similar to: Faxing thru freshtel

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

Hi All, I realise this is off topic, but its likely the best place to ask! I sent an email to snom support a few days ago but have yet to recieve a response.. Perhaps some one has found a solution to this problem already? I've searched the mailing lists and google and found nothing useful. I've also read Snom's mass deployment documentation but thats no real help in this case.

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

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

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

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

Hi, Has anyone else experienced problems as of the last couple of months when outbound calling through Freshtel? I've started getting a "No authority found" error. I've tried contacting them, and they seem to have some serious communication issues with their IT team, infact I think they have serious issues in their IT team full stop. First they can't find my account in

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 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

With a service like http://www.freshtel.net/?show=home that uses IAX and has servers in Australia, is it possible for the audio streams to take a different path than the call setup and control? In other words can it work like SIP with canreinvite where the two endpoint negotiate audio streams between themselves rather than though the FreshTel server? Thanks

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

What are your experiences with faxing through Asterisk to the PSTN? We are using g.711u as a codec, and are originating/terminating with Broadvox as well as through our own PSTN gateways. We have had some luck with incoming faxes coming into our network from Broadvox DIDs. They work 50% of the time. Not sure yet on PSTN incoming since nobody that is using FAX is in our local rate centers.

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

I have something going on that I don't fully understand after a weekend of looking for answers. I have an iax account with Tesco that works flawlessly with the Zoiper client - but is giving me trouble with inbound calls in Asterisk 1.6. After some playing I have ended up with an iax.conf file that looks like this: [general] calltokenoptional = 77.75.0.0/255.255.248.0 maxcallnumbers = 16382

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

Greetings! I have Asterisk 1.4.25.1 running as a Xen virtual machine using 64-bit domU and Dom0 with faxing and was curious about other users' experiences with it. So far receiving fax works just fine, but I'm curious if anyone else is doing international faxing with asterisk. If yes, could you please post your experiences? Were there any issues you may have encountered that I

I have been trying to get faxing working with stable but I have had no luck since cvs 1.0.4. I've tried 3 versions of SpanDSP and the system answers the fax but looks like it isn't training properly. Lee -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Bob Goddard Sent: 06 August 2005 23:44 To: