similar to: Asterisk - first impressions

I can't get my Asterisk to register/place calls with FWD. Here's what I have in my SIP.CONF: register => 11111@fwd.pulver.com/11111 [fwd] type=friend secret=somesecret host=fwd.pulver.com username=11111 fromuser=11111 fromdomain=fwd.pulver.com I'm using CVS version of Asterisk, checked it out last week. I get authenticate error when registering with fwd, and all my calls to

I'm trying to link up Cisco MGCP-enabled router (residential gateway) with Asterisk, and it looks like some sort of protocol mismatch, could it be MGCP 0.1 vs 1.0? Look at this (x.x.x.99 is the router, x.x.x.98 is Asterisk): MGCP read: NTFY 2 aaln/0@voip-gw1 MGCP 0.1 X: 0 O: hd from 192.168.154.99:2427MGCP read: NTFY 2 aaln/0@voip-gw1 MGCP 0.1 X: 0 O: hd from 192.168.154.99:2427Verb:

I'll have MGCP hardphone that needs to dial pre-defined number as soon as it goes off-hook. So far I'm lost as to how (if at all) this can be implemented in Asterisk. Any pointers? TIA, Peter

I'm trying to set up * server behind NAT. The box is set up as DMZ in my DSL router, i.e. all incoming connections without explicit port mapping are forwarded to *. So far I'm unable to get this setup to work for either IAX or SIP (tried IAXComm & XLite softphones on public IP address). Data seems to come in fine (IAX/SIP debug shows message interaction taking place), but there is no

I have some problems with busy detection and SIP. When I'm making a phonecall (out or internal) and someone else is calling me, the phone (Snom200) is ringing and leaving the first caller (no difference if I call someone or if someone calling me) in the background waiting. It doesn?t hang up the first call but the second one is overriding the first. Is there anyone that has experienced the

winetricks is about to change a bit; that big long menu is being split up into separate menus for dlls, fonts, settings, apps, and games. You can get the new version from http://kegel.com/wine/winetricks-alpha or the easier-to-remember http://winetricks.org/winetricks-alpha Please try it out and report your impressions here, I'd like to hear them before I make this the default version

After a long week agonizing over a scheduling library, and then specing it out rather quickly, I''ve got some work I''m pretty happy with. As a contrast to Test::Unit, I like the way RSpec spreads my attention around the codebase while I concentrate on implementing specific behaviours. In Test::Unit, I tend to write tests around the private methods of the object under test. I

(Win95, v62.3, Guido's port.) The graphapp based console seems a definite improvement. Screen output is much faster. History mechanism present. Can now build whole system because windres not required. Spin-off is that graphapp can be used for completely different applications. A couple of problems. When R started and then library(pkg), get message (not every time though) that pkg unable

I've checked everything (pwlib + openh323 + asterisk) out of CVS, compiled, and chan_h323 module does not load with "undefined symbol _ZTI19H323AudioCapability". What could be the problem? Peter

My Asterisk box also does NAT for internal network, and establishes site-to-site VPN tunnel(s). As a result I have several internal interfaces with private addresses on them, and only one public interface. By trial-and-error I've found out that FWD (SIP) won't work unless I disable my VPN tunnels - it would send the internal IP address to FWD's SIP server instead of public one. I

Generally speaking, I can say that Vorbis already has pretty good quality, better then MP3. Mode 6 (~350kbps) - sounds great, no high-frequency artifacts and no pre-echo in any of my 3 test cases (I use 3 tracks as pre-echo test cases: 1. castanets.wav from LAME homepage 2. drums from the begging of the song Metallica - The Small Hours and 3. bass guitar from the begging of the song Primus - My

Hi, I have only recently become aware of Dovecot and gave it a try. The previous 0.99.9.1 version didn't work well for me (OpenSSL), I dropped it, but 0.99.10 has come just in time (saw it on freshmeat) and I thought I'd give it another try if it promised SSL fixes, and it's sorta working for me (i. e. it works with mutt, Mozilla, sylpheed, but not cone -- but cone is beta and has SSL

Battery life sucks. This may be due to three factors: 1. It was the first charge that the machine had 2. Playing Vorbis takes more battery power than MP3 3. The battery is a bit rubbish More considered opinions to follow. Phil Hibbs Cap Gemini Ernst & Young Aston, UK <p>======================================================= This message contains information that may be privileged

Hello there Yesterday I updated my pwlib, openh323 and Asterisk from CVS. After making "clean opt" in pwlib and openh323 and make "clean install" in Asterisk i get an "Undefined symbol" error when I try to start Asterisk. As far as I can see its when loading the h323 channel driver the error occurs. Do I have to update other things as well, by reading the various

I tested Vorbis encoder - beta3 version, and here are my thoughts: - In comparison to beta2, subtle high-frequency artifacts seem to be gone (though they were small in beta2). Good job there! :-) - Velvet.wav also sounds better, but transparent quality is reached at -b256+. - Horn.wav still sounds very sucky, mode -b256 gives ~100kbps (this is understandable because in this sample practically

I cant seem to get outgoing calls to be placed properly .. No matter what I try I get an error from the PSTN company saying that the "call can not be completed as dialed" or "you need to dial a one..." The asterisk debugging seems to show the correct number being dialed out of the zap interface... the "9" is being stripped and there is a "1" where it is

I wonder how IAX compares to SIP bandwidth-wise? I've tried both over overseas IP connection, and somehow SIP seemed to work better. Peter

Dear UseRs!, Everything about UseR! 2010 was terrific! I really mean "everything" - the tutorials, invited talks, kaleidoscope sessions, focus sessions, breakfast, snacks, lunch, conference dinner, shuttle services, and the participants. The organization was fabulous. NIST were gracious hosts, and provided top notch facilities. The rousing speech by Antonio Possolo, who is the chief

Look Cisb... here's how to remove yourself from this list: First you need to know the PASSWORD that you signed up with. I'm sure you forgot it, so go here and put your email address ariel@cisb.mine.nu into the box that sais [EDIT OPTIONS]: http://lists.digium.com/mailman/listinfo/asterisk-users Then, on THAT page, if you know your password you can just enter it in & unsubscribe.

I have an interesting dilemma with sequencing in the dialplan. Up to now, I have assumed that the extensions in the dial plan were tested in the order that they appear in extensions.conf. In other words, I have the following fragment which was designed to dial toll free on the PSTN and all other long distance on VoIP: [longdistance] include => local