similar to: IVR No sound on other provider

Hi all, I've an asterisk 1.2.5 running very well for about a 9 months, and suddenly i cannot dial extensions 4XXX from SIP Phones. Now comes the wired stuff... I can dial this extensions from IAX phones as well as from Analogue extensions connected to our legacy pbx, that is installed on front of asterisk. So : Zapata Calls to SIP extensions 4XXX - OK IAX to SIP 4XXX-OK SIP to SIP 4XXX -

Hello veryone, I have a strange problem. I have an asterisk (latest from CVS) with latest oh323 channel driver. I place calls with DIAX. The H323 gateways only support G711A De DIAX only supports GSM When I perform an inbound call: H323 -> asterisk -> DIAX :: sound is ok. When I perform an outbound call: DIAX -> Asterisk -> h323 :: sound is terrible and CPU load is 80% When I

I've got the XTEN Lite soft phone mostly working with * but it's dropping out like a very bad cell phone call. The GSM codec is worst (unusable), G711u and G711a are best but not good enough to use. I don't think it's a lack of bandwidth. What tuning options or approaches should I be investigating to make this work. Also, what's the best soft phone(s) for Windows XP?

Hello; I certainly appreciate your response. In fact, I used that exact solution for three of the incoming lines. I setup ring groups and a silent ringtone for each phone. Unfortunately, the last incoming line is more complicated and uses an IVR with multiple input choices, so the solution is not as clear cut as for the other ones. That's why I was trying to look at other options. Best

Hi, I have installed two X-Lite phones and they're able to login successfully. The two phones plus the Asterisk system are all on the same LAN with private addresses assigned to each of them. When a call is initiated and is picked up on the other end, there is completely no sound at all (as in the line goes dead). The codecs set in the softphones are g711u, g711a, GSM, iLBC and SPX.

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

I know but this is not my sole decision. On 12.12.2017 16:17, Ron Wheeler wrote: > If your phone system goes down and you can not get it back up until > tomorrow afternoon because your support person is on another project, > you may wish you had an SLA.

Hi All; Using the FreePBX, after I added the extension from the GUI, I discover that it is automatically added in the extensions_additional.conf in the context [ext-local] and [from-did-direct-ivr] How I can change these context name? I need to determine this. How? Regards Bilal

Anyone can recommend a commercial large scale IVR with easy + pro management for asterisk? Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091124/53055c0c/attachment.htm

Dears Any one knows a standalone voip transcoder software name,not an ip pbx. What I want is to transcode the incoming sip calls from g711 to g723 or ilbc or g729 ..... and forward it to a media gateway .. Regards Khaled chehab ********************************************* No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by

Hi, I am working on a H.323 project which involves GnuGk and Asterisk My current goal is to provide IVR functionality for the H.323 users which register through GnuGk(eg. call credit information) I have successfully built a H.323 platform using GnuGk - it uses SQL accounting and authorisation. Now I am trying to integrate it with Asterisk in order to provide IVR functionality as I already

HI, I have in 'inbound route' a IVR, with press 1 or 2 the destination call is always a ring group called '600', my problem is that after press 1 (but this problem is present also with press 2) before that the inbound call is transfer to extension pass 10/11 seconds ! In attach log file about incoming call. I use Trixbox with Asterisk-1.6.0.10. Thanks. ------ Salvatore.

Hi, Bit of a puzzle this one - let's see if anybody else can shed some light... I Ghosted my Asterisk box to build one for a colleague. I added one HFC card to make the total two and amended zaptel.conf and zapata.conf accordingly. I tested it with my handsets on my ISDN lines and it worked AOK. I unplugged the box and took it to his house and plugged it in. He doesn't have any ISDN

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

I wonder if anybody know how to hire Alice or some professional voice-artist. I need to record 12 messages for a customer.

Hi, For a 20 users prospective customer, I'm wondering if any GUI would allow and end user to edit an Asterisk IVR tree ? For instance, I'm looking for something allowing to edit interactions like : "wait up to 20 seconds and say this "to reach sales department, type 1, to reach tech support type 2" message" Regards -------------- next part -------------- An HTML

Hi All, If i have this kind of setup, what do i need to make it's load balance. [ asterisk 1 ] -- [ asterisk 2 ] -- [ asterisk 3 ] -- [ asterisk 4 ] | | | | --------------------------------------------------------------------- | mysql cluster |

Hello list, I was wondering if it were possible for asterisk to do a voice recognition type IVR? Like you know how most banks and stuff do, where they ask you to say your selection or key it in? If this is possible, how can I set this up? I'm using FreePBX 2.11 on Linux, CentOS 6.7 32-bit, asterisk version 1.8.32.3. Thanks.

I'm thinking about using asterisk as an IVR system with an existing avaya index system. I've got 2x PRI 30 lines coming in to the Index, and I have 4 spare PRI cards in the Index. I was thinking about using a QUAD PRI card from Digium and having the calls come into the Index then transfer to Asterisk for IVR then back to the Index. That way if we get 60 inbound calls we'd in

hi all, i have gradwell DID i am using it for inbound dialing with IVR when ever customer call my DID some times other IVR is cumming on my IVR that IVR is not even related with my server .can u please help me on this -------------- next part -------------- An HTML attachment was scrubbed... URL: