search for: rtucker

...llowing: [rochny-audiocodes1] type=friend host=208.34.86.37 dtmfmode=info secret=[mumble] context=inbound I've tried various settings for dtmfmode, there and in the [general] section, to no avail. Any help or troubleshooting tips would be appreciated. Thanks! :-) -rt -- Ryan Tucker <rtucker@netacc.net>

I have a problem that sometimes lines will go into what I call never never land. The Asterisk system will put a line with <Zombi> on it when you type show channels it will make the analog phone line dead. And on the CLI it says: astsvr*CLI>Zap/1-2<ZOMBIE>(macro-twoline-exten,s,1)Up Dial Zap/1-2|20|r I have tried to release it with soft hangup Zap/1 & also soft hangup

Hi, I am translating the voice files of voicemail now. I don't know if the POUND and # are the same key in the telephone's keypad. If they are same, how could we understand the following message: %vm-msginstruct.gsm%To hear the next message press 6, to repeat this message press 5, to hear the previous message press 4, to delete or undelete this message press seven, to quite voicemail

I have a Mediatrix 1204 FXO gateway setup for SIP. I would like to know if anyone has gotten this item to work with Asterisk. I need to get a 2 or 4 port FX0 gateway working with asterisk. The Idea is the following. PBX at lets say any Hotel(Analog lines FXS) - FX0 Gateway(1204) -- {Internet} -- Asterisk - local IVR system. (IVR is not at present running Asterisk old dialogic system has FX0

Greetings... I have a TE410P with four T1's going into it. Things look roughly like this: #1 Goes to PBX -- we're responsible for timing #2 E&M span to telco 1 #3 PRI span to telco 1 #4 PRI span to telco 2 If I set primary sync source to span 2, users report strange echo, distortion, and crosstalk problems, which sound remarkably like frame slippage on spans 3 and 4. If I set

Greetings... We've been having some interoperability issues between Asterisk and an AudioCodes Mediant 2000, and, well, I gotta use the Mediant 2000 somewhere. So, I've been pondering using iptel.org's SIP server (SIP Express Router) as a "front end" for PSTN calls going out to the Mediant, while using Asterisk for everything else. Has anyone done something similar, or

Does anyone have a quick and dirty script for defragmenting mailboxes? i.e.: -rwx------ 1 root root 80553 Oct 20 16:27 msg0000.gsm -rw-r--r-- 1 root root 218 Oct 20 16:27 msg0000.txt -rwx------ 1 root root 781164 Oct 20 16:27 msg0000.wav -rwx------ 1 root root 79360 Oct 20 16:27 msg0000.WAV -rwx------ 1 root root 7260 Oct

Since it's all the craze, I might as well post our current Asterisk usage. :-) EQUIPMENT: - Beefyish box (dual Xeon 2.4GHz, gig of RAM, more-than-adequate disk space, etc) in a 1U chassis. - A second, slightly less beefyish box of specs I don't have handy right now, also in a 1U. - 2xTE410P CONNECTIONS: - 1 PRI to telco for local outbound/direct-dial inbound, 300 numbers