similar to: Not Able To tar zxvf zaptel-*.tar.gz

Hi Sorry to use this public place, but IRC and emails to webmaster at digium have not helped in the past. I have several issues with using the files server downloads.digium.com, which has replaced the simple ftp/http file server ftp.digium.com. In downloads.d.c the directory listing is served through a seperate per-directory script with an obscure name. Let's look at

I have downloaded and unpacked tarballs codec_g729a_v33_nocona.tar.gz.1 /pub/register/x86-64/register /pub/register/x86-64/asthostid When I run the utility register of asking me for the license number. I get the following. [root at fc7234153 modules]# /root/register Digium Product Registration - Version 3.0.0 Copyright (C) 2004-2007, Digium, Inc. Use the '-l' option to see license

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:

Dear All, I'm using Asterisk 1.4.16.2 with Zaptel 1.4.7 on Debian with kernel 2.6.18. I have two analog lines Zap/1 and Zap/2 as group 1 in zapata.conf. I'm using below context for dialing out. [outbound-local] exten => _9XXX,1,Dial(Zap/g1/${EXTEN:1},30,tTr) exten => _9XXXXX,1,Dial(Zap/g1/${EXTEN:1},30,tTr) exten => _9ZXXXXXX,1,Dial(Zap/g1/${EXTEN:1},30,tTr) exten =>

Hello list, hope some one could help me find the answer. Asterisk 1.4.16.2 installd as no-root user The main issue is that every now and then, cd * box seems to loose the user's registrations, there is nothing in the console, absolutely no messages, only when another friend trys to dial an extension I can see this on messages logs [Jan 26 10:35:46] WARNING[23015] app_dial.c: Unable to

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

Hi, I'm a brand newbie to asterisk trying to set it up for the first time and I can't get a softphone to connect, the connection times out. I had a trixbox pro install working, but I need more control and would like to learn to do it with asterisk. In /var/log/asterisk/messages I see: WARNING[17401] res_smdi.c: No SMDI interfaces are available to listen on, not starting SMDI

Hello everyone, I'm using very simple setup to make and receive external ISDN calls through a softphone (x-lite version 3.0 - Win32) via an asterisk box. Hardware setup: - Dialogic Diva BRI (lspci yields: Network controller: Eicon Technology Corporation DIVA Server BRI-2M/-2F (rev 01)) - ISDN BRI line Software setup: - Redhat 9 - asterisk-1.4.16.2 - chan_capi-1.0.2 Asterisk configuration

Hi, I have a really strange problem with a Polycom 430 phone and Asterisk 1.4.20. Currently If I dial the Polycom from my mobile phone answer the call on the Polycom and then hangup the mobile the call ends fine on the Polycom. But if I call from the Polycom to my mobile and then I hang up the mobile the Polycom thinks the call is still active. However doing a show sip channels shows the the

I am trying to use the GotoIfTime function and get a busy signal. What I am trying to accomplish is to have the system tell callers that we are closed after 5:00pm. Here is the code below. ; If we're open, then go to the open context ; We're open from 9am to 6pm Monday through Friday exten => 3200,1,GotoIfTime(09:00-17:59,mon-fri,*,*?open,3200,1) ; ; We're also late on Tuesday and

Hi Steve, > .. I'll try to sort all this out, and then I'll attack this > problem. Hopefully, I get it all into svn before the next release of > 1.4...! Just wondering if any new CDR functionality made it into the 1.4.16.2 release? I have looked through the ChangeLog for the 1.4.15 and 1.4.16.2 releases but didn't spot anything to do with changes in CDR handling. I for one

What's the status of ZRTP supported by Asterisk? There was some discussion on the -dev list and -users list, but it was inconclusive. At about the same timeframe, a bug (#0010024) was opened and updated for several months, but has been "suspended" since late 2007. Does any version (1.4.x, 1.6.x) of Asterisk support ZRTP with clients (or with other servers)? Any successful testing

Hi Paddy! This used to be 80 characters total (including all characters like channel type, '&' and '/'. Had the same issue in the past where I extended that in the code and recompiled. From what I understand there is basically no longer a hard limit in Dial since the recent change in the latest versions other than a single device must not exceed this but you can concatenate

Hi all R users, I'm trying to replicate the same results that are given in a published article after been granted the same data that the authors use. I'm having problems to determine the cointegration rank of my data set using the Johnasen's trace test. This trace test is already programmed in the package ur.ca and can be found in the function ca.jo(). After I run the ca.jo()

Well, that sounds pretty straight forward. I can do this and push it to gerrit. Do I need to create a ticket for this? With best regards Florian Floimair Innovation - Software-Development COMMEND INTERNATIONAL GMBH A-5020 Salzburg, Saalachstraße 51 http://www.commend.com <http://www.commend.com/> Security and Communication by Commend FN 178618z | LG Salzburg Am 22.08.19, 11:55

Does anybody know of these samba packages? http://ftp.cvut.cz/samba/samba-latest.tar.gz AFAICS they are faked and contain some kind of rootkit (you can see this in the history below. the server this history is from is taken offline for security reasons, and nobody is there till 7th Jan I can't give you more details) > 144 w > 145 cat /etc/issue > 146 uname -a > 147

https://bugzilla.netfilter.org/show_bug.cgi?id=824 Summary: --utc option is not working in iptables version 1.4.7/8 Product: iptables Version: unspecified Platform: i386 OS/Version: Fedora Status: NEW Severity: major Priority: P5 Component: iptables AssignedTo: netfilter-buglog