similar to: Re: PLEASE respond: how to get Asterisk to change coders on RTP handoff?? HELLO???

I have a hardware FXO/FXS which handle my voip calls, and they support G723 internally. Asterisk hands off these calls just fine, and everything works, as long as I don't want PBX menues available... The problem is, once I want it to return messages, it will only return them as GSM... which is fine, since my FXO/FXS support multiple coders. However, even though Asterisk lets me specify a

We have an external FXO/FXS, and use Asterisk as a call router. We want to use G723 for the actual phone calls, because we have limited bandwidth on our return direction. This has been working fine so far. However, now we want to set up Asterisk to handle PBX menues and accept extentions. Asterisk, of course, uses GSM for its messages, and cannot terminate G723 calls. So I want to tell

Hi, I've been playing round with Asterisk on Redhat 9 (2.4 Kernel) and was experiencing bad echo problems using Budgetone 100's when calling analogue lines in uk (Isdn4Linux / Digi Datafire). Calls to other ISDN and mobile network seemed ok although not much testing done. I've tried installing Asterisk on a faster processor (P4 3.0 GHZ) with a 2.6.8-24 kernel to see if that helps.

Sorry if this has already been posted and I have missed it. This is a great little reference I found that even veteran programmers can find useful. http://www.rubyinside.com/19-rails-tricks-most-rails-coders-dont- know-131.html Sunny

Devlounge is looking for advanced RoR coders who may be willing to share their knowledge by writting quality articles on Ruby and what it can accomplish. These can be tutorials, discussions, opinion pieces, getting started guides, etc. You''re articles will be seen by 350-500 unique people daily! Plus, you''ll get a great link back in our killer authors index. To apply to be

I haven't looked at the code so forgive me if this is already what it's doing.. Considering the problems with the current metadata implementation.. couldn't it be changed to just cache the data that comes from the source in order to update it if it changes, and then just send it down to the listening clients then in the form of an ID3V2 tag when they first connect, and then only

I read in todays (July 9, 1998) edition of the Linux Weekly News that "the Samba team could use coders, documenters and packagers as a result of a temporary dip in the volunteer resources available to them (caused by an unfortunate set of co-incidences, though we haven't heard the details yet)." and "This has slowed the Samba effort a bit." The full text of this news

I went to the Linux Solutions exhibition in Paris yesterday, visited the well stocked O'Reilly stand and saw a nice pile of Asterisk TFOT, 6 hours later there was only one left. It must say something, also it was the English version. -- Dave Cotton <dcotton@linuxautrement.com>

One thing I forgot to mention: I also cropped the registration and cut marks off the sides of the pages. If you want the uncropped version, get: http://www.alexburke.ca/asterisk-tfot-uncropped.pdf Sorry about the excessive noise, but I figured I should mention this. >Date: Mon, 20 Feb 2006 18:55:50 -0500 >To: asterisk-users@lists.digium.com >From: Alexander Burke

Hi, I have licensed the digium G.729A codec. But for some reason incoming and outgoing calls will ALWAYS use G.711a. When I force my phone to only accept G.729 then an incoming call from ZAP goes straight to my voicemailbox as the phone doesn't accept the codec Asterisk wants, even if I force it in sip.conf. Is there anywhere else I need to look to tell ZAP to use G.729 preferrably ? The

Hi all, I'm looking at some suggestions from you techies out there. Let me explain my scenario. Im a reseller to callshops. I need to take around 100 concurrent calls. Almost all endpoints are sending G723 codec and my peers take G729. Can anyone recommend the Server Specs that is ideal for this scenario. Im planning to lease a server. Calls are purely SIP or IAX2 only. Thanks in advance.

Hi, I am trying to replace my Cisco 5300 gateway with my new Zap TE405P card. Incoming calls and outgoing calls between my cisco and my SIP phone works fine on G.729. Recording messages in the asterisk voice-mailbox also works fine from both my SIP phone as well as PSTN -> Cisco -> Asterisk. I have licensed the digium G.729A codec. When I connect my ISDN PRI to my Zap card and I call

I'm hoping the members can guide me to someone looking for coding work in R. I'm running an on-line survey service bureau where we do 360-feedback (data is collected about a person's leadership/work behaviors from those all-around him/her 360-degrees). We've built the front end admin section in php and the survey builder is PHPSurveyor. I'm thinking R might be a good choice for

First encounter with * Just downloaded & installed aah-2.7 Started up AMP, but i can not find any reference towards isdn. I presume there has to be some configuration done for my Eicon-Diva-pro. Does aah actually support isdn-bri? On the mail-archive i found some references, but these are rather old ( they speak about the coming release of aah-2.1) aah-handbook (version 1.6) doesn't

Regarding my earlier post about labels and the 'n' priority: The TFOT book covers the use of these. See the box on page 81 entitled "Unnumbered Priorities." http://www.nufone.net/downloads/asteriskdocs/AsteriskTFOT.zip -MC

I'm using ezstream-0.1.2 KJ -----Oorspronkelijk bericht----- Van: Joel Ebel [mailto:jbebel@ncsu.edu] Verzonden: donderdag 3 februari 2005 21:05 Aan: Klaas Jan Wierenga Onderwerp: Re: [Icecast] Stream drops during handoff. Suggestions? Thanks. I'll have to try that. I wonder why ezstream would ever stop sending data for that long though. What version of ezstream are you running? Joel

Hello Asterisk users, Can someone thwack me with a clue stick please? I am following the Asterisk TFOT book Dial() example trying to get the limit and announcements to work as per below. These settings seem to have no effect. There are no warning messages after 4 minutes or every 30 secs thereafter and the call lasts longer than 5 minutes. gunner*CLI> show dialplan [ Context

How do I do this 1) ZAP-> * -> IAX(1) ------> IAX(2) -----> DG104S ------> Handset 2) No Answer on Handset 3) Back to IAX(1) 4) IAX(1) tries a cell phone 5) Still no Answer 6) Local * Voicemail. I have 1 working, and I had 4 working when there was only one box, i.e. when the handset did not answer the DG, asterisk went to the next step. Now that I have step 1 going to another

Sorry if this has been asked before, but I've searched high and low for the last couple days for an answer. An Internet radio station I DJ for is using Shoutcast and MP3, but we are considering moving to an Icecast/Ogg Vorbis combination instead. We work in 3-hour shifts. When we hand off, the DJ on-stream stops teh encoder, shouts "go" on IRC, and the DJ in line starts his

Hey All, I am setting up a network of Asterisk servers using IAX2. I am wondering if it is possible to disable the handoff feature? At the moment I have 4 asterisk machines, 3 are at SOHO offices and 1 is centrally hosted in a data centre. In addition the central machine has an IAX2 link to a VOIP provider (and might be set up with more in the future). All calls are routed through that