[asterisk-users] Conference topology

Hi, Does anyone have any guides, documents on best practice for "bridging" multiple Asterisk boxes together so no matter what box a person lands on, they can be on the same call? I assume the easiest would be to have one box dial out to all other boxes and bridge them. For example If we have room

Re: [asterisk-users] Conference bridge recording file name

That did it! I had missed that option. Thanks for the assistance! On Thu, Aug 26, 2021 at 9:50 AM Doug Lytle wrote: > According to the wiki, you can disable the timestamp > > record_file_timestamp > > Append the start time to the record_file name so that it is unique. > > >

Re: [asterisk-users] Conference bridge recording file name

According to the wiki, you can disable the timestamp record_file_timestamp Append the start time to the record_file name so that it is unique. https://wiki.asterisk.org/wiki/display/AST/Asterisk+16+Configuration_app_confbridge Doug --

[asterisk-users] Conference bridge recording file name

Hello, I'm attempting to enable conference bridge recording. I have it working, and I'm dynamically pushing the filename onto the bridge via the set CONFBRIDGE commands. But it seems regardless of what name I set, the actual filename is written as WHATIPROVIDED-uniqueid.wav. Example, I use the

[asterisk-users] Conference bridge profile does not exist

Hello, I have ab profile in /etc/asterisk/confbridge.conf but in my dialplan this profile is not found I tried a lot, but did no solution. What can be wrong? [out_bridge] type=bridge exten => ,n,ConfBridge(${conf_room},out_bridge) [Jun 5 19:27:09] WARNING[11008][C-0468]:

Re: [asterisk-users] Conference Call like conference done by mobile!

On Wed, Oct 5, 2016 at 11:46 PM, Mandar Khire wrote: > Hi, > Thanks for reply. > For use confbridge I follows link http://www.mytechrepublic.com/?p=418 > By it I manage to create Conference room & add members to it. > But each member has to dial conference Number. > In my

Re: [asterisk-users] Conference Call like conference done by mobile!

Hi, Thanks for reply. For use confbridge I follows link http://www.mytechrepublic.com/?p=418 By it I manage to create Conference room & add members to it. But each member has to dial conference Number. In my scenario Only first person dial second person's number. Example:- If Person1 has 6001,

Re: [asterisk-users] Conference Call like conference done by mobile!

On Wed, 2016-10-05 at 17:34 +0530, Mandar Khire wrote: > hi, > I trying to solve one scenario:- > As I can make call from mobile phone to my friend1. As he accept it, > I put him on hold, & dial friend2. > As he also accept it, I put him on hold & follow same procedure till > friend6. > The I

Re: [asterisk-users] Conference Call like conference done by mobile!

On 10/5/16 7:04 AM, Mandar Khire wrote: hi, I trying to solve one scenario:- As I can make call from mobile phone to my friend1. As he accept it, I put him on hold, & dial friend2. As he also accept it, I put him on hold & follow same procedure till friend6. The I click on 'Merge call' & I

[asterisk-users] Conference Call like conference done by mobile!

hi, I trying to solve one scenario:- As I can make call from mobile phone to my friend1. As he accept it, I put him on hold, & dial friend2. As he also accept it, I put him on hold & follow same procedure till friend6. The I click on 'Merge call' & I can talk to all 6 friends at a time & they can

Re: [asterisk-users] conference call stuttering / clocking issue (?) - ESXi virtual environment

a single conference. > Regards; > John V. > > -Original Message- > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin Long > Sent: Wednesday, March 09, 2016 2:23 PM > To: Asterisk Users Mailing List

Re: [asterisk-users] conference call stuttering / clocking issue (?) - ESXi virtual environment

risk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin Long Sent: Wednesday, March 09, 2016 2:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] conference call stuttering / clocking issue (?) - ESXi virtual e

[asterisk-users] conference call stuttering / clocking issue (?) - ESXi virtual environment

Title says it all - for the time being I am stuck deploying Asterisk in ESXi . We are also looking at Proxmox for our next round of servers.. Everything works fine except conference calls - very stuttery , have tried a few different codecs. I assume this is a granular clocking issue , and

[asterisk-users] Conference calls wont traverse my trunk

My conference call wont go thru my SIP trunk. I may be missing a dialplan configuration setting as my PCM phone to phone calls go over the (GSM) tunk. The server with the conference: exten = 5777,1,GoTo(conf-confDemo,join,1) [conf-confDemo] exten = join,1,ConfBridge(confDemo/S/1) The server

Re: [asterisk-users] Conference calls wont traverse my trunk

On Thu, Jun 27, 2013 at 9:53 AM, DadoMaker dadoma...@gmail.com wrote: My conference call wont go thru my SIP trunk. I may be missing a dialplan configuration setting as my PCM phone to phone calls go over the (GSM) tunk. The server with the conference: exten =

Re: [asterisk-users] Conference calls wont traverse my trunk

The cogerence works but doesnt go over my trunk. Its bypassing and the codec is PCM of phone. But in phone to phone call, the rtp traverses the trunk and I capture gsm packets to verify. The sip debug for conf call setup and leave: *CLI == Using SIP RTP CoS mark 5 -- Executing

Re: [asterisk-users] Conference calls wont traverse my trunk

Found a syntax err in my dialplan on the far side Asterisk config. Thanks, Dado On Thu, Jun 27, 2013 at 10:41 AM, DadoMaker dadoma...@gmail.com wrote: The cogerence works but doesnt go over my trunk. Its bypassing and the codec is PCM of phone. But in phone to phone call, the rtp traverses

Re: [asterisk-users] Conference solution to handle 10, 000 participants - possible at all?

Hi, I'd been thinking about such a huge conferencing system for about last few months. Like Steve suggested, my concept is almost similar but instead of making a central hub conference junction between multiple Conferences I was thinking of making a peer2peer runtime connection between

[asterisk-users] Conference solution to handle 10, 000 participants - possible at all?

Hello list, A client is asking to setup an asterisk based conferencing solution which could handle 10,000 participants (in one single conference or combined in multiple conferences), and later could be scaled to handle up to 50,000 participants. All callers will be over SIP, using g711. I

Re: [asterisk-users] Conference solution to handle 10, 000 participants - possible at all?

On Mon, 17 Oct 2011, VisionVoIP wrote: A client is asking to setup an asterisk based conferencing solution which could handle 10,000 participants (in one single conference or combined in multiple conferences), and later could be scaled to handle up to 50,000 participants. All callers will be

[asterisk-users] Conference calls through web-interface with moderation using Asterisk?

Good Morning, I have been researching this for a while, basically I'd like to have a website with the following functionality: • One-click call-in to show (after setting username, best-case scenario: sign-in through Drupal, use that name for conference-call) • Web-interface only (Flash/Flex,

Re: [asterisk-users] Conference feature

2011 22:25:00 -0300 From: rafaels...@gmail.com To: asterisk-users@lists.digium.com Subject: [asterisk-users] Conference feature Hi How to create the conference feature in Asterisk? Thank's Att, Rafael Saraiva

[asterisk-users] Conference feature

Hi How to create the conference feature in Asterisk? Thank's Att, Rafael Saraiva -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs:

Re: [asterisk-users] Conference feature

On Sun, 26 Jun 2011, Rafael dos Santos Saraiva wrote: How to create the conference feature in Asterisk? RTM, keeping your eyes open for references to 'meetme.' -- Thanks in advance, - Steve Edwards

Re: [asterisk-users] Conference feature

: [asterisk-users] Conference feature Hi How to create the conference feature in Asterisk? Thank'sAtt,Rafael Saraiva -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live

Re: [asterisk-users] Conference feature

Skype: flaviormiranda -- Date: Sun, 26 Jun 2011 22:25:00 -0300 From: rafaels...@gmail.com To: asterisk-users@lists.digium.com Subject: [asterisk-users] Conference feature Hi How to create the conference feature in Asterisk? Thank's Att, Rafael Saraiva

Re: [asterisk-users] Conference feature

On Sun, 26 Jun 2011, Rafael dos Santos Saraiva wrote: I am referring to 3-way conference With a little reading, you would discover that meetme can handle lots of participants. -- Thanks in advance, - Steve Edwards

Re: [asterisk-users] Conference feature

Steve Edwards wrote: On Sun, 26 Jun 2011, Rafael dos Santos Saraiva wrote: I am referring to 3-way conference With a little reading, you would discover that meetme can handle lots of participants. For those who know Telephony, 3 way conference and meet me conference are NOT the same.

Re: [asterisk-users] Conference feature

Does asterisk support it? On Sun, Jun 26, 2011 at 9:25 PM, Rafael dos Santos Saraiva rafaels...@gmail.com wrote: Hi How to create the conference feature in Asterisk? Thank's Att, Rafael Saraiva -- _ -- Bandwidth and

Re: [asterisk-users] Conference feature

I am given to understand that it does not. On 06/27/2011 12:13 AM, C F wrote: Does asterisk support it? On Sun, Jun 26, 2011 at 9:25 PM, Rafael dos Santos Saraiva rafaels...@gmail.com wrote: Hi How to create the conference feature in Asterisk? Thank's Att, Rafael Saraiva --

Re: [asterisk-users] Conference feature

-users] Conference feature I am given to understand that it does not. On 06/27/2011 12:13 AM, C F wrote: Does asterisk support it? On Sun, Jun 26, 2011 at 9:25 PM, Rafael dos Santos Saraiva rafaels...@gmail.com wrote: Hi How to create the conference feature in Asterisk? Thank's Att

[asterisk-users] CONFERENCE CONFIGURATION REQUIRE

Hi all, I am using asterisk1.2(vicidial). I am using like pbx . In this how can I confugure the internal conference calls. suppose I have A,B,C,D,E users these all peoples should be internal conferece . for them i was give 101,102,103,104,105 extensions. For this scenario what can I do exact

[asterisk-users] conference room ideas

Some neat conference room ideas that would be great to see incorporated into asterisk conference. https://imeet.com/support Cheers, Dean -- _ -- Bandwidth and Colocation Provided by

[asterisk-users] Conference Meetme

How many simultaneous conference meetme setups can be supported in the same time on Asterisk, and what are the corresponding server's specs for this. Thanks-- _ -- Bandwidth and Colocation Provided by

Re: [asterisk-users] Conference Meetme

On Wed, 14 Apr 2010, torinti...@hotmail.com wrote: How many simultaneous conference meetme setups can be supported in the same time on Asterisk, and what are the corresponding server's specs for this. How long is a piece of string? 0) A better subject yields better answers 1) A more

Re: [asterisk-users] Conference Meetme

Last year I did a lab test for a customer who wanted conferencing solution for his organization, on a 2 x dual core xeon with 4GB type server, which had 120 zap channels and I put all the channels in mutiple conferences, from 4 to 20 users per conference and let it running for two weeks. Munin

Re: [asterisk-users] Conference Meetme

Zakaria Sent: Wednesday, April 14, 2010 8:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Conference Meetme Last year I did a lab test for a customer who wanted conferencing solution for his organization, on a 2 x dual core xeon with 4GB type server

[asterisk-users] [Conference] Audio/Video

Hi guys, I'm planning of creating a speech/video conference application. This application will provide a system to see/listen to each personn present in the conference. So each ppl will have a audio and video stream. I'm wondering if you know a way to do this with asterisk or if it's supported

Re: [asterisk-users] Conference Meetme

Hi! I need the server to handle about 300 - 400 simultaneous meetme conferences, 5-10 participants in each, Actually I need to know, if I will get an IBM X3650 M2,QuadCore, 4-6 GB RAM, 8MB cache, how many simultaneous meetme conferences I can operate on a this server. There is no

Re: [asterisk-users] [Conference] Audio/Video

Le 04/15/2010 12:11 AM, Hans Witvliet a écrit : On Wed, 2010-04-14 at 22:55 +0200, Stéphane Bauland wrote: Hi guys, I'm planning of creating a speech/video conference application. This application will provide a system to see/listen to each personn present in the conference. So each ppl

Re: [asterisk-users] [Conference] Audio/Video

On Wed, Apr 14, 2010 at 4:55 PM, Stéphane Bauland baula...@epitech.net wrote: I'm planning of creating a speech/video conference application. This application will provide a system to see/listen to each personn present in the conference. Else, do you know any other way to do this ?

Re: [asterisk-users] [Conference] Audio/Video

http://www.projectdiastar.org/ looks promising... On Apr 14, 2010, at 7:04 PM, Stéphane Bauland wrote: Le 04/15/2010 12:11 AM, Hans Witvliet a écrit : On Wed, 2010-04-14 at 22:55 +0200, Stéphane Bauland wrote: Hi guys, I'm planning of creating a speech/video conference application. This

Re: [asterisk-users] [Conference] Audio/Video

We have used with success BBB (BigBlueButton - open source - http://bigbluebutton.org) and I recommend to try their demo in order to see if this solution gives all you need. Voice conf is based on Asterisk. HTH, Ioan Indreias www.modulo.ro On Thu, Apr 15, 2010 at 2:04 AM, Stéphane Bauland

[asterisk-users] Conference Calling

Hey All, I want to implement a conference calling scenario. Conference Call Procedure:User1 dial the User2. When call is connected put the current call on Hold and dial User3. When the call is connected between User1 and User3 join the User2 in a conference room!How I can implement this

Re: [asterisk-users] Conference Calling

Here is where to get you start with this. http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO -Tri From: Faheem faheem_...@yahoo.com To: asterisk-users@lists.digium.com Sent: Sat, February 27, 2010 12:08:24 PM Subject: [asterisk-users] Conference

Re: [asterisk-users] Conference Calling

Muhammad It is not really your scenario but the scenario to setup a conference call with three numbers could be to generate two call files that points to a local channel/a context/extension that route the leg into the conference room and have your own leg routed into the conference room

Re: [asterisk-users] Conference problem

The CM is sending the BYE messages.   Any ideas?   Christian --- On Wed, 4/22/09, Martin asteriskl...@callthem.info wrote: From: Martin asteriskl...@callthem.info Subject: Re: [asterisk-users] Conference problem To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users

[asterisk-users] Conference problem

Hello all,   I have some issues with the MeetMe application.   The working topology is as follows. The Asterisk (1.4.22-rc5) is connected through SIP trunk to a Call Manager (6.1.2) which is connected to a Cisco Voice Gateway. The Gateway is connected to PSTN through a PRI. The calls are

Re: [asterisk-users] Conference problem

run a sip debug and check whether it's asterisk disconnecting the calls (usually a SIP BYE message) or whether Asterisk is getting the disconnect from your Cisco GW Martin On Wed, Apr 22, 2009 at 10:56 AM, Cristi Iconaru cristi_icon...@yahoo.com wrote: Hello all, I have some issues with the

[asterisk-users] conference calling

Greetings listers. I'm running asterisk 1.4.21.2 on SUSE 11.0 using Polycom 501 phones. My outgoing connections are Zapata using a TDM401P. For the most part I can make and receive calls fine except for these 3 issues: 1. When I call an external conference, the

Re: [asterisk-users] conference calling

Turn off callprogres=yes or have it configured properly. It should fix your problem. regards Martin On Fri, Apr 3, 2009 at 2:42 PM, Danny Nicholas da...@debsinc.com wrote: Greetings listers. I’m running asterisk 1.4.21.2 on SUSE 11.0 using Polycom 501 phones.  My

[asterisk-users] conference function problems

The CLI shows zap is necessary for conference recording. Can I enable conference recording if using ztdummy or dahdi, how? ango -- Executing [...@owt_meetme:4] MeetMe(SIP/3601-c80b4520, 5599|rcixMP) in new stack == Parsing '/etc/asterisk/meetme.conf': Found -- Created MeetMe

Re: [asterisk-users] conference and wifi phones

On Tue, Mar 24, 2009 at 2:14 AM, Michael Graves mgra...@mstvp.com wrote: Amen to that! Unles you have some compelling reason for VoWifi it's not worthy of consideration. Especially for SOHO or small biz use. Too costly to do well. I have never understood why anyone would use wifi just to get

Re: [asterisk-users] conference and wifi phones

On Mon, 23 Mar 2009, Kelvin Chan wrote: One of our local companies here in the UK are trialling a new conference phone - the Konftel 300IP SIP however it's still as expensive as a Polycom, but that might be the $/£ exchange - might be cheaper where you are? It seems like an interesting

Re: [asterisk-users] conference and wifi phones

On Tue, 24 Mar 2009 01:51:36 + (UTC), Jeff LaCoursiere wrote: On Mon, 23 Mar 2009, Michael Graves wrote: On Mon, 23 Mar 2009 20:01:51 -0400, Dean Collins wrote: Siemens make a range of DECT handsets under the Gigaset model range. Yes they shit all over every wifi handset I have ever

Re: [asterisk-users] conference and wifi phones

I REALLY like the Snom M3 DECT SIP base. You can have up to 3 simultaneous calls through the base and you can have up to 8 phones registered with it. It's all web managed as well as http/s provisionable and has this nice phone to line matrix where you can set which phones ring on inbound calls and

Re: [asterisk-users] conference and wifi phones

2009/3/24 Steve Gladden aster...@michiganbroadband.com I REALLY like the Snom M3 DECT SIP base. Yeah - it's such a pitty that you always have to buy it bundled with one of these crappy handsets. Or is there a way to get only the base that I don't know? Chris

Re: [asterisk-users] conference and wifi phones

-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of randulo Sent: Tuesday, March 24, 2009 1:53 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] conference and wifi phones snip I have never understood why anyone

Re: [asterisk-users] conference and wifi phones

Discussion Subject: Re: [asterisk-users] conference and wifi phones snip I have never understood why anyone would use wifi just to get cordless facility when DECT works so much better. snip /r ___ -- Bandwidth and Colocation Provided by http

Re: [asterisk-users] conference and wifi phones

ok -Original Message- From: Frank Bulk frnk...@iname.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] conference

[asterisk-users] conference and wifi phones

Hi guys, I'm looking for a affordable conference phone and a wifi phone that has a cradle. Polycom seems to make pretty nice conf phones but the price is a bit crazy for us. I saw the recommendation with ATA plus an analog Polycom phone but I do prefer a SIP phone. All because it's just too

Re: [asterisk-users] conference and wifi phones

On Mon, 23 Mar 2009, Kelvin Chan wrote: Hi guys, I'm looking for a affordable conference phone and a wifi phone that has a cradle. Polycom seems to make pretty nice conf phones but the price is a bit crazy for us. I saw the recommendation with ATA plus an analog Polycom phone but I do

Re: [asterisk-users] conference and wifi phones

On Mon, 23 Mar 2009, Kelvin Chan wrote: Hi guys, I'm looking for a affordable conference phone and a wifi phone that has a cradle. Polycom seems to make pretty nice conf phones but the price is a bit crazy for us. I saw the recommendation with ATA plus an analog Polycom phone but I do

Re: [asterisk-users] conference and wifi phones

For wifi phone, I tried Linksys iPhone. It works well but lacks a cradle. My users often forget to charge it when they leave for the day and come back to a dead wifi phone for the next morning. I still don't get the market for this kind of phone. DECT cordless phones can be had for

Re: [asterisk-users] conference and wifi phones

One of our local companies here in the UK are trialling a new conference phone - the Konftel 300IP SIP however it's still as expensive as a Polycom, but that might be the $/£ exchange - might be cheaper where you are? It seems like an interesting product. Compared to Polycom 7000, it's

Re: [asterisk-users] conference and wifi phones

(London in-dial). -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kelvin Chan Sent: Monday, March 23, 2009 7:38 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users

Re: [asterisk-users] conference and wifi phones

On Mon, 23 Mar 2009 20:01:51 -0400, Dean Collins wrote: Siemens make a range of DECT handsets under the Gigaset model range. Yes they shit all over every wifi handset I have ever used. Dect is way better. Amen to that! Unles you have some compelling reason for VoWifi it's not worthy of

Re: [asterisk-users] conference and wifi phones

On Mon, 23 Mar 2009, Michael Graves wrote: On Mon, 23 Mar 2009 20:01:51 -0400, Dean Collins wrote: Siemens make a range of DECT handsets under the Gigaset model range. Yes they shit all over every wifi handset I have ever used. Dect is way better. Amen to that! Unles you have some

Re: [asterisk-users] Conference with an AGI inside Queue for password change

2008/12/19 Rajkumar S rajkum...@gmail.com Hi, I have a typical call center with queues and agents added via AddQueueMember. One of my requirement is to implement a forgot password function. If a caller does not remember the password, he calls up an unauthenticated line and the agent

Re: [asterisk-users] Conference with an AGI inside Queue for password change

maybe a simpler solution is set some variables to the caller channel trasfer to extencion where asterisk ask for the password put it in the data base and then transfer back to the agent. this is not so dificult to implement. you can use the mysql function or you can make a webservice and use CURL

[asterisk-users] Conference with an AGI inside Queue for password change

Hi, I have a typical call center with queues and agents added via AddQueueMember. One of my requirement is to implement a forgot password function. If a caller does not remember the password, he calls up an unauthenticated line and the agent manually authenticates him. Then the caller should have

Re: [asterisk-users] conference bridge

Nhadie Ramos wrote: Hi, How can i setup conference when i have 2 asterisk servers? my setup is 2 asterisk servers using realtime, i'm simply using DNS SRV just for redundancy (not really high availability). i have a web interface, wherein i can create extension, conference etc. adding

[asterisk-users] conference bridge

Hi, How can i setup conference when i have 2 asterisk servers? my setup is 2 asterisk servers using realtime, i'm simply using DNS SRV just for redundancy (not really high availability). i have a web interface, wherein i can create extension, conference etc. adding extension is ok, even if

[asterisk-users] Conference Hangup

Hi all, I have a question on asterisk conference. Now I use appl Meetme with option A x because when a marked person hangup I want to close all the conference. But what I have to do if I want two marked person and kill the conference when one of two hangup? Is possible? Thanks. Enrico. --

[asterisk-users] Conference Call Dial-Out to a participant

I have created a conference call solution for a client and works fine. The next challenge is to let the conference dial out the participant instead. Has anyone done this before or know the function to achieve this? Thanks. ___ --Bandwidth and Colocation

Re: [asterisk-users] Conference Call Dial-Out to a participant

On Sunday 18 November 2007 10:20:18 broadband Voice wrote: I have created a conference call solution for a client and works fine. The next challenge is to let the conference dial out the participant instead. Has anyone done this before or know the function to achieve this? Thanks. Please see

Re: [asterisk-users] Conference Call Dial-Out to a participant

I looked through /etc/asterisk and could not find the folder sampl.call. On 11/18/07, Tilghman Lesher [EMAIL PROTECTED] wrote: On Sunday 18 November 2007 10:20:18 broadband Voice wrote: I have created a conference call solution for a client and works fine. The next challenge is to let the

Re: [asterisk-users] Conference Call Dial-Out to a participant

You can find it enclosed sample.call Description: Binary data ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit:

Re: [asterisk-users] Conference Call Dial-Out to a participant

On Sun, 18 Nov 2007, broadband Voice wrote: I looked through /etc/asterisk and could not find the folder sampl.call. On 11/18/07, Tilghman Lesher [EMAIL PROTECTED] wrote: On Sunday 18 November 2007 10:20:18 broadband Voice wrote: I have created a conference call solution for a client and

Re: [asterisk-users] Conference Call Dial-Out to a participant

broadband Voice wrote: I looked through /etc/asterisk and could not find the folder sampl.call. That is the Asterisk configuration directory. You are looking for the Asterisk SOURCE CODE directory. If you installed from a package (.deb, .rpm, etc) then you will have to contact the packager

Re: [asterisk-users] Conference Call Dial-Out to a participant

On Sun, Nov 18, 2007 at 01:37:00PM -0600, Eric ManxPower Wieling wrote: broadband Voice wrote: I looked through /etc/asterisk and could not find the folder sampl.call. That is the Asterisk configuration directory. You are looking for the Asterisk SOURCE CODE directory. If you installed

[asterisk-users] Conference rooms

I all, I have a question about the use of conference rooms: can I, with a Voip telephone or softphone call some other telephone and invite them in a conference room? I read a lot of documentations about asterisk, but i can't find any example ! Thanks, best regard Fabio

Re: [asterisk-users] Conference rooms

Hi Fabio, Once you have an Asterisk box that have a conference room configured and a VoIP phone the supports forward you can easily forward your guests to the conference room. Moreover you can create a conference room extension available, via password, from the PSTN line. Hope this can help

[asterisk-users] Conference Calls with single-line SIP

Hi all, If I have 2 single-line SIP phones, I can still do a conference call using Asterisk, right? For example, two people in my office are on the call, along with 1 other person at a remote site. Regards, Zaheer ___ --Bandwidth and Colocation

Re: [asterisk-users] Conference Calls with single-line SIP

Yes, that will work fine Zaheer. On 16/10/07 1:32 AM, Zaheer Master [EMAIL PROTECTED] wrote: Hi all, If I have 2 single-line SIP phones, I can still do a conference call using Asterisk, right? For example, two people in my office are on the call, along with 1 other person at a remote site.

[asterisk-users] Conference call today at 12:30 PM EDT

Hey folks, Here's your chance to report in about Astricon, ask or answer general asterisk questions, talk about your asterisk-related (or voip-related) projects, sites, work, anything. We interested and listening. We have a great core group on these conferences, even though Indiana is

Re: [asterisk-users] Conference bridge.

On Thu, 13 Sep 2007, Paul Hales wrote: On Wed, 2007-09-12 at 16:44 -0400, Alex Balashov wrote: Any recommendations for an affordable SIP conference bridge unit? I mean one that isn't crappy; something where the duplex and cancellation functions that are traditionally built into such devices

[asterisk-users] Conference bridge.

Any recommendations for an affordable SIP conference bridge unit? I mean one that isn't crappy; something where the duplex and cancellation functions that are traditionally built into such devices actually work. I am referring to something that looks like this . . .

Re: [asterisk-users] Conference bridge.

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Alex Balashov wrote: Any recommendations for an affordable SIP conference bridge unit? I mean one that isn't crappy; something where the duplex and cancellation functions that are traditionally built into such devices actually work. Most

Re: [asterisk-users] Conference bridge.

On Wed, 2007-09-12 at 16:44 -0400, Alex Balashov wrote: Any recommendations for an affordable SIP conference bridge unit? I mean one that isn't crappy; something where the duplex and cancellation functions that are traditionally built into such devices actually work. Do you want something

Re: [asterisk-users] Asterisk Users Conference Friday @ 12:30PM EDT

ENUM and ISN You may be interested to know that John Todd was kind enough to come by at the last minute and give us a thorough grounding in ENUM and expand our knowledge about http://Freenum.org where you should run, not walk, to get yourself an ISN (ITAD Subscriber Number). You can listen to or

[asterisk-users] Asterisk Users Conference Friday @ 12:30PM EDT

FRIDAY September 7th at 12:30 PM EDT http://www.asteriskusersconference.org for more information on how to listen, talk, or both :) This week, ENUM is the main subject, although our friends at e164.org haven't been able to talk to us as planned. Come on by and share what you know about ENUM or

[asterisk-users] Friday Aug 10 @ 12:30 PM EDT - Asterisk Users Conference

This week, the second part of connecting to the outside world using TDM, ATA and even... IAX hardphones with compilable software. More on topics and guests: http://groups.google.com/group/asterisk-users-conference Instructions: http://www.AsteriskUsersConference.org IRC on freenode.net

[asterisk-users] Friday Aug 10th Asterisk Users Conference at 12:30 PM EDT

and scheduling of the conferences. If you feel like this is of interest, please join us: http://groups.google.com/group/asterisk-users-conference I hope we can make this a good way for you to know if topic of interest to you comes up. In the future, we'd like to get people using ENUM and DUNDI

Re: [asterisk-users] Next Friday at 12:30 PM EDT: Asterisk Users Conference TDM inside and outside the box

Steve, On 8/3/07, Steve Totaro [EMAIL PROTECTED] wrote: I just tried to call in after creating an account. After the call connects, enter the show id: 22622# and your_PIN# I dial in and am asked for the podcast id, I enter 22622# and am told that my passcode is not correct. I also tried

Re: [asterisk-users] Next Friday at 12:30 PM EDT: Asterisk Users Conference TDM inside and outside the box

Hi folks, The August 3 edition of our Friday conference call and podcast kicks off in just over and hour. I know the list isn't delivering properly but if a few people get this it'll be better than none. We are going to be talking today about TDM inside and outside the box. I own some

Re: [asterisk-users] Next Friday at 12:30 PM EDT: Asterisk Users Conference TDM inside and outside the box

I just tried to call in after creating an account. After the call connects, enter the show id: 22622# and your_PIN# I dial in and am asked for the podcast id, I enter 22622# and am told that my passcode is not correct. I also tried just entering my passcode but got the same error message.

[asterisk-users] Next Friday at 12:30 PM EDT: Asterisk Users Conference TDM inside and outside the box

Hi, I am going to be on the road for the next few days and with the variable delay on the mailing list, I am posting this now, 4 days before the conference. If you haven't yet listened or participated, please consider doing it. We have a great kernel of people at all levels of expertise and ideas

Re: [asterisk-users] Asterisk Users Conference Friday at 12:30 PM EDT

On 7/27/07, dave cantera [EMAIL PROTECTED] wrote: randulo, I could not get into the conference today... the SIP line was busy, no matter what I do, the website thinks I'm not logged in and gives me the login page. after I login, anything I want to do brings me back to the login page... so I

[asterisk-users] Asterisk Users Conference Friday at 12:30 PM EDT

You can listen or join the Asterisk Users Conference Fridays at 12:30 PM EDT Today's subject suggestions: FAX capabilities, what's your solution? Multiple asterisk server implimentation: ENUM, DUNDI or even two servers connected Your subjects? Share your ideas, ask your questions! See http

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