similar to: Astricon talk on wideband codecs

I saw several threads requesting an Asterisk forum to complement the email list. i.e. http://lists.digium.com/pipermail/asterisk-dev/2004-February/003103.html I recently created an Asterisk forum within TMC's popular VoIP forums for everyone to use. http://voip-forum.tmcnet.com/voip-forum/forum/forum_topics.asp?FID=15

Hi Folks, Are all the astricon presentations up? I'm especially after the one that tilghman did. I caught the tail end of the prez when I decided to skip the session I was attending and go for that one. :)

Hi all, I have the below extension defined under sip.conf: [2203] type=friend username=2203 secret=123456 host=192.168.0.164 mailbox=2203 context=intern canreinvite=yes dtmfmode=rfc2833 When trying to register from a softphone installed on a PC behind a nat with IP=192.168.0.164, I got 503 FOrbidden...Does anyone have any idea about what could be the issue? Regards -------------- next part

This is a fairly large list of hardware phones, features, and URLs, with some vendors I've never heard of before on it. I suspect that any phone that doesn't explicitly say "SIP" is an H.323 phone. http://www.tmcnet.com/it/0403/0403sg.htm#sidebar2 JT

Asterisk receives TMC Labs Internet Telephony Innovation Award http://www.tmcnet.com/it/0704/tmclabs.htm Jim James H. Thompson jht@lava.net -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040708/133dbbde/attachment.htm

Hello. I am new to this forum. If this issue has been discussed already, please inform me, but my STFW have indicated that it has not. If SS7 were available on * would it make commercial sense to use it? There are case studies on the web (courtesy of the Death Star :->) for ISPs with 10,000+ subscriber lines showing clear cost savings in transitioning from PRI to SS7 controlled IMT. My

Since it is Saturday evening (7PM EST) I am asking this on the list in case someone who knows sees it and can answer. Astricon is in my back yard for the first time, and I could hit you with a rock. I would always like to attend, and spoke at the 2007 Astricon in Phoenix but don't have the idle cycles. Question: Can I just go to Astricon and take the dCAP exam only? In and out? Cost? I

Starting around 10:00 AM EST. All services from them whether I connect by IP or DNS (both east coast and west). Anyone else? Fred Posner fred at teamforrest.com Main: +1 (212) 937-7844 Direct: +1 (503) 914-0999 www.teamforrest.com -------------- next part -------------- An HTML attachment was scrubbed... URL:

http://www.tmcnet.com/tmcnet/articles/2004/031704rt.htm Previous article by same author: http://www.tmcnet.com/it/0104/0104PO.htm

Is there any tool that can do LCR for Asterisk but also take into account MOS scores? Is it possible to automatically generate MOS scores on random "calls" so as to keep an updated database on a per provider, per destination, per time-of-day score? Hopefully, with that information we can create a better LCR module or script? Thanks, Daniel

Just a "heads-up" ... my home asterisk server is being flooded by someone from IP 184.73.17.150 which is an Amazon EC2 instance by the looks of it - they're trying to send SIP subscribes to one account - and they're flooding the requests in - it's averaging some 600Kbits/sec of incoming UDP data or about 200 a second )-: This is much worse than anything else I've

Hi All; How to increase the waiting time between entering the digits for the analoge phone device that is connected to fxs? Is it by DigitTimeout? But how it will be apply for analoge station if the user just pickup the handset and dialed the number? Any help? Regards Bilal

Steve Kann has developed a new jitterbuffer for IAX2, that hopefully will be integrated into Asterisk v1.1 soon, to be part of the 1.2 stable relase. Zoa and his bulgarian team is porting this buffer to SIP/RTP, but needs support in the form of funding in order to take the time to test this out and complete it in time. Please paypal your contribution to sponsor@astertest.com today. Every

I'm coming to Paris Dec 19. I was wondering if there was any interest in having an Asterisk get together in Paris sometime near there. Any one out there interested? Anyone in Paris who could help organize something like that? :) Mark

I have read that an IAX trunk requires a timing device. What wasn't clear to me was whether it is like TDM ie 1 timing device for the trunk, or if each end requires a timing device. I have a zaptel card in one server; do I have to have one in the second server in order to do an IAX trunk? I set up a trunk and so far calls can be made one way, but not the other. It is probably just not

This year we recorded quite a few of the AstriCon sessions - 3 out of the 4 tracks were video taped. The folks at TMC then went through a fairly painstaking process of synchronizing the video presentations to the slide decks that each presenter provided, so we have an index-able and fast-forward-able version of each talk. I'm really excited about this video presentation method,

We just uploaded the latest and greatest version of the idefisk iax2 softphone, version 1.24 Freely downloadable at: http://www.asteriskguru.com/tools/idefisk_beta.php Changes since the last release include: - history panel is working - receiving messages and urls (sendtext command in asterisk) - some bugfixes (the annoying hangup bug is finally gone!). A big thanks to everybody who sent us

List, We're working on an upcoming job that may require us to access a web service (WS). I'm curious to hear peoples thoughts on the best way to do this with asterisk. We'll be submitting a single number to the WS and it will return a success or error. One solution would be to write a simple perl script to interface into to the WS, and use SYSTEM() from asterisk to call it.

Hello, How many licenses to buy?? : From what we understood from digium website, we must buy as many licenses as the number of maximum simultaneous calls using G729 Codec we wish to make. For example, If we want to be able to make a maximum of 10 simultaneous calls using G729 Codec, we must buy 10 licenses. Is it right? Thanks you

I know Vonage doesn't officially have a "bring your own device" type program, but they do offer a softphone. Has anyone gotten Asterisk to connect directly to Vonage? This would be a great help!! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060329/5bc9f644/attachment.htm