similar to: asterisk voicemail to email and relaying

Is there a way to set/clear a BLF LED on a phone from the dialplan? I want to use one as an indicator of some state in the PBX - in this case it's "night mode" but I can think of other applications. I have BLFs working just fine for "normal" stuff, just wonderin if I can use them for more! Cheers, Gordon

I've recently set up two workstations running CentOS 6, one with an nVidia card and the elrepo drivers and one with an ATI Radeon card with the elrepo fglrx drivers. Both work well, but one aspect of the systems works different from CEntOS 5 on those systems: I cannot get a wallpaper image to span the two monitors. I have tried both with and without xinerama and there is no difference

Hello all, I'm currently writing a C program that uses the libvirt API and I need a way to obtain the pid of a given domain's QEMU process. Specifically, I'm writing an ivshmem server that uses SO_PEERCRED to get the pid of clients that connect to it, and I would like to use that pid to look up the domain in libvirt to determine the proper domain ID to return to the client. As

I'm relaying an icecast1 mp3 stream with icecast2. This works fine, and the song title is displayed correctly no matter if I use <relay-shoutcast-metadata>1</> or not on the relay. But, the stream name and other metadata provided by ices0 on the icecast1 server is not displayed in the listening client. Icecast1 source: [Id: 1] [Sock: 11] [Time of connect: 09/Aug/2004:10:01:24]

HI ALL; Is astersik enable to translate between different codecs. I have couple of SIP-UA , one with (a-law) and the other with (g729), registered with my astersik box.Can astersik translate between alaw-g729 and vice varsa. Regards mohammad -------------- next part -------------- An HTML attachment was scrubbed... URL:

Are there DUNDi CLI commands for Asterisk 1.4? I have searched google and I only see the dundi commands in Asterisk 1.6, although I see reference to them in older version's of Asterisk such as Asterisk 1.4 here: http://www.asteriskguru.com/tutorials/cli_cmd_14.html. When I view the CLI commands through help I don't see any of the dundi commands and there are errors when I run a command

Hi All, pl help me with this basic question. I have a users (soft clients) with usernames having Alphabetics. I want to use Asterisk as my server. How should I have the dial plans as there are no numbers involved . so How can I make the configuration to work ( with numbers I can get this done using extensions.conf) my expected result is : alice at pbx.com should be able to call bob at

Hi all, As we know we can configure in astersik like before 5:00pm calls go to reception and after 5:00 pm calls go to some mobile no. One of my client requested that he wants to manually shift the dial plan like above as he has flexiable timing sometime he finishes at 3:00pm some time 8pm. I can not give him freepbx access. Any idea or solution. Regards Farooq --

Hi Tom, Thanks for the feedback about my Shorewall evaluation I''ve published a blog today covering general things I''ve observed about the way to combine Shorewall with strongSwan: http://danielpocock.com/practical-linux-vpns-with-strongswan-shorewall-and-openwrt Please let me know if anything is inaccurate or if there is anything substantial that I missed and I''ll

Hello, Asterisk Real time database worked on astersik 1.6.2.0 but now i am working on Asterisk to latest version which is 1.6.2.2 ,there is a a warning [Feb 24 16:26:14] WARNING[4053]: config.c:2025 find_engine: Realtime mapping for 'sippeers' found to engine 'mysql', but the engine is not available [Feb 24 16:26:14] NOTICE[4053]: chan_sip.c:21500 handle_request_register:

I try to run epsxe in wine by dragging it into wine helper and it says "ePSXe.exe has exited with an error, have a look at the Log window..." It has said this for many other emulators I've tried to use and I'm just wonderin how do I fix this??

hi i am doing my final year project on Traffic Shaping .could any one please guide me how do i actually go about it. i mean the first step.i have read the lartc documentation for the same. what do i do next.please help thanks in advance _________________________________________________________________ How good are you in a Formula One car? Play now

Hi ; Have two handytone 486 and want to use them as digium TDM400 fxo-fxs card... I mean is it possible to direct pstn calls from astersik (extensions) to handytone line port directly and vice versa ?... Thanks in advance Betul Onemli not : Bu e-mail iletisi, sadece adreste belirtilen kisi veya kurulusun kullanimini hedeflemekte olup, mesajda yer alan bilgiler kisiye ozel ve gizli

Hello list i have configured astersik 1.4 with sip i have a question when i put in dial plan.conf exten => _0678922645.,1,Set(CALLERID(number)=520460587) exten => _0678922645 .,n,MixMonitor(zap_g1_${EXTEN}_${UNIQUEID}.wav|av(0}V(0)) exten => _0678922645 .,n,Dial(Zap/g1/${EXTEN},30,A(this-call-may-be-monitored-or-recorded)) exten => _067892264*5*,2,Hangup() i can not call my

Slash-dot is pointing to this article on Asterisk and Pingtel. http://www.theregister.co.uk/2005/05/22/pingtel_voip/ Paul Paul Mahler www.signate.com

An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031207/e5e9b8eb/attachment.htm -------------- next part -------------- Hello all, I am newbie to Telephony world (IP and PSTN). Please excuse me if you find my questions very dumb. I am trying to configure my IP 7940 with the Asterisk, when phone boots up it only shows the message "Phone

HI ALL; Is there anybody who use app_radius(astersik radius module)??????????? is it stable? Regards mohammad -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040803/8a096bfe/attachment.htm

re-posting the question. ----------- use case: when some one in my pbx calls 100.200, I have translations well defined, Media also (media via asterisk) --Works. when some one calls bob, or for any names I am adding Domain and call is been sent to the other party -- Works, no media... For the cases when it is talking to the external work, I want Astersik not to do anything with the SDP/payload.

Hi all, I have seen that R can be switched on to a Broker called IB. There is another one similar to IB that permits to make a "code" and send orders to broker to buy or sell stocks? Can be done trough R, writte in excel and trough API sell/buy to the broker? Can someone send me an example (easy example?) Many thanks. [[alternative HTML version deleted]]

Hoi All, I am trying to append collums to a data frame in a for loop. I read in tables, do some processing and then write the result to a data.frame. But, the thing I want is, that the results are appended to the data frame in stead of overwriting the results of the prevous table. It has to look something like this: After going trough the loop once: Array 1 1 2 3 4 5 After going