similar to: Choice of soundfile format

Hello List Does anybody have any experience with the FoneBridge line of products from RedFone? I think their HA implementation sounds interesting, and like the prospect of having dedicated hardware for our PRI connections. Kind Regards Jon Leren Sch?pzinsky

Hello List I am having a rather big problem with a sangoma A104 card, I just installed to replace a Digium TE410 card, that was acting up. But now we have a problem with the sangoma card. It runs great after being started, and calls proceed as normal, but after about 1 hour, it stops being able to make and receive calls. If I run wanpipemon debug, can see that the card still receives

Hello List Just want to check if anybody else is having this problem. Every time the PRI connections are disconnected, the card freezes, and I have to reload the driver, to make it work again. We are very seriously considering switching to Sangoma at this moment, due to this and other problems, but I want to know if there is a solution, and to make sure it isn't asterisk that's freezing

Hello List. We are having some problems using t.38 together with a Cisco voice router at one of our providers end. We are using the new digium asterisk fax module to generate the fax, and when we use together with our internal Audiocodes Mediant 2000 gateways, we have no issues what so ever, and the faxes go right through. When we send faxes to our other provider, who has cisco hardware

We tried them out early last year when we were looking at a large deployment and they gave us a lot of the redundancy that we wanted. However we did run into issues where calls seemed to get caught up in the system. It was as far as we could tell rather random. No consistency to it at all. Asterisk hung up the call but the telco side of the line didn't actually hang up. The channel was left

Hello List. I just got my new PS3 yesterday, and first thing I did was of course to install Linux, and then compile asterisk, and it worked without any problems. My question is this... Is anybody looking into using the Cell processor for G729 enc/dec? Using the 6 SPE processing units available, you should be able to enc/dec a whole lot of channels at one time. Looking at the

Asterisk Project Security Advisory - AST-2008-012 +------------------------------------------------------------------------+ | Product | Asterisk | |----------------------+-------------------------------------------------| | Summary | Remote crash vulnerability in IAX2 |

Hello I am currently having a problem, that threatens to drive me insane... I cannot understand how Asterisk matches up a sip request with a peer. Here is my example: INVITE sip:87654321@192.168.100.4 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.59:5060;branch=z9hG4bK00d97b99;rport From: "1088200336" <sip:1088200336@192.168.100.59>;tag=as54af3e4d To: <sip:87654321@192.168.100.4>

Asterisk Project Security Advisory - AST-2008-012 +------------------------------------------------------------------------+ | Product | Asterisk | |----------------------+-------------------------------------------------| | Summary | Remote crash vulnerability in IAX2 |

Hello List We are trying to redirect calls directly, instead of opening a new channel and dialing out. Etc: A calls B on our asterisk, and is directly redirected to C We have been told that this feature should be available on a PRI level, and is called Partial re-routing. Anybody has an idea of whether this is supported in Asterisk? Kind Regards Jon Sch?pzinsky Detele. No virus found in

That's just the thing, and it sucks, because the VoIP implementation actually works very good. Jon _____ Fra: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] P? vegne af list mail Sendt: 8. juni 2006 02:34 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: Re: SV: [Asterisk-Users] I can hear only one way when I use

I'm trying to play a wav-file on a channel. This is what I see in the asterisk debug console AGI Rx << STREAM FILE "test.wav" "12345" [Aug 20 16:10:19] WARNING[25219]: file.c:602 ast_openstream_full: File test.wav does not exist in any format So it doesn't find the file, or it's in a wrong format? I can listen to it with windows media player... it's a

Hello Olivier Ive been testing the E61 phone for some days now, and we need to have an inhouse asterisk server, connected to our main asterisk server, to get it to work. That means, that you cant just walk down to your local airport, and use the IP part of the phone on their network. You have to have a non nat local server, to get it to run. Other than that, the phone can accept calls both

Can you then inform me on what structures this information is stored in, in the asterisk code? Then ill try to do a quick dirty version of the replication. Jon -----Oprindelig meddelelse----- Fra: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] P? vegne af Kevin P. Fleming Sendt: 9. juni 2006 16:25 Til: Asterisk Users Mailing List - Non-Commercial

Hello list, is it possible with the monitor-command to record conversations and place the soundfile in a pre-defined directory per user ?! So when extension 200 presses '*#' to record the conversation, the resulting sound file is written to his home directory on the Samba-server. This way each user has his own directory with its recordings that no one else can access (as default rights

Hi Guys, I want to playback a sound file stored in mysql database in my perl script............pls can anyone help with an idea? response would be greatly appreciated Rgds _________________________________________________________________ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/

Hey, I've coded an OGG player for Win32 (it uses AL for playback so it's portable to Linux/Mac), but every time the program gets to the 'ov_open()' function, the app completely freezes, and I have to use the task-manager to kill it. I am supplying it with a valid file handle that was just opened (FILE*) and the vorbis file is also a pointer that is not in use (set to null). Any

Thank you, Robert! The announcementfile plays well, but at Dial-option "m" i have to specify a MoH class, that is something i cannot use (as i wrote in my post). Background command waits for a user input, but the caller should be connected to SIP Phone 100 after it has answered and the announcement has been played. Before connecting to SIP Phone 100 the caller should hear a

When I use option t and m together in the same dial statement the music on hold doesn't appear to work. Is this a normal operation? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040203/bfdd6806/attachment.htm

How can I copy all the contenent of the asterisk database to another machine? I want copy all the active sessions from one asterisk@home to another one and running on the second(this I can do using vrrp protocol, it isn't a problem), I want copy only all the active sessions and softphone registrations to another asterisk@home and then run on it. -------------- next part -------------- An HTML