similar to: SetLanguage application doesn;t seem to work in latest Asterisk

Hello everybody, I wonder if anyone can help me in something I am trying to do but have no clue on how to do it: I have an Asterisk installation and I would like to be able to assign certain extensions to people with NetMeetings that take dynamic IP address. Does any one know how I can get the IP of an incoming channel in order to be able to dial back to that channel after the call is hangup?

Hello again, I have noticed with Queues and roundrobin policy that if even if a timeout is set for a queue, Asterisk keeps ringing an available member of the queue after the timeout expires. This continues a few times before the next available agent is tried. I am using CVS of August 17 but I have read in the list that roundrobin worked fine since earlier in August. Does anyone know if this has

Hi all, I'm having some trouble. I've got an IVR with 4 different languages using the SETLANGUAGE command and I'm getting a 6 second delay when I make the first selection, after that all is fine. There's nothing in my dial plan that I can find that would be causing it. And the delay is driving me nuts! I have an IAX connection from a provider coming in. Could this be the cause?

Good morning everybody, Can someone explain to me the interconnection between these four things: indications.conf, SetLanguage(), zaptel.conf and ring-back ? If there is any !! :- ) I am having this case where some users cannot hear ring back from a DeadAGI script and it seems to be interconnected to these items. These users are from the iaxfriends table, they _can_ hear ring-back from a

I am trying to use language=it in asterisk I downloaded the sound package and installed it I added country=it in indications.conf language=it in sip.conf language=it in iax2.conf everything ok in call from sip and from iax The problem arises in outside call, coming trom CAPI Trunk I try language=it in capi.conf: no result: always language=en I found a german forum, and it seems to be a

On Wed, 1 Apr 2015, Prof Brian Ripley wrote: > I would start by trying LANGUAGE=en , e.g. More specifically, you can use en_US or en_GB. Rich [...] ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?

Hi, I've recently reinstalled a Diva in my asterisk server (alongside a QuadBRI :-) to test the nice features Armin has been adding in chan_capi. The capi.conf format has changed, so my question is how do I define a deflect= statement for different incoming MSN's? I've tried to define a section for each (group of) MSN with a different "deflect". Is that correct? [DIVA1]

Hi, I just installed OH323 Plugin and im now tryin to make simple Configuration to connect Openphone and Xlite to my Asterisk-Server. All works fine, i just wanna know if there's a better way to do it? Is there anything wrong with my Config? OH323.conf [general] listenAddress=0.0.0.0 listenPort=1720 connectPort=1720 tcpStart=10000 tcpEnd=20000 udpStart=8000 udpEnd=8005 fastStart=no

I am having several users complain about not being able to use the # button when dialing into IVR's, etc, because the # key prompts for transfering the call to another extension. Is there a way to still provide transfer capability, but not use the # key? I am using SNOM 200 phones so if anyone has any suggestions, I would greatly appreciate it. Thanks, Brian

? Mon, 26 Feb 2024 09:02:56 +0100 SIBYLLE ST?CKLI via R-help <r-help at r-project.org> ?????: > In the following code, which loads the tiff file, I get the following > error This warning is definitely worth investigating, but it shouldn't interrupt your code. Does the figure come out wrong after you see this warning? > In doTryCatch(return(expr), name, parentenv, handler) :

When I attempt to start asterisk with my modem setup listed it will not start attached are the error messages i get and also the modem.conf that i am currently using. Any assistance would be greatly appreciated. running CVS ver 12/7/03, modified only to allow the RxFax and TxFax to compile and run with it (from http://www.opencall.org) just e-mail me privately if you need more info Thanks in

Hello, does anyone know how to get the say number (say.c) agi "application" to work in french [assuming that I have the French voice files] I have looked in the code and about a 1/3 of the way thru there is : /*--- ast_say_number_full: call language-specific functions */ /* Called from AGI */ int ast_say_number_full(struct ast_channel *chan, int num, char *ints, char *language, char

this is not just playback recorded voice. this is let asterisk say chinese. how to do this. there have any ideas? -- Jeffery iaxtel Num: 1-700-576-1311 fwdnet Num: 728150

A customer of mine wants an IVR where the first 3 choices are 1 English 2 Spanish 3 French I can build the IVR but how do I get the system prompts to then speak the selected langauge. For example, a caller has selected Spanish and so is routed to the Spanish part of the IVR. At some point he breaks out of the IVR to leave a VM. How does the system know to continue offering him Spanish?

Hi, First i would like to thank the kind people of the list who have answered my previuos mail, but i am still stuck as asterisk still crashes upon startup, i have read the install article at http://www.voip-info.org/wiki-Asterisk+Step-by-step+Installation and i have search the asterisk archives, but i still cant get asterisk to work, i have tried reinstalling asterisk but it still complains and

Hello all, HI i am very new to asterisk and my boss needs me to investigate setting up asterisk for a new client. I have downloaded and installed (make, make install and make progdocs)asterisk on my personal computer and when i try to run it (./asterisk -vvvc) i get the following output below: NOTE: i am running REDHAT 9.0 on a 796MHz cpu machine: I am excited to be able to work with asterisk

Hello, Is it possible to set a variable in dialpan when the someone calls a particular DID number so that i can use that variable for calls coming to that number only. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100819/25402ade/attachment.htm

Just curious, I noticed that with SetLanguage() you can change it into a lot of other languages. Yes one can record them easily enough with "record", but don't like to re-invent the wheel.. Browsed through a lot of google-pages but failed to find any other languages (except for FR and ES on the digium site) Any pointers for german, dutch, greek, italian, .... prompts? Hans --

Hi A few day ago, I notice that some applications missed in asterisk 1.6.1 release even if *.so file which normally create them were compiled during Asterisk install. SetCallerID(), SetCIDNum(), SetCIDName(), SetLanguage() ... and maybe so more. anyone already notice that to ? If it's not normal, anyone have an solution to it ? -------------- next part -------------- An HTML attachment was

I have an Ubuntu system and went into Synaptic and checked asterisk for installation. Once installed, I started it with /usr/sbin/asterisk -vvvgc and got the following output with several errors and notices. Do I need to do more or are these ok? I expected to have some conf files in /etc/asterisk but there is nothing there. Thanks! Created by Mark Spencer <markster@digium.com>