similar to: Guide to better performance using * ?

How do I set configure my voicemail notification so that when I'm left a voicemail message it: 1) sends an e-mail to my inbox with the voicemail message attached 2) sends a message to my cellphone without the message attached I get notifications when I've got attachments turned off, but my cell doesn't like attachments in the messages and doesn't send them. An even better

Hello All, Recently I added some Nokia N95 customers and it worked pretty good. Now the customers are complaining about the dialing rules... They are used to dialing +12486543210 and +4479XXXXXX for long distance calls. Is there anyway to create a "+" sign dial plan which will allow them to dial a number with "+" sign. Cheers, Nitesh

Hi, I just want to ask if anyone has some experience with Alarmreceiver application in * 1.2? Is this application reliable (according to wiki it isn't)? I managed to communicate Asterisk (Alarmreceiver) with a burglar alarm, but it behaves very strange. Sometimes alarmreceiver is able to get some events but sometimes not. Maybe there are some other non commercial applications which work under

Using trixbox (or a custom dialplan if needed) has anyone been able to convert a number dialled like +612421100000 to something like 024221100000 ie (remove the +61 and replace with 0) i just dont know how to set it up, there seems to be no dialplan wildcard i can use to match +. I was thinking of something like .61XXXXXXXXXX but that still seems wrong to me. it could match other numbers.

I have a patton 1400 setup to handle the bri interface. As a trixbox user, I wanted a sip trunk rather than having to re-compile bri support into trixbix. Anyway, I have it working now so that asterisk can make calls and they are passed properly to the telephone network. Incoming calls however are another matter. I have (after turning on cli debug in the 1400) determined that its getting stuck

Thers a box we can get here in Australia that has an RJ45 plug, a built in web server that has a config page and URL's to close/open one of 4 included relays. I use phpagi to hit the url and open/close doors that way. If anyone is interested, ill let you know URL's but its being sold from our local Jaycar agent. Regards Kevin -----Original Message----- From:

Hey all - does RHES4/CentOS4 beta contain a recent integration of Subversion? I've been holding off trying to retro-fit CentOS 3.x on our CVS server, hoping that the official channels aready had it in store for the future. Not that you *can't* retrofit, I was just uncomfortable upgrading so many stock packages. Anyone have it installed, can take a look for me? thx, -te -- Troy

The recent thread about the mysql/postgres RPMS had me wonder -- is Redhat (ergo CentOS) moving to MySQL 4.x in the new RHES4 release? You folks with the beta, what's on the CD/ISOs? thx, -te -- Troy Engel | Systems Engineer Fluid, Inc | http://www.fluid.com

Another day, another apparant unexplained hardware incompatibility. I have a TE412P and a TDM400B living quite happily in a whitebox using an Intel motherboard: http://www.intel.com/design/servers/boards/se7230nh1-e/index.htm I tried to move to an IBM x3650 system. It uses a slightly newer chipset, but apparantly it's in the same family. The SE-7230 board has been EOL'd and the

Hi, I was able to implement T122p one port PRI and was able to call out, but I am planning to use TE412p (includes echo cancellation) 4 port digital card (PRI), I wanted to know can asterisk support 3 four port PRI cards (12 PRI connections) with proper hardware like dual core quadcore processor and 8gb RAM in one server? Also I was planning to implement using 64 bit architecture with Asterisk:

More than 60% of our outbound calls are now to mobiles, so the time has come to whack in a gsm channel bank. Does anyone have any preference of bank ? Do you use a PRI or VOIP connection from the bank to asterisk ? Real-world experiences are sooooo much better than marketing blurb ;) We currently have a TE412P with a free socket, so we have a choice either way. I am looking for up to 30

I have just upgraded my Polycom 501's from 1.6.2.0041 to 2.1.0.2708 to get the microbrowser. Almost everything is fine except when receiving calls from a BT200 (1.1.14 and earlier) the Polycom rings but when answered, drops out and the BT200 gets a busy tone. I have many PAP2T's and SPA3000's etc and they all cal call the Polycom without problem. Does anyone know what could be going

I am trying to solve the fax problem by installing an E1 channelbank (Megaplex MP-104) It's a box that has 8 x FXS ports and a single E1 port. The plan was to use one of my 4 E1 ports to connect to the Telstra onramp and one to the MP104. I have since discovered however that the MP104 only supports CAS signalling and I am having trouble getting asterisk to work in this mode. Currently we

Just to share some experience with everyone about what happened today to our Asterisk 1.4 box with Digium TE412P card. We had an unscheduled power outage which shut down the Asterisk box. When the power went up, Asterisk came back up okay but the ports on the card were all red. Zttool show red alarm and cat /proc/zaptel/1 show red alarm today. Both incoming and outgoing cannot be made. When a

Greetings list, I've been asked to provide a system for 200 extensions, most of which will be existing analogue POTS handsets, not IP handsets. I've not really had any experience with large channel banks in the past (since most of our deployments are strictly IP-only to the desk), so I'm at a loss as to which ones are worth looking at. If anyone's had experience using channel

Hello list, I'm currently looking into building a new Asterisk server, due to some codec problems i've got to transcode most of my channels between Alaw -- G729. Is there any indication on how many channels you would be able to transcode on a certain platform? I'm looking into dual Xeon or dual Opteron configurations, which of these platforms would perform better? And how much power

Can I run two instances of asterisk sharing a single te412p ? I want to be able to have several asterisk servers (for testing various scenarios) running on one server. I was wondering if these asterisk processes could share a zaptel/dahdi card nicely. Julian

Hi all, We need to get better echo cancellation on an Asterisk gateway. Currently it has two TE410P (1st gen) cards. So would it be possible to just buy two VPM450M cards ? Or do we need to buy two new TE412P cards ? In that case a Sangoma A108d card would be nice as well ? What configuration gives the best audio quality ? Thanks, Leon de Rooij leon at scarlet-internet.nl

Hi, I just read a pressrelease from VON that Digium will soon be releaseing a couple of new cards. What got me interested was: "The TE420P and TE415P support 128ms of G.168 (2002)-compliant echo cancellation across their entire 128 channels." Does anyone know when thease will be released and what they will cost when released? Thanks!

Hi I am not sure why this is happening or whether it has anything to do with my iaxmodem setup. When receiving a fax via iaxmodem, I got an error message saying *chan_iax2.c:7542 socket_process: Call rejected by 127.0.0.1: Busy*