similar to: How to get values of local channels context

Questions: Does anyone have a really STABLE asterisk system running about one year without need to restart the service or the SERVER ? Does anyone have a production Call Centre saled that don't lockup and is stable for 6 months ? I'm asking this questions because we have choose Asterisk for our call centre solution but, since the bugtracker only grows and people still want to stuck more

I have the following in the dialplan: [macro-systemrecording] exten => s,1,Goto(${ARG1},1) exten => dorecord,1,Record(/tmp/${CALLERID(number)}-ivrrecording:wav) exten => dorecord,n,Wait(1) exten => dorecord,n,Goto(confmenu,1) exten => docheck,1,Playback(/tmp/${CALLERID(number)}-ivrrecording) exten => docheck,n,Wait(1) exten => docheck,n,Goto(confmenu,1) exten =>

On attempting to make Zaptel 1.2.16 on FC5, I get the following messages: /usr/src/zaptel-1.2.16/xpp/xbus-core.c: In function 'debugfs_open': /usr/src/zaptel-1.2.16/xpp/xbus-core.c:171: error: 'struct inode' has no member named 'u' make[3]: *** [/usr/src/zaptel-1.2.16/xpp/xbus-core.o] Error 1 make[2]: *** [/usr/src/zaptel-1.2.16/xpp] Error 2 make[1]: ***

Hi, I am new to asterisk configuration. I want to get called number in features.conf. I am defining a feature in features.conf and that feature got executed on pressing a particular DTMF key sequence. As I want to execute my own application on pressing that key which will use called number. testfeature => 3,peer,AGI,StoreNumber|<CalledNumber> Here I want to use called number in place

>> The call that gets dropped had a retransmission of INVITE from UAC >> to UAS (and therefore retransmission of 200 OK from UAS to UAC). >> There is nothing wrong with the re-transmission as such, but I >> noticed a potential bug in Asterisk in the way it responds to an >> INVITE retransmission. Asterisk is bumping up the session version >> number in

Hi Tanya, I have tried the 2.6 pre-release on the following host : Windows XP pro SP2 with mingw/msys : uname -a MINGW32_NT-5.1 OLIVE 1.0.10(0.46/3/2) 2004-03-15 07:17 i686 unknown gcc -v Reading specs from d:/mingw/bin/../lib/gcc/mingw32/3.4.5/specs Configured with: ../gcc-3.4.5/configure --with-gcc --with-gnu-ld --with-gnu-as --host=mingw32 --target=mingw32 --prefix=/mingw --enable-threads

On Thu, Sep 3, 2009 at 6:24 AM, Olivier Meurant<meurant.olivier at gmail.com> wrote: > Hi Tanya, > > I have tried the 2.6 pre-release on the following host : > > Windows XP pro SP2 with mingw/msys : > > uname -a > MINGW32_NT-5.1 OLIVE 1.0.10(0.46/3/2) 2004-03-15 07:17 i686 unknown > > gcc -v > Reading specs from d:/mingw/bin/../lib/gcc/mingw32/3.4.5/specs

I replied to a post of mine a few days ago asking of anyone uses mailboxexists(). I haven't received any replies. Perhaps few use it or perhaps the reply was overlooked. I thought I'd post the question one last time before giving up on it for now... Thanks! -Michael

Hello Guys, I'm somewhat of a newbie and am desperately seeking for some help... I've managed to get asterisk up and running on my server, and signed up with a broadvoice account... I'm having no problem dialing and communicating between extensions, but whenever anyone tries to call my broadvoice account, they are greeted by no ring or anything, but rather simply a direct to

Hi, I've a problem; I've tryed to install wine 0.9.7 on a Mac Computer on Mac OS X System. I've downloaded source files from source-forge.net I have made configuration,and make dependecies and finally I did "make" !...but...at least appears a terrible message: #error unsupported cpu (minidump.c 172.2).... I've tryed this on two different computers: a IBM PowerPC G5

Hi all, I've note that the variable DIALEDPEERNUMBER is broken. Now i want to know if exist another method to retreive the called number on *, and, if it's possible, an example ;) Regards.

I am running asterisk 1.4 and the Digium GUI SVN-branch-2.0-r4489. When one phone calls another, I see the following on the console (here, 6223 dials 6123) -- Executing [6123 at DLPN_DefaultDialPlan:1] Macro("IAX2/6223-10489", "stdexten|6123|SIP/6123&IAX2/6123") in new stack -- Executing [s at macro-stdexten:1] Set("IAX2/6223-10489",

How many times do you ask for help here before getting a respone? Every single thing I do No matter what I get busy extensions. I am willing to pay someone to help here. Anybody got a clue? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050802/d0d1326c/attachment.htm

I am running R version 2.9.2 on Windows XP OS with RODBC version Version: 1.3-0. Has anyone out there in the R user community successfully appended records to a DB2 table on a remote database using the sqlSave function in the RODBC package? (or by any other means from R?) I posed a similar question a few months ago and unfortunately, did not receive a response. I was hoping recent upgrades to

Using 1.0.9: If I have: exten => s,1,Dial(SIP/5555&SIP/12345@192.168.1.1) How can I return the DIALSTATUS variable for the second SIP channel ONLY if the second SIP channel is busy, regardless of the dialstatus of the first SIP channel? What I want is, if the second SIP channel is busy go to n+1 or n+101 regardless of the status of the first SIP channel. tia

While reading about how to maximize capabilities in asterisk i have read about SER and OpenSER. The sites do not explain to newbies (maybe that's on purpose) what are the benefits of using those products tied with asterisk (or is SER an asterisk replacement??) Can someone give me an idea of what's the usage for open(ser) and asterisk? is it for scalability? should I run it in the same

Hi, I've got an NT4.0 domain that I'm planning to upgrade to Active Directory. I've got a couple of Samba member servers in the domain, various versions above 3.0. So I have two questions: 1. Are there any gotchas with this scenario or will the Samba clients just keep working as NT4-esque clients. 2. Is it possible to "upgrade" a Samba member server to be an Active

Hi: I want to make parking calls easier for my hard-working users. Is there a way to make the Polycom call park feature work with Asterisk? In case it just works out of the box, I haven't tried it yet; but the "call park" feature isn't enabled on the Polycom phones by default. -Stephen-

Dear all, I would like to know if it is possible to fit in R a Cox ph model with time-dependent covariates and to account for hierarchical effects at the same time. Additionally, I'd like also to know if it would be possible to perform any feature selection on this model fit. I have a data set that is composed by multiple marker measurements (and hundreds of covariates) at different time

Consider this dialplan fragment, where the call is being dialed into [macro-process-routing] over an iax2 channel from another (same build) Asterisk server: [macro-process-routing] ; This is the entrypoint of the debug call but is also refered to by Macro(process-routing) elsewhere in the dialplan ; XXX-NNN-6800 exten => _6800,1,Macro(6800-interceptor) ; This is matched when 8 is