similar to: Re: Asterisk-Users digest, Vol 1 #4230 - 13 msgs

I am having trouble getting transfers to work when a zap channel is part of the call. I have a couple SPA-2000's and some X100P cards as my setup. This is what I'm trying: Dial number from phone: -- Executing Dial("SIP/206-2c61", "Zap/1/#######") in new stack Currently on call: -- Called 1/####### Press flash to place call on hold with SPA-2000: --

I've been trying to get fax reception to work using an SPA-2000 to ring the fax machine or modem that's taking fax calls. I was curious if anyone else had tried something similar, and if so, had any luck getting it to work reliably. I've been able to get it to work, but it isn't reliable. (Pages/lines of black dots result more frequently than not.) The incoming lines are FXO

I've come across an odd dialing problem with my * setup. After * has been running for a while, if I try to dial out on any of my zap channels, (both are X100P cards) it picks up the line but never sends the DTMF. Has anyone heard of or seen this problem before? Right now I'm looking at 19 hours of uptime on * itself. If I restart it, everything works fine for a little while, then the

Because I'm tired of seeing Debian users look like idiots for using such horribly out of date versions just because they run "stable", I've set up a page with .deb packages of my 1.1.x rebuilds for use under lenny. http://www.rollernet.us/opensource/ Standard disclaimer applies: use at your own risk, no warranty, no support, if it eats your first born or erases your computer

Does anyone know how (if possible) to do three way calling on the UIP-200? There doesn't seem to be much info about this phone, but all the feature lists I've read says it can do conference calls. I can't seem to do it, though. Any help would be appreciated. -- Seth "et lux in tenebris lucet" Mattinen sethm@rollernet.us

I can't seem to make the "Hold" button function on the GS BudgeTone-100. I'm trying a procedure like this: 1) On a call 2) Press "Hold" button 3) Hang up phone What I expect is for the call to go on hold until I pick up the receiver again. (Like my SPA-2000's, except it's a flash, but I can hang up the phone and the call waits there for me to pick it up

I'm having an echo related problem with Zap channels that are answered before a dial takes place, such as for IVR menus or fax detection. Basically, it sounds like the volume gets turned up to maximum while * is ringing the internal extensions waiting for someone to pick up, so when you pick up the phone to talk, you get an over-amplified echo. The sequence is like this: exten =>

Perhaps I was a little too hasty in my conclusions of dysfunctional fax on the SPA-2000. It turns out I have a one way audio problem on line one of my SPA-2000. I have all the correct settings according to the comments in the wiki, but the problem persists. However, if I do a hook flash out of and back in to the call that isn't transmitting audio, it works fine. My sip.conf entry for the

Hi, I have a TDM400P (31B) in a PIV 2.8, 512Mb ram, CentOS 4.3, zaptel 1.2.5 and Asterisk 1.2.7.1 and a couple of standard analog phones with a flash button. A hook flash works fine for setting up a 3way call. But pressing the flash button doesn't do anything. The zapata config is below. Anyone have an idea what I'm doing wrong? [channels] context=local usercallerid=yes hidecallerid=no

Hi all I am in the process of migrating Courier to Dovecot, am using courier-dovecot-migrate.pl<http://www.dovecot.org/tools/courier-dovecot-migrate.pl>it works fine. I want to migrate each user as they login using post-login script (As i have huge no of users). We are using NFS storage, users are authenticated using mysql, User home directories as passed using variable mail ( usign mysql

If you want to run Dovecot on Debian Lenny for a Production System (with Fast Upgrade System for Security Patches), What would you recommend? Stick to the current port with Dovecot Version 1.0.15 (1:1.0.15-2.3) Use Backports Version 1.1.13 (1:1.1.13-2~bpo50+1) or Use any of the packages from:

I just installed CentOS 5 + Postfix + Dovecot. Now When I send mail via telnet from one user to another via Postfix (MTA), I have no problem. The recipient gets the message perfectly. Now this tells me that Postfix is working fine (I assume) but then I configured Dovecot to use IMAP4 and then configured my Thunderbird client. Now all the messages I previously sent using Telnet showed up in

These are the three cheap SIP phones that I've used. Grandstream BT10x $65/street Number only LCD Zultys ZIP 2 $100/retail No LCD Uniden UIP 200 $120/retail PoE, built-in switch -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 "In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss."

I am using procmail with deliver as recommended on the wiki to send mail snagged by SpamAssassin straight to a spam folder. It seemed to be working fine, but now I'm finding messages like this: Sep 17 12:13:47 hosty deliver(whoey): write() failed with mbox file /var/mail/whoey: File too large I don't think this is a setrlimit problem -- the host operating system is OpenBSD 4.0. My

Just starting to do the research on this one....I've got a customer who is showing interest in replacing any older Panasonic unit providing service to 96 tip/ring lines from a single PRI. Does anyone have any recent experience with a decent (as in, plays nice with * and has a reasonable per-port cost) channel bank or similar? Mediatrix only goes up to 24 port, as far as I can tell, which

I'm trying to switch from one call to another incoming call from PSTN. When I'm getting a "beep" I press flash but instead of swithing to the second call, I'm getting a dial tone. even if I press *0, I cannot connect to the second call. Anybody had this problem? Tx, Bogdan

Hi, I am the "IT guy" at a small startup based in UK. At the moment we have 3 analogue (PSTN) lines and we will be adding another 2 or 3 soon. Later on we should be changing to ISDN30. One of the partners mentioned getting an analogue PBX now, and when we move to ISDN, then get a digital PBX. I though of Asterisk. I have seen the website in the past and I know that it can do the job

When I reload the config, I see this error in the CLI. However, I don't see what I have done wrong: == Parsing '/etc/asterisk/zapata.conf': Found May 30 16:38:42 WARNING[12630]: chan_zap.c:10088 setup_zap: Ignoring signalling -- Reconfigured channel 1, FXO Kewlstart signalling May 30 16:38:42 WARNING[12630]: chan_zap.c:10088 setup_zap: Ignoring signalling -- Reconfigured

I have tried everything to get BV working outbound. All worked fine until the BV change last week. I called BV and they changed me to sip gen with a new password. I stripped my Asterisk server to one phone on Zap/1 until I get this working. The same BV account works fine with a SPA-3000 so I don't suspect a firewall problem. Symptoms: Asterisk registers with BV Ok Incoming calls work

I have had this imaptest[1] stall. I compared this command with what you have on the website and I was a bit surprised that what is on the website is working. What I have noticed is that if you remove this msgs argument, the imaptest stalls. I am not sure if this is according to design, but one might expect that specifying a secs, makes specifying msgs redundant? [1] imaptest - append=100,0