similar to: iaxtel down?

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

Hi all, I spent most of the last weekend testing and trying to diagnose some mostly incoming call issues. I'll start with the easy one in the hopes it might have a positive impact on the others. First- I have an account with IAXTel. I can place calls to other IAXTel subscribers and also through IAXTel to landline toll free numbers and all works great. iax2 show registry shows I am

Hi there I am using an IAX2 softphone built from the IaxClient library dialing into Meetme conferences. The IaxClient seems to use silence suppression, and not sure if this can be disabled. The client works fine through most routers, but for some it disconnects the client after about 5 minutes and gives this error in the asterisk logs: Chan_iax2.c:1480 attempt_transmit: Max retries exceeded to

G'day list, I am getting a lot[1] of traffic on my Internet link, ICMP messages from 69.73.19.178 telling me UDP port 5036 is unreachable (this IP address belongs to iaxtel.org). I see from the wiki that IAXtel supports only IAX2 from December 2003. Fine, however it looks like my * still wants to try and register using IAX1, and I can't find how to turn this off. This situation is

Hi guys, I am trying to use DISA. The scenario is - I call my home number (where X100P seats) from mobile phone, enter the password, enter international number and get connected via voiptel. It works perfectly when I call extension setup with DISA from X-PRO SIP phone, but when I dial into Zap, It seems that it does not detect DTMF tones. Here is a log and config files Please help

Hi We've recently set up and are using with success 1.0.7 using a Junghanns quadbri card to BRI ISDN, and Firefly with IAX2 protocol as softphones Works very well, however we're getting cases where sometimes the call just drops. >From setting more verbose modes we get a log which is shown below. The problem seems to be the maxretries message which comes from chan_iax2. We are using

Hi All I'm trying to get two Asterisk servers to talk to each other using IAX(2). I've read the WiKi and the docs and tried the examples..... I can't get it to work (I have 2 x 7960's registering on one server and 1 x 7960 registering on the other). I've set them up as follows... The two servers are set up as friends and have consecutive IP address's. The setup is

I created an account at IAXTEL.com to route 1-700-XXX-XXXX calls through. IAXTEL.com gave me a number (example) of 700-555-6226. I have made the following changes to my: /etc/asterisk/extensions.conf: [iaxtel700] exten => _81700XXXXXXX,1,Dial(IAX2/myusername:mypassword@iaxtel.com/${EXTEN:1}) exten => _81800NXXXXXX,1,Dial(IAX2/myusername:mypassword@iaxtel.com/${EXTEN:1})

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

I am somewhat unsure as to the definition of "Distinctive Ring". What I am trying to achieve is to have Zap connected phones (TDM400P) ring with different cadences depending on whether the call is incoming on the PSTN context or an IAX2 context. Googling, I find this from Mark: I've added distinctive ring support to Asterisk now (also I've added answer confirmation which is

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", "")

When I try to call iaxtel it goes to codec ADPCM even though I have define in iax.conf gsm Call accepted by 69.73.19.178 (format ADPCM) -- Format for call is ADPCM My settings: [general] port=4569 register => xxxx:xxxx@iaxtel.com bandwidth=high jitterbuffer=no tos=lowdelay [voipjet] type=peer host= xxx.xxx.xxx.xx secret= xxx auth=md5 notransfer=yes context=incoming disallow=all ;

I've not been able to register with iaxtel.com for the last couple of days. Is anyone else seeing this, or did I miss something?

I tried to connect my * to IAXtel, but i always get this errors. chan_iax2.c:5849 socket_read: Registration of 'mnetwork' rejected: Registration Refused On dial a iax number i get: chan_iax2.c:5526 socket_read: Call rejected by 69.73.19.178: No authority found chan_iax2.c:5528 socket_read: Immediately destroying 3, having received reject chan_iax2.c:2411 iax2_hangup: We're hanging

Hi list, i tried to write my first type and provider that should create logical volumes. Seems like i''m missing something as i get nothing when i use it: No errors and no logical volume :-( type/logicalvolume.rb: ================= Puppet::Type.newtype(:logicalvolume) do @doc = "Manage logical volumes" ensurable newparam(:lvname) do desc "The logcal

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

Hi all I've been trying to make * work with IAXtel to no avail, all seems ok in the config but am not getting anywhere This is what I'm getting from console (user/pass/dest # changed for obvious reasons): DEBUG[1133735216]: File chan_sip.c, Line 3841 (check_user): Setting NAT on RTP to 0 DEBUG[1133735216]: File chan_sip.c, Line 4891 (handle_request): Check for res for phone1

Hi, I have a fc14 box with two interfaces that I'm having some difficulty setting up a bridge properly. I have a few kvm guests set up and running, but I've switched from using NetworkManager to using standard networking with a bridge. I've modified the ifcfg-eth0 and created ifcfg-br0 as per the standard documentation, but I think I'm having routing problems. This is what the

Dear All, I''m trying to install Xen on Debian Sarge + testing (halfway updated system). Hardware P-IV, 256M Packages: -------- linux-image-2.6.17-2-xen-686 linux-modules-2.6.17-2-xen-686 xen-docs-3.0 xen-hypervisor-3.0-i386 xen-ioemu-3.0 xen-linux-system-2.6.17-2-xen-686 xen-tools xen-utils-3.0 /boot/grub/menu.lst: -------------------- title Debian Xen 3.0 root

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]