similar to: Problems with Digium TE410

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

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

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

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

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

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 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

Hello What soundfile format, is the one that uses least transcoding during playback? As I can see, I can choose wav or gsm. What sucks least cpu power, during playback to example a Zap channel? I would guess wav, but is this correct? Kind Regards Jon Leren Sch?pzinsky -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.408 / Virus Database: 268.13.11/496 -

Hi all. We are getting our first PRI line to use with Asterisk and one of the technical specifications is about framing, structured or unstructured. The main difference about them is almost clear for me: http://ckp.made-it.com/g704.html says: "G.704 is the framing specification for G.703. A carrier can 'steal' a 64kbps time slot (TS0) from a 2.048 Mbps line and use this to

worked came to one ring only now. Thank you very much. If I use TE410 or TE405 instead of X100P. do it make that first ring disappear? Shakil -----Original Message----- From: tony@softins.clara.co.uk [mailto:tony@softins.clara.co.uk] Sent: Tuesday, April 20, 2004 12:27 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Re: Auto Answering PSTN --> Asterisk using X100PCard In

Hello I can save you a lot of time, and tell you that it wont work. It does hold some registration information in the asterisk database, but most of the information is kept internally in Asterisk. Just FYI. Jon _____ Fra: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] P? vegne af Shenen Shenen Sendt: 9. juni 2006 11:37 Til:

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, I would like to have some comparison between E1 cards from Digium and those from Eicon for a VOIP - ISDN Gateway. How does they compare on the echo cancel point of view? Is the echocancellation code for E400 good enough for production environment? Best regards, Daniel -- Daniel ANDRE (mailto:dandre@iris-tech.fr) IRIS Technologies - http://www.iris-tech.com Serveur kwartz -

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

I just noticed that the new TE410 only supports 3.3v PCI. How do you know if it will work in your machine? It's not something commonly listed in your motherboard specs.... I was thinking about getting one for an Athlon 750 that is a spare box I'm doing testing on... (Original SlotA) -G

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

Hello Is it correct that IAX2 uses more CPU, than SIP? Also, can it be true that IAX2 is much more sensitive against high CPU loads? Also, does Asterisk support and use multiprocessor architectures, such as Xeon? ? Regards Jon -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.8.3/362 - Release Date: 12-06-2006