search for: heartthrob

...rent asterisk, zaptel and libpri and download new one and write new configuration. Regards, -------------- ITS IP Telephony and Contact Center Engineer Eng. Bilal Ghayad Mobile: 00965 9849460 ____________________________________________________________________________________ Be a better Heartthrob. Get better relationship answers from someone who knows. Yahoo! Answers - Check it out. http://answers.yahoo.com/dir/?link=list&sid=396545433

...es insecure=very disallow=all allow=g729 allow=g723 ....... extensions.conf: [to-sip] exten => _0011X., 1, Dial(SIP/${EXTEN:4}@out4) exten => _0011X., 2, Hangup() Any ideas? Vitaly ____________________________________________________________________________________ Be a better Heartthrob. Get better relationship answers from someone who knows. Yahoo! Answers - Check it out. http://answers.yahoo.com/dir/?link=list&sid=396545433

...pport (in case the Phone was damaged and need to replace, so the warantee), so which web can provide that? I do not need to buy from one and he is not responsible for support. Regards Bilal ____________________________________________________________________________________ Be a better Heartthrob. Get better relationship answers from someone who knows. Yahoo! Answers - Check it out. http://answers.yahoo.com/dir/?link=list&sid=396545433

...illing or AstBill or ASTCC? Also, from where I can download it and ready about its configuration? Regards ITS IP Telephony and Contact Center Engineer Eng. Bilal Ghayad Mobile: 009659849460 ____________________________________________________________________________________ Be a better Heartthrob. Get better relationship answers from someone who knows. Yahoo! Answers - Check it out. http://answers.yahoo.com/dir/?link=list&sid=396545433

Hi List; Where I determine the codec to be used for the SIP Trunk (between Asterik and another SIP softswitch)? Regards Bilal ____________________________________________________________________________________ Be a better Heartthrob. Get better relationship answers from someone who knows. Yahoo! Answers - Check it out. http://answers.yahoo.com/dir/?link=list&sid=396545433

Dear list Sorry to ask a user's question on developer list. I didn't find the user list. I am experimenting with --cuesheet and encoded a flac file with a cuesheet. Result is: * Totem on opensuse 10.2 opens the flac file with no meta data, no play list. "Next" button doesn't work; * Mplayer on opensuse 10.2 opens the flac file with no meta data,

Ralph reports: "I believe application/flac was used in analogy to application/ogg, and has never actually be registered. This might be an opportunity to register the more correct audio/flac. I don't know if an informational RFC on the flac format needs to be published for that to happen." What do you think, Josh? I'm in the process of writing a RFC regarding all MIME types

...g its port when I run vendor/plugins/acts_as_ferret/script/ferret_start. I get the ''bind error, address already in use''. Any help would be appreciated. Thanks! Dave ____________________________________________________________________________________ Be a better Heartthrob. Get better relationship answers from someone who knows. Yahoo! Answers - Check it out. http://answers.yahoo.com/dir/?link=list&sid=396545433 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://rubyforge.org/pipermail/ferret-talk/attachments/20070804/ec048f7f...

...gt; s,1,Set(CDR(accountCode)=1234) exten => s,n,Dial(SIP/1234) Hope this helps others. I was consulting some other people and they solved the problem before the digium-users could. -Matt ____________________________________________________________________________________ Be a better Heartthrob. Get better relationship answers from someone who knows. Yahoo! Answers - Check it out. http://answers.yahoo.com/dir/?link=list&sid=396545433

...X,102,Playback(all-allbusy) exten => _0.,1,Dial(SIP/mediant/${EXTEN:1}) exten => _0.,2,Congestion exten => _11.,1,Dial(SIP/mediant/${EXTEN:2}) exten => _11.,2,Congestion ______________________ Satish patel ______________________ --------------------------------- Be a better Heartthrob. Get better relationship answers from someone who knows. Yahoo! Answers - Check it out. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070808/b9fede52/attachment.htm

Hi: If any body use meetmemanager or conman or web-meetme please say how about is it.I'd appreciated any idea. Regards. --------------------------------- Be a better Heartthrob. Get better relationship answers from someone who knows. Yahoo! Answers - Check it out. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070822/894e1b87/attachment.htm

...e this post but I searched the archives and I didn't find this message which I posted last week (just in case someone sees it). I guess that my original post was dropped/lost/quarantined. ____________________________________________________________________________________ Be a better Heartthrob. Get better relationship answers from someone who knows. Yahoo! Answers - Check it out. http://answers.yahoo.com/dir/?link=list&sid=396545433

...ected to FXO) to an button on the Polycom IP Phone or Broadtel IP Phone, so if user select that button then he will be sure that his outside call will be via that specific line. Regards Bilal ____________________________________________________________________________________ Be a better Heartthrob. Get better relationship answers from someone who knows. Yahoo! Answers - Check it out. http://answers.yahoo.com/dir/?link=list&sid=396545433

Hi all Can centos have way to limit users to use resouce in the computer? thank you ____________________________________________________________________________________ Be a better Heartthrob. Get better relationship answers from someone who knows. Yahoo! Answers - Check it out. http://answers.yahoo.com/dir/?link=list&sid=396545433

...re is a KDC lookup problem, should it not occur globally ? -- To unsubscribe from this list go to the following URL and read the instructions: https://lists.samba.org/mailman/listinfo/samba ____________________________________________________________________________________ Be a better Heartthrob. Get better relationship answers from someone who knows. Yahoo! Answers - Check it out. http://answers.yahoo.com/dir/?link=list&sid=396545433

Maybe this has already been stated or is a feature request or I could be out of my mind. Is there anyway to access a remote mysql database from within let''s say...a puppet function? BC ____________________________________________________________________________________ Boardwalk for $500? In 2007? Ha! Play Monopoly Here and Now (it''s updated for today''s

Josh Coalson wrote: > the main reason I haven't swtiched is I'm not up on pkgconfig > and XIPTH_PATH_OGG has been working pretty well... can you tell > why it's not working for you? > > I'm always hesitant to mess with things in configure.in because > is such whack-a-mole problem (fixes things for some people and > breaks for others). I'll look into that

...I have to configure this on the sip.conf file or where exactly? And is it the same when I configure iax trunk? Should I determine the context in this case for this SIP trunk? Regards Bilal ____________________________________________________________________________________ Be a better Heartthrob. Get better relationship answers from someone who knows. Yahoo! Answers - Check it out. http://answers.yahoo.com/dir/?link=list&sid=396545433

...atus=no eventwhencalled=no joinempty=no leavewhenempty=yes maxlen=0 member=Local/4054 at from-internal/n,0 ; set by freepbx wrapuptime=0 extensions.conf: exten => 4050,n,Queue(4050,t,,) ____________________________________________________________________________________ Be a better Heartthrob. Get better relationship answers from someone who knows. Yahoo! Answers - Check it out. http://answers.yahoo.com/dir/?link=list&sid=396545433

Hi List; How can I configure asterisk to receive a call from SIP end point without being registered at asterisk and its IP address is dynamic, and authentication to be based on the username and password or any other string? I know that if I place the host with static IP then no need to register, but what if the voip gateway was having dynamic IP and I do not need to register on asterisk, but I