similar to: AMD setup in Astersik

Hi all, I am working on asterisk 1.2.18 zaptel 1.2.17 Polycom 650 polycom 430 SIP version 2.0.3.0131 for IP 650 SIP version for IP430 2.0.3.0127 freepbx 2.2.1 I am trying to configure BLF using asterisk but failed. I would be thankfull if somebody help me. Regards FArooq ********************************************** 1 ********************************************** in my

Hi, currently I''m using 48 class with htb & very stable Is there any maximum number of class I can create in a single linux box ? I need 500 or even 1000 class for campuss network. Any help appreciated thanks & regards Tino _______________________________________________ LARTC mailing list LARTC@mailman.ds9a.nl http://mailman.ds9a.nl/cgi-bin/mailman/listinfo/lartc

Hi, I''m new to rspec and wanted to translate some of my unit tests into rspecs. Unfortunately my first test fails with "Mysql not loaded". My application is running fine and can access the database. I''d appreciate any help, so I can get started. Tino -------------- next part -------------- An HTML attachment was scrubbed... URL:

I am running Asterisk and using Answer machine detection with call files on a virtual Vcloud server running Centos 5.3 and LAMP. I am finding that AMD is only detecting HUMAN or MACHINE for about 30% of the calls (I sent over 50,000 outbound calls last week, and 70% said NOTSURE). I have a suspicion that the problem may be due to the timing source on virtual server when its under load delivering

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

Hi, I am new to the Asterisk world. I don't know much about the architecture, but I am involved in installing and configuring the VoIP system. My requirement is to build a VoIP system using the 4 input lines (ISDN up0 telephone lines), it must be possible to receive calls from outside through the 4 ISDN up0 input lines, and also possible for outgoing calls, conferencing .etc. I

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, I have IVR and when I press 1, asterisk calls my mobile phone. If my mobile phone is offline, asterisk transfers to asterisk voicemail. I'd like asterisk detects my mobile voicemail and if my mobile voicemail answers, asterisk transfers to asterisk voicemail. For that, I used AMD. So I have problems ! Asterisk detects answering machine everytime! How do I do please ? extensions.conf

Hello, Is it possible to set a variable in dialpan when the someone calls a particular DID number so that i can use that variable for calls coming to that number only. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100819/25402ade/attachment.htm

Winbindd can see the NT-user, but samba can't work with the NT-user. My System: SuSE Linux 7.2 Enterprise Server Samba-2.2.3a I have install samba by the following steps: 1. ./configure --prefix=/opt/samba-2.2.3a --with-winbind 2. make 3. make install 4. cp /tmp/samba-2.2.3a/source/nsswitch/libnss_winbind.so /lib 5. ln -s /lib/libnss_winbind.so /lib/libnss_winbind.so.2 6. vi

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, There is a xen setup in which "brctl show" gives the following output. bridge name bridge id STP enabled interfaces eth1 8000.003048c9d4df no peth1 vif1.0 vif2.0

Hello, Is it possible to install Asterisk on Vmware(centos) from source. Is there any difference or disadvantage for this compared to asterisk running on physical machine. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100811/05a14968/attachment.htm