search for: elqredir

...y. On 12/28/07, broadband Voice <broadbandvoice at gmail.com> wrote: > > I successfully downloaded the Asterisk package from Digium but not able > tar zxvf zaptel-*.tar.gz. See log below. Thanks. > > > [root at fc7234153 src]# wget --passive-ftp http://www.digium.com/elqNow/elqRedir.htm?ref=http://downloads.digium.com/pub/asterisk/releases/asterisk-1.4.16.2.tar.gz > > --10:15:59-- http://www.digium.com/elqNow/elqRedir.htm?ref=http://downloads.digium.com/pub/asterisk/releases/asterisk-1.4.16.2.tar.gz > > => > `elqRedir.htm?ref=http:%2F%2Fdownloads....

...ephony/ I get a list of items. I have to guess which of them is a file and which is a directory. There is no proper date of change. Furthermore, I cannot follow links directly. Links are redirections. For instance, the link marked with "aadk" points to: http://www.digium.com/elqNow/elqRedir.htm?ref=http://downloads.digium.com/pub/telephony/aadk $ HEAD 'http://www.digium.com/elqNow/elqRedir.htm?ref=http://downloads.digium.com/pub/telephony/aadk' 200 OK Connection: close Date: Mon, 29 Oct 2007 10:05:54 GMT Accept-Ranges: bytes ETag: "26cb96-963-433e597412940" Server:...

Hi i have problem with AddQueueMember logic. I need login Agent(Member) in asterisk. use this option: for example: AddQueueMember(queuetest,SIP/ekiga,10,,Agent/13) and now i want to call to this Agent: exten => _1XX,1,Dial(Agent/${EXTEN:1}) call to 113 and asterisk should call to Agent => 13 on interface SIP/ekiga. This doesn't work, How can i do this on Asterisk 1.4(not

Hello Fellow Asterisk Mailing ListMembers, When I tried to download the latest version of Asterisk this is what I get: http://ftp.digium.com/pub/asterisk/releases/asterisk-1.4.0.tar.gz Opening fileinfo database failed http://ftp.digium.com/pub/asterisk/releases/asterisk-1.4.0.tar.gz Opening fileinfo database failed Where are all the latest Asterisk 1.4.x source files? Thanks in advance, -E

Dear Sir, Can someone help me please to find a free ebook talking about AGI scripting through asterisk? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090218/a59fc299/attachment.htm

Hi, all I want to add more than 200 sip accounts into sip.conf, username from 6000 to 6199, password is the same, i remember there is a better way to do this case, however, i have not searched the method yet. Anybody can tell me this method, TIA. BR Mike Li -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have an account with FWD and I have configured my SIP.conf with [fwd] type=friend secret=password username=901835 host=fwd.pulver.com But when I am trying to dial out my own DID , I dont see any call landing in asterisk. In extension.conf (vicidial) file I have exten => 2062036895 ,1,Ringing() exten => 2062036895 ,2,Wait(1) exten => 2062036895 ,3,Answer() exten => 2062036895