similar to: 25-50-100fxs

Does any have experience with E1 telephony support plus asterisk in costa rica ? Regards, Luis Morales -- --------------------------------------------------------------------------------- Luis Morales Consultor de Tecnologia Cel: +(58)416-4242091 --------------------------------------------------------------------------------- "Empieza por hacer lo necesario, luego lo que es posible... y

Hi, I'm no phone guy but I've been having quite a bit of fun messing with Asterisk. I just unwrapped my new Wildcard TDM400P (TDM40B) which is the new 4 port FXS PCI card. It has 4 RJ45 connectors and I'd like to hookup 4 regular phones which are RJ11. If I crip RJ45 connectors to the phone cords do I simple use the center two pins (pins 4 and 5)? Thanks, Randy Smith Tiger

Dear asterisk users, We want setup TE121 digium board: Model: Digium TE121: VoiceBus technology allows the TE121 to use an industry standard bus-mastering PCI Express interface. http://www.digium.com/en/products/digital/te121.php My platform Server: HP Proliant 150 G5 OS: UBUNTU x86_64 GNU/Linux Asterisk: 1.4.21.2 zaptel: SVN-branch-1.4-r4662M When we enable zaptel driver for TE121, the

We currently have a 15-phone system using Asterisk, a combination of analog phones/Grandstream HandyTone-286 and Grandstream BT101s, and 4 X100Ps connected to analog lines. The system works well except for the occasional echo problem. I have all the echo parameters configured, removed all the extra incoming analog lines except to the PBX, etc. following all the advice on the wiki and on the

Does any know what happens with svn repository on svn.digium.com ? -- --------------------------------------------------------------------------------- Luis Morales Consultor de Tecnologia Cel: +(58)416-4242091 --------------------------------------------------------------------------------- "Empieza por hacer lo necesario, luego lo que es posible... y de pronto estar?s haciendo lo

This is a global message to all to announce our callcenter / dialer / predictive dialer / vicidial program is now open. Codecs: G711, GSM, G729, G723 Protocols: SIP Duration Rate : 30/6 (6/6 with monthly minutes over 100,000) Channels : 100 to start with , more on demand. We are predictive dialer friendly , your account will not be shut off. Contact us to do a test run. Mike

I am hoping to replace my Nortel 8x24 switch with Asterisk. Right now my cabling comes from my outside phone box into my office and into a punchdown block and leaves the punchdown block as an amphenol connector which currently plugs into the Nortel swicth. A second amphenol connector them plugs into the switch and extends to another punchdown block that I believe carries the lines throughout the

Hi All I'm having some serious kernel panic while using digium cards. It may be related to IRQ shared. Can this cause a lot of drop call and bad voice quality ? Do you guys know if there is a way I can assign one IRQ for each digium card ? Thanks a lot. Here is the output of /var/log/syslog kernel: [ 3821.982893] Uhhuh. NMI received for unknown reason 20. kernel: [

Hello Everybody! I am currently setting up an Asterisk server for medium to high load (approximately 20-35 concurrent phone lines). Do you think the following specs will sufficiently satisfy this system? CPU: XeonQC3220 2.4GHZ 8M RAM: 2X2GB/800 Harddrive: 1X250GB I could add harddrives and partition them into /var and /log directories to help with diskdrive throughput. Thanks! Elliot

Anyone have a cookbook on configuring sendmail to work with Asterisk? Or,a few config examples.

Hi all, I hope you guys can help me out. I got a problem with using function CURL. I did Set(CURL=${CURL(URL)}); but the URL I was using is https, so when I generated the call, the CURL function could not get access to that https://URL server. What should I do with it? Thank you very much

Dear All, I would like to ask please if someone has a AGI script that select a value from a database and dial this value as a destination number Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090220/e2aa530c/attachment.htm

Is there a way in the dialplan to figure out which agent in a ring all queue answered a line? I'd like to take specific action based on the agent upon hangup. Ryan M. Colbert Director of Information Technology Rissman, Barrett, Hurt, Donahue & McLain, P.A. 201 E. Pine Street, Suite 1500 Orlando, FL 32801 (407) 517-3105 - Direct Telephone (407) 839-0120 - Main Office (407) 841-9726 - Fax

Ok, I've pretty much decided to try the Adit route. Somebody who has experience with these tell me if I'm missing something. I have 15 incoming PSTN lines. T1 is not an option at current location. I want to put in an Adit 600 with 2 8-port FXO boards. The adit will then connect to * via a digium t1 board. I configure zaptel.conf for the T1. What other parts would be needed? How do

A customer of mine has 3 TDM400P cards in a box running asterisk. On each card he has four FXO modules. I have set up the dialplan to dial via group 1 for an outgoing call. Channels 1-12 are in group 1. If he plugs a telephone cable into socket 2 or 3 etc, but not 1, when he dials out, it still tries to make the call via socket 1. Straight away the console says that it has dialed the

I have a TDM400P with 3 fxs and 1 fxo ports. I need to know which phone connector corresponds to which module and also which port number. If we are looking at the card with the PCI connector at the bottom, fxs/o modules at the top and the phone jacks on the left - do the phone jacks start at the top or the bottom (top is port 1 or bottom is port 1)? And which phone jack belongs to which

Hi, Does anyone have a suggestion how I can analyze the concurrent usage of ISDN channels? A client complains about their clients sometimes getting a busy tone when trying to call them. I suspect they just need to add an additional ISDN2 line but I need some conclusive information that they are indeed maxing out their ISDN channels. Thanks, Patrick

If you have an FXS port and would like to attach a PSTN analog line to it this device would do the job by converting the FXS port to FXO. It's a small external device. Works well with VOIP FXS and other FXS interfaces. Interface: 2 RJ11 Jacks (one for the FXS port and one for the PSTN outlet.) Cost: $35.00 with USPS regular mail included. Power: 9 to 20V dc power supply (Not

Hello List, our setup: Callcenter IBM Hardware, 1x TE420, 1x xircom analog switch, 4x different cellular providers on the xircom analog port, ~60 agents Debian 5.0.1 (Lenny) Asterisk 1.4.21.2 Debian Package recompiled with additional app_queue segfault fix Zaptel 1.4.11 Debian Package My Problem is I have two channels (Zap/9-1 and Zap/6-1) which have a duration of over 4 hours. I am

Hi all, We have already bought 30 Digium G.729 licenses to install in several machines, but Digium only has provided on key to use in registration. We suspect that the registration has assigned the whole 30 licenses to the same server. Do anyone know how to distribute the licenses among several servers? Thanks in advance. -- Arturo D?az Contact me on FWD: 870436 Skype: arturo.diaz.almagro