similar to: Loops and Variables

All - I've been wrestling with how to best structure the sip device accounts on a new asterisk server I'm deploying. All of the sip devices (currently only Linksys SPA941s) will reside on the same subnet as the server, and I have already set up a decent automatic provisioning system for the phones. When the rollout is complete, there will be about 100 SIP devices authenticating and

Resent to the list since I didn't think you would mind. Kevin wrote: > Eric, > > I have been using your vm outcall script for some time and it has worked > well. Thanks for your efforts. > > I am trying to re-install and I can't seem to get a call file generated. > I have set up postfix and in the log it appears that it pipes the > message to the vmoutcall

I am originating a call to a Local channel using an Originate Action: Action: Originate Channel: Local/dial at outdial Context: outdial Exten: answer Priority: 1 Timeout: 45000 ActionID: some_id In my dialplan, I have this: [outdial] exten => dial,1,Dial(${DIAL_STRING}, ${DIAL_TIMEOUT}) exten => dial,n,NoOp(Dial Status = ${DIALSTATUS}) exten =>

https://bugzilla.redhat.com/show_bug.cgi?id=466681 just to be sure that some one in xvm-team notices that before any bits from xvm-gate are putback ;) Florian

All, I sent a message for help a couple of weeks ago about a samba migration issue. With further testing, we have isolated the issue to Microsoft Outlook (versions 2007 and 2010). All other applications have no issues when they mount the new samba server reading, writing, etc. But, in Outlook, when you try to open an archive (pst file) on the new samba server, it pops up an error "the

Need your help! I'm very confised and tired, tried a lot of variants but still at the start, even lost all understanding how the permissions and masks works. Please write me a config for the following case: There is a directory: '/home/shared/door', owned by 'michael:office' shared via samba as '[door]' How can I configure the FS and samba to let 'michael'

Hello, Ive been days trying to make this game work as expected, finally today i found out that it was an AcceptEx issue, this is what the terminal says when someone wants to join my hosted warcraft 3 frozen throne games under wine: fixme:winsock:WS2_async_accept AcceptEx accepting socket closed but request was not cancelled My wine version is 1.4: wine --version wine-1.4 Regards, Octohedra

I can browse a site I developed/deployed to Apache 2.2.3/Mongrel from PC/s with Firefox/Explorer. I can''t browse from Mac Safari? I checked these setting and it still doesn''t work... http://docs.info.apple.com/article.html?artnum=151263 I don''t know Mac''s, what esle could this be? Jim

Hi, I''ve been using backgroundrb with some success up to now, but now when I do the start or setup tasks they just fail silently. Nothing in the log, no .pid file, and nothing in the process list. Rake with -trace just says, ** Invoke backgroundrb:setup (first_time) ** Execute backgroundrb:setup Any hints on how I can get more verbosity, or thoughts on what might be happening? (This

Greetings all. I have been running Wine under Kubuntu Feisty for about the last four months, keeping it updated with the latest wine releases on a weekly basis, I use it to run WordPerfect. Whilst WordPerfect didn't run flawlessly, I got it to work well enough to edit and save documents I work on at home. In four months none of the updates to Wine changed anything in how WordPerfect

Please add to the ProjectsWithLLVM/ page. Thanks. ==== Embedded System Language (ESL) ESL <code.google.com/p/esl/> is a new programming language designed to be used for efficient programming of embedded systems and other low-level system programming projects. ESL is a typed compiled language with features that allow the programmer to dictate the concrete representation of data values;

Good afternoon gentlemen (and ladies). A costumer of mine has many servers and each one maps their SIP extensions to the others via DUNDi. It works like a charm. SIP extensions can only register at one server, the one they "belong" to. In case one extension wants to call other that is registered in another server, DUNDi takes care of that by calling the other server using IAX2 and G.729

I use the following macro for my extensions. It only works with Zap channels and assumes that any Caller*ID number that is 4 digits is an internal call and all other calls are external calls. Use like this: exten => 1234,1,Macro(std-exten,Zap/4,20) [macro-std-exten] ; ; Caller*ID is 4 digits (internal call) ; exten => s/_XXXX,1,Dial(${ARG1}r2,${ARG2}) exten =>

hi folks. i'm experimenting with iaxmodem + hylafax using DID to determine where to send the fax to it's final destination. however i have difficulties passing the DID information from iaxmodem to hylafax. in extensions.conf: exten => _XXXX,1,Dial(IAX2/iaxmodem0/${EXTEN}|20|r) exten => _XXXX,n,Dial(IAX2/iaxmodem1/${EXTEN}|20|r) exten => _XXXX,n,Busy exten => _XXXX,n,Hangup

Hi, The call waiting tone is very annoying (you hear nothing while it plays the beep). I need callwaiting because of the queues (the phone has to ring as soon as you hangup) but I want to remove the beep on my dahdi channels, how can I do? Thanks, Niccol? -- http://www.linuxsystems.it

I just experienced something I'd rather not experience again. Using a SPA-841 SIP phone connected to our Asterisk server, someone dialed their own extension, answered, and then transferred the call using the phone's "XFER" soft key. This does a SIP REFER. Now, the phone has dropped out of the loop, and Asterisk has connected the two call legs into a loop, as far as I can tell.

I was wondering if there was a way of setting up the dialplan in a way that if you dial an extension that is NOT in the dialplan then it would play a not-in-service gsm file and then play congestion tones. I would rather like this better than just hearing a busy signal on my phones.. I DID search around on the wiki and using google and could not find anything. Thanks. -- Stephen Rosebush,

I am trying to implement a Centralized Call Waiting System. I have red some document about asterisk group features to manage group and category of a sip channel. I have done a lot of test about it but always it doesn't work correctly if I transfer the call. This is the macro code I use for inbound calls. [macro-test] ; ${ARG1} - technology something like SIP ; ${ARG2} - resource.

First off, I also posted this on the digium forums so if anyone here also reads those, sorry for the cross-post. When I place an outbound call using SIP to my cell phone, asterisk immediately starts processing the dialplan without waiting for the call to be answered. We could handle this on DAHDI using callprogress, but I don't know of a similar setting for SIP. Here is the contents of

Hi, I've noticed an unusual behavior on the dialplan execution: assume this DP: exten => _6XXX,1,NoOp(test1) exten => _XXXX,1,NoOp(test2) exten => _XXXX,2,NoOp(test3) If I call 6000 then test1 and test3 NoOps get executed, even though the pattern is different. I've always thought that if I call 6000 it would match the 6XXX pattern, that only has 1 priority, that would get