similar to: Sending voicemail with qmail

Dear all, I have some questions, I'm sure it's pretty stupid for most of you, but I need you guys to help me. Here are my questions: 1. Music On Hold, it doesn't play any sound on the parked call or hold call. But if I do ps-ax, it shows mpg123 .....( I forgot the exact line). I'm using slackware 9.1 2. I have fxs 3 port, and in my zapata.conf I have included callpickup=1-4,

Dear All, I''m using the kernel 2.6.6, iproute2-2.4.7.20020116, iptables v1.2.9, and gentoo. I have a leased-line 64 kbps. I can see the counter works in iptables, but in the htb, it doesn''t go to the right class (it always go to the default class). Any help will be appreciated here''s my htb conf #!/bin/bash tc qdisc del dev eth1 root tc qdisc add dev eth1 root

Dear all, I have a question regarding the configuration of *. I have 3 port FXS, and 2 port FXO. I have 4 users that use analog phone connected to FXS (I have 3 phones). I need to limit the user's capability (user A can call International, user B can call long distance, etc). I want to implement the password say to call , he/she needs to puch 9(for outgoing call)2-4 digits password,then

I am trying to use the MYSQL command to insert a record into a database and I can't seem to get it to work. I can do an UPDATE with no problem. Here is the line in my dialplan exten => s,12,MYSQL(QUERY resultid ${connid} INSERT INTO `member` ( `id` , `member_num` , `active` )VALUES ('',${number}' , '1')) Does anyone have an example of an INSERT INTO that I could look

Just wondering if there is a reason PPP support is compiled into zapata by *default*: # Uncomment for Generic PPP support (i.e. ZapRAS) # KFLAGS+=-DCONFIG_ZAPATA_PPP Especially since the comments imply that it should be commented out by default... The main reason I ask is because I usually try to re-compile the kernel to only include the bits that I need, and so I don't include PPP...

This newbie has been trying out Asterisk. It has been both a) surprisingly painful and b) impressive in terms of helpful support from other users. Having got two phones to communicate and then got voicemail MWI going (neither painlessly) I decided the next step was to implement call transfer as per nearly all commercial PBX systems i.e. hold call consult another extension either exit and let

> 3. Supposed I have 2 fxo cards (right now I have one already) and 3 > fxs, and one of the fxo will have two phone (running pararell), is > there any way for * to: > a. It always dial the first fxo, if the fxo is busy or is being used > (have other people conversation), will * be able to switch it to > other fxo? Here's the approximiate the conditions of the phone.

Dear All, I have been running an Asterisk 0.7.1 (patched with various agent applications) server for almost 2 years. We have a data center in the USA and a call center in the UK. All calls are routed to a group of central call queues in the USA. Agents from the data center, call center and from remote locations (London, Scotland, LA, Florida, and Maine) can log in, join the call queue and pick

Dear all, Is * capable to use qmail as a MTA? If so, how can I set it? I'm using asterisk v0.5, and TDM30B (FXS), Wildcard X100P(FXO), and x-lite(SIP softphone). In zapata.conf, I put already callwaiting=yes. My PSTN doesn't not support the callwaiting feature, so I don't expect the FXO is call waiting enabled. The question is can FXS and SIP support call waiting?? Cause everytime I

Dear all, I'm just finished installing the TDM (2 port) and X100P. I'm using X100P to pstn, and the TDM to the phone. I've loaded the module, and I can also list the card in the /proc/zaptel/ I'm a little confused now. in zapatel.conf, how do I know which channel is which. (TDM or X100P)? Thanks and pardon my English Isianto Istiadi

I am trying to download the latest release of Asterisk from: ftp://ftp.digium.com/pub/asterisk/ The link provided by Digium is incorrect for the Asterisk Tarball as there is no such file at ftp://ftp.digium.com/pub/asterisk/asterisk-1.0.7.tar.gz However the links for the Asterisk-Addons and other Tarballs is OK ftp://ftp.digium.com/pub/asterisk/asterisk/asterisk-addons-1.0.7.tar.gz Does anyone

Hy guys, I cannot understand How the asterisk works. I would like know how the h323.conf, sip.conf and extension.conf works. I don't understand the parameters and the [sections]. What I need to the asterisk get a SIP call and forward them to a H323 terminal. I working at the h323.conf and extension.conf but I cannot understand!!! Please someone can help me. I your can send me a example (with

For anybody who didn't know there is an article on asterisk in February's Linux Journal. AJ

I've got a nagios plugin making sure the * box is up, but I would like to do more than that. I need to make sure the PRIs connected to my box stay up and I need to make sure calls are not failing for any reason. Are there any * monitoring packages like this? -Daniel

Seems to have struck a small problem.. Using a t100p & Zhone channel bank.... one extension ringing another.....the following will appear WARNING[18448]: File chan_zap.c, Line 2685 (zt_handle_event): Didn't finish Caller-ID spill. Cancelling. if we are using defaultzone=au change it to us and the problem goes away..... any possible solutions ?? Gary .

Hello! I am trying to set up multi-link PPP using two T100P cards in one machine, and 1 T405P card (the 4-port one) in another machine. I have previously been able to get PPP working between the two T100P cards in separate machines.... The 4-port card seems to be my problem currently. I am trying to use the tor2 driver (from a fresh CVS download this morning). When I load the driver (or run

Hi all, I have call recording enabled via the Monitor command and it seems, the call stops being recorded after the call is transferred. Is this normal behavior? If so how can I continue recording of calls after they have been transferred....

I gave a queue setup like this, but I also have it setup so that if no agents are online, the caller cannot get in but I discovered that if that's the case, the call hangsup on the caller: [soportetecnico] ;Soporte Tecnico exten => 2,1,Playback(${SONIDOS}/transferringcall) exten => 2,2,Queue(Soporte-Tecnico) exten => 2-.,1,Playback(noagents) I want to play a message tothecaller

Hi! I've been trying to get ZapRAS or PPPD to work. Neither does! All i get is LCP: timeout sending Config-Requests But after trying, all voicelines get crazy! It sounds like robots when somebody calls! And since the zaptel drivers can't unload (the server hangs totaly if I try!), I have to reboot the whole server! The robot-voice is only on our side, it sounds fine at the other end.

Hi, I've updated my Asterisk 3 times with : cvs checkout -r v1-0 zaptel asterisk asterisk-addons and then do cd asterisk make clean && make && make install make samples make progdocs and then when I run Asterisk I get : Asterisk CVS-v1-0-02/11/05-01:46:25, Copyright (C) 1999-2004 Digium. Is this a bug in CVS handling or am I doing something wrong ? How to check which