similar to: Ping AGI Demo

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

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 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

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 a problem with asterisk when dial out to a SIP provider. Asterisk send a INVITE with no credentials, the provider reply with a 401 Unauthorized. However, Asterisk DOES NOT resend the invite again with credentials. But it hangs there (maybe waiting for a ok) It is this a bug in asterisk or the provider is supposed to send something else rather than a 401 as answer for a INVITE ?

Hi, I?m trying to get app_festival to work. I got the source from the Debian woody package of festival-1.4.2 and applied the patch that came with * sources it applied fine; then I made the debian package and installed it. I have this on extensions.conf: exten => 6700,1,Festival(Hi there how are you doing ?) When I dial 6700 I hear nothing and then * hangups: -- Executing

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

How do you get Cepstral working, they only offer windows versions. do I have to complie it to linux? http://www.cepstral.com

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 about to download Festival source, apply the astrisk diff's, and initiate basic testing. Thoughts are to download v1.4.3 (latest per the fesitval website. If anyone has an existing how-to, install notes, tips, or any suggestions I'd greatly appreciate it. Direct email is fine if you'd rather not post them. Thanks, Rich radamson@routers.com

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

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/

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

I am seeing strange behavior that I don't understand. Voicemail2 and voicemailmain2 work fine if I call from a sip phone directly connected to *, but if I call either of them from an analog line on the other side of a sip gateway, voicemail seems to ignore digits. If I am recording a message and press #, nothing happens except that it records the tone onto the message and I can't specify

Hi, I'm having two problems. First - I'm using the xten x-lite program to communicate with asterisk, and everything works fine except that DTMFs are not transferred. I've set DTMFMODE to inband on both the sip.conf file and the x-lite configuration, and still it doesn't work. Anyone had this problem before>? Second thing: I get a WARNING:[1209214400]: File dsp.c,

I'm about to begin working on an ivr project to do database backed scheduling. I would like to use text to speech in some places. What are the differences in using festival vs. Cepstral? How are they similar, how are they different? Is one really better than the other? How and Why? Thanks, Eric -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi All- I'm working on a project that will have remote (internet)access to an * server through SIP phones, either soft or hard ones. Does anyone have any experience to share about which SIP product they are using under similar conditions, as well as which codec is being used and bandwidth usage? TIA! PauloHM

Has anyone tried integrating Asterisk and Rhetorical's rVoice software? We're evaluating different approaches to system announcements via T2S. Has anyone gone down this route that could give some advice? I've installed festival and wasn't too impressed, the demo one the website seems far better quality and clarity then the defaults in the source package. However I must admit