similar to: Cisco IP phone 7975G + SCCP + Asterisk-1.4

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 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 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 asterisk every time my Agent login in queue and useing queue but i want to staticly map that agent in queue so how do it possible and what configuration required for it ??? ----PGP Signature-- Satish Patel mobile:- +91-9818875535 http://www.linuxbug.org --------------------------------- Looking for last minute shopping deals? Find them fast with Yahoo!

Dear all I have video confranceing deivice Codian and i want to intergrate asterisk box with codian so voice confrance is possible with codian users means some users have not codian endpoint so thay call join confranceing with SIP PHONE I have configures asterisk and register codian in asterisk now whn i call from asterisk to codian i got IVR and ask me to inter confrance

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

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:

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

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 and now i want to configure meetme confranceing but problem is when i dial confrance number i got message conf number is invalide and i got this error /dev/zap/psudo device not found what is this ??? --------------------------------- Need a vacation? Get great deals to amazing places on Yahoo! Travel. -------------- next part --------------

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 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

file is created you must place it in your web directory on the server.<br> &nbsp;<br> I chained the command and also wrote the output to an xml file in the = web directory.&nbsp; The command looks like this:<br> &nbsp;<br> &#8216;php /etc/asterisk/directory.php.txt &gt; /var/www/html/directory.xml&#8217;<br> &nbsp;<br>

Dear all I have samba + LDAP PDC setup and its working and xp client login in domain problem problem is when i login in XP Client with Administrator account or root account i can't not change any thing in XP client means Time setting and TCP setting . so what privilege i have to setup to give permission to administrator power user. i have seen this problem first time

Hi All, I'm thinking of creating a new asterisk server using the latest 1.4 stable release to replace my ageing Asterisk SVN-branch-1.2-r7231 (its been a while!). My only concern - my phones are Cisco 7960's (with sccp firmware 7.2 loaded) and to support them better, I remember compiling in a skinny(?) driver to replace the (from what I could tell) basic in built sccp support. After

Have a Cisco 7975g running SIP firmware version 8.3.4. Many things are broken with Asterisk. 1) BLF doesn't work 2) MWI doesn't work 3) Sometimes the calls get "stuck" on the display 4) Sometimes MOH works 5) Headset jack doesn't work Can anyone recommend a version of the SIP firmware for the Cisco 7975g that they know works well with Asterisk?