similar to: asterisk canreinvite=yes

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

Hey Guys! I have a stupid question about canreinvite. We are using asterisk 1.8.3.2 as a PBX we don't have NAT or firewall thing in between asterisk and phone. so i should use conreinvite=no right ? what is the default value of conreinvite in asterisk 1.8.3.2 ? i meant yes or no ? -S -------------- next part -------------- An HTML attachment was scrubbed... URL:

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!

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 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 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 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

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 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

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>

Hi, I'm trying to configure a server to use SAMBA and Winbind to authenticate to Active Directory.... I managed to get this portion up and running even the Dovecot portion. For reference something similar to this guide: http://www.whitneytechnologies.com/?p=119 However PAM is slightly different: # cat /etc/pam.d/dovecot auth sufficient pam_krb5.so no_user_check validate account

Dear all I have beginer in Voip and i have configured Asterisk server with 100 IP SIP phone ( SNOM ) everything is fine but problem is how to transfer call from one user to other means i call to some one and then someone want to transfer call to another person how it is possible i have also try with feartue.conf but it is now working i have also read document on voip-info website

2009/2/5 Olivier <oza-4h07 at myamail.com> > Hi, > > Here http://www.voip-info.org/tiki-index.php?page=Asterisk%20T.38 is a > table listing ATA/Gateways combinations. > Could anyone successfully set a Patton M-ATA to work with another one, > using Asterisk 1.4 ? > > Is reinvite (canreinvite=yes) necessary or not ? > > Regards > > Replying to myself, I