similar to: Asterisk and dial-up modems

The configuration I'm building replaces an existing PBX with Asterisk. There are 8 existing modems that people use to call in from the outside to connect to PC(s) on the inside to transfer data, etc. Callers access these modems by calling the main number and then dialing an extension for the modem they want to talk to. What are my options for supporting these modems with Asterisk? Here are

Guys, I have a customer that heavily uses modems, the problem they don't work reliably with some of the carriers I have used like Level3. This is somewhat expected due to the limits in VoIP so I need a better solution. If I set up an asterisk system on customer premise with an FXS card in it and have calls sent to another asterisk box with a PRI can I get this to be more reliable and better

I'm trying to use ZapRAS to enable ppp connection through my E1. After the ZapRAS command is executed, all sound is crappy on all lines! The only solution is to reboot the machine (or halt it, and then power it on since Digium's hardware doesn't like reboots). Anyone know how this can happen?! I'm using * 1.0.6 on Dell PowerEdge 1850 which are told (too late though) not to work

Don't buy this book for its content. It's a waste of $40. However, it is useful to wave in front of my customer's faces to show them that Asterisk is real. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040724/5ccf4012/attachment.htm

Hi! My application needs to look up (by spelling) the first and last names of a person and then insert the corresponding pre-recorded audio file to personalize the message. E.g. "Hi, John Brown. Your book is due back at the library." So I need "John" and "Brown" in audio files along with LOTS of other names - Do such databases of sound files already exist or do I

Hello, I was wondering if it was easy to setup a dialup internet account system in Asterisk somehow(like PPP) laptop with modem -> PSTN line -> Asterisk server -> Server's broadband internet connection Has anyone done this? Thanks, MATT---

Hi! I want to connect my asterisk system to the PSTN. Now I'm thinking about using an analog modem as FXO device. Which modems do run? How is the voice quality? (I looked into the wiki, but on <http://www.voip-info.org/tiki-index.php?page=Asterisk%20hardware> no regular modems were mentioned. Or should I use a special FXO-card? I don't want to spend 99$ for such a thing but

I upgraded to the latest stable version of 1.0 today and am still seeing the *8 problem where the phone that was originally dialed keeps on ringing even after another phone picks up. Are other people also seeing this? Has somebody figured out how to make this go away? Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, We are working on implementing a solution for a medium service provider. They were previously using a Cisco AS5300 gateway with some PRI trunks to receive modem calls, then route them out the Internet. The Telco they were buying the trunks to discovered this configuration and restricted them due to legal conventions, and stated that in order to continue doing this, they would have to talk

Hi, I'm trying to get ZapRAS working but not getting very far.. Asterisk CLI shows: WARNING[3355]: app_zapras.c:173 run_ras: wait4 returned -1: No child processes and /var/log/messages shows: using the plugin option requires root privilege Can anyone shed any light on this and any fix? Googling the error doesn't find much.. I'm not sure what 'plugin' it is talking about,

hi all Where can I find documentation about how to setup ZapRAS? What I want to do (optimally) is to allow for automatic dial-up to external sites, each having an ISDN router. Today we use a small ISDN router for this, but it'd be a lot better, IMHO, to have asterisk do this (functioning as a ISDN router), as we may cancel our BRIs then. Is this possible? And if so, how can I do it? I

My need to do this through asterisk is simply the ability to provide me access with no additional cost to my customer. It seems like a nice thing to include as long as authentication is done well. I have worked on a dozen or more types of switches and all of them have supported this or had the capabilty through hardware or licensing. I am trying to get around opening and closing the firewall,

hi! I'm trying to get zapras working in GSM csd network. Whenever a dialup call is initiated from the mobile to the * gateway the following appears in the log and zapras terminates. Phone gives the error dialup not answered. ==> /var/log/messages <== pppd[2310]: Plugin zaptel.so loaded. pppd[2310]: Zaptel Plugin Initialized pppd[2310]: Using zaptel device 'stdin' pppd[2310]:

Hello all! Sirrix AG, Saarbr?cken (manufacturer of the Sirrix.PCI4S0 4-port ISDN card for Asterisk) has released the new Asterisk dialplan command PPPD (app_pppd). It allows to connect a Linux PPP daemon to an arbitrary digital (ISDN) Asterisk channel to provide RAS dialin and dialout. The PPPD command has successfully been tested with Sirrix.PCI4S0 cards and a standard ISDN4Linux ipppd "on

I have a pri comming into a t100p in my asterisk box. I have a second t100p configured as pri_net connected to a nas server. I can route modem calls to the NAS with no problem, but I am concerned about isdn data connections. Will asterisk route 64k isdn calls between 2 t100p's? Also, should I expect worse modem connections than normal? Chad Sawyer -------------- next part --------------

Hi All, I'm about to setup an Asterisk install to take over an old legacy PBX system. At present, the legacy system has modules in it which provides 4 * data ISDN links to the video conferencing unit (Tandberg 3000 MXP) on site, these use the ISDN30 (uk) that the normal voice calls go over. Is it possible to emulate this in asterisk? I've seen zapras but I'm not sure if that's

Hello my name is Ilija Poznic and I have a problem. My configuration is 1. Digium TDM4000P with one FXS. 2. AVM Fritz ISDN adapter (configured with capi). When I connect to my ISP and then start *. Asterisks is registering me to SIP provider iconnect. After that I can call international call trough VoIP. My problem is that I want to dialout to ISP only when I have a international call.

I''m looking at an example where a stub seems to work sometimes, and sometimes appears to become "unstubbed". I haven''t boiled it down to a minimal example, but it goes something like this: ---- the model: class Premise << ActiveRecord::Base def lookup_stuff_on_the_web $stderr.puts("entering lookup_stuff_on_the_web with #{self}") ... end end

Hi, Last weekend I was planning to buy a physical PBX system, but instead I have been blown away by the fact that VoIP really works, that Asterisk is so easy to set up and use... and free! We're in Australia, so as I understand it, we aren't allowed to use the Zaptel cards. We need to set up our system to route incoming 56K data calls to the PPP daemon on our Linux box or to our

I've configured my [*] to dial the pstn which is working like a charm. I've also configured an extension to ring when the PSTN line is ringing which is also working brilliantly, but, sometimes it doesn't detect that the call has been hungup. I've had a look on voip-info and checked the conf files but I can't see anything that I've missed. Cheers Matt