similar to: AGI STREAM FILE command

How will this fare? I am planning on putting an asterisk box for my brother in the Philippines but they only have dialup internet. I want them to be able to use a telephone set on a phonejack or linejack card and call me and vice versa via VOIP. My setup in the US is working already with a broadband cable connection. I am thinking that dialup may not work because of the bandwidth required

Hi! I'm trying to transfer a incomming call from a Queue to another extension. I'm receiving a call from a queue with the AgentCallbackLogin. The queu is as following: Queue(sales|t) Which should allow transfers. So as soon as the call is answered I would like to be able to transfer it When the agent presses the # I get the dialtone but as soon as I press any digit Asterisk tells me

Don't feal bad about not reading. I yell at my 10 y.o. about it all the time. READ, NO more TV, READ!!! > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > Damon Estep > Sent: Tuesday, January 31, 2006 10:23 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion >

> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Troy Settle > Sent: Wednesday, September 14, 2005 7:03 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] MAX PRI for single server (was:Not > enoughlinesavailable for Asterisk implemetation)

Yeah the "variable stays there" because the channel is never up to be cleared. If you do something like exten => _X.,1,Wait(1) exten => _X.,2,Hangup You will see the same behavior. Can you confirm?? I am running CVS from about a week ago... Alex > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com >

> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > Damon Estep > Sent: Tuesday, January 31, 2006 1:48 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: RE: [Asterisk-Users] dialing 2 channels at the > sametimewithdifferentcaller ID number? > >

What is CFU and CFNR? > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Michel Koenen > Sent: Tuesday, August 30, 2005 1:46 AM > To: asterisk-users@lists.digium.com > Subject: RE: [Asterisk-Users] How to use * and # as part of > numberindialcommand > > > From: "Damon

The issue I have had with all other FC3 kernels apart from the 2.6.9 one was that the zaptel build would throw lots of warnings up. This would have the knock on of hanging the system, spinlock I think the problem was, on a modprobe -r. Lee ________________________________ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Damon Estep

> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Damon Estep > Sent: Tuesday, January 31, 2006 8:09 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: RE: [Asterisk-Users] dialing 2 channels > atthesametimewithdifferentcaller ID number? > > > >

In the real world (or at least in my world) we use undersubscribed internet connections that come with a service level agreement (SLA) that guarantees that the jitter, delay, and packet loss with be within defined parameters in the service agreement. With most DSL and Cable you will not get a SLA, with the cheapest T1s you might get one, but the only penalty to the ISP if they do not meet is a

I tried that, you have to ANSWER before you can clear it, which is not a good idea... > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Alexander Lopez > Sent: Tuesday, September 13, 2005 9:27 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: RE: [Asterisk-Users]

I se what you are talking about I an able to reproduce!!! However your PRI may be in a Round-Robin picking order, that would cycle through all of the channels until it reaches an end and then it repeats. I set our PRI to first available hunting instead of RR and it will use the same channel over and over again regardless if the call exists. If anything it's a feature!!! Unassigned DID will

But it does indicated that a variable is staying assigned that should not be, which could have other impact over time??? The behavior is very different for c call where there is a dialplan match for the dialed number, when the call completes the channel extension variable is cleared. If you do not mind please ad a bug note that you experienced the same thing! The bug marshals think I am nuts.

Joost Witteveen (joost@iliana.nl) wrote: > > So, each UDP package with 20 bytes speex-data, we send: > > 20 bytes speex > 12 bytes ogg headers (and others?) > 28 bytes UDP/IP headers (2 IP numerbers, 2 portnumbers, checksum, etc, etc) > > and, if it goes over the phone, each package has a few ppp headers. > > Am I overlooking something, or does this fixed frame

Michel Send me the same output for a dial string that only sends the *31* Is this an ISDN line? What type of card/signalling/switchtype are you using? It looks as if the PSTN switch accepts the *31* and then hangs up so you can make the NEXT call with the *31* feature enabled. If so I assume the *31* feature will be enabled for the next call on the ENTIRE SPAN if it is an ISDN trunk group. If

I have considered opening a bug report on this, but wanted to get some feedback and make sure I am not missing something in the way of a simple work around. What is the scenario in which this impacts your implementation? Ours is the desire to use the same realtime SIP database for many asterisk servers, and route the call based on a "home server" value in the realtime database. The

I found that only the kernel is installed. I'd avoid 2.6.12 for now as I had problem with the zaptel driver and stay with 2.6.9. Regards Lee ________________________________ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Damon Estep Sent: 24 August 2005 22:33 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] fedora

What was the issue with zaptel and 2.6.12? ________________________________ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Lee Archer Sent: Thursday, August 25, 2005 1:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] fedora core 3 kernel source - could someonethrowthe dog a bone! I

> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Damon Estep > Sent: Monday, January 30, 2006 11:20 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: RE: [Asterisk-Users] dialing 2 channels at the same > timewithdifferentcaller ID number? > > >Exten

> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Kevin P. Fleming > Sent: Tuesday, January 31, 2006 3:41 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] RE: [Asterisk-Announce] Asterisk 1.2.4 and > Zaptel1.2.3 > > Damon Estep