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Dear List; I am trying to create a link between Asterisk and My softswitch, the link to be SIP Trunk. I did the below configuration and I do not know if any one can help me and advise me to have better configuration to be sure that link is fine. But I do not know how to determine the SIP username to be sent for my softswitch as sometimes the softswitch need to check it. Also, does asterisk

Hi list; I am looking for an application that can be used with call center, in this application we can integrate the telephony part of the call center (like CTI Client ad so on), any one can advise for a good application to be used with Asterisk Call Center? - Note: The application to be customized easy, to be able to use it with Banking, Telecom, Oil, .. etc. Regards Bilal

--- Erik de Castro Lopo <erikd-flac@mega-nerd.com> wrote: > Harry Sack wrote: > > > Hi FLAC dev's list, > > > > I'm looking for a library for the C# language (Microsoft .Net > > Framework 2.0or higher) to play FLAC files and/or maybe do some > other > > things like getting > > the file duration, file properties, ... of FLAC files. >

Dear all I have install asterisk 1.2.x and it is working fine my setup is like [*]-------[Mediant2k]------------[Avaya] Now i want to transfer call in internal extension i have read more document on www.voip-info.com but it is now so much clear so if u have any sample configuration file and doucment plz suggest me i have configure feature.conf and extention.conf for this task

Dear all I have asterisk 1.2 configuration and it is working fine but thing is that i have alread Avaya setup and i have intergrate my Linuxbox asterik with Avaya system avaya already use 4 digit dialplan (1644 example ) and in asterisk i have configure 2 digit dialplan ( 44 example ) now i want to configure 4 digit dialplan in asterik without any change in avaya or asterisk so

Hi: I want to have conference call with asterisknow and need 2 ports E1.Which Digium card is better?TE212 or TE220.I haven't problem with motherboard. Regards. --------------------------------- Get the Yahoo! toolbar and be alerted to new email wherever you're surfing. -------------- next part -------------- An HTML attachment was scrubbed... URL:

2007/6/14, Josh Coalson <xflac@yahoo.com>: > > --- Erik de Castro Lopo <erikd-flac@mega-nerd.com> wrote: > > Harry Sack wrote: > > > > > Hi FLAC dev's list, > > > > > > I'm looking for a library for the C# language (Microsoft .Net > > > Framework 2.0or higher) to play FLAC files and/or maybe do some > > other > >

Dear all i have one confusion about how to dial outgoing call through asterisk like when i press 0 i got dial ton of exchange for outgoing call my setup is [sip_phone]-----[*]----[mediant2k]-----[Avaya_PBX]------e1-----[Exchange_PSTN] now i want to setup whn i press 0 in my sip phone i got dialton of PSTN so i can call outside people is there any special configuration to give

Hi, I have a Digium B410P 4-port BRI card. I installed misdn 1.1.3 with hfcmulti driver and misdnuser 1.1.3. I configured the card "correctly" as misdnportinfo reports: # misdnportinfo Port 1: TE-mode BRI S/T interface line (for phone lines) -> Interface is Poin-To-Point. -> Protocol: DSS1 (Euro ISDN) -> childcnt: 2 -------- Port 2: TE-mode BRI S/T interface line (for

Besides using an exec line with a case statement(to determine the specific os''s disable command), does puppet come with a buit in method to disable a user account? Thanks!. --------------------------------- Get the Yahoo! toolbar and be alerted to new email wherever you''re surfing. _______________________________________________ Puppet-users mailing list

Dear sir I have setup Avaya with mediant with asterisk [sip_phone]---[ * ]---[mediant]---E1-trunk--[Avaya]---[analog_phone] This is my configuration when i call from SIP phone i got ringing sound of phone but whn i call from analog_phone behind avaya i didn't get ring sound of ring but SIP phone speaker ring why i am not able to hear ring sound from analog phone Regards

Hi, I have a set of 4 graphs and I need to add a legend that is shared by those 4 graphs. This is what I tried: >locator(1) # I placed the cursor in the center of the 4 graphs $x [1] 9.299001 $y [1] 226.3201 >legend(9.3,226.3,"and the rest of the legend arguments")# but the legend didn't show. The legend only appears when I place in inside any of the for plots. How

Hi, I've just turned from Fedora Core to Centos 5, And would like to know the 'official' way/mechanism to upgrade a bunch of Centos 5 machines. My basic situation is: hundred of machines will be installed with Centos 5.0 by means of kickstart. and then the machines will always uses Centos 5.0 kickstart images for initial installation, not Centos 5.1, Centos 5.2, etc. So my

Hi List; I need to configure a softphone to be client and use it with Asterisk, which is the recommended one? Is it iax2? Regards Bilal ____________________________________________________________________________________ Boardwalk for $500? In 2007? Ha! Play Monopoly Here and Now (it's updated for today's economy) at Yahoo! Games.

dear R-helper, I tried to find out a package in which i can have penalized maximum likelihood estimator applying on generalized extreme value distribution with beta function) but could not. would you please help me to know the name of the package. thanks for your help. S.Murshed --- r-help-request at stat.math.ethz.ch wrote: > Send R-help mailing list submissions to > r-help at

Hi List; How can I convert some digits to another digits, and how I can insert in the end or in the begining some digits, for example: If I have a number like 11336784888, then I need to replace each digit of value 1 by 5, how? Also how can I add digits to the numbers like adding 00 in the beginning, so the dialed number becoming: 0011336784888 or adding digits (like 99) in the end of the

Hi List; I did yum install kernel and yum install kernel-devel, now when I type 'uname' -a I have the following: [root@localhost /]# 'uname' -a Linux localhost.localdomain 2.6.15-1.2054_FC5smp #1 SMP Tue Mar 14 16:05:46 EST 2006 i686 i686 i386 GNU/Linux And when I type rpm -q kernel, then I have the followig: [root@localhost /]# rpm - q kernel kernel-2.6.20-1.2319.fc5 So the

help me in another issue related also to registering asterisk with another softswitch: A) If nat=yes, then I have to set canreinvite=no to be able to register, correct? B) In case of using firefly softphone, how it possible to set it to have nat=yes (at the firefly it self and not at the sip user configuration section)? As most of the sip endpoint give an option to set nat=yes and so on, how it

I got one email from eric asked me to Lower the rxgain and txgain on your Zap channels. But actually it is already the voice volume is low and I was looking to increase the gain (currently it is 0.0), so I do not know if eric was mean to reduce it less than 0.0, but I can not do that due to the low volume that is already existed, so any more reduce will make the voice not hearable well, even if

I''m developing a site that will need to display images that will be stored in a location outside the site. What would be the easiest way to do this? The site will be running on a linux machine. Thanks, Will --------------------------------- You snooze, you lose. Get messages ASAP with AutoCheck in the all-new Yahoo! Mail Beta.