similar to: [1.4] Minimal installation?

Hello I'm trying to use ChanIsAvail() to check when the landline is back to idle after a call, but for some reason, Asterisk doesn't detect that the callee has hung up after listening to MoH for a few seconds: ========== extensions.conf ;Play MoH for a few seconds, hang up, and ;check ChanIsAvail() able to detect when line idle again exten => 8888,1,Answer() exten =>

Hello About every three months, my dad's little Asterisk server that handles his business phone line with an OpenVox PCI card stops taking calls. To check if it's the cause, I'd like to run a CRON job every night to restart Zaptel and Asterisk. Before I go ahead, I'd like to know if I can just send the following commands, or if there are issues I should know about:

Hello I read a whole book on OpenVPN, but still can't figure how to configure the server + client so that the the client connects and sends SIP/RTP data through the tunnel. To get started, I'd rather use a shared key instead of X509 (certificates + keys). The server is running on a uClinux appliance, with /dev/net/tun, and OpenVPN is 2.0.9. The clients will be Windows hosts connecting

Is the new NIN Ghosts music (free download) safe for MOH? Justin ____________________________________________________________________________________ Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello List, I have found that asterisk supports only forwards in-dialog MESSAGE method. That is, if the MESSAGE method is sent within an active call. But according our requirement we need to send MESSAGE method to the other leg without being in a call (general stateless proxy forward). Is it possible to do this in asterisk using some tricks? Regards, Rajib Deka SIEMENS Ltd. Robert V Chandran

Dear all, I am from Saudi Arabiya and I am using digium hardware with asterisk 1.6. When I am executing following AMI originate API. Orginate start to execute extenstion without knowing of PSTN(FXO) channel is ringing. Any one can help me to resolve this issue ? Action: Originate Channel: Dahdi/g0/2923878 Context: outbound-ivr Exten: 1234 Priority: 1 ActionID: ABC45678901234567890

Hello I'd like to configure Asterisk so that... 1. I ring it from my cellphone with CID number displayed, just to notify Asterisk that I wish to make a call 2. Asterisk waits until I hang up, calls me back, and prompts me for the number I wish to call 3. Asterisk puts me on hold through Flash(), which is apparently the equivalent of hitting the R key on European handsets 4. Asterisk calls the

Hello My ISP provides an FXS port to plug a handset, which can be used to make free calls to (GSM) cellphones, similar to the Billion ADSL modems: http://au.billion.com/product/voip.php My plan is to install an SIP client on a smartphone, so that when I'm travelling, I can connect to a good wifi hotspot, register with an Asterisk server at home which has an FXO card, tell Asterisk the

Hi, I'm using a two-years old installation script for the first time on a Squeeze (linux 2.6.32) platform. For an unknown reason (might be an obvious one), Dahdi can't be loaded anymore. 1. First of all, it seems /dev/dahdi content was previously (ie in Lenny) owned by asterisk:asterisk (asterisk is run as asterisk). Now it is owned by root. Any clue about this ? 2. Secondly, I changed

Hi, Xorcom astribanks get initialized straight on when using Ubuntu 11.10 packages but I am having a hard time to get the same result running in a qemu/libvirt image. The first difficulty is that astribanks devices get different usb device ids during their initialisation process, requiring hot plug support. I have figured out how to solve this issue using the technique described in this post :

Hello I need Asterisk (1.4.39.2) to simulate a flash hook (ie. hitting the "R" key on European handsets) so I can put a call on hold, dial a second number, and set up a conference call. By default, linux/include/dahdi/kernel.h sets the flashtime to 750ms, which appears to be too long for European telcos, as they seem to expect a line cut of about 100ms. After editing the

As best i could figure it out, I've installed dahdi and rc4. My TDM400P doesn't answer fxo or fxs. /etc/dahdi/system.conf: loadzone = us defaultzone=us fxoks=1,2 fxsks=4 /etc/asterisk/chan_dahdi.conf: [house-phones] context=internal ; Uses the [internal] context in extensions.conf signalling=fxo_ks ; fxo_ks Use FXO signalling for an FXS chanel dahdichan => 1 ;

Hi, I wonder if I may freely use the default soundfiles that came with asterisk (fpm-world-mix, fpm-calm-river and fpm-sunshine) on production server? Are there any official sources of royalty free music? -- Mvh, Aurimas Skirgaila -------------- next part -------------- An HTML attachment was scrubbed... URL:

Still trying to upgrade to 1.6.0.9 for 1.4. It worked - it worked all day yesterday, but this morning: -- Executing [646xxxyyyy at longdistance:1] Answer("SIP/172-08276a60", "") in new stack .......... -- Executing [646xxxyyy at longdistance:6] Dial("SIP/172-08276a60", ""DAHDI/g2"/1646xxxyyyy") in new stack May 27 09:56:57]

Hello Incoming calls from the FXO trigger an AGI script which simply NOOP data sent by Asterisk through stdin. The first two NOOP work fine, but after this, Asterisk isn't happy: ============= extensions.conf ... [from_fxo] exten => s,1,Wait(2) exten => s,n,Set(CID=${CALLERID(num)}) exten => s,n,AGI(/var/tmp/test.lua) exten => s,n,Wait(5) exten => s,n,Hangup =============

I am posting here since you guys are my last hope. I am trying to configure a Polycom Soundpoint IP 335 with MWI. Is there any way to eliminate the scrolling messages and Msgs softkey? I am trying to get it where it's just the light that indicates the new messages. I don't know if Asterisk has to send a different notification or what have you. Thanks, --Eric -------------- next

Hello The following, dead simple Bash script ran as AGI doesn't reply to Asterisk: ============= extensions.conf [from_fxo] exten => s,1,Wait(2) exten => s,n,Set(CID=${CALLERID(num)}) exten => s,n,AGI(/var/tmp/basic.agi) exten => s,n,Hangup() ============= /var/tmp/basic.agi #!/bin/bash #Ripped from #http://lists.digium.com/pipermail/asterisk-users/2003-July/008554.html while

Is anyone using asterisk with fail2ban? I have it working except it takes way more break-in attempts than what is set in "maxretry" in jail.conf For example, I get an email saying: "The IP 199.204.45.19 has just been banned by Fail2Ban after 181 attempts against ASTERISK." when "maxretry = 5" in jail.conf Perhaps someone else is experiencing this or has resolved it,

Hi, I have a X101P card and am running Asterisk SVN-trunk-r226890. The X101P card seems to be identified properly: debian:/usr/src/dahdi-tools# dahdi_hardware pci:0000:01:01.0 wcfxo+ e159:0001 Wildcard X101P debian:/usr/src/dahdi-tools# dahdi_cfg -vv DAHDI Tools Version - SVN-trunk-r7409 DAHDI Version: SVN-trunk-r7445 Echo Canceller(s): MG2 Configuration ======================

Hello I'd like to install Asterisk and Dahdi on a Ubuntu host using packages instead of compiling from the source. Are the following packages enough for this? ========== asterisk - Open Source Private Branch Exchange (PBX) asterisk-config - Configuration files for Asterisk dahdi - utilities for using the DAHDI kernel modules dahdi-linux - DAHDI telephony interface - Linux userspace parts