similar to: Asterisk External Resources Page

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 want to have a list of companies providing services via IAX on my Asterisk web page. If you know of a company that does this or run a company that does this please e-mail me at eric@fnords.org with 1) web site, 2) contact info and 3) services provided. I don't want to put pricing info on the page. Thank you. --Eric -- Sample configs, scripts, more : http://www.fnords.org/~eric/asterisk/

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 =>

Can anybody recommend a good web interface for asterisk that actually works. I am looking for a web interface that can show how many callers are on the phone, should be able to transfer the calls and disconnect. I have tried using the flash operator but has been unsuccessful in making it work. thanks -----Original Message----- From: asterisk-users-admin@lists.digium.com

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

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

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

Everyone we have a few new things to give back to the asterisk community. http://bugs.digium.com/bug_view_page.php?bug_id=0002379 http://bugs.digium.com/bug_view_page.php?bug_id=0002380 http://bugs.digium.com/bug_view_page.php?bug_id=0002381 These include app_chanspy, the ability to spy on ANY bridged call taking place inside asterisk. NOT just ZAP as with ZapScan/Barge. Native format_* files

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!

In response to the SIP and NAT discussion, I have updated the ticket on the subject that seemed to be getting the most attention: #104. There are enough clueful people here that perhaps someone can come up with a patch that handles NAT in the elegant way that I describe in the bugnotes, as I am but a mere integrator who has limited C skills. In the absence of such a patch being offered, we

Hi there, I have serious problems with MGCP and Swissvoice ip10s, and it appears that recent CVS also introduced trouble for other MGCP users. Please check and add comments in the bugtracker so that we can get a clearer picture - thanks! Also comment if things are working fine for you. http://bugs.digium.com/bug_view_page.php?bug_id=0001542

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

Can anyone tell me how I can implement the features added in the following link for call transfer? The authors seem to feel they are finished but it doesn't appear to have been integrated into what everyone can download. It is referred to as a patch but I don't understand how it could be applied. Here is the link: http://bugs.digium.com/bug_view_page.php?bug_id=0002010 I guess I just

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: