similar to: Astersik with ISDN up0

Dear Customer Support, i connected the asterisk to a e1 interface of our hipath4000. outgoing calls from a sip peer of my asterisk to an up0 telephone which iss connected to the hipath4000 are working. If you want to dial from an up0 device to the e1 interface where asterisk is connected to, you have to use the prefix 83. But when you enter the 3rd cipher this error appears at the cli

What are the general reason why a samba share might be slow..? Especially those mapped network drive? What would probably be the cause of slowness in the access and mapping time? Need help Kumaran

hi, how to configure mongrel_cluster in windows. mongrel_rails cluster::configure -e development -p 3000 -N 2 i have used this one its configured correctly then if i start the server it is throwing error.if any one knows how to configure help me with regards shiva -- Posted via http://www.ruby-forum.com/.

HI; Thanks for your reply. The reason for why I am going through asterisk in such case is just "using asterisk voicemail service" I mean: ATA1 calls ATA2, suppose ATA2 is unreachable or he is not at the office, then the call reroute (my GK is able to reroute calls if the first route is not valid) to atersik for voicemail service. Do you think I can handle it with asterisk native

Slash-dot is pointing to this article on Asterisk and Pingtel. http://www.theregister.co.uk/2005/05/22/pingtel_voip/ Paul Paul Mahler www.signate.com

Hello, I am writing a program based on Astersik Manager which needs to put calls on hold and to redirect them to others extensions. I haven't funded any action able to do this. Is there a way to place calls on hold using Asterisk Manager Actions? Amaury -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello All: Does the Asterisk support to insert an off the board transcoder for a call? Thanks, Charles ____________________________________________________________________________________ Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping -------------- next part -------------- An

Hi to all!! I'm successful to connect Asterisk to PostgreSQL database... If it's possible, can anyone learn me how to store sip user in PostgreSQL database and how to configure voicemail?? Thanks for all!!!

Dear Sirs, I had compiled PWlib and OpenH323 correctly in my Fedora Core 2. But when I try to compile asterisk-oh323 I get the following error: `__use_AST_MUTEX_DEFINE_STATIC_rather_than_AST_MUTEX_INITIALIZER__' How can I solve it? Thank you for your help. Juanjo

We are planning to install a voip system based on asterisk for 2000-3000 retail locations and up to 6000-8000 sip accounts/users. Instead of setting up a new, centralized PSTN gateway, we are intend to use a CISCO gateway/router of an existing CISCO voip solution in the headquarter and we must able to call all CISCO based voip phones in the headquarter running together with a CCM. SIP-Phones

Hi everyone, I want to call from one Asterisk to another Asterisk via SIP, but i dn't know how. I have found out something in these links: http://www.voip-info.org/wiki/index.php?page=Asterisk+config+sip.conf http://www.voip-info.org/tiki-index.php?page=Asterisk%20SIP%20Channels but I don't understand them very well. At first, I tried simply doing this: In SIP Client:

Hi all, I'm building a karaoke service. Asterisk will play a music file, people can detect the point when they want to sing and record by press * key during the music is playing, and press # key to stop recording. I use 2 functions: ast_streamfile and ast_seekstream to play audio file, and function ast_waitstream_fr to detect whenever people press DTMF key. The problems is that, Asterisk

Hi there, quick question...do digium make any BRI cards (ISDN2) or even better a quad port BRI, maybe im going blind, but I cant see any on their website Cheers Gary

In my Asterisk server following things have been done to detect answering machines before the answered call connects to the agents in queue. In extension_additional.conf ============================== [ext-queues] include => ext-queues-custom exten => 5000,20,Macro(user-callerid,) ; changed the priority to 20 ............... ============================== In extension_custom.conf

Hello, I've got very annoying behaviour from our asterisk PBX. We have 12 channels T1 e&m wink start for TDM and using iax softphones internally (iaxcomm, but tried firefly-thirdparty and discarded for bad sound quality). Slackware 9.1 w/ kernel 2.4.26+ digium TE110P card. In some cases when call is placed from softphone to TDM, system does not detect call answered on Zap channel and

Hello sir very Greeting to u......... Right now i have using puppy linux (ver 3) how to install ssh packages & configured ( i have already got openssh 5.1 but i got some errors means root can't access via putty so what can i do...) Kindly tel step by step as soon as... im waiting for ur Reply Thanking you......... -- ????????z

Hi All, The meetme.conf have created as below: [rooms] conf => 101 conf => 102 and extensions.conf as below: exten => _1XX,1,MeetMe,${EXTEN} why the warning printed when I called 101. WARNING[27660]: File pbx.c, Line 1051 (pbx_extension_helper): No application 'MeetMe' for extension (ipcentrex, 101, 1) And I found asterisk have not load the meetme.conf when it starts up.

Hi all, I am facing one problem in configuration of httpd-vhosts. my requirment is if url comes from servername (i.e http://sutra) it should redirect to my home page.if url comes from serverAlias(i.e http://sutrateam) it should go to advance search page....if any one knows how to do pls help me below file is my vhost file <VirtualHost *:80> ServerName sutra ServerAlias

Hi, Iam trying to setup Samba version 3.2.3 on Redhat (RHEL5) server to use Active Directory for authentication. I followed the instructions from article in following website: http://technet.microsoft.com/en-au/magazine/dd228986.aspx Setup Winbind + Samba + Kerberos and it seems to work fine. I can see the users in Active Directory through winbind as well as authenticate users using NTLM

Yes, i know that there are many ISDN card on the market. But when i spend money for ISDN card, i prefer to be Digiums, to get all support and help Asterisk :).