similar to: Answer Confirmation with SIP/AIX channels

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

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

Hey, I have a few dids with callcentric. They seem to work fine most of the time but at some points I get "handle_request_invite: Failed to authenticate user <sip:PSTNnumber" This happens intermittently. The way I understand it the insecure=port,invite should tell asterisk not to authenticate users coming from that host. But its not working for some reason. This is my sip.conf

I was looking at the sample "follow me" config (http://www.voip-info.org/wiki-Asterisk+Tips+follow+me <http://www.voip-info.org/wiki-Asterisk+Tips+follow+me> ) which uses a dial modifier 'c' to enable Answer confirmation - "If the letter c follows, then "Answer Confirmation" is requested, in which the call is not considered answered until the called user

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

Hi, I am working on calling 2 registrations of same user on 2 different ip or ports. It works fine and both phones ring simultaneously. the problem is that there is one way audio, calling party can hear me but i can't hear calling party. here is the scenario.. SIP/XYZ at 192.168.0.20:5060 SIP/XYZ at 192.168.0.10:5678 i dial using following dial string Dial(SIP/XYZ at

Is there a way to get answer confirmation via IAX and not only via ZAP? We get our outbound service via an IAX trunk to our provider so we aren't in control of their ZAP configs, but ideally we'd like to be able to achieve the answer confirmation functionality regardless, especially in the case of follow-me that dials SIP and IAX extensions simultaneously when the IAX dials a

Hi, Pretty sure I'm missing something simple, but I've seen references to this feature but not found documentation for it: I have a queue set up so that many people are contacted (ringall) when a call comes in. I would like the answering party to confirm that he is a human being rather than cellphone voicemaill by pressing a digit. This is somewhat similar to the 2nd macro example

Take a look at freepbx 2.2 beta. We have made both ringgroups and follow-me have a call confirmation option. When used, the ringgroup/follow-me extensions that are outside lines (like your cell phone) must confirm they want the call (press 1 to accept, 2 to decline). All the while the caller hears ringing (or MoH if chosen). If no answer, they are sent on to where ever else you want them to go

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

Using Digium's "Asterisk Developer's Kit (TDM)", I've been trying to make an outside call by copying sample.call to /var/spool/asterisk/outgoing. I want the VoiceMailMain to run when the call is answered. The call is made correctly but, as you probably know, the application starts as soon as the call is made. I see there are two solutions: Using callprogress=yes in

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

Hi all, Lets say I have my incoming calls transfered to my mobile phone. When a call comes in, Asterisk will answer the call and ask the caller to hold the line while the call is being transfered. I know how to do this, but i dont want the caller to hear me answer the mobile phone. They can hear some music on hold. When I answer Asterisk will read the callerID to me and I can then decide if this

I just read on Slashdot (at http://yro.slashdot.org/article.pl?sid=07/12/16/222243 ) that Trixbox "has been phoning home with statistics about their installations", as a Trixbox user exposed in "Trixbox Phones Home" at http://www.trixbox.org/forums/trixbox-forums/open-discussion/trixbox-phones-home . -- (C) Matthew Rubenstein

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 forwarded to Asterisk by the CM. ? The problem is that some users who are calling in from PSTN are getting

> Hello All, > > History ~ > I recently took a position with a call center. At the time they had > about 50 agents in a call queue. The queue was setup to ringall. The > agents use Eyebeam softphones. Everything is local lan, no routers, > everything connected via Cisco 3600 10/100 switches. > > Now we are up to about 150 agents, and I have kept everything pretty

I've been using Asterisk 1.2.6 with a 4 port FXO Sangoma A200 card for the past month without many problems (other than the fact that the Sangoma card doesn't disconnect hung up calls immediately, which I posted about in another thread, and has still not been fixed), however, I had a call from one of our clients today and they complained that our phone system kept ringing through when they

I'm trying to get dynamic agents/queues working for any type of telephone with a DID. I need an application or a method to inhibit a channel/technology from responding with an Answer() until the queue member accepts the call by hitting '#'. This way I can use any POTS line as a queue member, saving costs on ATA's for home workers who only spend a minority of their time at home.

I yesterday brought up to date the version of * the CVS and now I have a problem. I cannot effect the RELOAD that * it breaks. Somebody can help or say as to load new users without stopping * ? Thank?s Excuse my English Joao Carlos Moura

Hello list I don't get to compile h323. I have the mistake: asteriskaudio.cxx: In destructor `virtual PAsteriskSoundChannel::~PAsteriskSoundChannel()': asteriskaudio.cxx:167: `baseChannel' undeclared (first use this function) asteriskaudio.cxx:167: (Each undeclared identifier is reported only once for each function it appears in.) make[1]: ** [asteriskaudio.o] Erro 1 make[1]: