similar to: Inbound IAXTel failing?

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

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

Is there any way on the iaxtel.com web site to see if my asterisk is registering and what 700 number is associated with my iaxtel account? I registered many months ago but never used it. My asterisk shows registered, but I can't seem to receive any calls (callers get a the user is not registered message) -- BTEL Consulting 850-484-4535 x2111 (Office) 504-595-3916 x2111 (Experimental)

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

Hi all I've been trying to make * work with IAXtel to no avail, all seems ok in the config but am not getting anywhere This is what I'm getting from console (user/pass/dest # changed for obvious reasons): DEBUG[1133735216]: File chan_sip.c, Line 3841 (check_user): Setting NAT on RTP to 0 DEBUG[1133735216]: File chan_sip.c, Line 4891 (handle_request): Check for res for phone1

I have tried for over a month off and on to get iaxtel for inbound to work... and tonight after alot of troubleshooting we noticed this: iaxtel inbound will use the last entry in your iax.conf to auth against. So if [iaxtel] is at the top and say [voicepulse] at the bottom. An inbound call will try to auth against that [voicepulse] entry even with the [iaxtel] entry at the top of the file. Has

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

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

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 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

Does anyone know if the 1-800 iaxtel gateway is down? I've been trying to use it all day today and asterisk says it's ringing: Channel (Context Extension Pri ) State Appl. Data IAX2[iaxtel]/1 ( s 1 ) Ringing AppDial (Outgoing Line) SIP/2201-a253 (home 18888476626 1 ) Ring Dial IAX2/XXX:YYYY@iaxtel.com/18888476626@iaxtel But I

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!

Last night Mark and I made some changes to the IAXTEL tollfree outbound, and inbound access. The inbound access number has changed to: 248-724-0700. (This number is in Pontiac, MI Ratecenter, and is supplied by Telesthetic LLC, a next gen phone compnay) This number will say "Please dial your number now" at that point you can dial your 1-700-XXX-XXXX IAXTEL number assigned. In 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

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 :-/

<BLOCKQUOTE style="PADDING-LEFT: 8px; MARGIN-LEFT: 8px; BORDER-LEFT: blue 2px solid"> <DIV>Hello,</DIV> <DIV>&nbsp;can someone tell me what's wrong with this? I can't make toll free calls via iaxtel. Here's the definition in my extensions.conf</DIV> <DIV>&nbsp;</DIV><FONT size=2> <DIV>[iaxtel-trunks]</DIV>