similar to: G729 on PS3 Cell

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, I am interested in optimizing some of R's vector math functions to utilize the SPE units of the Cell processor (commonly found in the Playstation 3) and I am wondering if anyone has already done any work in that area. I can't find anything using the search page or Google. (Admittedly it is difficult to search for information on a one-letter-named programming language whose

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

Joost Schuur wrote: > My primary concern is the use of the word 'unstable' on the current > download page for 1.2b2. One of our major devloper partners in > particular saw that reference and opted to use 1.0.5 on their PS3 title, > which is why we based our work for them on 1.0.5. The kind of commercial > game developers that are our customers aren't going to have the

Please correct me if I am wrong, Jean-Marc, but I do not think that any patches are getting applied to 1.0.5 anyway. Also, if you expect a patch to be applied, you will need to provide the changes as a patch, not as a modified copy of the source tree. The 1.2 branch includes a mechanism for private memory allocation from a static buffer. You provide a usermisc.h file that replaces the

Saad Nader wrote: > We needed to use a release version of the SDK because this part of > production code for our Sony implementations. I'm not sure how stable > that branch is. Is the encoding and decoding already stable? In other > words, what is not stable about that branch? It's listed as "unstable" in the sense of "I might change the API" and that is

Joost Schuur wrote: > Saad has been keeping me in the loop on your recent discussions. > Since all of our testing has been against 1.0.5, based on that being the > last non-beta version, that's the particular scope of the task that Saad > is working on right now. I'm just saying it's useless to do any work on 1.0.5. It is inferior to the latest beta in all aspects,

Hi Jean-Marc, We needed to use a release version of the SDK because this part of production code for our Sony implementations. I'm not sure how stable that branch is. Is the encoding and decoding already stable? In other words, what is not stable about that branch? Thanks, --------------------------- Saad Nader Middleware Engineer Powered By Gamespy, IGN -----Original Message----- From:

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

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 -

I've been trying to install R on a PS3 for several weeks now without success and have read various posts here, and elsewhere, on this and installing R on other PPC64 architecture machines which have given me some pointers. Yesterday, after much frustration, I finally succeeded and it runs well. As several people have posted questions about this here, and elsewhere on this, I thought I'd

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 |

Skipped content of type multipart/alternative-------------- next part -------------- A non-text attachment was scrubbed... Name: speex-1.0.5.zip Type: application/x-zip-compressed Size: 641376 bytes Desc: speex-1.0.5.zip Url : http://lists.xiph.org/pipermail/speex-dev/attachments/20071022/2447cf50/speex-1.0.5-0001.bin

Hi Jean-Marc, Jim, Saad has been keeping me in the loop on your recent discussions. Since all of our testing has been against 1.0.5, based on that being the last non-beta version, that's the particular scope of the task that Saad is working on right now. I like what I'm reading about as far as encoder/decoder quality improvements e.g. in the 1.2 betas, and am going to push for

Hi, On 10/26/07, Jean-Marc Valin <jean-marc.valin@usherbrooke.ca> wrote: > Joost Schuur wrote: > > My primary concern is the use of the word 'unstable' on the current > > download page for 1.2b2. One of our major devloper partners in > > particular saw that reference and opted to use 1.0.5 on their PS3 title, > > which is why we based our work for them

Hi Jim, Jim Crichton wrote: > Please correct me if I am wrong, Jean-Marc, but I do not think that > any patches are getting applied to 1.0.5 anyway. Also, if you expect > a patch to be applied, you will need to provide the changes as a > patch, not as a modified copy of the source tree. That's right. I won't be applying anything 1.0.5 unless it's a security issue or

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