similar to: external IVR

Dear All I want to use asterisk for the following Senario and Need help to find a SAMPLE extension.conf Incoming call >>>>>>>>>>>Asterisk >>>>>>>>>>>>>>GSM Termination Gw first leg second leg What I want to do is putting first call leg on

HI ALL; I found in the following page: http://www.voip-info.org/wiki-Asterisk+G.729+Licensing 1-If I could record all IVR promts in G729 format 2-If I could record voicemail in g279 format with """format_g729.c""""" then I donot need any g729 license (I suppose all my clients have g729 ip phones) My question is, how

Hi all, I'm running 1.6.2.0-rc6, and I'm running into a problem: sometimes the audio vanishes in the middle of listening to an IVR background prompt. This happens with both analog (Digium card) and IAX2 incoming calls. The prompts are stored in ulaw format (and the IAX2 calls use ulaw). The asterisk console claims that the IVR prompts are proceeding in the expected fashion, but I

Is anyone else experiencing problems with Asterisk 1.2, the ivr does not work. I have tried it on Linksys RT31P2 and Grandstream Handytone 496. After a call goes through you're not able to enter any of the prompts on a IVR. and cannot enter pin numbers when using a calling card or anything that requires you to enter into an ivr system. I already set my dtmf mode in asterisk. --------------

can i use MP3 files as an IVR prompts directly without converting to .gsm format? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091010/67205e07/attachment.htm

Hi all! Is anyone using the $subject setup? What I would like to do the following setup: 1. OXE is setup for receiving calls, handling Agents 2. Asterisk as external IVR on extension 9xxx connected with ISDN (Q.931) PRI The incoming calling route: 1. OXE handles incoming calls, answer 2. Transfer to extension 9xxx 3. Asterisk answer (using one channel) 4. IVR is handling calls 5. If needed IVR

A customer of mine wants an IVR where the first 3 choices are 1 English 2 Spanish 3 French I can build the IVR but how do I get the system prompts to then speak the selected langauge. For example, a caller has selected Spanish and so is routed to the Spanish part of the IVR. At some point he breaks out of the IVR to leave a VM. How does the system know to continue offering him Spanish?

Hi Is it possible to receive video calls using Asterisk and then process them as an IVR ? One of our clients wants to set-up a video IVR system in the US and we are evaluation possible options. Also, what is the bandwidth of receiving a video call in US ? What protocols and codecs are supported and does it work on DID numbers ? Can I rent a hosted solution for this ? Thanks in anticipation of

Hello, I'm trying to understand why when I make a call from a SIP phone to an external number who has an IVR system in which I've to choose some options using the dialpad, it does'nt recognise the key pressed and remains still waiting for my choose. I'm tryng using Grandstream 102, and i've tryed with all the 3 modes possibile: Dtmf inband, rfc2833 and INFO (obviously

On Wed, 2006-12-06 at 23:51 -0700, asterisk-users-request@lists.digium.com wrote: > Date: Wed, 06 Dec 2006 22:37:01 -0500 > From: Steve Prior <sprior@geekster.com> > Subject: Re: [asterisk-users] Is there any Asterisk controllable > thermostat? > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> >

I am using Asterisk 1.06 Stable. When I dial my Mobile Number to check Voice Mail or my Bank Account Phone Access Number, the IVR System on the other end asks me to enter *2378 to transfer to an attendant. But When I press *2378, Asterisk tells me that it cannot transfer the calls and gives an error on CLI saying Extension '' does not exist in the dial plan. What is the trick to make

After over a year of alphas, betas, and release candidates I'm happy to announce that Version 1.0 of SafiServer and SafiWorkshop has just been released under the open source license GPL (ver 3). You can download installers from our site www.safisystems.com and the source code can be downloaded from Sourceforge (more details available on our site). If you're not familiar with our system

<FONT face="Default Sans Serif,Verdana,Arial,Helvetica,sans-serif" size=2><DIV><BR><DIV>Hi!</DIV><DIV>&nbsp;</DIV><DIV>Does anyone have any ideas on how to build an interactive IVR where questions are asked by Asterisk (pre-recorded prompts), the caller answers the questions, and the system records the answers and emails the whole

Got 1 issue I can't seem to knock out of this particular box. The IVR works fine on the zap channels and the incoming SIP routes. But coming in via the IAX2 route leaves me with a silent phone. The prompts all work still letting me navigate the menu. But just can't hear anything. This is with A@H 2.8 (Asterisk 1.2.7.1, with FreePBX 2.1.0 also installed) Any thoughts on where to

HI ALL; Anybody can explain the difference between "call parking " vs "call transfer" Regards mohammad -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040720/2975991b/attachment.htm

Can someone point me to some reasonable example / starting point to implement a basic IVR menu? Looking for something rather simple like the press 1 for sales, 2 for tech support, and probably an option to list the voicemail directory kind of thing. Nothing elaborate needed, just basic menu. (Yes, I did look at the wiki and google searched for "ivr menu".)

HI ALL; I have problem converting a windows .wav file to .gsm format by Sox. Could anyone help. Cheers, Mohammad -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050706/3408bfd5/attachment.htm

HI ALL HI MICHAEL; My name is mohammad and I am iranian.I have been trying to install oh323 channel but I come up with dead end. In fact it makes me crazy. plz help me michael. I saw mailing list and I trid serevel CVS headers such as , 2004-06-07( seven of june) 0r 2004-07-02( second of july) besides I use: 1-openh323 v1.12.2 2-pwlib v1.5.2 3- asterisk CVS (2004-06-07,

HI All; I want to use Asterisk for just Voicemail Server and I need a Dynamic creation of Mailboxes. My users 's Mailboxes are same as "Extensions" but I donot want to add mailboxes in "Voicemail.conf" Is there any way to create mailbox from Asterisk dial-plan ? Appreciate any suggestions Mohammad Mirzaee -------------- next part -------------- An HTML attachment was

Hi All, I have setup 2 trunks using 2 different voip providers using sip. the first one i have no problem calling inbound then redirected to an IVR, i can hear the IVR. the second one has issues, inbound works going to IVR as i can see it on the CLI, but i don't hear anything. i tried redirecting it to an extension not an IVR just to see if inbound really works, and it rings the