similar to: Zap Auto disconnect after xx seconds of silence

We've got a Toshiba DK system w/ analog ports that went to a voicemail server. I swapped in an Asterisk box with a Digium 4-port fxo card. It /almost/ worked perfectly. The problem is that Zap channels never hang up. They have to time out. I set up MeetMe, but all Zap channels hung forever. Very annoying. Same thing for FXO-to-FXO bridges. I figured out today why and fixed it.

I was playing around with conferencing tonight. I was able to place a bunch of SIP phones and a couple of my Zap FXS phones into a conference. So I thought, "Let's see what it's like when people come in from outside." So I called a friend and had him call in on one of my Zap channels, WHICH IS CONNECTED TO MY POTS LINE THAT DOESN'T DO DISCONNECT SUPERVISION. When he

Hellos, I am having issues with my meetme conferencing. When I dial the conferencing number, It hangs after a few seconds.I have read somewhere that I need to enable ztdummy, which I have done but still no changes. Here is my log ~=~=~=~=~=~=~=~=~=~= PuTTY log 2009.08.12 18:44:43 =~=~=~=~=~=~=~=~=~=~=~= -- Executing [1;36;40mMacro[0;37;40m("[1;35;40mSIP/1215-fc5b[0;37;40m",

My Inbound calls lands , but line get disconnect in exactly 3 secs. Here goes my extension.conf setting : [from-ipkall] exten => 901835,1,Ringing ; call ringing exten => 901835,2,Wait(1) ; Wait 1 second for CID delivery from PRI exten => 901835,3,Answer ; Answer the line exten =>

/Hello Users, Good Morning, In Conferemcing How to Disconnect the phone while in between the Conference ..... When *I press the ' # ' key for Disconnecting the Conference.......... Below the Following to shows some Warning, ( in Red Color ) from-sip en *CLI> -- Executing Playback("SIP/9002-08f9feb8", "conf-hasentered") in new stack

Hello, We have an * installation that is causing us fits. The problems we are seeing: 1) In the middle of a call the call gets dumped and the caller hears a dial tone. 2) While talking on a call the caller hears nothing for 5 to 10 seconds. The person on the other end of the call hears everything just fine. Then the call returns to normal and both parties can hear. our network:

We are using G711 for phones to talk to Asterisk and G729 licenses at asterisk to talk to ITSP Could you please suggest transcoder to use from G711 and G729 and which is comptible with Asterisk. We will like to avoid using TDM if possible Also i remember that initially we didn't have G729 and were using only 711 for with vicidial but then also we had same problems. at that time it was only 2

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 =>

Hi all, My scenario is such that I have three users connected to a conference. CLI> meetme list 1234 User #: 01 9176502096 <no name> Channel: Zap/23-1 (unmonitored)00:00:32 User #: 02 john john Channel: SIP/john-b7800468 (unmonitored) 00:00:28 User #: 03 6463875998 <no name> Channel: Zap/22-1 (unmonitored)00:00:19 3 users in that

When someone calls into our asterisk server over a PSTN line, dials an extension and then hangs up, the SIP phone related to the given extension will ring about 4 or 5 times before asterisk shows that the channel has been hung up in the console. This isn't such a big deal on its own, but what's happening now is that if a user calls in from a PSTN line, gets voicemail on the extension, and

Here's the scenario: 1. I call out through * using a X100P card to somebody. Then I transfer them to a MeetMe conference and that all works. 2. After the conference is over everybody hangs up but "show channels" shows that the Zap/1-1 channel is still in use by MeetMe and the analog line is not freed up for re-use. Ever. Any clues? Thanks!

> -----Original Message----- > From: Andrew Kohlsmith [mailto:akohlsmith-asterisk@benshaw.com] > Is the channel physically being hung up before the * tone is heard? Good question. If it is, Asterisk doesn't detect it -- the PBX doesn't support Kewlstart-style disconnect notification. The sequence I hear on the extension, when I plug in an analog phone, is the click of the

I've been asked by my company to setup a conferencing system to support up to 400 people on a conference calls, where all users will be dialling in frpm the PSTN. I am exploring using Asterisk meetme to do this. I have two questions in relation to this:- For Meetme conferences is it better to have all participants to dial in via SIP provider terminating to Asterisk via SIP/IAX, or use

Hi List, I'm having a bit of an odd problem with asterisk and outgoing zap calls. Tzafrir has been kind enough to help me get the logging sorted out so I have some idea of what's going wrong, but I'm a little flummoxed. Essentially the symptoms are as follows; Make a SIP call from Cisco 7960 or 7940 to asterisk, where it is routed out on a ZAP (x100p) line. After

When I call meetme: exten => 1000,1,Answer exten => 1000,n,Meetme(|||d) Asterisk is complaing with: -- Executing Answer("IAX2/xxx.yyy.142.204:4569-2", "") in new stack -- Executing MeetMe("IAX2/xxx.yyy.142.204:4569-2", "|||d") in new stack -- Playing 'conf-getconfno' (language 'en') Warning, flexible rate not

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

I looks like someone decided to bundle a patch that hasn't been merged yet. Good for testing, not so good for initial impressions. In /etc/asterisk/asterisk.conf add or uncomment this: [options] ;silence_suppression=yes And see if that helps. You need a timing source for it to work, which is why it is disabled by default, but the logging might be a bit chatty in any case. Dan

Running * version 1.6.1.17. My meetme conferences automagically disconnect users approximately 5-15 seconds after the user is connected. This occurs regardless of whether music on hold is active or not. [Mar 23 11:34:36] -- Executing Macro("SIP/SDN_TMCKEE-000000e9", "confroom,1808") [Mar 23 11:34:36] -- Executing [s at macro-confroom:1]

Hi asterisk-users, We are using Asterisk 1.2.12.1, and are trying to use the Page application. It seems to work but after approx 4-5 seconds the call is hung up. The dialplan code look like this: exten => _*2XX,1,AGI(get-paging-devices.agi,${EXTEN:2}) exten => _*2XX,n,GotoIf($[ "${PAGING_DEVICES}" = "invalid" ]?i,1) exten => _*2XX,n,SIPAddHeader(Call-Info:

One of our users recently had a powerfail while connected to our meetme gateway. (Asterisk 1.4.17 on debian 4.0) Through the course of it, asterisk never hung up. His system came back up, and started sending ICMP port unreachables, but the stream went on, flooding him with "silence" media stream packets (there was nobody else in the conference). Is asterisk aware of ICMP