similar to: Channel is rsrvd and does not turn off

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

I have two Adtran 750's connecting our analog phones to asterisk. On occasion, I get a channel that gets "stuck" off hook. 'show channels' shows: Zap/27-1 (longdistance s 1 ) Rsrvd (None) (None) And will just stay like that until the phone is manually picked up and hung up again (or asterisk is stopped/started). I guess this is a function of an unclean hangup (being

Good afternoon list. I am experiencing a problem with the CDR and callfiles. What is happening is this: When generating a call with a callfile, everything works perfectly, but the CDR is recorded in the table when they answer the call destination. The field disposition is being recorded correctly, but the duration field is marked with the ring time and billsec is marked with 0. This just happens

Guys, I saw in the Asterisk documentation (queues.conf) that members can register with technologies such as SIP, Dahdi and Location. But I have a specific need for members to be registered as Khompchannel. Ex: member => Khomp/b0L1/9200 But reloading module app_queue.so when I run the command "queue show", the member registered as Khomp appears as invalid:

Hi, We are using Vicidial and sometime even when agent disconnects, outgoing call originated by dialer is still active. Since call was initiated by dialer and then bought into meetme conference of agent and we can't corelate this call to any agent channel. When agents are dialing, channels doesn't show calls vicidial2*CLI> show channels Channel Location

Greetings. I've plugged 3 analog lines on an ethernet cable in an Khomp card to receive it's incoming calls. Without any configuration, when I call those numbers the asterisk server automatically answer the call and play the default music. The problem is: I need to discern the lines and redirect each one to his respective extension. Since they doesn't got any Caller ID Service the

Acrescentando o report do Dell, os equipamentos da Khomp s?o homologados pela Anatel - funcionamento normalmente nas implementa??es de Asterisk puro, FreePBX ou Elastix. Caso desejem mais informa??es sobre equipamentos da Khomp, consultem a CAM Tecnologia. A CAM Tecnologia atua com revenda ou venda direta da khomp para o cliente final. Contato: Rubens Duarte de Andrade Tel: (21) 3189-1050

Sauda??es. Como que se faz um conector RJ45 em uma ponta e RJ11 e outra. Pretendo conectar a linha de um ATA em uma placa Khomp KFXO IP. A ponta que tem o conector RJ45 est? crimpada com a sequencia 568B e vai ser conectada na placa Khomp, mas a ponta RJ11 eu n?o sei como deve ficar. Li alguns manuais na internet mas n?o entendi ao certo como tem que ser feito. -- Att.* *** Luis H. Forchesatto

Hi experts. Recently I've insalled a PCI Khomp Pane on my server and inserted 4 chips to make call with it. The calls are good and no issue was noticed but I got reports that when someone call the chips the call volume is uncommonly low for both sides and they deploy some failures on the audio, only when the call comes from outside. When an extension at the same network makes a call that goes

I have a small system based on one TDM400 card with - 2 FXO modules for incoming lines - 1 FXS module for one phone The system was working fine in the past. The motherboard was exchanged and during the switch, the phones line was rewired with a mistake and a incoming phone line was connected to the FXS module, and there was ringing voltage on the module. Now, the system kinda works but the FXS

I'm running Asterisk CVS-v1-0-12/20/04. I'm using PHP with Manager API Here is the code: #################################################################### # Make call #################################################################### $socket = fsockopen($ask_db,"5038", $errno, $errstr, $timeout); if (!$socket) { echo "$errstr ($errno)<br /\n"; } else {

Hello I need to write a script that will dial a list of customers and play a message. I couldn't find a way to tell Asterisk/Zaptel to wait until the callee has actually picked up the phone before proceeding with Playback(): ============ ;call made through Dial(): Doesn't proceed after off-hook/hangup [internal] exten => 8888,1,Dial(Zap/1/${IPPI}) exten => 8888,n,NoOp(We never

Prezado Fernando, Muito obrigado por sua complementa??o na resposta! Surgiram algumas d?vidas agora: A ?nica forma de retornar os dados num header field, como o Rafael dos Santos Saraiva sugeriu envolve criar outro channel? Ou seja, o que eu preciso ? que a mesma execu??o do dia plan obtenha um valor recebido do Sip Client, execute uma query num banco de dados e em seguida inclua a resposta

Hi to all. I have a problem in the AMI. Sometimes the AMI don't generate the event NewState when the exten of destiny is Ringing and sometimes don't show me the callerid in this events. The event NewState what i refer: Event: Newstate Privilege: call,all Channel: SIP/17-00006fd6 ChannelState: 5 ChannelStateDesc: Ringing CallerIDNum: 4191920902 CallerIDName: 4191920902 Uniqueid:

My system is linked to a legacy PBX via Q-SIG and most of my tests so far have been from that PBX. I created a number of MeetMe conference rooms and they work fine when called from the legacy PBX. However, when there's a MeetMe room with a legacy caller and a SIP phone, the SIP phone can hear the legacy caller. But the legacy caller can't hear the SIP phone. However, "meetme show

Hi With an FXO module + Zaptel, I'd like to know if there are ways to know when the remote party has answered the phone, whether calling through a callfile or by sending DTMF's. I read about {CHANNEL(state), ChanIsAvail(), and ${DIALSTATUS}: Are those reliable ways to know when the channel is available for dialing out and the call has been answered?

First of all - Many, many thanks to Mark for his troubleshooting and fix of bug 1320 (FXO_KS signalled Zap Channels on Adtran 750 Channel Bank Stuck in Rsrvd State). I have heard complaints that once every couple weeks, when a user picks up their analog phone (t1 span off of a TE410P into an Adtran 750 with 6 FXS cards), they don't get dialtone, but instead, hear another conversation.

Hello, I'm using an AMI Originate command to send a fax. The fax is sent by a script, and I'd like my script to send the fax, wait until it has succeeded or failed, then exit with an appropriate error code (it is driven by a mail system, so the exit code will tell the mail system whether to retry the fax later). The script works great if the fax succeeds, or if the line is busy or

Hi; If my landline service provider does not provide the ability to send the SMS, and I need to send SMS from asterisk, then what is the required? How? Is it possible to send SMS from asterisk using SIM card to be connected via GSM adaptor connected to FXS ports? Or HOW?

hello, I got those extensions: exten => 555,1,MeetMeCount(500|count) exten => 555,2,Gotoif,$[${count} = 1]?6 exten => 555,3,Meetme,500|pMs|1234 exten => 555,4,Playback,goodbye exten => 555,5,Hangup exten => 555,6,Goto(from-internal-custom,556,1) exten => 555,7,hangup exten => 556,1,System(/bin/cp /etc/asterisk/1-test /var/spool/asterisk/outgoing/) exten =>