similar to: BudgeTone hold?

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

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

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

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

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 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: --

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

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:

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

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

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

I have a P3-800 with two IDE drives in a software RAID1 configuration. Each drive is on a separate IDE channel. Now anytime there is HD activity, I hear "beeps" and "cutting out" on a call using the X100P card. I ran the zttest program, and discovered HD activity would drop the accuracy down to between 2% and 50%. However I noticed if I disabled one drive in the RAID1

I have a strange problem in that I have put a budgetone out on the internet that connects to my * server that's behind a firewall. They can call me I can call them and it works fine. However, I have setup a link to sipdiscount on my * server. If the budgetone user calls via my * box to sipdiscount all the budgetone user hears is silence and the called person hears silence as well when they

Hello, I have been working on getting Asterisk up and running with a couple Grandstream hard phones (Budgetone 100s) and a couple softphones. Things work pretty well but we have discovered an odd problem. A calls B and then A can put the call on hold and pick it back up. A calls B and B can put the call on hold but cannot pick it back up. These tests are internal on the hard phones. The

I have two problems with Asterisk and my Budgetone 100s. 1. Sometimes when outside callers dial my Budgetone it will ring twice, then the caller will receive a busy signal. This happens intermittently and I cannot pin down the circumstances of when it works and when it doesn't. Asterisk logging doesn't give me too much info when this happens: *CLI> -- Executing

On Jun 20, 2004, at 9:16 PM, asterisk-users-request@lists.digium.com wrote: > I am new to Asterisk, but though that I would need calls being > answered in > different contexts. > > How can I direct one line to a given context and the other one to > another, > or is there a better way??? > I use this at the end of my zapata.conf file to do exactly what you're

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 =>