similar to: Automatically Logging Out Queue Members

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

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 =>

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

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 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 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 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'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!

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

Does anyone know of any inexpensive alternatives to the four port analog module offered by Digium ($305) what work seamlessly with asterisk? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040105/1644638e/attachment.htm

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. I just tried updating asterisk and I guess I broke something. Here's the log: Unable to find a path from G729A to SLINR Unable to find a path from ULAW to G729A Any ideas on what I should try? I tried nuking all the zaptel stuff in the system and the source and started over agian. Also nuked the asterisk config files.... I saw this asked once before but there was no reply :-/

Hi, Do the callers in USA dialling from USA Telco lines always have to prefix the CITY/AREA code with "1" in order To successfully make a call to other USA destinations? ---- I have not been to USA (yet) :) Ta SJ

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

Assume you have the messages button on your Cisco phone set to dial 3009. Here's an sample dialplan entry that will make the "DND" and "ToVM" and "Messages" button work as expected. This should work for both -stable and -head. exten => 3009,1,GoToIf($[X${RDNIS} != X]3009,4) exten => 3009,2,VoicemailMain() exten => 3009,3,Hangup exten =>