similar to: Call recording stopped when call transferred

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...

you can do that. But are u installing qmail and * on same box. i wont recommend that. i use qmail and *. qmail is strictly for internet email. * is on separate server not exposed to Internet. * box also has sendmail. i hv configured sendmail to use smart host (qmail server). This way its safe and secure. HTH, -B ----- Original Message ----- From: "Ing Isianto Istiadi"

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

Hi, I am working on a project to record agent calls when completing specific transactions with customers. Since these calls do not go through the asterisk box (They go through a lucent G3), we're thinking that service observe would be the easiest way to accomplish our goal. Here's what I need: On demand, I need to be able to attach to the switch, dial the service observe code, make

Guys.. Why is it that when a call comes to a call queue and in term gets assigned to an agent, if that agent tries to xfer the call using # or any other feature, it doesn't do anything? I just hear the "pleeps" on the phone but asterisk doesn't intervene with the "Transfer" prompt. Am I missing something? Thx!

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

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

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

Hi all, I want to be able to play messages while someone is in a queue. From the documentation the Background() command should work: From wiki: """Starts playing a given sound file, but immediately returns, permitting the sound file to play in the background while the next commands (if any) execute.""" But in my example below, it does not return until it has

Hi! I need your advice. My problem is that I have very bad sound quality calling to cellular phone via asterisk router. There are some kind of noises and echo effects when you try to speak louder. I have the following components in my call routing schema: - PBX with E1 port. - asterisk router with TE405P card(32bit/4 E1 ports). - Teles server with PRI interface card(3 E1 ports) and VTM

Dear all, I have had an experience which I would run by all of you to see if this is normal. I am running a few asterisk servers with 512M RAM memory, and as I have mentioned in previous notes, I have experienced frequent crashes when faced with more than 15-20 simultaneous calls. I have tried to find out if it could be due to (a) Xeon chip running HT, (b) old Kernel version 2.4.18-3, (c) old

Hi, Yet another question, now that I have callerid working correctly, I'm trying to work out how to utilise the different numbers I have. I have a 100 number range allocated to my E1/PRI/OnRamp service. My incoming calls are handled like this: Advertised/published number is an analogue line terminating on a X101P. If the analog line is busy, it has a call diversion to the PRI on a TE405P

Hello, I thought that my Digium TDM400P would be the right hardware to support the zaptel timer, and put the following IAX.CONF entry to test, (trunk=yes) in the example below [VHAX] type=peer auth=md5 username=whoknows jitterbuffer=yes ;trunk=yes secret=terriblesecret host=4.5.6.7 qualify=1200 disallow=all allow=ulaw allow=gsm ;allow=g711u ;allow=g711a But, it didn't work. So I had to

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

I had CC readers going over the internet (with pings over 80ms) connected to Linksys PAP2. It was only successful once every 3 attempts. I had 100% reliability when it was connected on LAN. Timing is an issue, if you doing everything on LAN it should not be a problem. Just make sure you use G.711 protocol. > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com

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

Hi there Just a quick question. I have been playing around with asterisk CVS-1.0.02 on fedora core 1 (2.4 kernel) and I would like to have a look at asterisk v 1.0.6 but am still a little uncertain which linux kernel is best to run on ?, can I use Fedora Core 3 (is it the preferred kernel) or should I stick with FC1 Ps - the only additional hardware in the box will be a digium single port E1