search for: voxbox

...transfer target party is unavailable (ie: voicemail), there appears to be no way to get the original caller back. If it's a known limitation, has anyone come up with a functional work around? Thank -- .................................. Ryan Courtnage Coalescent Systems Inc 403.244.8089 www.voxbox.ca

...mit: sip_xmit of 0x80f680c (len 465) to 66.18.203.117 returned -1: Bad file descriptor My question is ... "Is there a better way to do this, without the use of STUN or proxy servers?" Thanks -- .................................. Ryan Courtnage Coalescent Systems Inc 403.244.8089 www.voxbox.ca

...y reproduce a problem where my client's * box cannot receive a fax sent from Windows-based HotFax. HotFax hangs up with error "Phase B Error". Watching the verbose * console, it appears that there is some useful communication between spandsp and the HotFax (output is here: http://voxbox.ca/tmp/rxfax.html). Oddly enough, faxing between HotFax and spandsp _used_ to work, before I switched her * box over from an x100p to a tdm400p+fxo (although I can't be 100% certain the card change is to blame, since zaptel and asterisk have also both been updated). Any comments on either...

...ve been able to duplicate these problems both on a customer site, and in my own test environment. Has anyone else using the UIP200 experience any of these issues? Still waiting for Uniden Support to return my call. Thanks .................. Ryan Courtnage Coalescent Systems 403.244.8089 www.voxbox.ca

hello is there a gui that would allow me to configure everything from phones, to extentions, to voice mail to basicly everything that asterisk can do? I did go to www.voip-info.org and none of the guis I saw there do the trick and the ones that come close aren't downloadable just wanted to see status on this thanks hank ---------------------------------------- My Inbox is protected by

Hello all, The "Secret Agent" final release of the Asterisk Management Portal is now available for download: http://amp.coalescentsystems.ca/ This exciting new release adds a great deal of functionality and flexibility. Thank you for all the contributions and feedback! 1.10.007 - Added AMP Users (multi-department, basic multi-tenant) - Added incremental upgrade script

...at least one other, is just like me and struggling to find a solution ;-) Digium support has yet to find a solution. I'm really out of ideas here - I'll take any bone you can throw me. Thanks -- .................................. Ryan Courtnage Coalescent Systems Inc 403.244.8089 www.voxbox.ca

I compiled libtiff version 3.6.1 and spandsp and spandsp version k. When trying to load asterisk I get the folloein error: Jun 7 10:15:03 WARNING[16384]: loader.c:408 load_modules: Loading module app_dtmftotext.so failed! Ouch ... error while writing audio data: : Broken pipe [root@zapata root]# Warning, flexible rate not heavily tested! Please help! -- Manuel Marin Garcia TRANSTELCO S.A.

...og lines, there are splitters to unused analog phones ('bat-phones' .. just in case) ... is it possible that this is the source of the problem? Zaptel from 2004.06.14 CVS. Asterisk from 2004.06.02 CVS Thank you -- .................. Ryan Courtnage Coalescent Systems 403.830.9410 ryan@voxbox.ca

I have Asterisk CVS-HEAD-05/25/04-17:13:22, Copyright (C) 1999-2004 Digium. Usign exclusively digium hardware. 3 TDM400P cards. 1 4xFXO 1 4xFXS 1 1xFX0 & 3xFXS When * is attending FXO calls, bridged to FXS calls, natively ofcourse, at a random time, the call hangus up. Also, for example, if a call is done, and an other extension hangup, there are some probability that the other extension

Hello, Since outgoinglimit is EOL'd, I've implemented SetGroup/GetGroupCount to ensure that SIP clients will only have a single call at any time. Works perfectly for simple calls using Dial(). I'm now struggling to find a way to similarily limit 2nd calls to SIP clients that are Agents, who receive their calls from a Queue(). Is there any way to accomplish this (without writing

Does anyone know how (if possible) to do three way calling on the UIP-200? There doesn't seem to be much info about this phone, but all the feature lists I've read says it can do conference calls. I can't seem to do it, though. Any help would be appreciated. -- Seth "et lux in tenebris lucet" Mattinen sethm@rollernet.us

...or zapata.conf that suggest this is supported on fxo interfaces. If it's not supported, is this something that could be achieved via changes to the zaptel drivers (without re-engineering the card/modules)? Thanks -- Ryan Courtnage Director & CTO Coalescent Systems Inc 403.244.8089 www.voxbox.ca

...and the ATA FXS <--> TDM FXO link 'goes dead'. Has anyone had any measure of success doing this? Primus' service is becoming very popular in Canada, and some customers are wanting to do this. Thanks -- Ryan Courtnage Director & CTO Coalescent Systems Inc 403.244.8089 www.voxbox.ca

I have a TDM400P card all installed, and I can call out from a sip phone, but when I call from the PSTN into the asterisk machine, as soon as the Answer() gets called, the dial tone changes and is sounds like there is a lot of static on the line. Below is the part of the dial plan for answering the call. exten => s,1,Wait,1 exten => s,2,Answer exten => s,3,Dial(Sip/pfriedel,20,tT)

I'm trying to set up basic monitoring for a specific extension (5004) to record all outgoing and incoming calls and save them as WAV files. I've set this in the extensions.conf file: exten => 5004,1,Answer exten => 5004,2,Wait,1 exten => 5004,3,SetVar(CALLFILENAME=/var/spool/asterisk/MONITOR-${TIMESTAMP}-${CA LLERIDNUM}) exten => 5004,4,Monitor,wav|${CALLFILENAME} But it

Hi all, I know that this has been passed around before, and I know that it happens about every 3 months or so, but evertime the answers change, so I thought I would pass it around again. A company I work for has 3 incomming lines and 4 phones. They require voicemail and MOH. Their phone systems VM hard drive died today, they were quoted a $2000 to replace it. I started to talk to them about

I'm finishing up deploying an Asterisk (Trixbox) box at work. Wow, I thought Asterisk was cool by itself, but Trixbox has made just about everything turnkey. Great stuff! So... we're using Grandstream GXP-2000 handsets to connect to the Trixbox, which sits on our DMZ with a single public IP. I need the phones to work from random places behind NAT, as well as in the office. I'm using

Hello all, Can I execute linux command like(ls, mkdir) through the Manager interface? I can't seem to access the manual at digium.com. I keep getting 'Forbidden' error. Looks like they are upgrading or something. CF

I got a Uniden UIP200 and started to configure it and I am lost.... I have a tftp server setup on my * server and have the files unidencom.txt and uniden<mac>.txt there. But it doesn't quite work yet. It registers as a sip phone (sip show peers), but I cann't dial it and the display shows #1 disconnected all the time. It has firmware version BS4.59a in it. I have no idea if I