similar to: Making list of IAX providers

I have a ALPHA version of my new ping AGI demo available. Access via: IAXTel 1-700-923-3645 or Dial(IAX2/guest@ext.fnords.org) When asked for an extension, enter 2101. This will bring you to the System Services menu. The Cepstral version of the ping is option 28, the Festival version of the ping is option 32. Please report problems and/or issues directly to me. I'm trying to get

Some of you may know me as ManxPower from #Asterisk at irc.freenode.,net I've posted my demp weather report Asterisk AGI script at http://www.fnords.org/~eric/asterisk/downloads/ I have no affiliation with Cepstral. Below is the README: Contact: Eric Wieling <eric@fnords.org> If you want a demo of this AGI script you may call via IAXtel 1-700-923-3645 extension 2101. Option 23 is

I'm trying to come up with good jitterbuffer related settings for my Asterisk boxes. I ran 4 pings for about 2 days from my main Asterisk server to remote Asterisk servers. During that time there were some large file uploads which caused the max rtt to be quite large. Here are the results: pkts loss min avg max mdev 132013 %0 70.36 78.13 1967.37 36.04 132013 %0 98.95 120.46 2419.24 111.26

Is anyone else having trouble receiving IAXTel calls? I don't know if it's my config that's broken or IAXTel that broken. Several people have given me their IAXTel numbers and calls to them all fail. I can call FWD numbers via IAXTel just fine. --Eric -- Sample configs, scripts, more : http://www.fnords.org/~eric/asterisk/ BTEL Consulting 504-899-1387 or 850-484-4545 or

I have the following in my extensions.conf: exten => 21,1,NoOp(${CALLERIDNUM}) exten => 21,2,GotoIf($[${CALLERIDNUM} = ""]?21|4:21|9) exten => 21,4,Playback(/etc/asterisk/interactive-services/no-callerid) exten => 21,5,Wait(1) exten => 21,6,Playback(/etc/asterisk/interactive-services/no-callerid) exten => 21,7,Wait(1) exten => 21,8,Goto(10,4) exten =>

I've submitted http://bugs.digium.com/bug_view_page.php?bug_id=0000434 requesting that Digium put up a page with links with external Asterisk related resources. If you have a web site with Asterisk related information, patches, samples, documentation, etc, please add a bugnote to the above URL. There is a lot of good information out there, but time and time again I hear complaints that

Can anyone confirm that the feature listed below works? I'm using AgentCallbackLogin and it never seems to log the agent off if they don't answer. /etc/asterisk/agents.conf ; Define autologoff times if appropriate. This is how long ; the phone has to ring with no answer before the agent is ; automatically logged off (in seconds) ; autologoff=15 -- Go to

How do I make a queue member (added with AddQueueMember) automatically be logged out the queue if they don't pick up? --Eric -- Go to http://www.digium.com/index.php?menu=documentation and look at the "Unofficial Links" section. This section has links to a wide variety of 3rd party Asterisk related pages. My page is the "Asterisk Resource Pages". BTEL Consulting

This is a new install so it's a fresh checkout of everything. Does anyone know what might cause this error? [chan_sip.so]WARNING[16384]: File loader.c, Line 239 (ast_load_resource): /usr/lib/asterisk/modules/chan_sip.so: undefined symbol: ast_moh_stop -- BTEL Consulting 504-899-1387 or 850-484-4545 or 877-677-9643

I have posted a link to the tarball of my rather simple AGI script that allows a user to input a Zip Code (USA only) via DTMF and have the current weather conditions spoken to them. This is the first release and I'm sure it will have some bugs. It requires a few modules from CPAN and the asterisk-perl AGI interface. It's a very small script. Available at

I have a quick question... In the previous thread http://www.marko.net/asterisk/archives/0210/0306.html it is mentioned Mark added support for MWI to the chan_zap. Is this in the zapata.conf and if so, if stutter is turned on then the MWI is turned on with it? Geoff

I have an Aastra 390 ADSI phone. It's not locked. I can call ADSIProg without a problem and it programs my phone. Calling Voicemail2 also programs my phone. However, in order for the VMail option to appear on the screen I have to go into the Services menu, pick Asterisk PBX and pick Select. Then the VMail softbutton appears on the screen, but any time I make a call it goes back to the

I'm using app_festival to speak some text to callers. I'm having two problems with this. The first is with IAX calls (I've not tried others) the first few seconds of the speech is garbled. The second problem I'm having is the the volume of the speech IS VERY LOUD. I tried putting the following in the siteinit.scm but it didn't seem to make any difference. (set!

The cheap ones on EBay won't work with the SC420 server. I have one and can't make any of the clones work. I do have one TDM40B card for analog stations that works well. The problem with the SC420 is that it won't let you set the interrupts yourself and you end up with interrupts being shared. =============================================================== Message: 26 Date:

Hello, I have that problem with codec G729. Please can somebody help me! WARNING[16384]: File codec_g729b.c, Line 413 (load_module): Unable to initialize va stuff: -1 == Detected 4 licensed G.729 transcoders WARNING[16384]: File translate.c, Line 219 (calc_cost): Translator 'g729tolinb' does not produce sample frames. == Registered translator 'g729tolinb' from format G729A to

I'm getting the following when sending to a specific fax machine. Any ideas? File name is '/var/spool/asterisk/email2fax/7F2SOeYJiU.tif' Changed from phase 0 to 2 Slow carrier up Slow carrier down Slow carrier up <<< NSF: 20 00 00 11 80 00 8a 49 10 53 54 49 52 4c 49 4e 47 20 43 4f 56 49 4e 47 54 00 67 00 80 80 80 0c 01 02 NSF without final frame tag The remote is made by

Hi All Has anything been done to fix the issue where the * box is sat behind a nat firewall? Regards Mark

Hi there, I'm very new here and would like to know if anyone has reccomendations on fundamental reading (other than the handbook) whick might prevent me from asking some really dumb questions (after this one of course). What I'm trying to do: I have a SOHO... very SO in fact. I would like to build a simple system with two analog lines. I do recording and can't always have a cell

Hello, I am trying to use 2 netjet cards under asterisk and isdn4linux. I am having a hard time trying to get them to work in terms of dial out. Does anyone have a working config I could look at for even one card (tried that, not much luck either). When i dial out: -- Accepting AUTHENTICATED call from 172.16.11.2, requested format = 2, actual format = 2 -- Executing

Is there anyway for a sip station to play MoH out of the speaker? I know I can do it by calling the station and putting it on hold. For example: On a samsung pbx with MoH, if you goto one of the workstaions and press a button The moh plays out of the speaker. Is there any way to do this with asterisk? Kevin, Honeycomb Internet Services