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Full_Name: Lyndon Drake Version: Version 0.65.0 Under development (unstable) (August 07, 1999) OS: Linux 2.2.2 Submission from: (NULL) (202.14.102.52) I was attempting to install the DSE package from a directory whose pathname includes spaces. The INSTALL script interprets spaces in the pathname as a separator....

Hi all, I want to Connect to IBM MQ from Rails/Ruby, can it be done? and get the messages from the Queue, then i want to use this message in ruby/rails. Is there a direct way of doing the above job?? If so any packages are available? If there is not a way currently, and any thoughts of how we can do the above job indirectly through some other means or round about ways or workarounds can be

Well, it looks like AstriCon 2006 is going to be big. We've sold out the entire Westin Park Central -- every last room. So, here are some nearby hotels to check if you're planning on coming down to Dallas for the big Asterisk-fest. Wyndham Garden Hotel-Park Central 8051 Lyndon B Johnson Fwy Dallas, TX 972-680-3000 Residence Inn LBJ/Park Central 7642 Lyndon B Johnson Fwy Dallas, TX 972-503-1333 Clarion Hotel Park Central 8102 Lyndon B Johnson Fwy Dallas, TX 972-960-6555 Amerisuites Dallas/Park Central 12411 North Central Expressway Dallas, TX 972-458-1224 S...

I want to be able to allow a caller to dial a ddi, system to verify identity etc (this is all done) I then want them to sit listening to music, until an event happens. When this (external) event happens, I want to play a certain file to the caller, using playback (so that they have ff / rw etc), and when finished, go back to the music. 1) I thought of redirecting to an extension that played the

...you get a message for companyB call centre Dial 555-1141 and you get a message for companyC call centre etc. etc. Once you have listened to the message welcome to blah blah your call is important etc etc. you are put through to an operator If you need any clarification please let me know Thanks Lyndon -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060404/955fab97/attachment.htm

I cannot reach voip-info - is it just me or is the site not available ? Julian

...his students. He wants to put R on it, but he's having trouble compiling R and using an RPM on RH 6.0. It appears to be either kernel or glibc related, but I don't have the time right now to figure out what the problem is (exams...). Does anyone have an RPM of R for RedHat 6.0? Thanks, Lyndon -.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.- r-help mailing list -- Read http://www.ci.tuwien.ac.at/~hornik/R/R-FAQ.html Send "info", "help", or "[un]subscribe" (in the "body", not the subject !) To: r-help-request at stat....

Apart from religious grounds (!), is there any pros or cons why I should choose one over the other for a new install of asterisk ? Julian

I experience random crash of machine (full hang, requiring a hard reset) after trying to test run Asterisk 11. The machine is a centos 5.8 32 bits pc with 1G ram. Asterisk is compiled from the source and no other software has been installed Anyone experience similar situation? -------------- next part -------------- An HTML attachment was scrubbed... URL:

I upgraded my Cisco7960 to SIP 8-2 from 7-4. Everything seems ok, works fine. Except that when I make an outbound call, I get a double-ring sound. I also found that if the target number is engaged, I get a ring sound and at the same time get a busy sound. If I revert back to 7-4, there is no problem. Anyone else had this, or any clues on how to fix it ? All of our other phones are still on

Has anyone in the UK got ANI to work on an inbound call ? Using asterisk 1.4 trunk and zaptel 1.4 trunk, with a Euro-ISDN 30 Julian ______________________________________________________________________ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email

I've set up a red hat linux 7.3 server running samba 2.2.3a. Windows clients often freeze when opening excel files stores on the server. Any ideas? Thanks Lyndon Davies Information Systems Manager ldavies@wandsworth.gov.uk 020 8871 6569 ===================================================================== IMPORTANT: Confidentiality: This e-mail and its attachments are intended for the above named only and may be confidential. If they have come to you...

I have sent a few emails over the past couple of days that simply have not arrived on the list (or so it seems). Is anyone else encountering this ? Julian

...3 intervals, by interval, or by species by interval. For now, it would be step forward to get any of these working. Apart from me learning how to do it, the import of this is that, if the CI's are doing what I hope they might, there should be little residual spatial correlation. Thanks, Lyndon.

.... When I call the queue, the call is not passed onto the member, and there is no activity on the cli. Eventually the call times out. If I add SIP/706 instead of Local/706@AgentQ then it all works as expected. Any clues or help ? Many thanks ! Julian. Kevin P. Fleming wrote: > ----- Julian Lyndon-Smith <asterisk@dotr.com> wrote: >> Now, I want to be able to use a device, rather than agents. So I can >> use addQueueMember and add my SIP device. However, I still want to do a couple of things before the device is called. > > This is what the Local channel (chan_loc...

I hope this isn't an invalid topic for this list. I'm on so many lists and I hate to join another one just to get help on one thing. Apologies if it's not. I am able to use ssh-keygen to generate keys so that I can ssh from my Mac to any of my SuSE systems or ssh from my Mac to any of my FreeBSD systems, without having to enter my password. When I try the same thing from a SuSE system

If I have the following in the dialplan exten => foo,n,Dial(SIP/1234&Zap/G1c/55443322) and SIP/5432 calls this extension, is it possible to show different callerid numbers to each of the target numbers ? The reason I ask is that if the call is from an internal sip phone, I want to show the internal callerid (5432) to the SIP phone on 1234, and the DDI of the 5432 extension

...s an additional patch: ?https://lists.alioth.debian.org/pipermail/nut-upsuser/2012-September/007922.html I'm happy to test an updated build, but have no idea about how to go about doing a build in the first place. Could someone help me in the process of creating a snapshot build for FreeNAS? Lyndon See also the following forum threads: ?http://forums.freenas.org/threads/nut-needs-a-patch.14402/ ?http://forums.freenas.org/threads/upc-back-ups-bk500ei-problem.14193/ And a github commit which I think is related: ?https://github.com/networkupstools/nut/commit/7ac6ad3143b3dec83450b70e435d7ffc6...

Just for some information really : How many of you use multiple sip lines on a phone ?. I'm sitting here looking at my 7960, with it's 6 lines. I've every only used one line, and I was wondering if I was a weirdo ;) The only time I've ever found a use was when I had two systems (production and test) and it caused so much grief (could have been asterisk or cisco) I simply use a

Using svn trunk, I was trying to see what the astdb entry in the sip.conf file does. Nothing :) I presume that it's meant to create an entry in the astdb. so, I have astdb=chan2ext/SIP/grandstream1=1234 in sip.conf But database show only gives *CLI> database show /SIP/Registry/706 : 192.168.0.200:5060:3600:706:sip:706@192.168.0.200:5060