search for: akrall

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

...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 Tel. 01-800-INTRUDER (01-800-468-7833) Email: akrall at intruder.com.mx Como lo estoy haciendo? Contacte a mi Director:

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. Anybody know why sometimes on some servers Asterisk shows more than once while doing a ps? [root@server2 akrall]# ps -ax|grep asterisk 20555 ? S 0:00 /bin/sh /usr/sbin/safe_asterisk 20557 ? S 0:00 asterisk -vvvg -c 20558 ? S 0:00 asterisk -vvvg -c 20560 ? S 0:00 asterisk -vvvg -c 20561 ? S 0:00 asterisk -vvvg -c 20562 ? S 0:00 asterisk -...

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.

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

...at 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 [mailto:akrall-lists@intruder.com.mx] Sent: Wednesday, May 25, 2005 10:41 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] CRM integration (was RE: CallerID) As one of the members said. The idea behind this is not to have to install any app on the computer b...

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?

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 just downloaded the zip file from grandstreams website to upgrade my gxp2000 firmware from 1.0.0.3 to the latest but seems there are some files missing on the zip file... Anybody been able to upgrade their firmware? My website shows this files as missing: 201.133.125.152 - - [11/May/2005:16:47:16 -0500] "GET /firmware/ring1.bin HTTP/1.0" 200 12737 "-" "Grandstream

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 > |[mailto:asterisk-users-bounces@lists.d...

Anybody doing email to fax using spandsp?

Guys. I have an issue with a te110p card and also some tdm04b cards on the same system: Zttest returns this for the tdm04b cards: [root@mollendo ~]# /usr/src/zaptel-1.2.4/zttest 38 -v Opened pseudo zap interface, measuring accuracy... 8192 samples in 8191 sample intervals 99.987793% 8192 samples in 8192 sample intervals 100.000000% 8192 samples in 8192 sample intervals 100.000000% 8192 samples

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? :)

Hey, For the bridge issue, take a look at 'notransfer=yes' option in your iax.conf. It'll force * to stay in the path http://www.mail-archive.com/asterisk-users@lists.digium.com/msg42262.html

Hello All, Wondering what sort of real world mileage people are getting out of different internet connecions - i.e. different DSL connection speeds, cable modems, etc... Is it reasonable to hope to carry 10 - 15 concurrent calls on a 768K DSL? I'm not talking about theoretical BW or looking for any difinitive absolute guarantee... With DSL and Cable - there is no guarantee, so I'm

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

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

Guys. Anybody has developed and asterisk brochure for commercial purposes (consultant, etc) that I might be able to take a look at?

Guys. Im having a big problem transfering incoming calls thru zap channels to some other extension. If the call is made by me to the outside via zap channels, no problem, hitting # gets me the transfer prompt, but if the call comes in thru zap and eventhough I am sending the call from the zap channel to my sip ata (GS ata 286) using Dial with wtWT as parameters, when hitting # I don't hear