similar to: Asterisk-1.2.1.tar on Suse Linux 9

Hi Atif make is a Unix's command which uses Makefile file for package's compilation. So after installing the complete development package from distribution disk, launch make. Ciao mauro

Hi, I am facing problem in sync time within DomU with Dom0. After gone through mailing lists, I have set /proc/sys/xen/independent_wallclock to 0. But still DomU are not syncing their time :-S Machine using kernel-xen-2.6.19-1.2911.fc6 xen-3.0.3-3.fc6 xen-libs-3.0.3-3.fc6 Please advice to solve this weird problem, Thanks _______________________________________________ Xen-users mailing list

What are people's thoughts on asterisk 1.2.26? Any show stopping bugs? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080117/57d1002d/attachment.htm

Hi, I found some issues with Snom 320 message retrieve key. This button works only when the MWI does not blink! If MWI blinks and I do press retrieve button I get "Unknown" on display and busy tone. From the sip debug it looks like that Snom does not send credentials to Asterisk which responds with 407 Proxy Auth required. I have loaded Snom with latest 5 firmware. No change. I'm

Respected All, I am writing a program in R and facing some problem with applying "if statment". Program first draw random numbers from bivariate normal distribution then compute variable say Pi and Pij from that sample and then further computation ..... ..... ..... In some samples Pij is appearing with negitive sign and ultimately resulted in an negative variance estimator.

on the asterisk site, it was stated while ago, how to download stable version. like cvs checkout -r v1-0_stable asterisk-addons zaptel libpri but now it's not their. is stable-version removed from the CVS ? or is their some different procedure ? thank you -- Atif

Hello everybody!!! I really need your help guys, I am using the AGI mode in meetme application, and I want that AGI should wait for an input from the client/user i.e. a digit and then proceed, but I have used that AGI function agi->wait_for_digit(), but no use....my agi just passes, or ignores this function, where AGI should stop here and wait for the input.... .....my extension in my

Respected All, I hope you are enjoying good health, I am tring to write a program in R but could not be very sucessful. My program draws random sample form bivariate normal distribution and then compute a variable PIJ. For certian samples some entries of variable PIJ is apearing as negative, which result in negative variance estimator. I want to introduce a loop in my program that verify the each

Please suggest a way out to the following problem. I have a T by n data matrix (say Y) where coulmns are time series of length T. To do some analysis in WinBUGS I need to construct my data as follows. yy<-rep(Y,k) ## this will be a vector Yk<-array(yy,dim=c(T,n,k)) ## data array Here the definition of dim indices is first index: T rows second index: n columns third index: for kth T by n

I am using gentoo Linux and Asterisk CVS-HEAD-09/23/04-19:57. I have tested both KPhone and IaxComm for linux but receiving no audio from asterisk. sound is working fine, as I can listen playing files using PLAY or APLAY. KPhone is configured with DTMFmode=inband and codec is ulaw and IaxComm is configured with ilbc if somebody can sort out this Thank you regards, -- Atif

I am configuring a sip-phone, receing calls, excellent voice quality. but it does not place calls, please, can some one sort out. here is my debug output, and below that is sip-debug, Jul 16 11:34:32 DEBUG[163850]: Setting NAT on RTP to 0 Jul 16 11:34:32 DEBUG[163850]: Stopping retransmission on 'iiasPlzFribMJMcW' of Response 1: Found Jul 16 11:34:32 DEBUG[163850]: Setting NAT on RTP to

Hello there! Somebody tried the meetme|b which initiates the conf-background AGI. Actually I want to originate another call from a conference.my AGI originates the call and connects it to the conference, but the calleeee is nowhere My extension exten => 21,1,meetme(21|pb) and my AGI **************************************************************************** #!/usr/bin/perl -w

HI all, I have the following setup running: EP<--->Calling Asterisk<--->Relaying Asterisk<--->Softswitch<---> PSTN The Endpoint EP is registered with the Calling Asterisk. Calls are forwarded from this machine to Relaying Asterisk which in turn forwards it to the Softswitch. In addition, this machine also relays back responses from the Softswitch to the Calling

this is from my extensions.conf, the first three patterns are for toll-free numbers, and fourth pattern is for other numbers, where an AGI is called for authentication. now when I dial 011448000664327 if falls into the fourth pattern, where as it should be matched by the first pattern. Any suggestions 1 - exten => _01144800XXXXXXX,1,Dial(${MAG}/${EXTEN:3},45,tT) 2 - exten =>

Hello everyone, I''m wondering what the current status of hardware performance counter usability in Xen is. I see some old posts describing the diffculties of virtualizing hardware counters within dom0 and the domUs, but not much else. Have they been implemented or are they in the process of being implemented? Or are there no future plans for implementation? Any help would be

All, I have AAH 1.0 installed using Digium TDM04B and Grandstream GXP2000 phones. All seems well other than the phones have to be reset up to 5 times per day. It is like they lose thier ip connection or maybe thier SIP connection. Has anyone else experienced this issue? I have the phones set for static IP addresses and that doesnt seem to help either. Any help would be greatly

Respected All, I need some efficient program or package to draw all possible samples of size two without replacement. I am using "combinat" package to list all possible samples but it hangs my computer for larger populations say 10,000 (i.e. 49995000 all possible samples). I wish to even work for larger populations then this and replicate this procedure for many times. Kindly can anyone

Wow. My message made it to the list after more than 3 hours. Philipp -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? -> http://www.das-asterisk-buch.de Gesch?ftsf?hrer: Stefan Wintermeyer Handelsregister: Neuwied B 14998

I've added lines like this: speeddial = 123,test speeddial = 260,Bob in the [device] section for my 7921, but the speed dials do NOT appear on the menu (click right from the main screen). Am I missing something obvious here? Thanks MD -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi everybody :) I am a new member here and hope that someone gives me a hint for my problem: Let's say I am at work and my SIP phone (KPhone in my case) is connected to my private Asterisk. I want to call my wife at home so her SIP phone rings. She does not pick up the phone (maybe she is somewhere in the house and has to run to the phone) so after 15 seconds her cell phone should ring.