similar to: Dialout from SIP to PSTN

Hi all ! Have anyone a resource / link for documentation to configure Asterisk to act as a SIP 2 PSTN gateway (ISDN PRI) ? Thx. Regads, Andreas. -- "If you want to pray. Go to the sea." ---------------------------------------------------------------- Andreas Czerniak <cognac@amcs.net> PGPkey http://pgp5.ai.mit.edu:11371/pks/lookup?op=get&search=0xEDB224EC

What I've got... Software: Linux: Slackware 9.1 Asterisk: out of CVS - so its new. isdn4k-utils: to test the ISDN Card Hardware: PII Pentium 400Mhz (Its a test of concept machine) with 320Kb RAM 1 x ISDN BRI Card - DIVA EICON (Installed + working) 2 x Grandstream (Barbie?) BT100 SIP Phones. What Works.. I can call from one phone to the other... get read voicemail... I can

The card is an ASUSCOM ISDNLink PCI (passive) and the circuit is from Qwest (in the US). I will be using this circuit only for voice (I'm doing this because of the poor quality of my POTS lines). I've compiled Hisax (as a module) into my 2.4.25 kernel, and with 'modprobe hisax type=35 protocol=4 id=hisax' I get the following: Apr 7 10:34:24 dev kernel: HiSax: Linux Driver for

Hi, Having seen the messages on the dev list a couple of weeks ago about chan_sebi I thought I would try to get it going on my system. I am using 1.6.1.4 so I first upgraded the driver to work with 1.6. I think it is mostly correct and the bit I have wrong shouldn't be causing me the problems I am seeing. If your are interested my chan_sebi 1.4 to 1.6 patch is at

Here is the output of R -d gdb: [emile@bb emile]$ R -d gdb GNU gdb 4.18 Copyright 1998 Free Software Foundation, Inc. GDB is free software, covered by the GNU General Public License, and you are welcome to change it and/or distribute copies of it under certain conditions. Type "show copying" to see the conditions. There is absolutely no warranty for GDB. Type "show warranty"

Hi all, I tried samba 4.8.3, 4.8.4 and 4.8.5 to build a domain. In the domain I firstly create a machine acconut and set it's password. Then I get a computer that own this machine account's name. I use the mechod NetJoinDomain to get this computer authencated to the domain. It failed with returncode 1326. Besides, all the process above is avaliable in samba 4.5.16. So does any default

Hi, I'm not the maintainter for the "megatec_usb" driver (in fact, I'm no longer maintainer for the base "megatec" either), so I don't really know the status for that particular UPS. I'm forwarding this to the NUT mailing-list though. Regards, -- Carlos Rodrigues On Fri, Dec 5, 2008 at 10:30 AM, Adrian Czerniak <adrian.czerniak at becomo.com> wrote:

Hi, I'm not the maintainter for the "megatec_usb" driver (in fact, I'm no longer maintainer for the base "megatec" either), so I don't really know the status for that particular UPS. I'm forwarding this to the NUT mailing-list though. Regards, -- Carlos Rodrigues On Fri, Dec 5, 2008 at 10:30 AM, Adrian Czerniak <adrian.czerniak at becomo.com> wrote:

Send CentOS-announce mailing list submissions to centos-announce at centos.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.centos.org/mailman/listinfo/centos-announce or, via email, send a message with subject or body 'help' to centos-announce-request at centos.org You can reach the person managing the list at centos-announce-owner at centos.org When

I haven't managed to work this one out yet, so any assistance appreciated ... We want to be able to set the outgoing caller-id on, BRI according to the extension but haven't worked how with asterisk ? we have several hundred inbound numbers on these BRI so we are able o use any one these to sett on outdial. One other point I have been told should work, bu have no way of trying... In

I've been reading the Linejack strikes again messages, and have another Newbie question is it possible to use a Voip Product as a Dialout line for * ? I have a Vegastream 100 Voip to PRI. box. With * can I use that as a Dialout / dialin box? The Vega100 does either sip or h.323. Thanks. Bradley Greep

Hi there, I?ve been out for some months now, haven?t been checking the list at all. Does anyone know if the problem with the Quicknet Linejack (FXO) card dial out to PSTN with asterisk was solved? Is anybody working on it? Cheers, -Z -- __________________________________________________________ Sign-up for your own personalized E-mail at Mail.com http://www.mail.com/?sr=signup

I need to make automatic voice calls from a Linux server, so when the system receive special "signals" it must use a wave (or .au) audio file, dial the number to call a person, and "speak" using the audio file. What can i use for this subject? I need a specific hardware device or a normal analogic voice-modem is ok? which software can i use (i need to invoke from a Perl

Hi, When I try to dialout over chan_capi everything works fine when I settle for msn=* in my capi.conf and use the primary msn of my ISDN-line. But trying to configure a different MSN the chan_capi doesn't dial and comes with: No one is available to answer at this time What can be the prob? -- Thanks, Marc aka IzNoGood

Dear all I read on the list back in 2003 that * does not support IXJ LineJACK dialout yet is this still the case? Thanks

If you set "dialout=fromvm" in your voicemail.conf, how do you then go about being able to dial back out? Is there a service feature code?

Hello! How can one select outgoing MSN when dialing out from ttyI-interfaces? I have successfully done this with CAPI e.g... exten => _XXXXX.,2,Dial,CAPI/60:bBYEXTENSION ...in extensions.conf. Currently correponding for my ISDN modem interface is... exten => _XXXXX.,2,Dial(Modem/g1:${EXTEN}) ...but this selects only MSN of outgoing group g1 for dialout MSN number. I also tried to

Specs FC2, Asterisk 1.0.0, Zaptel 1.0.0 TDM400P Port 1 FXS Port 4 FXO Standard analogue handset plugged in with pstn line. Problem: When I go to dialout it drops numbers on the outgoing number. Keys dialed from handset were 9 0418800185 I tried hitting the keys slowly as well as at my normal speed, all tones are heard in the handset for all numbers.

Hi List, I have set up the following in my extensions.conf ; local numbers look like 0262XXXXXX ; but must be dialed 262 262XXXXXX exten => _0262XXXXXX,1,Dial,IAX2/543@voipjet/011262262${EXTEN:4} exten => _0262XXXXXX,2,Dial,IAX2/jhiver@NuFone/011262262${EXTEN:4} exten => _0262XXXXXX,3,Congestion It did work for a while, however when dialing I get: stargate*CLI> -- Executing

Hello asterisk-users. I have the following dial-plan: [test] exten => 482,1,Dial(OH323/106@192.168.2.73,10) exten => 482,n,Dial(OH323/102@192.168.2.73,10) exten => 482,n,Dial(OH323/103@192.168.2.73,10) exten => 482,n,Dial(OH323/104@192.168.2.73,10) exten => 482,n,Dial(OH323/105@192.168.2.73,10) exten => 482,n,Dial(OH323/106@192.168.2.73,10) When I call exten =>