similar to: PhpAgiTutrial

Hello list partners you know about a softphone made in java attachable in a web page? GNU! Thaks in advance!______________________________ Visita http://www.tutopia.com y comienza a navegar m?s r?pido en Internet. Tutopia es Internet para todos. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, i am from Mexico , i like to install icecast2 server on mandriva , i have tryed but i could not, i hope yo can help me please, is very important for me , i need a guide , the steps or something , tanks a lot. Luis Enrique Perez Rios ______________________________ Visita http://www.tutopia.com y comienza a navegar m?s r?pido en Internet. Tutopia es Internet para todos.

hi we need a predictive dialer which can be used with asterisk software. Is it possible? Bye Owais Bin Zuber --------------------------------- Do you Yahoo!? Yahoo! Search - Find what you’re looking for faster. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040310/a49a4b5c/attachment.htm

I am Jerry Richmond CEO of ByVolution LLC. We have purched some did's from you that we use to test with, weare going to order our first batch of 250 this week. John Blackman is not with us any more. I need for some one to call me on my cell phone because our office no. 9198270713 to 9198270720 can not except outside calls. Brian Sponaugle tells me he is not getting the help he needs from

I've been using realtime for sip users information. I noticed that when you are doing this, if you do a 'reload' or restart asterisk, the information in a 'sip show peers' goes away. When I do this, MWI stops working. I always though MWI used the astdb file ('database show') to determine where to send MWI but it must be using 'sip show peers' because when this

Hi, After following the suggestions on the mailing lists and the wiki I'm still experiencing choppy moh. The song plays but with frequent noise parts. - I'm using asterisk 1.2.4 on our production server and 1.2.7 on the test server. - native moh with .gsm and .pcm formats (according to http://astrecipes.net/?n=152) - compiled ztdummy as a timing source any pointers on how to dig deeper

Hello, i've recently switched over from 1.0.9 to 1.2.3. I've experienced some (to me) weird behaviour. This is the config for an example queue.conf: [654] wrapuptime=30 timeout=20 strategy=ringall retry=5 queue-youarenext=queue-youarenext queue-thereare=queue-thereare queue-thankyou=queue-thankyou queue-callswaiting=queue-callswaiting music=default monitor-join=yes monitor-format=

Hi, one of the sip-extensions we created always returns busy when someone tries to call the phone. The extension itself can place calls. We're using snom360 phones with the latest firmware. On every one of those phones when we register with the sip-extension, we've experienced the same problem. This is the output from sip show channels: Peer User/ANR Call ID Seq

Hi All, I was wondering if there's a way to clear a SIP channel ('sip show channels') thru the asterisk CLI? kind regards, Bart Bart van Daal Network Operations Van Landeghemstraat 20 9100 SINT-NIKLAAS b.vandaal@edpnet.net www.edpnet.be T +32 (0)3 265 67 00 F +32 (0)3 265 67 01

hi All Any body already setup asteriks call routing to Cisco AS5300 with SIP Server ? i need informations sample config for that, or can show how to route docs . thanks Dirgan --------------------------------- Meet your soulmate! Yahoo! Asia presents Meetic - where millions of singles gather -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi.. Can anybody help me.. here is my problem: I have a little LAN with a Samba Box (It's my PDC), and some DOS PCs, running my core bussiness application (Foxpro 2.6) All the Boxes are Intel 440BX2 whit Celeron 600MHZ, HD 10GB Maxtor 5400RMP, 64MB RAM and 3Com 905B Nic. The DOS Box have MSDOS 6.22 with MSClient 3.0 everything it's Ok about the connection and autentication with

Take advantage of a 20% discount on the most recent R books from Chapman & Hall/CRC! We are pleased to offer our latest R books at a 20% discount through our website. To take advantage of this offer, simply visit www.crcpress.com, choose your titles and insert code AZL02 in the 'Promotion Code' field at checkout. Standard Shipping is always FREE on all orders from CRCPress.com!

Sorry for crossposting initial message to both lists. I think the topic may be interesting for both - users and developers. Any volunteers to help to translate the documentation to languages other than English? This is very big job, so there can be more than one person, working on the same language. Among responses to my call for volunteers I got response from Pedro Restrepo suggesting his help

aight than thanks Geoff i'll just use version 2.2 for right now.. thanks for the update (u might say) 73's and 88's DE TODD KC?CCB DN84FK ICQ #: 71473037 MYSPACE : http://www.myspace.com/kc0ccb NET: http://www.webspawner.com/users/hamradio/index.html TEMP WEB SERVER: http://209.159.208.18:8000/stream.m3u ----- Original Message ----- From: <icecast-request@xiph.org> To:

When I read variables in AGI scripts, I see only the follwing 13 variables agi_request agi_channel agi_language agi_type agi_uniqueid agi_callerid agi_dnid agi_rdnis agi_context agi_extension agi_priority agi_enhanced agi_accountcode beside these, I found following variables documented on several sites. agi_calleridname agi_callingpres agi_callingani2 agi_callington agi_callingtns Where can I

Hello, I'm running asterisk-1.0.6 on a centos3.4 box. I'm still in testing phase and so far everything is running smoothly. I'm now trying to play a soundfile or an mp3file using 'MP3Player', 'Playback' or the 'Background' commands, but don't get any sound. The logfile says: -- Executing BackGround("SIP/joa-9def", "tt-weasels") in

Hello, I am building the SPEC 2006 Benchmark with Clang as the compiler. I have written a function pass in LLVM and I am trying to run that for SPEC by invoking the pass in the build options of SPEC. The build options of SPEC are in a *.cfg config file, which allows us to specify the choice of compiler while building SPEC. (https://www.spec.org/cpu2006/Docs/install-guide-unix.html) The pass

Hello, I know that Mog was trying to get the bug-tracker cleaned up as the number of bugs has increased substantially over the past few months. I figured that I would do my part to bring attention to a couple of bugs that are interesting and have some wide reaching impact. That being said: http://bugs.digium.com/view.php?id=5970 - This has a patch created by Corydon, but it has segfaulted on

Hi, Can I escape a call queue by pressing a '*' or do I have to use a digit?? Thanks

Is anybody using 1.2 (or 1.2.1) in a production network using Realtime (voicemail, sip or extensions) with 100+ SIP phones? If so, what are your experiences? We've been running 1.0.3 for about a year and it's been rock-solid. We'd like to upgrade to Realtime and 1.2, but I'm afraid of killing our stability. Obviously, we'd do it in stages (upgrade to 1.2, then realtime