search for: krisedward

Is anyone running Lumenvox on Gentoo? My asterisk install has been running like a champ for a few years now and I really hate the thoughts of changing distros just for Lumenvox. Here is my issue: The engine needs the libs from boost. I emerged boost and noticed that there were four libs that the engine were looking for that were not installed via portage. libboost_regex.so.2

Compiled from CVS today and no more dropping outbound calls after 40 secs. :D (was using cvs from 3 days ago) (just thought I'd pass it along in case anyone is still strugling with broadvoice calls)

I've seen some posts about ppl using gnomemeeting via oh323, but is anyone using it w/ sip?? (only their cvs supports sip, but I figured somebody was trying it.. I'm grabbing it now :)

How ISPs May Quietly Kill VoIP Communications Posted by Zonk on Friday March 18, @10:59PM from the poor-kids-never-had-a-chance dept. ravenII writes "PBS's i'Cringley's informative piece gives an eye-opening look at the anticompetitive behavior of some ISPs who are showing up late to the VoIP game. This is not something that could be easily mandated, and the beauty of this

I've noticed that if I load the alsa drivers, I can't transfer from the console.. Is that normal? It says the app is loaded, but when I type it in, it says there's no such application 'transfer'. I typically leave my console up on a transparent shell on my desktop and use that instead of a sip or iax client connected to it (I don't have a spare box to run asterisk on).

I have asterisk running on my personal computer and am using Kphone to connect to it. My provider is broadvoice which is Ulaw and I had kphone connected as GSM. The lag was terrible coming from Pots-->--Broadvoice-->Kphone. About 2.5 seconds! Going the other direction seemed fine. I did a: show translation recalc 200 and see that the translation time should be about 2 ms. When I do the

I actually got rid of my phone service.. no more pots line in my house. But, I miss my call screen macro. Any way to do this with a SIP channel? (Obviously the parking isn't the problem, but rather recording their name). I set it up so they should only have to record their name once provided they call from the same number. Anyway, I think I'm gonna get a big "NO" on

(Sorry if this has hit the list before. I sent it yesterday, but never saw it come through) I actually got rid of my phone service.. no more pots line in my house. But, I miss my call screen macro. Any way to do this with a SIP channel? (Obviously the parking isn't the problem, but rather recording their name). I set it up so they should only have to record their name once provided they

Sorry, but I can't tell if my messages are coming through. I don't see them, but perhaps the mailer doesn't send your own message back to you. If they are making it, please excuse this one. Thanks, Kris

I've posted about this before, but it's been so long I thought I'd see if there is a new solution (can't find anything on google or wiki) I use the Record() app on my incoming zap calls to record a persons name if their caller id is not in the db. After the name is recorded, the call is parked and ParkAndAnnounce announces that a call is holding from ${SCREENNAME}. Works fine

I have ParkAndAnnounce set up to call my cell instead of Console/DSP and this works fine. If a caller tries to get me on my cell, they are parked... I am called and told where they are holding, but then ParkAndAnnounce hangs up. So I have to call my system back and dial the parked extension to be bridged with the caller. Is there anyway to make the announce jumpto something else so I can just

obviously i'm looking for a direct connect w/ asterisk (SIP or IAX), so no proprietary equip, but if you provide, or know of a provider that has any of these available, please let me know. (Tennessee) 423-869-xxxx (Kentucky) 606-337-xxxx 606-248-xxxx 606-242-xxxx

Hi list, I have some on hold "activities" that I would like to implement and I'm just wondering if there is a way to do it. Here's what I'm thinking: While a caller is on hold, they could have the option to do things like retrieve weather, news, play blackjack( ala tellme), etc. These are all things I could do in the conventional method of sending someone to a context,

Regarding my earlier email (for some reason, I don't get my own emails from the list), I looked at the code and although I'm no programmer, I see that this is meant to hangup after the announcement. If I comment that line out, the call remains on the line, but I'm in limbo. I tried to add a chan->priority = 101 in place of the hangup and get this on the cli: = Spawn extension

I have Meetme2 (as well as the web ui) installed and am having some difficulty with the admin features. I've set up two extensions pointing to the same conference, one with the admin flag (1234|Maps) and another with (1234|Mmps). My issues: If the admin presses the * key, it goes to an endless loop of "enter the conf no. followed by pound key" If the user presses the * key, the

I have been trying for sometime to upgrade this phone to either sip, or the latest sccp (at this point I don't care) with no success. It has an sccp image (don't know the version off the top of my head) that is incompatible with chan_sccp. When I boot up the phone, it asks the tfpt server for the .tlv file, then the SEP<mac> file which it receives successfully. It then asks for

Just wondering if there are any Zaurus owners out there using there zaurus as a voip phone?? I'm trying to decide which on to buy. The sl-6000 is perfect for phone use from what I've read, but it's not very pocket friendly (http://www.sharpusa.com/images/hpc_SL6000_pic1.jpg) The clamshell models (http://conics.net/shp/pda/zaurus-sl-c700/sl-c3000.html) are much nicer but would

Ok, I've been away from the list for sometime now and feel as though I'm going to ask something that's been asked many times before, yet I find nothing in the wiki, so here goes: I notice that allison has done recordings for weather forecasts, yet I find no agi's that parse forecasts and use this recordings to piece together the forecast. I find plenty of weather agi's, but

I'm still not getting my outbound to work. I've seen two patches relevant to broadvoice for chan_sip.c which apparently have already been added to CVS. I'm dropping all outgoing calls after ~30 secs. Asterisk doesn't seem to know they're gone though. I called my cell w/ broadvoice and turned on sip debug AFTer the call had physically dropped: *CLI> sip show registry