similar to: does astcanary really work?

Hi, I only run a home-based asterisk (v1.4.18), and have never patched it, so I'm a unfamiliar with what time frame to expect for patches being implimented. I just downloaded (April 14) svn asterisk V1.6.1 r188415, on a "play" machine and noticed that when I stop asterisk, the astcanary module does not exit - when I restart asterisk, a new copy of astcanary also starts. In browsing

hi, I have strange problems with HTB and several hundred classes flat structure i.e. root |--50kbps |--30kbps |--50kbps |--80kbps |--100kbps .... several hundred classes like this Ceil is the same as rate. The machine get no more than 2-3% average cpu(2.4Ghz pentium). What happens is that from time to time the traffic got "stalled". I tried numerous things to solve

Hi, according to discussion on asterisk IRC, where people said, that macros will be depracated, I tried to migrate from macros to contexts and Gosub but if I try to use gosub in extensions.ael, ael compiler complains, that I shouln't use Gosub app, but I can't find ael keyword, that will be Gosub equivalent, or can I ignore this ael warnings? thanks PJ LOG: lev:3 file:pval.c

Hey, I have a few dids with callcentric. They seem to work fine most of the time but at some points I get "handle_request_invite: Failed to authenticate user <sip:PSTNnumber" This happens intermittently. The way I understand it the insecure=port,invite should tell asterisk not to authenticate users coming from that host. But its not working for some reason. This is my sip.conf

Hi, I have asterisk on public IP and phones in two locations behind firewall/nat, - when I have nat=yes and canreinvite=no, this is working fine, but rtp stream must go _always_ through asterisk, even if phones talk inside their locations - when I have nat=yes and canreinvite=yes, phones can speak only inside their location and rtp stream is connected directly between phones (this is, imho,

Hi Users, Administrators and Pavel Jezek, You prefer chan_h323 from asterisk tree and it's of course that use channels by tree is very good. But in 1.2.x, the chan_h323 is very simple and the chan_oh323 is so bad. And I work with chan_ooh323, that it's too from Digium and work good! And I am Studing one possible change to Asterisk 1.4.x , but in 1.4.x the oh323 channel don't have more,

The Asterisk Development Team is pleased to announce the release of Asterisk 1.6.0.7. Asterisk 1.6.0.7 is available for immediate download at http://downloads.digium.com/pub/asterisk/ This release resolves an issue where IMAP voicemail message retrieval and Message Waiting Indication (MWI) would not work properly with the same mailbox name in multiple voicemail contexts. This release also fixes a

Lets talk about SIP and IAX2 1. The good and bad of both 2. What is the better one and why 3. and any other information that maybe use full -- Best regards, Al Bochter Bochter Services (Voip PBX) Toll Free: 866-638-1254 EXT: 250 (Voip PBX) Free World DialUp: 780217 EXT: 250 (Voip) Cellular: 712-432-5401 http://www.BochterServices.com/?t=Email BUY and sell Coins, Silver and Gold

Hi, have you some idea, how to make "roaming line" with Asterisk? i.e. is possible to have phone line assigned to user if migrating from one office to another? thanks PJ

Hello, if somebody using this scenario in production successfully, please send me info, which ISDN card for asterisk server is usefull for me (Digium, Sangoma)? my crucial requirement is "caller id name" transfer/display between ISDN (Siemens PBX) and IP phone connected to asterisk I'm using PRI interface and Q.SIG signaling. thank you PJ

Hello I would like to define a trunk from my Asterisk to a VoIP provider, but I want to make it secure, because its through the Internet. I want to be sure no one makes calls as being me, and that my calls aren't intercepted. Is it possible to define encrypted trunks? And should I define the trunk in SIP, IAX or something else? Thanks Joao Pereira

Well, I'm using Hardy Heron from the last release of Ubuntu, and everything runs almost perfect with wine ... :? But I'm Having problems when I'm trying games.. Dunno, for example, I've tried Call Of Cthulhu: Dark Corners of the Earth, and got a little problem (mostly for my video card that is a little old :( ). The First problem came with the error of the installer of the game,

When i start a program like excel or something else it works. But i can't save my work to hd. I think there is something wrong with te read/write permissions of the hd.... How can i correct that ? Sent via Deja.com http://www.deja.com/

- I play the game atlantica online - Tried for a few days to make it work on vmware, read all about making the host enable for 3d games - Works fine, a bit slow but all 100% -------------------- - Tried on wine, and omg, runs 200% and fast, extremelly fast - Problem is the mouse - had same problem on vmware, but i found the answer on: > vmmouse.present = FALSE > > This line disables

I'm having problem powering Cisco phone's (7940 and 7905) on Linksys SRW224P switch (with PoE functionality). I have tested three phone's, one is working (7905) and two aren't (7905 and 7940). I have plugged all three phones on same switch port with same cable! Do I need to change anything in phone configuration? Is there something wrong with Linksys switch? How can I troubleshoot

Hello! I'm looking for TEXT sending over IAX2, more precisely sending text messages between IAX2 softphone and asterisk. Currently I'm trying iaxComm and I was able to send a text message from asterisk to iaxComm (with AGI SEND TEXT). After some searching I got some questions: 1. is it possible to send text messages during a call (established or during ringing)? 2. is it possible to send

Hi all!! I have downloaded the asterisk from svn checkout http://svn.digium.com/svn/asterisk/trunk asterisk-trunk (is the asterisk 1.4 subversion). I also downloaded the patch for cellphone and make it work fine. Then I bought the tdm11b board to have phone connection in my computer. I installed the hardware for zapte and the libpri modules in my Mandriva 2007 and the lights of the pci card

Hello, I can use directed call pickup using pickup application (between sip, iax, skinny cals), but unable to pickup call that is ringing on phone behind h323 gateway (using original h323 channel in asterisk), is this even suported? thx PJ exten => _*7.,1,Pickup(${EXTEN:2}) console log, when trying o pickup ringing line 324 (h323), from skinny phone (953) -- Executing

The Asterisk Development Team has announced the third release candidate of Asterisk 10.0.0. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 10.0.0-rc3 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the

The Asterisk Development Team has announced the third release candidate of Asterisk 10.0.0. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 10.0.0-rc3 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the