search for: bicom

Bicom Systems releases outCALL, an Asterisk open source Outlook integration LONDON, UK (11th April 2007) - Bicom Systems announced today it has released outCALL, an open source desktop application allowing integration Microsoft Outlook. OutCALL allows users an easy way for placing and receiving phone ca...

Are there any switchvox/fonality type Asterisk based PBXs where I can buy just the software? I don't want to buy their 'bundles' that come with junky PC hardware. I just want their software/GUI to run on my hardware. Does Asterisk BE come with a GUI management console for managing phones, queues, VM and the like? -Matt -- Matthew S. Crocker Vice President Crocker

...sire to build this, fullfills an immediate need for business. If your intention is just to build a GUI for Asterisk, read no further. If your desire is to build something more purposeful, your best bet would be to see the existing commercial GUI/HostedPBX offerings like Pbxware and Switchware from bicomsystems.com ( http://www.bicomsystems.com) and Thirdlane Technologies (http://www.thirdlane.com/opensource.htm) and the Open Source software like AMP and try to emulate (or preferably improve upon) them. My suggestion is to create a "VOIP Business in a Box System" that has inter-alia fo...

...ariable DIALSTATUS 200 result=1 (NOANSWER) Arrgh. Well, it looks like that DIALSTATUS variable is not getting updated properly. Since we can't tell what the AGI script is really doing or how it works, it would be darn difficult to fix it. Anyone for reverse compiling? I don't think Bicom would like that so much, though, so maybe you can make a feature request with them.

I'm working with the final details of the Astricon agenda. I haven't got anything so far on Asterisk GUI's and there are plenty of projects out there. I would like to invite developer's of Asterisk GUI's, both open source and commercial, to participate. What I'm thinking of is giving each GUI a slot of 10-15 minutes for a presentation and then a panel discussion on the GUI

>> We have bought PBXware GUI from Bicom systems and configured >> extensions >> with Polycom Phones as UAs. >> >> The Polycom Phones can dial out and make calls but I cannot make >> extension to extension calling. >> >> Googling did not help much. >> >> As you may be aware PBXware is...

...terisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Noah Miller Sent: Thursday, March 10, 2005 4:52 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Re: Polycom phones do not talk to each other >> We have bought PBXware GUI from Bicom systems and configured >> extensions with Polycom Phones as UAs. >> >> The Polycom Phones can dial out and make calls but I cannot make >> extension to extension calling. >> >> Googling did not help much. >> >> As you may be aware PBXware is a close...

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Has anyone done any integration with this? All I know so far is that it appears to use some non standard form of SIP. Any pointers? - -- Kind Regards, Matt Riddell Director _______________________________________________ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News -

Hi all I think this could be en old question. I would like to do a realtime billing prepaid system, mainly using asterisk. I have found few things; I can not get CDR function into agi because asterisk set them once the call is absolutely finish (at least main values for the main porpouse, billsec,duration, etc..) There is a patch that allow you to use CDR

Hi! I got some strange problem with routing loadbalancing. I cannot get the full speed from my ISPs until I get some big files from close ftp server. I have server with one connection to internal network and 3 to ISPs: __________ | eth1|---- ISP1 | | internal--|eth0 eth2|---- ISP2 net | | (~300 | eth3|---- ISP3 hosts

...other andcannot answer when we pickup Never used pbxware, but the context the sip phones dial out using specified in sip.conf needs to include the dialplan context of the phones in extensions.conf. On Thu, 2005-03-10 at 15:08, Kanuri, Seshu (Company IT) wrote: We have bought PBXware GUI from Bicom systems and configured extensions with Polycom Phones as UAs. The Polycom Phones can dial out and make calls but I cannot make extension to extension calling. Googling did not help much. As you may be aware PBXware is a closed source software GUI from Bicom Systems for configuring exten...

Hi I have been playing with trying to get voicemail.conf from DB, I am using cvs-head, but when I start asterisk, it dies a horrible death, because it cant load any voicemil setting. I looked at my mysql logs to see what query was being sent, and I get SELECT category, var_name, var_val, cat_metric FROM voicemail_users WHERE filename='voicemail.conf' and commented=0 ORDER BY

Hi List, Does anyone know of a good dual wan router that can handle SIP well and can failover between connections if there is a SIP issue on one of the lines (meaning there still is a connection however there isnt enough bandwith or sip packets arent going thru etc.) ? Thanks. Dovid -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have recently noticed that druid @ http://www.voiceroute.org has created an open source edition of their platform. I downloaded it today and installed it on a play system where I have about 20 ip phones ranging from cisco, polycom and aastra phones. I didn't even have to configure them as the system automatically did it for me. I have been using trixbox/freepbx combination for over that last

Hello! How do I handle calls when they reach a queue that has no members? Currently, the callers are thrown out, because of the autofallthrough. The message is app_queue.c:2094 queue_exec: Unable to join queue 'queue-name' == Auto fallthrough, channel 'Zap/3-1' status is 'UNKNOWN' It seems that Queue() won't continue at a specific priority - like n+101 - if

Hi list; I am looking for an application that can be used with call center, in this application we can integrate the telephony part of the call center (like CTI Client ad so on), any one can advise for a good application to be used with Asterisk Call Center? - Note: The application to be customized easy, to be able to use it with Banking, Telecom, Oil, .. etc. Regards Bilal

Hi All, Has anyone passed the tests using Asterisk: http://www.btwholesale.com/pages/static/Products/Converged_Voice/IP_Exchange.html I presume the same rules apply for scaling and possibly have OpenSIPS/Kamailio on the front? Thanks. -- http://www.suretecsystems.com/services/openldap/ http://www.suretectelecom.com

Please can someone look at my last two posts and try and shed some light onto why my system is dropping calls.. If I don't get it right we will be forced to drop Asterisk which I really don't want to do.. Thanks..

Hi, My users are currently using an operator console interface like this: see it at: http://www.whssf.org/interface.jpg which came with a Praxon PDX we got about 5 years ago, which is now unsupported, it works very good and converts any analog phone plugged into the system into a powerful console, (provided you have a computer next to it) you just provide the box ip, user login, user pass,

Hello, Ever since I started using Asterisk I always get this error: Jan 10 15:39:26 NOTICE[4501]: res_musiconhold.c:463 monmp3thread: Request to schedule in the past?!?! I have a dedicated system system that really runs only Asterisk: - Pentium III 500Mhz - 128MB of RAM - 10GB of Disk Space - SuSE v9.2 - MySQL - Apache (only for use with Asterisk) - NTP client for clock synch There is no X