similar to: Can audio streams go client to cleint with IAX?

Someone submitted a bug about wav49 voicemail problems with the Windows Media Player here http://bugs.digium.com/bug_view_page.php?bug_id=0000254 bkw918 changed the status of the bug to resolved because he could not reproduce the error with his version of Windows Media Player. I am having the same problem as the original bug poster. I am using WMP 9.00.00.3075 running on Windows XP and using

Okay, I've read as much as I can, and I think i've followed instructions, but I'm still having problems with * and firefly... I can get outgoing to other freshtel working, but not incoming (I get the "not available" voicemail), or outgoing to landline. I'm using the debian asterisk package (0.9.1-RC1-4) My iax.conf has in general (under my FWD register, which

Hi, I'm having some problems getting calls to go out via freshtel. There dosn't seem to be any specific information on how to get it working anywhere. The only information I've found is here: http://www.voip-info.org/wiki-Freshtel and that dosn't give you any idea of how to actually get it working. I've tried adapting information from other IAX2 provider examples but have

Ladies and Gents.... Please be patient as I try to explain what I am trying to achieve.. I have a PSTN line and a Freshtel account, what I want to do is have the PSTN line as the first choice for outgoing calls for local calls and Freshtel as the second choice. The problem is that it's easy enough to set up both individually but how do I get the "second choice" drop the leading

I found version 2.0.0 of Firefly on the Freshtel site, but it only has the network setup options for the Freshtel network, despite the final statement on the page http://www.freshtel.net/firefly/download/ that says: ----------------- Standalone SIP / IAX mode: If you want to use Firefly on our network (with your own voicemail etc.) you will need to register a Firefly number. However, you can

Ok I have spent the last week working on getting my small PBX to work. I will in the end only have 4 SIP extensions being either softphones of IP phones. Currently only 1 SIP config for testing. And at the this point it should be all fairly easy with all inbound and outbound to PSTN will be going Via Firefly/Freshtel.net in Australia via IAX. Inbound does work in it's current basic state.

Hi, I'm looking at connecting an analog fax to asterisk via an FXO card. The plan is to send faxes thru freshtel. Has anyone done faxing with freshtel? Cheers, -Shaun

Hi All, I am getting this error during make. please help me./ speexec.c: In function `speex_ec_process': speexec.c:112: syntax error before "noise" cc1: warnings being treated as errors speexec.c:133: warning: implicit declaration of function `speex_echo_state_reset' speexec.c:148: warning: passing arg 5 of `speex_echo_cancel' makes pointer from integer without a cast

Well here is something simple, well I think it is for the smarty's out there :) I got a connection to freshtel and want to get the iax working. I have config'ed up iax.conf with the register line and get in return in the cli> -- Registered to '202.168.7.130', who sees us as 203.29.98.221:4569 So that appears to be connected. When I call the DID number I get the Voicemail

Ok I have an * server finally setup and acepting calls from freshtel and I am VERY impressed at how well the Freshtel.net service works but thats another subject :) I have it all setup so that I can Dial my DID number on freshtel and that gets set to my * via IAX. At the moment I have the demo configured so that I can test it all and make sure it is all working. The problem is that I

Why don't you just try it? From what others have been reporting, it shouldn't take you long to get it running. You can use speexenc and speexdec for testing. On Nov 12, 2008, at 2:26, "Vincent Burel" <vincent.burel at vb-audio.com> wrote: > ok, thanks for these precision, and do you have some measure about > CPU load > ? > i really would like to get a

Hello All, I have been looking for a commercial version of an IAX2 Softphone for Windows but the ones I have came across (i.e. Iaxcomm, Iaxphone, Diax) do not seem to have an updated version since April 2004 in some cases. We looked at Firefly but we sent emails to Virbiage/Freshtel with questions and could never get a response from them. Has anyone got any recommendations for commercial

Okay, you've answered part of my question, which is "What value equals silence?". I assume then that a (decoded) frame of silence would be a buffer the size of my frame (320 bytes) full of 0's. Passing this frame (a frame of all 0's) through the encoder causes it to blowup though.. In response to your answer below, I don't think I want to overwrite the decoded

Ok I have a challange that I can't seem to find a way to fix it. My Voicemail in * timesout after 30secs without fail everytime no matter what I do. I have incomming calls comming in through Freshtel IAX2, if it goes to SIP extension when it is online it can hang on for what ever time the call goes for. If however it goes to the Voicemail it will timeout at 30sec and I can't seem to

Disclaimer: Not trolling. Cross-posting to -users to gague support. -users : Straw poll - if an XML based Manager Interface was avaliable as an option in asterisk.conf, would that be a good thing, or a stupid thing? >Have you ever tried initiating a session via XML with a terminal that >doesn't support backspace... I'm actually proposing that an XML I/F be avaliable as an option

Hi All, I am getting this error during make. please help me./ speexec.c: In function `speex_ec_process': speexec.c:112: syntax error before "noise" cc1: warnings being treated as errors speexec.c:133: warning: implicit declaration of function `speex_echo_state_reset' speexec.c:148: warning: passing arg 5 of `speex_echo_cancel' makes pointer from integer without a cast

Hi, Most companies we work with, have 'designated' crisis management teams. These vary from the insignificant crisis', through to life-threatening crisis'. There is always an assigned emergency services contact, whose job it is in an emergency, to maintain communication with the emergency services. One of our corporate functions is crisis management - so we have to consider

Hi all, Does anybody know any USB phone that I can use as an IAX Client? Thanks. Ronaldo.

I agree with Brian! Robert's post is off topic or may be just a marketing effort, to push their site. Anyone who wants freshtel.net for US/Canada calling at 6.9 Cents a minute, raise their hands? ... I see none Seshu -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Brian Capouch Sent: Wednesday, July

Hi, I've got two H.323 Client connected to Asterisk, when one of them requests boeing connected to the other I use CALL application and both get in touch trhough asterisk, but using Call Asterisk stays on the middle and the sound quality gets poor. Is there any way to 'transfer' the call so Asterisk doesn't stay in the middle ?? I use OpenH323GK (www.gnugk.org