similar to: Speex and bandwidth usage on Asterisk's IAX

Hi, We are using MixMonitor to record the call. When the call is bridged, the latency is significant. We tried to increase the internet speed and the server RAM and processor speed and still we are having that issue. We use VoiceTrading and Gafachi's Termination minutes to make calls. As we are in US and VoiceTrading in Europe, somebody suggested to move the termination minute provider

The Asterisk Development Team has announced the release of Asterisk 13.15.0-rc1. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.15.0-rc1 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release:

The Asterisk Development Team has announced the release of Asterisk 14.4.0-rc1. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 14.4.0-rc1 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release:

Has anyone done a modification where you can Interrupt Asterisk's SayDigits(). This will be helpful in order to be able to interrupt an announce and dial digits without waiting to hear all the announcements. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081009/06185059/attachment.htm

Folks, everything's in the subject, i've got a Linux Box with a Digium E100P E1 Card, modules are loaded, but I don't know which signalling to put in my zapata.conf... Thanks for you help. -- Nicolas Cartron nc@ncartron.org

hello, can anyone installing/configuring asterisk's on SLES9 if someone can share his/her views experiences . Thanks In Advance.

Hi list, Is this supposed to be a joke? It doesn't sound very professional. comedian n 1: a professional performer who tells jokes and performs comical acts Moby Thesaurus words for "comedian": banana, buffoon, burlesquer, card, caricaturist, choreographer, clown, comedienne, comic, cutup, dramatist, dramatizer, dramaturge, droll, epigrammatist,

Hi guys, I have a linux box. It has two network cards. I configured it as a router. It uses its WAN to register onto its own sip proxy and its clients use its LAN to register onto it. Whenever it cannot register through its WAN its sip clients(extentions) will logon failed. That is, if asterisk cannot be a sip client it cannot be a sip server, either. I want it to be a working sip server

Hello Everybody, I wished to know that the .c files in Asterisk in the /usr/src/asterisk directory, can they be modified to change the behavior of Asterisk? If yes, could you please tell me as how does one go about modifying the code and also which are the files that are modified? It would be very kind of you if could mail me a sample file with the changes made and the

Type "swift" at the command line so you can see the -options. Then modify the line to use the correct switch and specify the name of the voice you want to use. Thanks, Steve -----Original Message----- From: Pimjai Wesnarat [mailto:pw@nummerndirekt.de] Sent: Fri 4/21/2006 6:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: [Asterisk-Users] How

What i need is to recieve a call in a console! I mean i can call from CLI...but can i recieve calls too? If this is possible how is the console identificated and where! Actually i need to call from one Asterisc server console to another(i know what is asterisc server for, but this is a specific task)! Thanks! --------------------------------- Don't pick lemons. See all the new 2007 cars

Hello, I tried to configure a very simple case of Asterisk using SIP userA --- Asterisk server ---- userB sip.conf [userA] type=friend username=userA host=dynamic nat=no context=test [userB] type=friend username=userB host=dynamic nat=no context=test In extensions.conf [test] exten => 1000,1,Dial(SIP/userA) exten => 2000,1,Dial(SIP/userB) I make a call from userA to userB, it works,

Hi, all For some work i'am trying to connect to Asterisk's PBX using isdn cards. But I don't know anything about it. So i'll be pleased to get some information about it! Thanks, Best rgrds!!

Hi, Am running a freepbx install and created trunks, extensions and groups. Now I'd like to hand out the Asterisk phonebook to the phones (all VVX 310's). Is there an easy way to do this? Best, Stanley

hello everyone. I am running plain asterisk and I am using asterisk's internal database for: -phonebook -blacklist numbers instead of having to update the database of new entry or delete an entry, is it possible to have it in an external file such as txt? so every new entry/deletion will take place there. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, I want to make initial VoIP authentication process from asterisk server to be based on EAP-SIM authentication of Freeradius server (so it will be not necessary to insert account datas in the asterisk database). Is there any way of doing that from Freeradius and Asterisk? Or at least, is there any way to sync the EAP-SIM data on Freeradius to asterisk server? thank you -------------- next

So I think that that final result is pretty amazing. That's 35% less bandwidth usage than G.729. Even without the preprocessors help you do 17% better. Regards, Steve Davies _______________________________________________ Speex-dev mailing list Speex-dev@xiph.org http://lists.xiph.org/mailman/listinfo/speex-dev

Hello everyone, After several years of using Asterisk I have always been frustrated by the support for DNS. I have seen all kinds of strange behavior when Asterisk is used on a system with "iffy" DNS servers: - no failover to other DNS servers in /etc/resolv.conf (might be a C library thing) - chan_sip will sometimes mark even local SIP peers as unreachable during/after any DNS

I'm monitoring Asterisk with Nagios. Nagios constantly alerts because of too many zombie processes. I eventually had to disable the notification for the alert but why does Asterisk create so many zombie processes, I've see more than 30 at times and it generally stays in the 20s... just seems unusual and wondering if it's harmful, thanks in advance. -------------- next part

Hi, I'm using Cepstral as a TTS Engine for Asterisk with Swift application. It works fine when I have just 1 voice installed. Now I have 2 voices in the same language installed but I can't seem to find the way to select which voice to use in Swift's application in Asterisk. Does anyone know?? Thank you, Pim