisk-users] Autoreply: Re: SunRocket / ALLO / etc special offer

I'll take either Actually now that I have had a chance to think about what I did (sorry bad week here). Yes, I will admit I did patrionize the users list... sorry if I offended anyone. I just figured I'd try to help any SunRocket users out that may not be on the biz list.If you review

[asterisk-users] How to catch isdn progress message

Hi All, I would like to ask for some advice how to solve following situation: We have to record and decode isdn PROGRESS message and when particular message is found call should be hang up and dialplan should continue. So far we have come up with two ways we think solve the problem and would ver

Re: [asterisk-users] How to catch isdn progress message

> Message: 12 > Date: Thu, 10 Sep 2009 13:11:27 +0200 > From: "Jay R. Worthington" > Subject: Re: [asterisk-users] How to catch isdn progress message > To: asterisk-users@lists.digium.com > Message-ID: > <710bae00909100411u450d406ao1655c200e4d84...@mail.gmail.com> > Content-Type: text/plain;