search for: courtnag

...t-transfer a caller to another extension. If the 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

...all answering options. - Queues can now play a "welcome" message to callers upon joining. - HINT priorities for FIXED devices - Interface translated to French, German, Italian, Spanish - FOP .21 - FOP button layout can now be sorted by last name or extension number Regards -- Ryan Courtnage Director & CTO Coalescent Systems Inc. Enabling Open Source Telephony 403.244.2790 www.coalescentsystems.ca

...operator - provide customization points for all AMP generated extension contexts. - Upgrade to Flash Operator Panel 0.20 - Upgrade Asterisk-Stat to v2.0 - Added cvs2cl generated ChangeLog (see this for all changes and bug fixes) Regards, Ryan _______________________________________________ Ryan Courtnage Coalescent Systems Inc Director & CTO Enabling Open Source Telephony 403.244.8089 www.coalescentsystems.ca

...that a nightly restart of the server helps. And 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

...51:28 WARNING[-1275102288]: chan_sip.c:590 __sip_xmit: 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

...2/3/4 in your case say. I have X100P's and I seem to be having similar sounding problems. I noticed that the above command shows the channel to be off-hook at all times when a phone line is plugged in. I don't know why or if it is a bug in the application reporting the status. dbc. Ryan Courtnage wrote: > On July 8, 2004 03:22 am, Nicholas Bachmann wrote: > >>Ryan Courtnage wrote: >> >>>Hello, >>> >>>Over the past several weeks, we have been having an intermittant problem >>>with our deployment of a TDM400P card (4 fxo). We have tried...

...t on the tdm400p. I have 2 theories: #1 - the ring voltage for some calls is below acceptable levels #2 - the tdm400p card is bad Assuming #1, can the zaptel driver be tweaked to be more sensitive to ringing? Any other ideas or experiences? Running asterisk/zaptel v1.0.2 Thank you, -- Ryan Courtnage Director & CTO Coalescent Systems Inc 403.244.8089 www.coalescentsystems.ca

...* (ie: vmail) do work (using rfc2833). I'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

...to home/cell without tying up 2 lines. I haven't seen options for 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

...ead, a couple of people suggested that this does not work reliabley, 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

Hello, I've found a couple of previous posts on this subject, but with no posted resolution... I'm attempting to run * as a non-root user (asterisk), following the guidelines on the wiki: http://voip-info.org/tiki-index.php?page=Asterisk%20non-root I can run * as my new user with "/usr/sbin/asterisk -vvvvc" without problem. However, I'm unable to run * using

Hi All, I'm using an X100P to connect to PSTN ( context=from-pstn ). I'm trying to get fax detection to work. Using the simplest dialplan, I cannot get * to detect fax tones: [from-pstn] exten => s,1,Answer exten => fax,1,Goto(ext-fax,999,1) The fax is never detected (ie: Goto never executed) All that I see on the * -vvvvc console is: -- Executing Answer("Zap/1-1",

...n 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 of these issues is appreciated, Thanks RC 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.

...these hangups? On each of the incoming analog 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

Hi all, I've been using the following method to limit calls to sip clients to 1: exten => 200,1,SetGroup(200) exten => 200,2,CheckGroup(1) exten => 200,3,Dial(SIP/200) exten => 200,103,Busy This works fine for a single extension. However, I also need to dial groups of sip clients. It appears that SetGroup can only be used once per channel. This (useless) example would not

Hi all, Theoretically, I know it's possible, but is any using multiple tdm400ps (fxo) in single * box? In a production environment? Any gotchas aside form irq sharing? Thanks Ryan

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