similar to: Problems compiling on mandrake

Hi, I have a latest asterisk cvs sources on my box, the rest of enviroment is: gcc version 3.3.3 20040412 (Red Hat Linux 3.3.3-7) linux 2.6.7 and after checkout I get: In file included from chan_phone.c:36: /usr/include/linux/ixjuser.h:353: error: syntax error before '*' token make[1]: *** [chan_phone.o] Error 1 make[1]: Leaving directory

Guys, Im looking for a good text file editor for asterisk config files that can be embedded on a web page for online editing (on an interface), any recommendations? ________________________________________________________________ Anton Krall Direccion General Intruder Consulting A Division of IntruderEnterprises S.A. de C.V. www.Intruder.com.mx www.IntruderStore.com.mx Tel. 3872-2200 ext. 201

The method we use for web popups on incoming calls in the astGUIclient client app that we are working on for release next week is to use AJAX(Javascript + XMLHTTPRequest) It works in Firefox and IE5+ and doesn't require any META refreshes. We've been using this internally for the last month and it works great. MATT--- -----Original Message----- From: Anton Krall

Guys... Ive been having problems with my callerid and I have no more clues as to what I could be.. dates and times stamped on voicemail and info received on the phones display are off by +6 hours and also the date for example today is Jan 02 :) What can I do to modify this? __________________________________________________________________ Anton Krall

I have a weird problem... very puzzling.. Yesterday I had sound problems with the voice prompts, I couldnt hear them, so I rebooted the system and voila, I was able to hear everything.. so I went to bad.. and I just woke up and tried the system again and its back!!! I dial the voicemail system and I cant hear the voice welcome.. I can hear any voice prompts.... Has anybody had this kind of

I am also very interested in CRM integration. Anything I can do to help? One thing I don't understand is how is the browser being launched on the person's PC. Or is it not launched automatically? Anyone know of a simple app running on the desktop to do this? I looked into IPSwithcBoard and it appears like it should be able to do the job, but: a) It is pretty heavy - it does a lot of

Guys.. which free softphone is the best,grandest,most recommended one out there? based on your own experiences..

Hi all, I have read and done the LARTC HOW-to and almost everything work''s execpt from the inside network. When load balancing over the two access it only work on the linux router/nat machine 1) I am running stock mandrake kernel 2.6.8.1-12mdk and using stock shorewall config 2.0.8 2) I have two internet link and two internals link. Load balance setup work for the linux router/nat

Guys. How do those callcenter apps work with Asterisk where a call comes in and * send a URL and some screen popup up based on callerid or something or username or id and shows all the customers info? Anybody done that? What do you need to do that? If you are using ATAs or IP Phones, how do those integrate with the PC so the screen would popup?

Guys, Im new to asterisk and voip but Im have a couple of questions regarding the initial setup. 1. Im going to install an asterisk server at home, where I have 2 phone lines, what kind of card do I need to get? I was thinking about 2 X100P Cards, so 1 can have 2 FXO ports and regarding phones, what else do I need? Ive seen the Grandstream HandyTone HT-286, I guess that servers as and FXS devide

Guys.. Ive noticed that I have 2 mpg123 processes running, is that ok? also... can you make MOH random? Also, I dont know if there is a problem with my config but when listening to MOH, every 3 or so second I get a click sound which notices because music gets a hickup every 3 or so seconds... is this ok? __________________________________________________________________ Anton Krall

Greetings: We have been running a CVS HEAD version of asterisk from Mar 10, 2005 on ix86 (PIII-600) Linux 2.4.27 with ixj (chan_phone) hardware. In a hope of getting better 'chan_skinny' support (to attempt using a Cisco 7920 IP phone) I built asterisk 1.2.40 on this box. Initial tests verify that our previous dialplan is working (iax2 trunks, register sip phones, registering withour SER

Hi, I was just wondering if anyone knows how predictive dialers detect voicemail and answering machines, and if they could explain to me how that works. Thanks! Brian.

Hello Im trying to install an old version of Asterisk. But it isnt working: when I run "make install": gcc -o gentone gentone.c -lm ./gentone busy 480 620 Wavelength 1 (in samples): 16.66667 Minimum samples (1): 50 (3.000000.3 wavelengths) Wavelength 1 (in samples): 12.90323 Minimum samples (1): 400 (31.000000.3 wavelengths) Need 400 samples Wrote busy.h ./gentone ringtone 440 480

Guys, anybody has a 1.2.x compatible app_voicemail patched for Spanish prompts that can handle for example, instead of saying "trabajo mensjes" would say "mensajes de trabajo o mensajes trabajo" (inverse)? Also can handle singular and plural (mensaje vs. mensajes)? Anton

Uys, Steve Underwood I just got a Sangoma A101 card and Im using unicall 0.0.3.pre9 for R2MFC, I get the far and local end unblocked but as soon as I try to make a call I get dialing and then protocol failure.. Do you guys know if there are any issues with sangoma and unicall? Anybody has an a101 card working with unicall and r2mfc? Are you out there Steve? :)

Hi: Iam running kernel is 2.6.8.1-12mdk but the modules of zaptel are being installed to /lib/modules/2.6.8.1-12mdkcustom how can i fix this up, any one have an idea? Best Regards; Wissam _________________________________________________________________ Search from any Web page with powerful protection. Get the FREE Windows Live Toolbar Today! http://www.toolbar.live.com

Hi, I am a newbie in asterisk, i could compile it and run it with no problem on a RedHat 9. In the same box, i got a linejack and a phonejack cards and i downloaded the CVS driver from quicknet. This 2 card were working in a openh323 (openphone and pstn) project with gnugk on a RedHat 9. I am using the default samples, and i tried /dev/phone0 and /dev/phone1, but when i run asterisk, i get this

Could not find your post for 4 months ago. -------------- Original message -------------- From: "Anton Krall" <akrall-lists@intruder.com.mx> > Yes, check a post that I made about 4 months ago, I posted the cofig for > setting the speaker, handset and ring volumes .. > > |-----Original Message----- > |From: asterisk-users-bounces@lists.digium.com >

Guys I just read on the wiki: "2005-07-19 - long awaited extension lights (hint priority) and call pickup on various phones work with newly released asterisk patch digium bugtracker - feel free to test and report findings to the bugtracker to have this commited to cvs." How does this work? And will it work only on certain phones or can it work with the gxp2000?