similar to: Everyone is busy/congested at this time

Hi All, I have this codec problem, I use "gsm" in my iax.conf file and in teliax settings also, but the error is still appearing as below. please help me. Kumara Starting simple switch on 'Zap/1-1' -- Executing Dial("Zap/1-1","IAX2/kumara@teliax/01194777070239|30|tr") in new stack -- Called kumara@teliax/01194777070239 -- Call accepted by

Hi all, again, with another question ( may be the final one) I have come up to this point, means when I dial a number in my analogue (panasonic) phone I hear the ring at the end through my asterisk box (via TDM20B card) that uses IAX2 over teliax and after time-out, it gives this message. Starting simple switch on 'Zap/1-1' -- Executing

Hi all, Does anyone run Asterisk on FC4? with Digium's TDM40B cards. any success stories? my Intel 865 M'd+ Intel 3.0GHz freezee during installation (FC4). Please any comments? Kumara

Greeting!, I read somewhere that without cdr, Mysql etc it is possible to take outgoing-call-logs to a text file. (I am not sure please). is it really possible ? if so, how do I do it? any links to refer? Thank you. Kumara

Dear all, I am happy to tell you that I received a Digium's TDM20B card for my Asterisk box today. but the phone jack is RJ45 instead of RJ11. So Please I need precise instructions to connect a phone to this card. please, assume that I have a phone (a normal analouge phone connected to the one end of a cable with an RJ11 jack (at the phone side). and now I want to connect the other end to the

Greetings! to all Now, with some hard time and help from many genurous people's in the list, I have come to this point with my TDM20B card & my teliax's IAX2 account. I hope someone may help me with this issue mentioned below. I have already selected my codec as gms in my iax.conf as well as in teliax's "my account page" but still i have the same error when I attempt

I trying to receive a SIP call and have ring a analog phone. I have a TDM11B card with FXS(green) module in line 1. I have Sip server "SER" setup to accept a SIP call, add a 970 extension to uri and set to asterisk SIP server on port 5065. When I send a SIP call from "kphone a soft SIP phone" running to sip://wally.world@cci.net "SER" picks call ok and changes uri

Hi, I have a problem with IAX accounts... I set up a few months ago an Asterisk server, with mysql support to load iax accounts. Settings seems fine because apparently the system works as expected. Yesterday I tried to add another iax account in the iax.conf directly. And I have a problem with this new account (named 444). I can authenticate from 444 to the server, and I can receive calls from

On Sat, Mar 14, 2020 at 2:02 PM John Roman <john at dev1ce.com> wrote: > greetings asterisk users :) > ive just deployed version 17 and migrated as best I can to pjsip. I can > receive calls, and get to my mailbox prompt, however placing calls seems > impossible with the following error on dial: > > Connected to Asterisk GIT-master-0cde95ec89 currently running on dunkel

I am using Teliax to terminate my calls, and I have 3 licenses' for g729 from Digium. "show translations" verifies that the registration took place. When I place a call, having "allow=g729" as the only allow option in iax.conf, I get the following error: WARNING[361]: chan_iax2.c:6017 socket_read: Call rejected by 208.139.204.228: Unable to negotiate codec If I place a

See: http://lists.digium.com/pipermail/asterisk-users/2004-September/063348.h tml -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Kumara Jayaweera Sent: Tuesday, June 14, 2005 1:46 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] RJ45 instead RJ11 in Digium's TDM20B card help meplease

I am planning on dabbling with some VOIP providers. I was thinking of Teliax first. My thinking is that the first LD call would go to teliax and the second (etc.) calls would go out to the PSTN. I could then verify bandwidth and quality to decide when to add more trunks and to Internet connections. I have been doing some concept testing with FWD for toll free calls, but I am using 393 as a

ive enabled logging. aside from a realm error i see on my endpoint, im still not sure whats up Asterisk GIT-master-0cde95ec89, Copyright (C) 1999 - 2018, Digium, Inc. and others. Created by Mark Spencer <markster at digium.com> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General

I sent the below message out last Friday when the list seemed to be having issues. Never got any responses and not sure if it just no one knows or if it did not get through. Please don't flog me too bad for reposting... :-) ------------------------------------------------------------ Hi all, I am having some audio quality issues with a provider under sip. The issue I am having is

Has anyone had problems getting their preffered codecs on the Teliax web interface taking effect? I have two accounts, two separate yet similarly configured * servers. On one account the settings took right away - on another server I am getting no result. In fact, no matter what I change the settings to, only the old codecs are usable (otherwise * says it can't negotiate a codec). Teliax

Could someone please look at this information and help me decipher what it should actually mean to me? I've found a bit of information here and there but I'd like to know what I'm supposed to be reading into this information: pbx*CLI> iax2 show channels Channel Peer Username ID (Lo/Rem) Seq (Tx/Rx) Lag Jitter JitBuf Format IAX2/teliax-4

greetings asterisk users :) ive just deployed version 17 and migrated as best I can to pjsip. I can receive calls, and get to my mailbox prompt, however placing calls seems impossible with the following error on dial: Connected to Asterisk GIT-master-0cde95ec89 currently running on dunkel (pid = 517890) dunkel*CLI> dunkel*CLI> == Setting global variable 'SIPDOMAIN' to

I use the Teliax service with the IAX2 protocol. I noticed 2 days ago that I was not registered with the Teliax server. I used the "iax2 show registry" command and found I was not registered with Teliax. I issued a reload command in asterisk in order to connect again. I went to the Teliax website and noticed a message which stated clients may need to reboot due to changes made with

hi all i have a TE110P connected to my PBX when i try to call a extension number in other location 3525 the asterisk give me a error -- User entered '3525' -- Executing [450 at lacnicuy:4] GotoIf("Zap/31-1", "0?6:5") in new stack -- Goto (lacnicuy,450,5) -- Executing [450 at lacnicuy:5] Dial("Zap/31-1", "IAX2/lacnic:splacnic at

> > Funny. The first thing I always do after a reboot is call in from my > cell to make sure things work. But last night I rebooted and immediately > tried dialing out (with a TDM842B) and got: > > WARNING[2720]: app_dial.c:1721 dial_exec_full: Unable to create channel of > type 'DAHDI' (cause 34 - Circuit/channel congestion) > > Restarting all of Asterisk or