similar to: Retain call control: Avoid letting call get

Try this subject line if you will. On 11/14/06, joe a. <joea@j4computers.com> wrote: > > Did not know how to make up a subject line for this. > > I have a dial plan that allows a caller can try my cell phone. And that's > fine. If the call cannot be made, it sends caller back to voice menu. > > However, I'd like a way for the caller to elect to go back to the

I love TrixBox, with the custom config files you can tweak pretty much with TrixBox too, I have at least done some. Plan to do a plain Asterisk install later, but for now I learn a lot about the config files just with TrixBox. Some things might be a bit harder with TrixBox due to some of the premade dial plans, but can get it to work :-) _____ Fra: asterisk-users-bounces@lists.digium.com

Hi, I'm having problem with call parking. When I park call, either via transfer to xten or park digit sequence from features.conf, I hear the parking lot number read to me and the user gets transferred. However, MOH stops for the caller the moment user is transferred. The user can be retrieved by dialing the parked extension and voice resumes. If the parked user hangs up, the channel state

We're looking at using 4 or 8 port T1 cards with echo cancellation and are evaluating brands to go with. We know that Sangoma has excellent solutions especially when it comes to echo. But we still have to hear about actual performance of a Digium card using the same Octasic DSP echo canceller. Would appreciate hearing something on this. --------------------------------- Sponsored Link

Hi Friends, Thank you for your response. Yesterday only, I configured my Nokia E70 mobile and its working fine. For group members convenience, here I am giving the configuration: Configuring the Nokia E70: Go to Menu - Tools - Settings - Connection - Sip Settings - Profile name: Olivetalk Service Profile: IETF Default Access Point: Olive Public user name: sip:102@202.xxx.xxx.xxx Use

From: "Time Bandit" <timebandit001@gmail.com> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Date: Tue, 7 Nov 2006 08:53:51 -0500 Subject: Re: [asterisk-users] Follow Me problems > Today we appear to have discovered our first bug. We have an extension > setup to "followme" by ringing that extension

Hi Friends, Recently, I bought NOKIA E70 mobile. I have configured my mobile to connect with my Asterisk server depends on the information available in Internet. But, its telling that "Registration failed". If anybody configured this mobile for Asterisk server, please tell me the step by step configuration or please tell me a good website link to do this. Looking forward to your

Hi all, I want to initiate a call from the asterisk to an extension, where I will forward the asterisk side to another extension later (to the conference extension). I can initiate a call uning originate call from an extension to the desired extension, but it would need someone from the originator extension to answer the phone. How can i register an extension to asterisk where it

Hi, I have asterisk installed. When a connection comes from the outside one of our phones rings for about 45 seconds. Is it possible to another phone fetch the call while it's ringing on the first phone? I don't want to use ringgroups because the second phone would be ringing also. Thanks Nuno Fernandes -------------- next part -------------- A non-text attachment was scrubbed...

Hello, I have a default installation of asterisk 1.6.1.9-2 When i create a user in users.conf via asterisk-gui, calls, voicemail etc works. But if i create a user realtime (and my realtime caching is available too) i can see the realtime user with sip show peers. But, my local dial rules does not work. I can call from realtime user to static users(the ones in users.conf) and if they are not

Hello all, Asterisk Management Portal 1.10.009 has now been released. This exciting new version has several notable additions (listed below). The AMP homepage is http://amp.coalescentsystems.ca. Here you'll find links to the download, install guide, and documentation wiki. As usual, please use amportal-users mailing list for discussions about AMP:

Hi list, For months everything worked super here in our setup. This week I implemented some new idea in our webbased calendar system. I thought it would be nice to have an option that tells asterisk you are not available for calls during an appointment. For this to work I could no longer use the ringgroup setup: Dial(SIP/10&SIP/11&SIP/12,40,tr) So I thought, why not use the Local channel

Hello, we want to setup the following scenario: - each user has a softphone AND a hardphone - the softphone is started with the operating system - the hardphone is connected all the time using SIP - only ONE extension for each user Both phones should ring when the user is called. We've setup an asterisk 1.4.18 and at the moment only the last registered client rings. In Asterisk 1.2 the

Hello all, The Asterisk Management Portal (AMP) is now known as FreePBX. FreePBX 2.0.1 is now available for download. A **BIG** thank you goes out to the project developers for all their hard work, and to beta testers for running FreePBX through it's paces! This exciting new release boasts a better user experience, additional functionality, and a new module system. The module system is

Ok here is what I''m trying to accomplish. Right now we''ve only 3/4 T for internet access. This new program they want requires a full T just for it. My proposed solution is to bring in a DSL or wireless link for some cheap access for this new application. What I would like to do is setup a linux box with 3 nics. Traffic would go like so client-->linux box-->firewall

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

Hello, I need to activate the permissive parameter for a asterisk PCI card. But this parameter doesn''t exist in /sys/module/pciback/parameters/ . I tried to activate this option in the boot option (pciback.permissive) ... When I want to use my PCI card in a domU : Nov 20 10:41:55 myhost kernel: pciback 0000:07:01.0: Driver tried to write to a read-only configuration space field at

Hi, I set up a ring group. I would like for people who select a certain voice menu option to ring a list of extensions (I have just one extension in there at the moment) and if it doesn't answer to go to an extension's voice mail. I am using a version of asterisk from CVS, last updated a couple of weeks ago. This line in extensions_addtional.conf sends the call to ringgroup 3 if

We need incoming calls to simultaneously ring SIP phones, and a cell phone which is called via a SIP or IAX trunk. When the cell phone answers we'd like a brief prompt played (e.g. "press # to accept call") and if # is pressed connect the incoming call to the cell phone. ZAP trunks have some of this functionality with the call confirmation option, but we must use SIP or IAX trunks.

Hi everyone! I'm quite a newbie at this Asterisk stuff so please bear with me. We've recently decided to start training in Asterisk via AsteriskNow! Asterisk version is 1.4.18.1 through AsteriskNow! 1.02 The box we have is paired with a Digium TE110P and we've managed to get it to the point where incoming calls via a single DID (from NTT Japan) can be received and answered