similar to: VoIP Dialout issues

I am using Asterisk CVS from last week and have been using Realtime SIP for a couple weeks now without any problems. Yesterday I decided to turn on Realtime IAX but I am having problems dialing to my long distance providers like Voicepulse, Sixtel or Nufone. I get the following: -- Executing Dial("SIP/2001-3761", "IAX2/password@voicepulse/19566680301") in new stack

Hi, After a few extra days of hard work, debugging, and many coffees, I am pround to announce that Asterisk::LCR has been released on CPAN. Asterisk::LCR is an open-source, Perl-based collection of tools to help you manage efficiently multiple VoIP providers with your Asterisk installation. It is capable of importing providers rates from multiple providers, comparing these rates, and

Hello I just settup a realtime mysql table for sip_peers. All peers (friends) is autenticateing but when i want to initiate a call between them i got the following error. Someone have some ideea? Thank you. ---<Cut Here>--- pbx*CLI>console dial 1014 == Console is full duplex -- Executing [1014@default:1] Dial("OSS/dsp", "SIP/1014|40|t") in new stack

Is iaxtel down? Ive been getting this: May 21 19:23:42 NOTICE[29984]: chan_iax2.c:2782 auto_congest: Auto-congesting call due to slow response -- IAX2/Iaxtel-12 is circuit-busy -- Hungup 'IAX2/Iaxtel-12' is it down or am I doing something wrong?

I am experiencing a very wired problem. Some of my cdr are lost. I use logging cdr to csv, mysql and odbc. But some of them are lost. They miss in csv mysql and odbc, so i'm pretty sure it is related to asterisk functioning. I am running asterisk 1.0.7; this is simple configuration file: extensions.conf [general] static=yes writeprotect=no [macro-gw-voipjet] exten =>

Hello all, Here is my problem, I try to place a call to FWD (free world dialup) trough my asterisk PBX. my config is as follow: extensions.conf ---------------- [internal] exten => 613,1,Dial(IAX2/iaxfwd-outbound/613) (service echo de FWD) exten => xxxxxx,1,Dial(IAX2/iaxfwd-outbound/xxxxxx) mon numero FWD exten => yyyyyy,1,Dial(IAX2/iaxfwd-outbound/yyyyyy) celui d'un ami FWD

Hello, I Have an agent on a queue, evry thing works normally, but after a time (about 5 minutes) my agent is pauses (agent is still regitred but can't takes calls), on Astrisk console i have the message : [Jun 20 11:55:12] NOTICE[8803]: chan_sip.c:2757 auto_congest: Auto-congesting SIP/anna-08215f68 -- SIP/anna-08215f68 is circuit-busy -- Nobody picked up in 8000 ms I think

I try to connect to voipjet, but I get always busy, ... it worked yesterday, ... no changes on my side.... -- Executing SetGroup("SIP/615-829b", "iax-voipjet") in new stack -- Executing Dial("SIP/615-829b", "IAX2/17567@voipjet/011886228357765") in new stack May 20 18:16:26 NOTICE[9733]: app_dial.c:973 dial_exec_full: Unable to create channel

Hi All, I am using Asterisk 1.4.26.2 and I am getting the following problem making connections to this server. My other servers are Version 1.2.x which have no problems and this 1.4.26.2 server can call the other 1.2.x servers. The error is: chan_iax2.c:4251 handle_call_token: Call rejected, CallToken Support required. If unexpected, resolve by placing address 192.168.25.250 in the

So far my experience with TOS has been that most of them are pretty odd. No one wants the liability of a stock trade gone foul or a call to the doctor that gets disconnected. Essentially, those things in the TOS are just a CYA. They are un-enforced but should someone decide to attempt to sue based upon a financial loss, the ITSP is covered. So, yep. That is weird but not unexpected. Heaven

I just had a situation where I could not get calls from or out to one of my IAX2 boxes to another. The one which seemed to have a problem didn't show the server in its "iax2 show registry" list. I reloaded and the register showed up. Looking at the server, when I called the number I got the message: Jun 2 20:48:34 NOTICE[25542]: chan_iax2.c:2209 auto_congest: Auto-congesting

Hi - I'm running CVS-HEAD from 2005-08-11 20:17:17 UTC, and I'm trying to set up some redundancy on IAX connections between locations. I have two IAX peers set up that work correctly by themselves: "ast551-out" and "ast551-out-backup": [ast551-out] type=peer secret=secret username=ast551 host=X.X.X.X qualify=1000 disallow=all allow=gsm allow=ulaw trunk=no

Ok so I've just built and installed a CVS (HEAD) version of asterisk on RHFC2 running a 2.6.13.3 kernel.org kernel. I installed the samples via "make samples". Everything seems to work except one thing. I'm trying to do the connect to the Digium IAX demo server portion of the demo (dial 500) and I just get the following messages. I am behind a NAT server and did NOT change

Hello, sometimes I'm experiencing autocongest error due "slow response", anyone knows, what this means? Second or third attempt after that happens pass successfully... this happens ever in fastethernet lan, so no problem with lag in wan environment, I'm using idefisk 1.32 on client side (winxp or linux)... PJ -- Executing Dial("IAX2/bill-7",

Hy all! Your Asterisk server is return this log : *CLI> -- Executing Dial("Khomp/B0C0", "IAX2/*.*.*.*/9834|30|r") in new stack -- Called *.*.*.*/9834 Mar 14 15:35:40 NOTICE[4212]: chan_iax2.c:2836 auto_congest: Auto-congesting call due to slow response -- IAX2/*.*.*.*:4569-1 is circuit-busy -- Hungup 'IAX2/*.*.*.*:4569-1' == Everyone is

Hi, I have an FWD account and it's configured in asterisk. I can be called by people using FWD, but I cannot make FWD calls myself. Every number dialed with a 8 prefix goes to FWD, if for example I call the echo servie I get this: Connected to Asterisk 1.2.13 currently running on asterisk (pid = 2865) Verbosity is at least 35 -- Executing SetCallerID("SIP/timothy-08224f08",

Hello, I've got various phones (mostly SPA-922) behind NAT registered to Asterisk. I've set nat=yes and canreinvite=no, and everything seemed to work great with 1.2.17. After upgrading to 1.4.2 using users.conf and macro-stdexten my spa-922 can't call other extensions. -- Executing [23@default:1] Macro("SIP/22-b72006f0", "stdexten|23| SIP/23") in new stack

Hi All, I am stuck with an issue in the Openser+Asterisk Forking. In this solution we are using Openser as the Registrar. Hence it will store all the contact bindings along with the q values for a given user, say ua1. The current setup is such that the INVITEs are sent to Asterisk by Openser and Asterisk sends out the INVITE. Now if ua1 is registered with two different contacts having

Hi List, I'm using VoIPJet and NuFone as a fallback, and it seems that both of them are circuit busy! Also it seems that VoIPJet takes forever to return 'circuit busy' while NuFone does it instantly. At any rate, is there like a reliable third VoIP provider I can use for fallback when the two others are busy? Cheers, Jean-Michel.

I've just setup a new asterisk box (cvs HEAD) with a digium tdm411 and a couple computers with eyebeam. I have one small. I cannot call the eyebeam clients from the phone connected the fxs port. I can call the phone from the eyebeem clients. And, I get both the fxs phone and eyebeam clients to ring when a call comes in through the fxo port. I have been trying to get this straightened out