similar to: busydetect and random hangups

I have a couple of X100P's in my system and while on calls they just randomly hang up for no reason. I have tried messing with the busydetect and callprogress setting them to yes and no same and still random hangups. Is there another setting I should be looking at? My zap config looks like. context = inbound-work include => extensions signalling

FYI to anyone else who may be experiencing random hangups; I removed the busydetect=yes lines from the conf files on my asterisk servers, and haven't had a hangup since. I had done that once before and it didn't seem to have much of an effect, so I'm not breaking out the champagne yet. But so far over dozens of calls both made and received since I took that line out, I

Hi, I'm trying to build a call center application that allows attendants to come in the morning and dial a certain extension to make their extension available. I wouldn't like to use the AgentLogin app because their line would need to stay off-hook (is this correct?) Is there any SET channel status command that would allow me to do something like this? PauloHM -------------- next

Hi guys. I have a XP100P Clone , and the busydetect dont work for me.. PSTN---Asterisk---Sip---Asterisk----PBX Any call from pstn side dont disconnect ... I have no disconnect supervision and busydetect dont work... Please Help me. Zapata.conf [channels] echocancel=yes usecallerid=no hidecallerid=no rxgain=0.0 txgain=0.0 signalling=fxs_ks callprogress=no context=entrada channel=>1

Count me in too. -----Original Message----- From: sip [mailto:sip@intology.com] Sent: Friday, October 17, 2003 1:56 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Beta testers for visual configuration tool for asterisk count me in ----- Original Message ----- From: "Paulo Mannheimer" <paulohm@instant.com.br> To: <asterisk-users@lists.digium.com>

Hi All, We've been developing for a while an IDE for Asterisk, and the time has come to open it for beta testers. You can check at www.instant.com.br/viv.html for a snapshot of the application. Current modules are Dialplan and VoiceMail configuration. As you may see, it is all-visual, with drag and drop support and integrated sound recording, saving and cross-checking, so you dialpland

Hi All, This isn't really a question, but it's an issue I experienced that was driving me crazy for a few days, so I thought it might be good for the archives. Basically what was happening was everytime a particular customer called (long distance), the line would disconnect immediately after answering. I thought it might have been the phone, so I swapped the phone with another - still

Hi All, Can anyone tell me what are the <dev1> <dev2> parameters that I should use to run pridump? I took a look at the source code but couldn't figure this one out. Best, PauloHM

Hi, I would like * to treat a SIP extension as a normal extension, when it comes to the busy functionality. In other words, if someone tries to call the SIP phone and there is already an ongoing conversation, the new caller should get a busy message/tone Is there any parameter that I can set? Is this something that should be configured at my softphone? Best, PHM

The subject pretty much says it all. I have a customer who would like to have an option where a caller can leave a voicemail in such a fashion that it would be simultaneously delivered to a set of mailboxes all at once--the idea is "trouble ticket" type operation where multiple technicians will *each* get the vm. He prefers that, if we can do it, to a "shared mailbox"

Hi, I'm trying to use iaxcomm. I can place a call from the softphone, but when I place a call to it, when I answer I get ... NOTICE[16401]: File channel.c, Line 1094 (ast_read): Dropping incompatible voice frame on IAX2[paulohm]/3 of format GSM since our native format has changed to ALAW My iax.conf looks like this .. [paulohm] type=friend host=dynamic username=... secret=...

Hi everybody, I would like to create SIP call flow Diagram under Windows. Is anybody know a program to perform it? I have already Ethereal and I would like an explicit diagram just to show where something have problems... Thanks Ignace

I have a very strange problem I'm hoping someone has encountered already. I've scoured the internet for an answer to this one. My phone company provides no disconnect supervision. Hence I'm forced to use the busydetect feature. I have a TDM400 with two FXO ports. If I call from an internal extension to a landline and then hangup the landline Asterisk detects the busy signal

Hi. I was wondering how can I make incoming calls to wait if the phone operator is busy. I've 8 incoming lines, with 30 extensions. What I need is if the operator is busy with call nr #1 , the new incoming call waits until the op. is free. Looking into app_queue seems the way to go. So I want to ask if I'm right or wrong: I set up only a queue , is to say operatorq, where the only member

I was wondering if the accountcode flag works with sip channels. I was looking into the debug and ,even if I have the line accountcode=XXX into the users sections of my sip.conf, I don't see it logged into the cdr. Matteo Brancaleoni mbrancaleoni@espia.it Emmegi System Administrator EspiA - EMMEGI Srl - e*solution provider Uffici: Via Pascoli, 37 20129 Milano - Italy Sede Legale: Corso

Hello, For those interested in overcoming the problem with some NATs and Firewalls in regards to tftp. I found a nice little tftp server here: http://freshmeat.net/projects/jtftp/?topic_id=87 I tried it and it works great. Regards, Andres. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I'm testing an E1 with E&M signaling. Some of the problems I'm running into are the following: 1) if I try to configure any channel above channel 15, I start getting a "multiframe alignment error" on my telco test equipment. So I have my zaptel file only configured for 15 channels, like this span=1,1,0,cas,hdb3 e&m=1-15 2) When the test equipment tries to send me

Please, don't hate me because I use Redhat. I am aware that I am asking for problems in running Asterisk on Redhat. I recently aquired a nifty server, moved my digium cards, and installed asterisk. I noticed that one of the four processors was being used at 100% and nothing was working. I tracked CPU utilization back to the Asterisk process. Please, help. James

Hi all, We are about to make our first channel bank install. This will be a one PRI outside connection and up to 70 extensions. As the schedule (and the budget) is pretty tight, I would like to learn a little bit more about general experiences with channel banks, like echo cancellation problems, Caller ID usage, etc. TIA, Paulohm

Hi , I''m getting random hangups on zap channels with long calls. It seems that the hungup happens after 10 minutes or so. AbsoluteTimeout is set to 0. Any other thing I should be configuring? Thanks! PHM