similar to: Basic Asterisk capabilities question

All, is anyone using Bayonne in conjunction with Asterisk? I'm currently using only Bayonne, but I'm investigating the possibilities of switching the telephony frontend over to Asterisk, and have Asterisk route the IVR tasks to Bayonne through H323. Anyone care to share his views on this approach? Any pointers or do's and don'ts? All info is greatly appreciated! Regards,

I found an old Dialogic card in an abandoned PC, that I think is a Dialogic D/41EPCI based on some googling.. The lspci output says: 00:09.0 Bridge: PLX Technology, Inc. PCI <-> IOBus Bridge (rev 01) Subsystem: Dialogic Corp: Unknown device 0529 I'm just getting started with Asterisk to build a SOHO system. Any suggestions for what I can do with this card? It would be

Hello, I had two quad-E1 Tormenta2 cards in my system and yesterday i had to replace the CPU. I noticed the rotary switch on each card that sets the card number or ID... Both switches were set up on 0 and everything worked fine. Should i set one card to 0 and the other one to 1, or... "if it works, don't fix it!" ? I had some nightmares with IDs on Dialogic hardware with Bayonne

Hi, I'm trying to connect a PC with a TE410P to an E1/IP equipment. Unfortunately I keep getting a yellow alarm from zaptel (in zttool) and a Loss of Framing alarm on the remote equipment. The E1/IP is connected on the other side to a PRI interface on a GSM MSC. I have configured the span as: span=1,1,0,ccs,hdb3 (also tried span=1,0,0,ccs,hdb3) and the channels as: bchan=1-15,17-31

Hi i have an application developed with bayonne. Recentely i'm experiencing some problems and i am planning to migrate to asterisk. I would like to know if i can do these things whit asterisk: - IVR integration with database (mysql, insert,delete,update,select) - TTS - record exploration (for example, check if some resources are available in the database, and list them to the user (via

I was trying to do a little searching to see if there has even been a comparison between Asterisk and VOCAL or any of the other OSS packages? "Practical Voice Over IP using VOCAL" published by O'Reilly and Associates, attempts to make a strong case about how scalable VOCAL. Of course, considering that the book is written by the makers of VOCAL, it tends to have a one sided slant.

Hello, We are trying to deploy a new asterisk server with a Wildcard T400P (quad T1) card. It uses a custom voice recording app written in the perl AGI. Now that the machine has been in production, it seems to lock up within 24 hours of reboot! When it locks, we can ping the machine, but we cannot log in using telnet or ssh. Asterisk stops answering the phone and our Big Brother monitoring

Hi everybody, I need to know about sound quailty issues from those who have experience with Tormenta2 PRI Interface. Also how to make it work with new versions of Asterisk and Zaptel. And also suggestion if it is a good idea to switch to some newer card from Sangoma or Digium, or Tormenta should work fine. I have sound breaking many times a day over the trunks. Server is AMD Athlon 2.4 GHz with

Folks, I love Asterisk, have been using it for a while now. I'd like to know if anyone has some good comparison points on Asterisk vs. Vocal (Vovida) vs. GNU Bayonne. I know only a little about the later two. Also, one drawback I've hard about Asterisk (not for me, but for general consumption/deployment) is easy of configuration -- people like GUIs. They want point-n-click. I'm a

Hi, I using asterisk 1.2.4 on a CentOS with Linux 2.6.9-22.0.2.ELsmp kernel. I've two x100p cards connected, only one card is reconigzed by asterisk. 02:01.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface 02:02.0 Ethernet controller: Davicom Semiconductor, Inc. 21x4x DEC-Tulip compatible 10/100 Ethernet (rev 31) 02:03.0 Communication controller: Tiger Jet

Anyone care to elaborate on the differences between the T400P and the TE405P? -Darren -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050908/e817f8bb/attachment.htm

Hi, After upgrading to new version of Asterisk and Zaptel, my Tormenta2 card has stopped working. Zaptel doesn't detect it. What should I do to make it work again. It is enabled in /etc/sysconfig/zaptel. Where else I have to enable it so zaptel can detect it and make it work. zaptel.conf and zapata.conf are the same as I had them before. I tried googling but couldn't find any help. --

Hi List, Asterisk 's *ControlPlayback* will used for play any recorded file as an audio player. Is it possible that we can use it for multiple forward and rewind ? ex:- original: ControlPlayback(filename,skipms,ff,rew,stop,pause) expected ControlPlayback(filename,skip1,skip2,skip3,forward1,rewind1,forward2,rewind2,forward3,rewind3,stop,pause) : ----- Thanks and regards Virendra Bhati

I am projecting to migrate applications written with the Dialogic API to Asterisk. There are a few things that I do with Dialogic that I am still not sure are possible with Asterisk : - play a file from an offset (so a user can press a key to rewind 3 sec., pause, etc.) - dynamic volume control (the user can press a key to increase or decrease the volume) Thanks if anyone has a clue... Yves

Im trying to create a custom application via the AGI. I want to authenticate the users that dial in with a userid and pin. However, the number of digits in the PIN and userid are variable, and therefore I need to allow the user to "press enter" by hitting the pound key. How would I accomplish this in the AGI? stream_file doesnt seem to work, since it only allows one digit to be

Basically, have an old IVR application running under Apex's Omnivox software on a box with 4 old intel dialogic D240SCs, and would like to allow remote clients to gain access to aforementioned IVR application via softphone, 7960, ata, etc via asterisk with a TE410P. Unfortunately, all I know about the dialogics is that they're configured for D4/AMI, which I believe I have configured right

I upgraded from Tormenta2 to Sangoma A101. I followed the instructions, and installation was successful. zttool, ztcfg, all show card is installed properly. I copied the parameters from my old working zaptel.conf, zapata.conf and zapata-auto.conf. Verified on Sangoma website that these files are correct. Also configured wanpipe1.conf. But doing all this didn't start the PRI channels. It says

> -----Original Message----- > From: asterisk-users-admin@lists.digium.com > [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of > Gary Franczyk > Sent: Thursday, January 15, 2004 10:37 AM > To: asterisk-users@lists.digium.com > Subject: RE: [Asterisk-Users] Basic Asterisk capabilities question > > > Whaaa?? So, to allow 24+ lines of dial in access, how

ClueCon is coming in 2 weeks so we urge everyone who plans on attending to register today so we get a proper headcount! ClueCon was put together by Asterlink, the same team of people who helped shape Asterisk into what it is today by writing features, fixing bugs, offering IRC support and assisting with the management of the development effort. We have produced several real-world solutions based

Hi there, Regarding the calling card system, I would like to know what software tools are suitable to write the CDR system? 1) Perl (but what kind of database should be used?) 2) PHP + MySQL 3) or others suggestion? Also, which one is easier to access AGI? Any source and information would be appreciated. Best regards, Max Chow -------------- next part --------------