search for: kesh

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2005 May 17
0
Problem with getting the value of variable DIALSTATUS in AGI script
...?. I wrote a small perl script, that just calls to the specified number and then receives the information about +the status of the call. This script is below: #!/usr/bin/perl use Asterisk::AGI; $AGI = new Asterisk::AGI; my %input = $AGI->ReadParse(); open (TEST,">>/home/kesha/ast_test"); $AGI->exec("DIAL OH323/232313124\@ip.address.of.the.gateway"); $status=$AGI->get_variable("DIALSTATUS"); print TEST $status; close (TEST); When I call to the specified upper number, I am successfully connecting with the subscriber and have...
2007 Aug 20
1
Disabling Asterisk Authentication
Hello, I have a small LAN network connected through an Asterisk Server. When I try to make a call between two of the user pc's on this network I get a "401 Unauthorized" error. Would anyone know how to remove the Asterisk Authorization/Authentication? I am not sure if this can be done with an entry into the sip.conf file, or by other means. My sip.conf file is shown below: ;
2007 Dec 27
3
Grandtream Conference issue
...he Line1 and Line2 Option. When I'm initiating the conference at that time, IP phone is sending the G711ulaw for the conference call, while in my phone I've set the all codec option to PCMU only. Due to this I'm facing issue. Any solution for this problem, please let me know. Regards, Keshav Regards, Kesh " Lets change the future...lets change the world." --------------------------------- Never miss a thing. Make Yahoo your homepage. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users...
2007 Jun 29
0
nway call
...ling the third pasrt number, at this time the first person( second party) is listening the music on hold. The next number is geeting connect but i m not able to take back that person in conference by pressing the required keys *0. Any if any of you have made that then please let me know. Regards, Kesh --------------------------------- Ready for the edge of your seat? Check out tonight's top picks on Yahoo! TV. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070629/e168a261/attachment.htm
2008 Jan 21
1
asterisk-addons-1.6.0-beta1---Error
...IENT_MULTI_RESULTS' undeclared (first use in this function) make[1]: *** [app_addon_sql_mysql.o] Error 1 make[1]: Leaving directory `/usr/src/asterisk/asterisk-addons-1.6.0-beta1' make: *** [all] Error 2 Earlier 1.4 addons are woking fine. Anybody have any idea about this error. Regards Keshav Regards, Kesh " Lets change the future...lets change the world." --------------------------------- Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.dig...
2007 Jul 19
2
Upgrade Procedure
Hello All, I would like to upgrade my recently installed Asterisk 1.2.21.1 to Asterisk 1.4.8? My OS is CentOS 4.5 with Linux 2.6.9-55.0.2.plus.c4smp #1 SMP Fri Jul 6 05:25:07 EDT 2007 i686 i686 i386 GNU/Linux Is there any detail step by step procedure to uninstall the current version and install Asterisk 1.4.8, Zaptel 1.4.4, Libpri 1.4.1, Addons 1.4.2? Cheers, Nitesh
2007 Jul 18
3
how to use call transfer
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
2008 Jan 20
6
IAX softphone
Hi All; I tried Firefly softphone with IAX and it gave very poor quality. Any one advise a good strong softphone that can work with IAX fine? Regards Bilal ____________________________________________________________________________________ Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ
2007 Jun 22
10
Query
Hi all, Can anybody tell me that wether I should install DIGIUM-TE120P card on redhat9.0 2.4.20-8 or 2.6.18 kernel. I am using kernel 2.6.18 but facing a very serious problem of modutils and iptable. Can anybody help me out. Thanx and Regards sanchal singh
2008 Oct 10
1
Switching Servers now can't retrieve mail
I have just switched my mail server. I have copied (I think) the config files from the old to new. Exim is work fine on the new machine as I can see email file showing up in /var/mail/lists/new . However, dovecot reports nothing when I check mail . I am have turn on logging on both machines and I do not see anything that looks like an error. I have also looked to see if the dovecot.conf file is
2007 Jul 18
1
Issue in insatlling addons-1.4.2
...: No such file or directory make[1]: *** [install] Error 1 make[1]: Leaving directory `/usr/src/asterisk/asterisk-addons-1.4.2/asterisk-ooh323c' make: *** [install] Error 2 Have anyone any Idea how to solve this issue.. Please suggest me how to solve this problem, or this is a bug?? Regards, Keshav --------------------------------- Park yourself in front of a world of choices in alternative vehicles. Visit the Yahoo! Auto Green Center. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070...