search for: nathanpralle

...inserted, which has advanced error checking (yay) and also changes the CallerID in Asterisk if it performs a successful lookup in the HooDaHek database. Thanks to Steven BerkHolz for pointing out that rather obvious tidbit. As always, information and download linkage available here: http://www.nathanpralle.com/software/hoodahek.html Nathan -- ------------------------------------- Nathan E. Pralle Give the Director a Serpent Deflector www.nathanpralle.com -------------------------------------

I see that the regular site for the Asterisk Perl AGI module is down (or something), so I've made an alternate download from my site: http://www.nathanpralle.com/downloads/asterisk-perl-0.08.tar.gz (just in case somebody was looking for it) Nathan -- ------------------------------------- Nathan E. Pralle Give the Director a Serpent Deflector www.nathanpralle.com -------------------------------------

Is there a guide anywhere which runs through how to set up asterisk with mysql? I've looked and almost all the document misses out relevant information. Thanks Dan Journo -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050920/1ce45adb/attachment.htm

I am wondering if anyone can help me figure out how to do something. I'm running a simple perl AGI which at the end of the call, creates a MySQL row with some of the tasks done during the AGI session. However, when the call is unexpectedly dropped (possibly the caller hangs up before finishing the entire AGI flow), I don't get the MySQL row created. I'm wondering if there is

...;t want to pass the Call Waiting pips along to the engaged user. IE: I want Asterisk to detect that CW is currently being transmitted on the line, and track it, but not pass it on. Is there a way to do this? TIA, Nathan -- ---------------------------- Interesting things abide: http://www.nathanpralle.com ----------------------------

Hi Group, I am trying to capture CDR records from Asterisk into PGSQL database. The problem is that the CDR information does not get populated until the call is hangup. Does anyone know if there is a way to get CDR info into database before call ends??? Thanks, Prakash

...tones from other answers were being cached and then dumped on the call to get_data. Anyone ever experienced this before? I have to do some major exploring, but nothing comes to mind yet. Nathan -- ------------------------------------- Nathan E. Pralle Give the Director a Serpent Deflector www.nathanpralle.com -------------------------------------

...#39;d ask the list. If we found something that lots of people wanted, it probably couldn't hurt to contact a company and ask for bulk deals. Whadya think? Anyone tried this before? Nathan -- ------------------------------------- Nathan E. Pralle Give the Director a Serpent Deflector www.nathanpralle.com -------------------------------------

Hi all. Done some research, Googled a lot, but can't find out if there is a USB FXO adapter that works well with Asterisk. If someone knows of one or has used one, I'd be very interested to hear about it. Many thanks, Nathan -- -------------------- www.nathanpralle.com --------------------

Hello, Can i identify the sound files that are played in the asterisk console? I defined the verbose to 100 but i can not see the sound files that are played in some situations... :( For example, I need to know what files are played for the message: "Extension xxx is unavailable...". The goal is to translate that to Portuguese (pt_pt)... Thanks in advance, PS.

Dear Asterisk Users, I'm a Japanese and now configuring Voicemail. Now I need to modify the way cmd VoicemailMain works to fix language difference and other my conveniences. What I want to do are... 1) Add words used in message retrieving guidance. I need to add different suffixes to numeric words due to Japanese way of mentioning time. (e.g. in English, you can say "Five

I have an asterisk box and SIP / IAX2 phones. To call out, users have to add 0 (zero) before a real telephone number. That means, that if they want to call someone that has a number 123456, they have to call 0-123456. Simple, right? This has a serious drawback though - when someone calls us from the number 123456, we see the callerID 123456, and we're unable to use the callback/redial

We are running our * server as a virtual PBX for 6 companies. I am having all of the Allison prompts plus our own custom IVR prompts being re-recorded for each company, in a different voice (marketing thing) with a different personality (perky, corporate, earthy) . I'm curious if someone could point out a dirty trick to get the voice to play right, for internal and external callers,

Hi, What do I need to do to get asterisk to NOT pickup a Zap channel when it rings? The channel in question is used for outbound calls only, and all incoming calls are answered by an analog phone elsewhere in the building that does not run through asterisk... so.. either make it not answer.. or make it delay for like 90 seconds.. I've tried wait's.. but it still seems to pickup the