similar to: Is there a problem with iaxtel?

[ Sorry, I incorrectly copied some Reference headers into this post and tacked it onto the wrong thread. -Steve ] So far, I have been able to receive incoming iaxtel calls via my assigned 1-700-xxx-xxxx number, but only when using md5 authentication in iax.conf: [iaxtel] type=user ; Incoming calls only context=incoming auth=md5 secret=<mysecret> ; Required for

Hi all, I'm trying to get IAX2 trunking between two * boxes and am having extreme difficulty :) What happens is when the sending * server (the one initiating the call) receives the ACCEPT back from the receiving server it immediately replies with INVAL. I've checked the code and it seems to be not matching the accept packet with the relevant item in the iaxs array due to the following

Hi All, I am having trouble with receiving calls from FWD via IAX. I know this isn't a FWD support forum, but I suspect the problem is my asterisk setup. The problem is that I can dial out to fwd subscribers, even myself but they can't dial me using my FWD number. I don't know much about IAX, but it would seem to me like a registration problem, but I get no errors or warnings in

I didn't get any response at all to my last "request for status" on IAXTEL. So, when this happens, I attribute it to one of a number of things: 1. No-one knows. 2. No-one cares. 3. Everyone knows, but are too busy to reply. At any rate, my investigative side kicks in and I began searching thru the digest's I've gotten, looking for references to IAXTEL. Mostly it is

Does anyone know if the 1-800 iaxtel gateway is down? I've been trying to use it all day today and asterisk says it's ringing: Channel (Context Extension Pri ) State Appl. Data IAX2[iaxtel]/1 ( s 1 ) Ringing AppDial (Outgoing Line) SIP/2201-a253 (home 18888476626 1 ) Ring Dial IAX2/XXX:YYYY@iaxtel.com/18888476626@iaxtel But I

I finally got around to signing up with Iaxtel and Free World Dialup... the price was right, as far as that goes! I've gotten and placed calls via FWD just fine. But I can't seem to get registered, or stay registered, with Iaxtel. My logs show the story; at startup I see: Jan 20 07:14:44 VERBOSE[14121]: -- Registered to '65.39.205.121', who sees us as ... yada yada... As

I've searched WIKI and Archives but nothing. Im getting: -- Called username:password@iaxtel.com/1800somenumber@iaxtel Jun 21 17:04:12 WARNING[1158883520]: chan_iax2.c:5097 socket_read: Call rejected by 69.73.19.178: Unable to negotiate codec -- Hungup 'IAX2[Iaxtel]/8' == No one is available to answer at this time -- Executing Hangup("SIP/104-b8eb", "")

<BLOCKQUOTE style="PADDING-LEFT: 8px; MARGIN-LEFT: 8px; BORDER-LEFT: blue 2px solid"> <DIV>Hello,</DIV> <DIV>&nbsp;can someone tell me what's wrong with this? I can't make toll free calls via iaxtel. Here's the definition in my extensions.conf</DIV> <DIV>&nbsp;</DIV><FONT size=2> <DIV>[iaxtel-trunks]</DIV>

I'll admit it. I'm a asterisk newbie (but no stranger to telephony). The setup is simple: two Grandstream BudgeTel 100 phones (SIPPhone specials) on a private segment calling to a Linux box acting as the segment's firewall with a leg on our public network. The phones are setup as SIP/phone1 (x1000) and SIP/phone2 (x1001), respectively (thanks to the Asterisk HOWTO). Getting IAX

Through my Asterisk server, I am trying to use IAXTel to dial 800-type numbers (yes, I know I can do the same thing with FWD and others via SIP, but I wanted to play with IAX a little). It appears I'm running into some sort of a codec mismatch or something because it's not working right. The SIP client is a SPA-3000. In iax.conf, I have something like the following: [General]

Hi All, Is this working for any of you? We're supposed to be able to dial FWD accounts through IAXTel by dialing 170099+FWD number. I don't ever seem to be able to do this even though I can dial 800 numbers through IAXTel. Michael -- Michael Graves mgraves@pixelpower.com Sr. Product Specialist www.pixelpower.com Pixel Power Inc.

Is anyone other than me having trouble dialing out via IAXTEL? I havn't changed my config files in weeks but seems that IAXTel calls (800 and FWD) stopped working in the past week sometime. Robert

I have set up FWD via IAX service. I have tested the IAX service with 613, echo test, and 612, saytime. It all works well. However when ringing a FWD user, I got this error all the time: Connected to Asterisk CVS-v1-0-02/01/05-09:34:45 currently running on chat (pid = 8282) chat*CLI> Verbosity is at least 3 -- Executing SetCallerID("SIP/1001-a1fb", ""David

I'm attempting to get two Asterisk 0.4.0 PBXes to communicate with one another using IAX/IAX2 trunks. I've managed to get a semi-functional NAT Firewall working as a PBX (with Asterisk running directly on the firewall itself), but there are issues with bind()ing to various interfaces which is causing outbound SIP issues. To get around these issues, the idea is to do something like

Hi, I've been setting up * recently and slowly getting to grips with it, however I'm getting rather confused with all the different configs for IAX calls, FWD calls iaxtel etc etc. What I think I need it a basic understanding or even a comparison of these different voip systems (if thats what they are?) I'd like to be able to make calls to other voip users, both in the UK and abroad

When somebody call me on my pstn # cable connected to my fxo card it does not work when I check my computer the following error shows Connected to Asterisk CVS-v1-0-12/05/04-19:46:25 currently running on asterisk1 (pid = 2160) Verbosity is atleast 3 -- Remote UNIX connection -- Starting simple switch on 'Zap/1-1' == Starting Zap/1-1 at incoming,s,1 failed so falling

Hi List, While trying to calling Toll Free numbers using IAXTEL, the call connects, I hear about 2 seconds of voice and then the voice drops off and I get the following error message which keeps scrolling across my console screen. WARNING[-167797840]: chan_iax2.c:5967 socket_read: Received mini frame before first full voice frame Asterisk shows that the format of this call is GSM. However if I

Hi, im trying to setup ASTCC but I'm getting it difficult. I've correctly set up the mysql database astcc and added a brand, trunk, route and a card as follows: brands +------+----------+------+--------------+------+--------+------+------+ | name | language | inc | publishednum | did | markup | days | fee | +------+----------+------+--------------+------+--------+------+------+ | FWD

I'm trying to make a call from an IAXPhone client - through the * PBX to an 888 number using the IAXTel link. I'm using the basic conf files for extensions and iax. I get successfully connected (at the "Attempting native bridge" line of the output) and am then able to talk both ways for 20 to 30 seconds and then the IAX phone appears off line. If I wait on the PSTN line for

hello, I have a test setup with 2 asterisk servers, each having a one snom 100 via sip using it. I`m experimenting on how trunking between them would work. I have them setup for RSA authentication which I plan to use in the future. So I`ve setup the keys and servers seem authenticate to each other. One is named phila and other hurricane. Here is what I see on phila: -- Registered