similar to: Agent auto congesting

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

Hello, I have a queue with only one element and one agent member. I want that my call leave the queue after 30s. My problem is that my call stays 60s in the queue and my agent is called 2 times. Can you say me how can i do it please?? -------------------------------- [queue] music=default strategy=ringall timeout=30 maxlen=1 context=mbdsys announce-frequency=0 announce-holdtime=no

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 List, I have set up the following in my extensions.conf ; local numbers look like 0262XXXXXX ; but must be dialed 262 262XXXXXX exten => _0262XXXXXX,1,Dial,IAX2/543@voipjet/011262262${EXTEN:4} exten => _0262XXXXXX,2,Dial,IAX2/jhiver@NuFone/011262262${EXTEN:4} exten => _0262XXXXXX,3,Congestion It did work for a while, however when dialing I get: stargate*CLI> -- Executing

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

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

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?

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

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

Hi, I had R working since version 1.4. Then I bought a new HD and installed a RH 7.3 on it and since then I can no longer install any R package. Here is the failure message I obtain: ... g77 -fPIC -O2 -m486 -fno-strength-reduce -g -c sortm.f -o sortm.o gcc -shared -o fields.so css.o csstr.o cvrcss.o cvrf.o dchold.o dcopy.o ddot.o dlv.o

Just a word to say thanks for your help. Yes, in order to install R packages I needed ncurses-devel which was not installed. Many thanks Rachid -- Dr. Rachid Cheddadi Centre universitaire Arles Tel: 00.33.(0)4.90.96.18.18 European Pollen Database Fax: 00.33.(0)4.90.93.98.03 CNRS - UMR 6116 rachid.cheddadi at wanadoo.fr 13200 Arles - France rachid.cheddadi at

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

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",

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, I have a 4 column file (long/lat/elev/variable) and I tried to fit the values of my variable to the XYZ space using Tps and I keep getting the following message: Warning messages: 1: GCV search gives a minumum at the endpoints of the grid search in: Krig.find.gcvmin(info, lambda.grid, gcv.grid$GCV, Krig.fgcv, 2: GCV search gives a minumum at the endpoints of the grid search in:

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

I'm new to Asterisk, but, Managed to get it working for outound calls from my ATA --> Asterisk --> Cisco 2620 using SIP. However, I'm having problems with Inbound calls from the Cisco.. Cisco 2620 --> Asterisk --> ATA .. In fact, voice mail won't even work.. This is a snippet of what I'm getting when I try to call the ATA -- Executing