similar to: Asterisk Qeueu with static agent

Dear all I am going to setup Asterisk Call center solution and i have setup my queue and agent i have 2 SNOM ip phone but when i call to queue my agent phone is rining without musicnhold or when both phone is busy then i call to queue its directy hangup without musicnhole means my call not goes in to queue what is the problem my queue.conf [root at pbx asterisk]# cat

Dear all I have TDM808B 8 port FXO it is configure perfectly but i got some problem of incomming phone Hangup and callerid display problem i am going to explain you the issue i have install asterisk 1.4 and i have 100 of SIP phone now everything is fine but problem is when i incoming call on FXO and dial sip extention SIP phone is rining but when i disconnect my incoming

Dear all I have special requirement of samba domain security...i want to bind user with machine so that use only ...and only able to login with that same machine ...means user can not login in to any other PC or machine only access on own machine...is it possible with ldap attirbutes ..? $ cat ~/satish/url.txt

Dear all I have smb+ ldap setup not everything is fine but i want to assign some right to perticuler Group so they can change TCP/IP properties and change system time and do some other right Is it possible to give some privilages to normal users ??? $ cat ~/satish/url.txt http://www.linuxbug.org

Dear all I have asterisk 1.4.11 on Cent 4.3 i have faceing some problem i have TDM800P 8 port FXO card when i terminate PSTN line on this port can make outgoing call it is working fine but incomming call not handling ...when i call from outside to this line it is rinning but no one call land on my asterisk no debug in asterisk some time it land but most of time not .....

Dear all I have small lan and i have configure hardphone with my asterisk with one E1 PSTN line now i have configue to use canreinvite=yes in sip.conf If i user conreinvite=no then my RTP goes throgh asterisk means asterisk come in media path and if i user conreinvite=yes then RTP path would be sip phone to sip phone ??? My all phone in LAN not behind the NAT so guessest

What is SIP jitter buffer how can i test it ??? ----PGP Signature-- Satish Patel mobile:- +91-9818875535 http://www.linuxbug.org --------------------------------- Get easy, one-click access to your favorites. Make Yahoo! your homepage. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Dear all I have Asterisk which is connected with avaya through E1 back 2 back now i have on asterisk side G.711 codec and Avaya also useing G.711 codec everything fine. I need G.729 on my asterisk side. can i have lots of SIP phone on my lan and issue is i have 2 to 3 building so problem is LAN is congested thats why i need G.729 Now testing perpose i have download

Can anyone tell me what might cause callers on hold in a queue to drop into agents voicemail boxes? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071217/ae6cba5a/attachment.htm

Dear all I have install samba + ldap and it is successfully joing the domain but problem is when i login in XP machine with Administrator account of samba i cannot change anything in XP even not system time so is it problem of privileges ?? $ cat ~/satish/url.txt http://www.linuxbug.org

Dear all I have configure Asterisk 1.4 with sccp and configure Cisco phone 7975G model with Asterisk and it is working fine but i have one problem when i going to confance like when i call to someone and put it on hold and want to take in another New call to new person and want to take both live call in confrance how to do it in asterisk is there any configuration XML

Dear all I have asterisk with TDM808B FXO port with i call in asterisk and i promt IVR then user dial extention for user then my SIP phone rining but i disconnect or hangup my mobile phone but still SIP phone rining and stop rining after timeout is there any PSTN problme or FXO signalling problme i have configuraed singalling=fxs_ks ----PGP Signature-- Satish

Dear all anybody have idea of this 2 card and performance vise which one is suggestable ??? ----PGP Signature-- Satish Patel mobile:- +91-9818875535 http://www.linuxbug.org --------------------------------- Be a better sports nut! Let your teams follow you with Yahoo Mobile. Try it now. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Dear all I want to deny multiple login in domain with single user account what is the solution for it ?? $ cat ~/satish/url.txt http://www.linuxbug.org _____________________________________________________________________________________________________ --------------------------------- Get the freedom to save as

Hey guys, For some reason, when I enable smb and nmb to start at boot time... I get error messages in my log.smbd. [2008/02/01 18:47:00, 1] lib/smbldap.c:another_ldap_try(1153) Connection to LDAP server failed for the 1 try! [2008/02/01 18:47:04, 1] lib/smbldap.c:another_ldap_try(1153) Connection to LDAP server failed for the 2 try! From this you'd think that either I hadn't

Hi all, Some of you may recall a little while ago I posted a fairly complex request about integrating samba into a solaris 10 fileserver and using Open LDAP as the primary authentication source. Well, I got no responses as I believe it may have been too complicated-a-request/ question. So, a simplification and some different questions. 1. I have a Solaris 10 x86 host that I'd like to

Dear all anyone have idea about connect asterisk with mediant 2000 audiocode configuration ... anybody have configuration about it --------------------------------- Get your own web address. Have a HUGE year through Yahoo! Small Business. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Good morning, I have a client that needs some shaping done. Im only familiar with the HTB qeueu and was hopeing someone could give me a few pointers if possible. I need to shape outgoing and incoming as far as possible though I don''t know if it will be possible to do so without policeing the incoming bandwidth. They have a link and their service provider allows them to burts as

I've set up my Samba PDC to act as a WINS server, because I need remote access to it (through a VPN connection). I put "wins support = Yes" into the smb.conf file. As far as I know, WINS doesn't use a unique port number, so enabling the usual NetBIOS ports (137/udp, 138/udp, 139/tcp) on the firewall should be enough to make WINS work. I did this already. I tried to connect to a

created you must place it in your web directory on the server. =20 I chained the command and also wrote the output to an xml file in the web directory. The command looks like this: =20 'php /etc/asterisk/directory.php.txt > /var/www/html/directory.xml' =20 System Speeddials using Services Button =20 =20 For speed dials I modified the php code to look to a specific file in the