similar to: ILBC to G711 transcoding experince ?

I assume most people here know what a joke DIDX is -- but in case you didn't already know, please avoid these people. Basic features of their service don't work, their tech support refuses/drags their feet to fix them for a month and if you post publicly about them, they terminate your service. Instead of investing their effort in reading mailinglists to terminate customers maybe they

Dear Asterisk Users Mailing List - Non-Commercial Discussion, I joined VirtualPhoneLine.Com service and am really enjoying the use of it. VirtualPhoneLine.Com allows me to buy virtual numbers from anywhere in the world and then forwards it to my Mobile Number, Regular Phone, MSN Messenger, Google Talk or an IP Phone. Have a look at the http://www.virtualphoneline.com/faq and

Hello We are developing an application to be deployed on E1 lines (inbound and outbound calls) What is the best way to fully test the application if we do not have E1 lines in the development environment? Is there some kind of software tester to test IVR/Callcenter applications virtually?? thanks and best regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, all Can someone tell me where to tell asterisk no to use 127.0.0.1 IP (localhost)? When I am registering with VoIP providers, they get my info as s@127.0.0.1. (This is SIP registration). Also, in SIP logs, when calling I am getting things like this: Executing SetCallerID("SIP/phone2-22c3", ""CID Name" <CIDNUMBER>") > in new stack > -- Executing

Dear Asterisk Fans, I'm an Asterisk consultant in Argentina and want to make an spanish wiki (something like voip-info.org). I have the idea and some concepts about this project. It won't be a comercial project, it would be free and it's target would be spanish talking asterisk enthusiasts. I'm wrinting these for the sake of finding contributors, people who want to help me

On Thu, Jan 22, 2009 at 19:34, Rehan Allah Wala <rehan at supertec.com> wrote: > Your service is still up and working, Because Suzanne Bowen has better judgment than you. > You did charge back on the payment to us, That is correct. There is $86 balance in my account I did not expect to get back by just asking for it. > We are being nice to you and you do not understand the

Dear All, For those who celebrate Eid. I would like to wish you a very Happy Eid Mubarak. For those who do not know what it is, Its a Prayer in memory of Ishmael son of Abraham, when he attempted to sacrifice his beloved son on request by god. Muslim's celebrate it with a sacrifice of a goat every year. I belive Christians & Jew's belive in the same. Peace and Harmony to all.

All of the emails I get from the list have the correct time with the exception of the typical list slowness. All of your emails (and only your emails and spam) are approximately 11 or twelve hours in the future. The email I am responding to has the correct day but the time reads 11:13 PM. I am using Thunderbird 2.0.0.5. If using Outlook, I think the time is correct. Thanks, Steve Rehan

Hi All, I have two Asterisks, one on the LAN that handles the internal calls with a PSTN interface and one on the DMZ with a public interface. I would like to route SIP calls from the internal users to the Internet over IAX2 to the DMZ and onwards. All users have soft phones so they would enter sip:someuser@somevoip.org to get a connection. I would like to avoid having number prefixes to dial

This seems to have been reported before, but I've seen no resolution: http://lists.digium.com/pipermail/asterisk-dev/2003-August/001470.html http://lists.digium.com/pipermail/asterisk-users/2003-August/019091.html http://lists.digium.com/pipermail/asterisk-users/2003-October/024472.html When forcing use of Speex, no sound comes out at all (Speex-1.0.3 on the Asterisk server) When forcing

Does anyone have any views on the best software base SIP client to use that normal users could use with Asterisk without being too techie ? I have tried the X-Lite client with varying success. The first version worked OK but music on hold broke the voice paths and the slightly newer version initiated the call but failed to make the voice connect in both directions. The SJphone client works but

All, Has anybody else experienced garbled voice between a phone using alaw/ulaw and one using iLBC? I have a Nokia E series phone with a preference to use iLBC and this works fine in Asterisk 1.2. However, since moving to 1.4 - I get garbled voice on Inbound (g711->iLBC). Outbound voice seems fine (iLBC->g711) though. It's not a 20/30ms framing issue as the phone uses 30ms

atlast after working of 7 hours i got voange soft account working on asterisk.

I have a connect to * via H.323/g711 from device A and want to connect to B which want for H.323/g729 h323.conf contains disallow=all allow=alaw allow=g729 but outgoing faststart/TCS contains only g711 (from h323_request(format) i think) and so no codec negotiation and no voice. Howto run up g711/H323 -> * -> g729/H323 PS intel's g729 was used. ast 1.0.3-6 PPS stupid -

We're having a strange problem with our receptionist. She runs an xpro softphone and we're using a queue to handle incoming calls. It seems nearly all of the calls that come in through the queue get dropped. At first we thought it might have been human error (clicking the wrong button in xpro or something) or that the person waiting in the queue just gave up and hungup, however it

Hi, I tested asterisk server with Xpro program, and all the function working well ( like 3 way calling, transfer.... ). But on the VOIP phone, I don't know press which key for 3 way calling function and transfer function... Can anybody teach me ? thanks

The scenario we have is fax send/recieve software that ONLY talks T38 and an asterisk box. We have ITSP providers that do NOT talk T38 but G711 only. Does asterisk have the capability to take the T38 call from an ATA or T38 software then bridge/transcode it and do G711 out to the PSTN providers? If not is there another product PAID or FREE software or hardware that can do this easily and

Hi, I?m trying to get app_festival to work. I got the source from the Debian woody package of festival-1.4.2 and applied the patch that came with * sources it applied fine; then I made the debian package and installed it. I have this on extensions.conf: exten => 6700,1,Festival(Hi there how are you doing ?) When I dial 6700 I hear nothing and then * hangups: -- Executing

Here is my Configuration PSTN -> Cisco AS5350 -> SIP -> ASTERISK -> SIP -> ATA186 When I call from the pstn to the ATA, the ATA rings but I don't hear anything on the calling side until the call is picked up. When I call from the ATA, everything seems to work fine. When I bypassed ASTERISK, everything seems to work fine. Anyone know what I might have configured wrong?

Hi, I tested asterisk server with Xpro program, and all the function working well ( like 3 way calling, transfer.... ). But on the VOIP phone, I don't know press which key for 3 way calling function and transfer function... Can anybody teach me ? thanks