Re: [asterisk-users] BT IP Exchange interconnect
Gavin Henry wrote: 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. We have, asterisk 100% + Kamailio... 9 months of work (maybe less now) :) -- Regards, Senad Jordanovic, CEO Bicom Systems Ltd, +1 619-760-7770 +44 20 3399 8877 se...@bicomsystems.com www.bicomsystems.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Russia Calls Skype/VoIP Security Threat
Alex Balashov wrote: Good luck with that. Brent Davidson wrote: Anybody seen this article yet? Looks like Russian Telecom business have decided that VoIP is going to put a dent in their profits so their pitching it as a threat to Russia's national security and working to get laws put into place to make sure the government controls VoIP providers operating in or providing services to Russia. http://www.reuters.com/article/technologyNews/idUSTRE56N41I20090724?feedType=RSSfeedName=technologyNewsrpc=22sp=true ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Agree to @mission impossible@ Regards, Senad Jordanovic, CEO Bicom Systems Ltd, +1 619-760-7770 +44 20 3399 8877 se...@bicomsystems.com www.bicomsystems.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Announcement: Howler-optimised G.729A Solution for Asterisk
Jay Fenton wrote: [ Optimised G.729A 'Howlet' for Asterisk FreSWITCH ] Howler Technologies are proud to announce today the launch of their fully indemnified and highly optimised G.729A solution for Asterisk, including a unique floating license model. Why would someone buy it instead of Digium g729 codec? Senad ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Current state of Asterisk and Virtualization?
Gavin Henry wrote: Hi all, In a pure VoIP env, what is the current state of do's and don't s of virtualizing * in order to provide multiple separate instances, say for hosting lots of Asterisk-gui/FreePBX/a-n-other gui? I've read lots of threads going back to 2007 and I'm in the general option that kvm is the way to go now, if at all. If dadhi_dummy/zt_dummy is still an issue for conferencing etc. a conference box could be put along side the vm hardware and have a card in it. Thoughts, experiences and being told to shut up are all very much appreciated. Thanks. http://www.bicomsystems.com/products/C/P/797/411/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] multiple asterisks in a server
Danny Nicholas wrote: It is possible but not easy. Virtualization isn't necessarily the answer because of sharing the physical device(s) - If you're a SIP-only environment, then that wouldn't be a problem, but most * installs (IMO) use some flavor of Zap/DAHDI which has to be addressed/locked. Maybe you can use SERVERware... http://www.bicomsystems.com/products/ Senad -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Backeberg Sent: Tuesday, February 24, 2009 10:46 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] multiple asterisks in a server On Tue, Feb 24, 2009 at 2:59 AM, Rilawich Ango maillist...@gmail.com wrote: Hi all, Is it possible to install more than 1 asterisk in a single server? Can somebody help me understand why you would want to do this? I suppose development versus production, but wouldn't you also want better separation, like virtualization? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] fax / t38 gateway
Olivier wrote: 2008/10/24 Brendan Martens [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] Do you have any recommendations for good ones, or, non-buggy ones? Some of or resellers are using 2102 apparently with no issues :) Senad www.bicomsystems.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Softphone Framework or Libraries
Ricardo Melendez wrote: Hi to all, I have a project for Customer Relationship Management interfaced with asterisk, I need send the CallerID to my application (via http or tcp/ip), When the phone rings I need to launch a pop-up windows to the Call Center Agent to display customer info, do you know a framework/libraries to make this, if is possible a softphone embedded into html page for the same function. I need to choice one to suit it to my needs Thanks in advance. Ricardo Melendez Ricardo Try if outcall will do the job you need: http://outcall.sourceforge.net/ Senad www.bicomsystems.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6.0 CDR billsec and duration not working from h extension
Eric Chamberlain wrote:
Can someone tell me what I am doing wrong? Why doesn't CDR(duration)
or CDR(billsec) return the correct values?
cdr.conf
endbeforehexten=yes
extensions.conf
[macro-Dial]
; ${ARG1} - Dial String
exten = s,1,Dial(${ARG1},,M(post-dial))
exten = h,1,NoOp(Call was hung up - ${CDR(duration)} seconds long,
billed for ${CDR(billsec)} seconds)
The log shows:
-- Executing [EMAIL PROTECTED]:1] NoOp(SIP/10.10.10.170-b7d94f78,
Call was hung up - 0 seconds long, billed for 0 seconds) in new stack
But cdr-csv/Master.csv has logged time values for duration and billsec:
,510555,+410001,pop-inbound,1510555
510555,SIP/10.10.10.170-b7d94f78,SIP/
voipprovider.com-089ae8a0,Dial,SIP/1510555:password::[EMAIL
PROTECTED]
,,M(post-dial),2008-10-09 20:59:00,2008-10-09
20:59:03,2008-10-09 20:59:08,
8,5,ANSWERED,DOCUMENTATION,1223585940.35
--
Eric Chamberlain
Eric,
Asterisk CDR @logging@ is just no less then short of shite. An absolute
DEAD end.
WHO ever is in charge of Digium is not doing its JOB.
Senad
www.bicomsystems.com
(Morao sam da im kazem :) )
___
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Re: [asterisk-users] Short question: CPU hardware requirements for Asterisk
Steve Edwards wrote: On Tue, 23 Sep 2008, Alejandro Cabrera Obed wrote: Dear all, just a short question: What is the best CPU hardware requirements (CPU, memory, hard drive) to install Asterisk with SIP/RTP protocol for 100-150 users, and routing the RTP traffic by itself (no direct RTP traffic client-to-client) Hi Maybe below document will help you with an idea what is @[EMAIL PROTECTED] http://www.bicomsystems.com/files/whitepapers/report-officeBOX-testing.pdf Senad www.bicomsystems.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Callcenter monitoring tool
voip crazy wrote: Hello all, Anyone expecialized with call center monitoring and reporting solution based on asterisk. A client of us, want to install a call center reporting solution for an asterisk server but I do not know which could be the best tool for that. I need a tool for reporting queue calls, agent calls, and disconnect cause. Any clue will be appreciated. Thanks in advance. VoipCrazy http://www.bicomsystems.com/products/C/P/798/154_2573/ Senad ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk end-user GUI?
Ken D'Ambrosio wrote: I badly want to roll out Asterisk at my job. Unfortunately, my boss is dazzled by shiny objects. We had a vendor in today who showed us their system which, honestly, didn't suck -- but boy, is it going to be expensive! One major component of the eye candy was an end-user interface that allowed the user to initiate calls to a contact list, check for presence, create conferences, etc. Is there anything like that, aimed at end-users (as opposed to admins) for Asterisk? I'd even be willing to go with proprietary; I just don't want a wholly-proprietary, hobbled, licensed-to-Heck-and-back system, which is where it looks like my boss is leaning. Thanks! -Ken The original, the very first, and stil the best used by SMEs, corporations and goverments worlwide... PBXware :) http://www.bicomsystems.com/products/C/P/798/154/ Senad ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco vs Asterisk
Patrick wrote: Al Baker wrote: Quote Yet amazingly (if this is, indeed, a source of amazement for you), CCM and other Cisco software can be just as buggy as anything OSS, if not worse. This is simply NOT TRUE and shows a complete lack of understanding of modern software development. CISCO software is developed in a CMM environment. It has a formal test methodology and uses Automated Testing on EACH new release to ensure that 100% of the software that functioned in the Last Release, actually works in this release. Further, there is mandatory soak-testing for all new software. Sorry, anyone who wants to compare Professional TELCO GRADE software development with Open Source is just Completely and Totally freakin clueless. I don't know where you got this idea but I've worked in the telco grade equipment business for years and I can assure you that I've seen bug riddled, jaw dropping releases that were borderline pathetic. Besides Benoit's examples of the CSR-1 and IOS releases, ask anyone that had the pleasure of using Cisco's early CCM releases (iirc those still ran on Windows). Maybe this comes as a shock but many vendors actually use their customers as a testing platform. They sell them stuff that has some, more or many bugs and fix stuff moving forward. They might even charge their customers for the latest releases with the bug fixes. Check out the changelogs of Cisco SIP firmware releases which you can only get legally when you pay for a SmartNet contract. A reason one might *think* that vendors have this elaborate development and testing methodology in place and that their stuff rocks in the stability and no-bugs-found-here department is to give oneself some piece of mind over the crapload of money forked over for the product and another crapload of money for the support contract/SLA. Not sure what the term is in English but I think it is positive cognitive dissonance. /me steps down from soapbox now :) Regards, Patrick Patrick, Well presented ... thank you...:) I will add one sentence: A software without a bug, is DEAD software. Regards, Senad ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco vs Asterisk
T G wrote: I'm a CCIE and CCVP. I have worked in the Cisco TSBU on both CCM and Telepresence systems I have two IP patents for the VoiP Lite protocols and have been designing and building OSS IPBXs for companies including Google going back to 2001. I'm not mentioning any of that to be jerk I mentioned it to say I'm as qualified as anyone to to compare the CCM and OSS servers. The only fair way to compare the two is a list of weights features, for example if cost is your biggest feature then OSS is better, if support is your biggest feature than Cisco wins. When a customer is comparing the costly (TCO) and best supported systems in the world with hundreds of thousands installed systems for the large global companies on the planted backed by 54,000 employees and over $25b in the bank vs, a FREE system with one layer of support maybe two layers of support, the features don't even come in the evaluation in my opinion. I once asked a manager why did you buy the CCM and he said no one ever got fired for buying Cisco if anything wrong, If push the OSS and it goes I could loose my job. I would get a list of the important features, because there is no answer to your question of which is better. What you mentioned above is mostly correct presuming you are referencing OSS being provided by an organisation with limited resources and perhaps limited experience in OS. Spin that into a perspective of a well organised company harvesting full potential of OS, adding its own proprietary software level allowing it to offer value products and EXCELLENT support, then I will strongly disagree with you. In particular where customer solution isn't just a solution, but rather its products and people becomes your business's communications partner. Senad www.bicomsystems.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [asterisk-biz] New faxing protocol. Good/Bad ?
Dovid Bender wrote: Hi List, I was thinking the other day that even with T.38 there are still some issues with faxing. I was thinking of a protocol that instead of just sending down the fax tones an ATA or VOIP fax machine would get the entire fax convert it into some sort of image and pass it down the line to the receiving end. I got the idea from RFC2833. Yes I know that fax machines send bit by bit and get a conformation on it but maybe this would work a bit better. Send the entire image over and then get a response when it is done. This way if there is issues along the way the packets can be re-sent with out any issue. Dovid (Cross posted to Biz for those that aren't on the users list - I want their onion too ;) ). Hi For those interested in above, PBXware had it implemented 2 years ago. Regards, Senad ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk virtualization on VMWARE SX infrastructure
Michiel van Baak wrote: On 14:42, Sat 17 May 08, Steve Totaro wrote: On Sat, May 17, 2008 at 2:18 PM, nik600 [EMAIL PROTECTED] wrote: Hi what about asterisk virtualization on VMWARE XS infrastructure? The system installed will manage a call center with 50 operator, queues, CDR logging on external database. the protocol used is SIP, probably with G711 codec. Virtualization of Asterisk i a risk regarding performance? Thanks to all I wouldn't do it. Maybe in a lab but certainly not for a 50 seat call center. I would ;) We run asterisk under vmware in production and have no problem with it. This is in a pure voip setup. So do many of our customers handling many concurrent calls. Senad ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk virtualization on VMWARESX infrastructure
Sam Tam wrote: Why if you have 50 operator then I would even consider using dual server running backup So the idea of using vmware may really be very risky, let alone not talk about performance issue Sam What experience do you have experience with a virtulisation with VoIP? Senad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Senad Jordanovic Sent: Friday, May 23, 2008 3:37 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] asterisk virtualization on VMWARESX infrastructure Michiel van Baak wrote: On 14:42, Sat 17 May 08, Steve Totaro wrote: On Sat, May 17, 2008 at 2:18 PM, nik600 [EMAIL PROTECTED] wrote: Hi what about asterisk virtualization on VMWARE XS infrastructure? The system installed will manage a call center with 50 operator, queues, CDR logging on external database. the protocol used is SIP, probably with G711 codec. Virtualization of Asterisk i a risk regarding performance? Thanks to all I wouldn't do it. Maybe in a lab but certainly not for a 50 seat call center. I would ;) We run asterisk under vmware in production and have no problem with it. This is in a pure voip setup. So do many of our customers handling many concurrent calls. Senad ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is Asterisk ready for Prime-Time?
John Novack wrote: Bill Andersen wrote: This is not a troll. I've used my real email because I want this taken seriously. I'm not trying to make anyone mad, I just want some real discussion on this issue. Please bare with me... I'm a USER of Asterisk. We purchased 3 commercially available Asterisk Based PBXs a little over a year ago. (I won't mention which one at this point - I don't want to bad mouth them - yet!) Two of the systems are very small (5 SIP lines/6 Polycom phones). The third is on a PRI with 30 Polycom phones. My smaller sites work pretty good. I've only had to restart Asterisk every month or so. However, my 30 station system is a continuous headache. I average a restart at least once a week. Sometimes a couple of times in the week. I'm always being called to fix something that just stopped working. I DON'T WANT TO GET INTO A Well, don't just complain, tell us your setup and we can help you get it working. This list HAS helped me figure out some of the issues. THANK YOU! But the purpose of this post is more of a fact finding mission. 1) Was choosing Asterisk for our company the wrong decision... a) IF... I expect a phone system to just work. Once it is configured, a phone system should just work with very little attention. My previous system was a Comdial with external voice mail on a DOS based PC. I LITERALLY WENT OVER 4 YEARS WITHOUT HAVING TO REMOVE POWER TO THE COMDIAL CONTROL OR RE-BOOT THE VOICE MAIL PC. b) IF... I really only need a phone system that allows an operator to answer each call and transfer them to the appropriate person. I need voice mail, but very little auto attendant features (mostly after hours). All the bells and whistles that Asterisk offers are cool, but don't bring that much to the table for our purpose. c) IF... Stability is more of an issue than high end features? 2) Are there any users out there that really DO have an Asterisk system that just works like clockwork? I'm saying, once setup, run for a year (or more) without any issues? 3) If SO, Should I simply consider a different vendor? 4) If NOT, and if my expectations are that a system SHOULD just run and run without any problems. Is Asterisk simply not my solution. Is Asterisk not REALLY ready for production. Because in my mind (as a user of phone services), dealing with the phone system, even on a MONTHLY basis, means that the system is NOT really production ready... Before we installed an Asterisk based PBX, I spent maybe 4 hours per YEAR with phone issues (setting up a new station?). Since we moved to an Asterisk based PBX, I spend 4 hours (or more) every WEEK! Am I expecting too much? Bill For those of us who have spent many a year in telephony, I tend to agree with you. Asterisk is NOT ready for prime time Total cost of ownership for a supply house system ( Comdial, now Vertical, with a Keyvoice DOS based VM ) or an NEC DX series with VM on a CF card) in a small to medium sized office simply hangs on the wall and works, for years and years, and has many more features than most offices need or use. Last month I replaced a 20 year old system that finally failed, I have other systems installed and working in excess of 10 years, and seldom have any service issues. Mostly are user reeducation on mailboxes and the like when people leave and no one knows a password. Square Hybrid Key systems ( Shared Line Appearance ) have worked flawlessly for 20 years, and a host of other features that Asterisk is still struggling to get working. Many of these systems are more affordable than Asterisk at either the wholesale or retail level as well. The current fad is IP or VOIP and regrettably many businesses jump into the deep end of that pool without the faintest idea of where they will land. That said, there is also a place for Asterisk or a like system, and many of the users on this list have them in place and doing the job, but the system is not hang it on the wall and forget it. PC based systems in general from a hardware perspective are NOT as reliable, nor is the operating system or the application. They DO need to be restarted from time to time. In fact in my experience the system should have an automatic reboot once a week at a quiet time. Many versions of Asterisk can get insane and be cured by a simple reboot that seems to give the real Linux experts the heebiejeebies. A reboot should not be considered blasphemy, though I have only seen one hang on the wall system that needed that in 20 years, and that was strictly due to a timing issue with short term power outages. Too many select the equipment and system they know, rather than what is right for the customer. I am in a group of antique telephone equipment collectors that use Asterisk as an interface to a world
Re: [asterisk-users] Is Asterisk ready for Prime-Time?
130 physical extensions including 24x7 inbound call centre Debian on Dell server [EMAIL PROTECTED]:~# uptime 13:15:31 up 192 days, 23:49, 2 users, load average: 0.00, 0.01, 0.00 here is one more running multi tenant Hosted PBXes: saul ~ # uptime 18:59:11 up 263 days, 23:50, 1 user, load average: 0.96, 0.49, 0.35 saul ~ # Senad ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is Asterisk ready for Prime-Time?
Bill Andersen wrote: Senad Jordanovic wrote: However, I will say that it is not asterisk but people/company deploying it. Generally speaking after deployment, and as long users are using the system normally, no reboot is required. I'm thinking part of the problem IS the company deploying the commercial product we purchased. I really like their GUI. I'm an IT guy and I'd say out of the last 10 or so issues we have had with the product, I'm the one that figured out why it wasn't working correctly. They had to fix it, (their code), but I would see the symptoms and say Hey, could it be this?. I had one email from their programmer that said Good catch. Well, thanks for the Kudos, but why the hell am I paying an annual fee to catch your bugs! Yeah.. unfortunately that happens. Customers do find bugs, but but as always it is about how the software maker reacts to it :) Also, a friend of mine once said Software without a bug is dead software :) Regards, Senad ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Druid Open Source Edition
Steve Totaro wrote: On Sun, Mar 16, 2008 at 4:00 PM, Senad Jordanovic [EMAIL PROTECTED] wrote: Steve Totaro wrote: On Sun, Mar 16, 2008 at 3:27 PM, Senad Jordanovic [EMAIL PROTECTED] wrote: Joshua Wilson wrote: It is the same whether you are using trixbox, switchbox, pbxware or any other system roughly. You have to use their system. Sometimes, this is the only way to make sure everything they support is installed and integrated properly without problems. Hi Joshua, PBXware has tarball delivery method allowing you to use whatever Linux distro you choose. This was actually its very first delivery method before CD, appliance of VPSes. Regards, Senad www.bicomsystems.com It's funny how Senad and Bicom show up in any GUI thread ;-) Absolutely. It also hopefully shows how dedicated we are at what we do. Senad That, and you like to hijack threads. When someone mentions PBXware, as it is the very first ever GUI for asterisk or it is related to feature/function it is just normal for someone to comment. While we are on subject of Druid... Look at Druids trunks administration fields, you will find remarkable similarities in trunks administration with PBXware... (this is the last time I bothered to look) I wonder why? :) Senad ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Druid Open Source Edition
While we are on subject of Druid... Look at Druids trunks administration fields, you will find remarkable similarities in trunks administration with PBXware... (this is the last time I bothered to look) I wonder why? :) Senad Yes, but this Druid is Opensource and PBXware is not (this is the last time I bothered to look). Since the *entire* subject IS Druid (Druid Open Source Edition), if I were going to design a GUI at this point, I would take all the strongest parts of each GUI. It only makes sense. Many different people would say, hey that looks remarkably like our reporting, provisioning, trunk, CRM integration So you are saying that is OK to ignore IP (intellectual property)? Senad ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Druid Open Source Edition
Outback Dingo wrote: this I definatley concur Sure no problem. I have no problem with that. Steve do you want to continue of the pitch? Senad ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Druid Open Source Edition
Steve Totaro wrote: No pitch here. It is you making the sales pitch. This is the Asterisk Users Mailing List - Non-Commercial Discussion list which you and Dean repeatedly use for commercial purposes. I have stayed away and declared will not continue on this subject any further as other members expressed their concern. I have even invited you to take up the matter outside the pitch. The sport pitch dear Steve (soccer for example)... You chose to ignore it for whatever reason ticks you not to be!!! That SHOWS a lot about you. Should you choose to reply further, I will ignore your posts to allow you to SHOW your self further. Senad ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Druid Open Source Edition
Joshua Wilson wrote: It is the same whether you are using trixbox, switchbox, pbxware or any other system roughly. You have to use their system. Sometimes, this is the only way to make sure everything they support is installed and integrated properly without problems. Hi Joshua, PBXware has tarball delivery method allowing you to use whatever Linux distro you choose. This was actually its very first delivery method before CD, appliance of VPSes. Regards, Senad www.bicomsystems.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Druid Open Source Edition
Steve Totaro wrote: On Sun, Mar 16, 2008 at 3:27 PM, Senad Jordanovic [EMAIL PROTECTED] wrote: Joshua Wilson wrote: It is the same whether you are using trixbox, switchbox, pbxware or any other system roughly. You have to use their system. Sometimes, this is the only way to make sure everything they support is installed and integrated properly without problems. Hi Joshua, PBXware has tarball delivery method allowing you to use whatever Linux distro you choose. This was actually its very first delivery method before CD, appliance of VPSes. Regards, Senad www.bicomsystems.com It's funny how Senad and Bicom show up in any GUI thread ;-) Absolutely. It also hopefully shows how dedicated we are at what we do. Senad ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Microsoft Office Communications Server
Razza wrote: Imagine, repairing an engine of your brand new car you just bought? Imagine restarting your TV because it just froze? What if your shoes have just changed colour to blue screen? It will just not pass, will it? ... You will DEMAND a service for your car/TV,shoes or you may return it or whatever. So.. Imagine how much your business will be affected with a phone SYSTEM based on a such operating system, one which can not even meet basic desktop user requirements let alone crucial every day in/out business communications tool like a phone system. At the end, if you do not answer a call some else will!!! Senad Jordanovic www.bicomsystems.com http://www.bicomsystems.com What utter stereotypical dross. I would suggest to you learning how to use text emails and quoting first then you may have some responses that may be your worth while. Senad ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Microsoft Office Communications Server
Milton Calnek wrote: Tzafrir Cohen wrote: TV sets and such are simple enough. But when the device gets more sofficticated then, yes: reboot tends to become a first reaction. You say first reaction like there's some other choice with Windows. Right. Asterisk never crashes. Asterisk is completely solid. It's amazing what happens when you say Sure, look under the hood!! The free software community is full of examples of open source being more stable with fewer bugs than their closed source, commercial competitors. Of course it crashes... in wrongs hands :) Senad ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Microsoft Office Communications Server
Razza wrote: On 11/03/2008, *Senad Jordanovic* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I would suggest to you learning how to use text emails and quoting first then you may have some responses that may be your worth while. Clearly, you keep responding! Oh and move out of the dark ages and get a decent mail reader. there is a say... never start a silly fight first, but always make sure you finish it Guess, which one are you? Senad ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Microsoft Office Communications Server
Kristian Kielhofner wrote: On Mon, Mar 10, 2008 at 6:38 PM, [EMAIL PROTECTED] wrote: What is the logic of them using SIP over TCP? Is this a broad industry trend? Or just the latest attempt to get around SIP/NAT issues? Michael Graves mgraves at mstvp.com o(713) 861-4005 c(713) 201-1262 sip:[EMAIL PROTECTED] skype mjgraves FWD 54245 I would imagine it's because they plan on doing all kinds of neat stuff with SIP including video, TXT, Windows Updates, who knows... SIP over UDP has some pretty serious packet fragmentation issues. If you end up with a large enough SDP or something that causes a SIP packet to grow larger than the smallest MTU in the path between your two endpoints it doesn't work (no fragmentation support with SIP over UDP). SIP over TCP does not have this problem. Also, you need SIP TCP support for TLS... Well... I have been a MS windows desktop user for a while as many other people have. It mostly works except at times one needs to maintain/repair what one bought. I have switched :) Imagine, repairing an engine of your brand new car you just bought? Imagine restarting your TV because it just froze? What if your shoes have just changed colour to blue screen? It will just not pass, will it? ... You will DEMAND a service for your car/TV,shoes or you may return it or whatever. So.. Imagine how much your business will be affected with a phone SYSTEM based on a such operating system, one which can not even meet basic desktop user requirements let alone crucial every day in/out business communications tool like a phone system. At the end, if you do not answer a call some else will!!! Senad Jordanovic www.bicomsystems.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] I need the least expensive way to do this
John Novack wrote: You get what you pay for, if you are lucky Go with a used hybrid PBX, best bet is a Panasonic that suppots both POTS phones and prop phones. It will outlast ANY micro-computer based desktop hardware. Perfect for a church or very small office or retail business. You are mentioning technology currently desperately trying to update its firmware to 21st century environments!!! I just replaced one that was installed 20 years ago, and NEVER needed any attention until it finally failed. We regularly replace such systems with: http://www.bicomsystems.com/products/C/P/797/255_3649/ Senad www.bicomsystems.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk as a softswith for a small ISP
Sigma Networks wrote: you should take a look at Thirdlane's MultiTenant Edition management solution to partition Asterisk easily. http://www.thirdlane.com/solutions/service-providers Feel free to contact me offline. Well... If you need hardware all complete solution we have vSWITCH: http://www.bicomsystems.com/products/C/P/797/255_2883/ Or.. software/solution, look at http://www.bicomsystems.com/products/C/P/798/154/ Senad ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Looking for PSTN provider with unlimited inbound/outbound plan
Steve Finkelstein wrote: Hi Senad, Did you happen to find out if it was indeed anywhere in the US48? Thanks! - sf Hi Steve, Yes it is. Contact me of the list if you need to. Regards, Senad On 1/2/08, *Senad Jordanovic* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Dovid B wrote: Senad, You can get unlimited as in FREE to any where in US48 or just local ? As far I know it is anywhere to US48. I will find out and get back to you. Senad - Original Message - From: Senad Jordanovic [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com mailto:asterisk-users@lists.digium.com Sent: Monday, December 31, 2007 3:10 PM Subject: Re: [asterisk-users] Looking for PSTN provider with unlimited inbound/outbound plan Justin Case wrote: Tell me when to stop laughing. Multiple channels and unlimited minutes ? No sane person will give that to you. Yap I agree... but but for about $900 per month one could get T1 (24 channels) unlimited in/out as far I seen last time our providers rates. Senad On Dec 30, 2007 2:16 AM, Steve Finkelstein [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi all, I have a budget to work with and was wondering if there are any folks providing SIP/IAX2 trunking for unlimited inbound/outbound for a flat rate? We're in the budget range of roughly $5,000 a month and we need multiple channels per DID. I'm not sure if something like this is feasible in the world of VoIP -- and I only need to be able to make domestic/USA calls. Thanks for any potential leads. Happy holidays! - sf ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users . ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Looking for PSTN provider with unlimited inbound/outbound plan
Dovid B wrote: Senad, You can get unlimited as in FREE to any where in US48 or just local ? As far I know it is anywhere to US48. I will find out and get back to you. Senad - Original Message - From: Senad Jordanovic [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, December 31, 2007 3:10 PM Subject: Re: [asterisk-users] Looking for PSTN provider with unlimited inbound/outbound plan Justin Case wrote: Tell me when to stop laughing. Multiple channels and unlimited minutes ? No sane person will give that to you. Yap I agree... but but for about $900 per month one could get T1 (24 channels) unlimited in/out as far I seen last time our providers rates. Senad On Dec 30, 2007 2:16 AM, Steve Finkelstein [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi all, I have a budget to work with and was wondering if there are any folks providing SIP/IAX2 trunking for unlimited inbound/outbound for a flat rate? We're in the budget range of roughly $5,000 a month and we need multiple channels per DID. I'm not sure if something like this is feasible in the world of VoIP -- and I only need to be able to make domestic/USA calls. Thanks for any potential leads. Happy holidays! - sf ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users . ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Looking for PSTN provider with unlimited inbound/outbound plan
Justin Case wrote: Tell me when to stop laughing. Multiple channels and unlimited minutes ? No sane person will give that to you. Yap I agree... but but for about $900 per month one could get T1 (24 channels) unlimited in/out as far I seen last time our providers rates. Senad On Dec 30, 2007 2:16 AM, Steve Finkelstein [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi all, I have a budget to work with and was wondering if there are any folks providing SIP/IAX2 trunking for unlimited inbound/outbound for a flat rate? We're in the budget range of roughly $5,000 a month and we need multiple channels per DID. I'm not sure if something like this is feasible in the world of VoIP -- and I only need to be able to make domestic/USA calls. Thanks for any potential leads. Happy holidays! - sf ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Center Setup on asterisk
Godson Gera wrote: Contact me off list ;-) On Dec 12, 2007 5:39 PM, satish patel [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Dear all I need call center setup on asterisk so i need do doucment and book .is it available on net Satish, What sort of requirements do you have? Inbound/Outbound? etc... Senad www.bicomsystems.com PGP Signature-- Satish Patel mobile:- +91-9818875535 http://www.linuxbug.org Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://us.rd.yahoo.com/evt=51734/*http://tools.search.yahoo.com/newsearch/category.php?category=shopping ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users -- Godson Gera, http://godson.auroinfo.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Mystery phone!
Brian Hutchinson wrote: The web site is Russian (Serbian I think). Company is Hybird Systems (Hibridni System AD). Best I can tell which probably does not help much except to say it is a legit company that has been around a long time making computer stuff since the 60's. Here is original manufacturer: http://www.tecomproduct.com/IP2008.htm Enjoy!!! Senad Jordanovic (Central European Time) Bicom Systems Ltd +1 619-760- ex 7001 +44 20 7043 3480 ex 7001 [EMAIL PROTECTED] www.bicomsystems.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Virtual Appliances
Zaheer Master wrote: Hi All, Does anyone know of a good virtual appliance for Asterisk under VMware? I am very interested in the JEOS concept for reducing the attack surface of a machine, so I think an appliance might be a good way to do this. BTW, I'll be using this with direct SIP Trunking and Snom 370/360 IP phones, so no hardware card is necessary. Thanks in advance! Regards, Zaheer K. Master President, Adamant Security Inc. http://www.vmware.com/appliances/directory/576 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CALEA enforcement guidelines according to Comcast
Kristian Kielhofner wrote: Sounds like Comcast's manual for CALEA compliance was leaked. Pretty interesting read if you are curious: http://www.fas.org/blog/secrecy/ Direct link (PDF): http://www.fas.org/blog/secrecy/docs/handbook.pdf nice one Kris :) Senad ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Click to Talk Web Applications with Asterisk
zoachien wrote: Google for mexuar. Zoa Or look at one that works with MS Windows, Linux or Apple http://www.bicomsystems.com/products/C/P/319/382/ Senad ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multitenant or Multiple virtual machines
Any opinions/comments/recommendations? Before anyone recommends just buying the virtual PBX service from someone else, we _really_ want to do this in-house :) I am all for using VPS (virtual private servers) instead of the classic multitenant. Here are some reasons: - a VPS provides a Linux environment to each client. This is a big plus for some clients knowing they are not boxed - Any custom development, new features can be easily applied to individual clients VPSes without destroying/affecting other clients. Remember, these clients must have their phone lines up in order to trade :) - Firmware updates, bug fixes failures etc, will not affect other tenants. - I can move clients VPS to another server, another data centre across the world if necessary with a couple commands. - One can allocate specific resources to each tenant. This is very important for call centres for example. All of the above, and a lot more commercial reasons have made me think of developing administration high availability solution for VPSes which we and our customers use extensively. Thanks Senad www.bicomsystems.com ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multitenant or Multiple virtual machines
Tzafrir Cohen wrote: On Thu, Sep 06, 2007 at 04:38:40PM +0100, Senad Jordanovic wrote: Any opinions/comments/recommendations? Before anyone recommends just buying the virtual PBX service from someone else, we _really_ want to do this in-house :) I am all for using VPS (virtual private servers) instead of the classic multitenant. Here are some reasons: - a VPS provides a Linux environment to each client. This is a big plus for some clients knowing they are not boxed - Any custom development, new features can be easily applied to individual clients VPSes without destroying/affecting other clients. Remember, these clients must have their phone lines up in order to trade :) - Firmware updates, bug fixes failures etc, will not affect other tenants. But have to be tested and applied separately to each one = more work. Since this is custom development for client... Client is happy to pay for it... Next.. - I can move clients VPS to another server, another data centre across the world if necessary with a couple commands. - One can allocate specific resources to each tenant. This is very important for call centres for example. Allocating resources means that the global pool, which is normally not used, can't easily be shared. This can be a pain. Deviding the memory of a 2GB server between 8 tenants gives you 8 258MB servers. True, but done corectly it is huge benefit/saving. Just the fact that virtual machines/VPS technologies is now supported in kernels of many operating systems tells A LOT... Senad ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multitenant or Multiple virtual machines
Edgar Guadamuz wrote: A question. are the clients going to be able to manage the PBX? Yes... or are you going to give them the PBX service without access to each server? Up to you... Senad ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GUI for Asterisk realtime
Mike Clark wrote: Are there any nice GUIs out there for Asterisk Realtime? Google doesn't yield much. I spent a day trying to get VoiceOne to work without much success. Thanks, Mike Clark Are you looking for open source or commercial? Senad ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GUI for Asterisk realtime
Thanks. I had googled as well and found basically the same links. We are building a DUNDi/Realtime cluster and need gui management. Mike.. http://www.bicomsystems.com/products/ Senad ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Outcall 1.40 released
Hi OutCALL 1.40 is released. It is available in two flavours: - Without extension authentication - With extension authentication Changelog: OutCALL 1.40 (2007-06-29): - Multi-language support (French-Canada is included in the setup, while the English PO file is distributed with OutCALL setup which can be translated and added into OutCALL in run-time) Please use http://www.poedit.net/ for translation - Support for Skinny protocol - It is possible to define prefix for outgoing calls (Settings-General) - It is possible to define one or more prefixes which will be deleted from the incoming CallerID (Settings-General) - In Settings dialog, after you Apply changes, OutCALL automatically reconnects using new Server details (if those are changed) - It is possible to Import Contacts from CSV file which is generated using Outlook Export Wizard ( File-Export-Comma Separated Values (Windows) ) BUG fixes: - Critical BUG when Loading Outlook Contacts (some contacts would not be loaded if Contact's info contains some escaping characters) - Settings and other dialogs cannot be opened twice - Settings and other dialogs can now be accessed from the taskbar - Added all DLL dependencies into the setup Available at: http://outcall.sourceforge.net/ Regards, Senad Jordanovic www.bicomsystems.com [EMAIL PROTECTED] +1 (212) 400 7921 +44 (20) 7043 3488 Regards, Senad Jordanovic www.bicomsystems.com [EMAIL PROTECTED] +1 (212) 400 7921 +44 (20) 7043 3488 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Blip every 30 seconds?
Joe acquisto wrote: Telephone conversations that are being recorded, are supposed to beep periodically, to alert/remind the recorded person that the conversation is being recorded. Perhaps that is what you are hearing? joe a. On 8/2/2007 at 8:47 AM, Matt [EMAIL PROTECTED] wrote: Strange issue when I record a file from a phone to the asterisk system I get a blip in the recording every 30 seconds. It's a very small blip, but it is there.It seems like it's only if I'm recording, not when I'm playing back that the issue happens. My SATA drives, ETH0, and my Sangoma card are all on seperate IRQs. Any thoughts on what might be causing this and how to stop it? What version of asterisk are you using? Senad ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hi ability solution
Any High availiability solution for asterisk? VoipCrazy http://www.bicomsystems.com/files/projects/serverware/SERVERware.pdf ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hi ability solution
Steve Totaro wrote: Senad Jordanovic wrote: Any High availiability solution for asterisk? VoipCrazy http://www.bicomsystems.com/files/projects/serverware/SERVERware.pdf Is this free? Benefits over opensource packages? Thanks, Steve Totaro Steve... (and anyone else)I made a mistake replying to asterisk-users list thinking it is asterisk-biz list. Anything else please contact me of this list. Thanks Senad ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk GUI
For example, a user could post a message to the list asking I'm new to Linux and Asterisk. Should edit my dialplan by hand, use FreePBX, or buy a commercial solution? Imagine the response as you tried to convince them to buy PBXWare, FreePBX users try to convince them that they should start out using FreePBX, and others go on about how hand coding a dialplan is the one-true-wayR to learn Asterisk. Generally, the original poster is just looking to get everyone stirred up over nothing. In other words, Paul's original post of GUI bad! CLI good! was just the sort of post that is going to get folks fired up re-re-restarting the age-old discussion of which is better: CLI or GUI. Basically, it could be like posting any of the following: - Which is better: emacs or vi? - Which linux distribution is the best? - Which is better: Macs or Windows? All of these questions share the following: 1.) They have no right answer (macs are better for some, Windows for others, and linux for others still, not to mention OS/2, BSD, etc) 2.) People on the various sides of the debate have extremely strong feelings on the matter 3.) Nobody is likely to be convinced that the other side is right and that they are wrong. 4.) They have all been discussed thousands of times before, and nothing new is likely to be said on the matter. 5.) The only purpose served by the discussion, due to the reasons above, is to clutter up the mailing list. 6.) Any discussion thread regarding these sorts of topics is best avoided. For a more thorough description of an internet troll, see the following wikipedia article:http://en.wikipedia.org/wiki/Troll_% 28internet%29 In other words, if you see a post that is just going to result in a re-rehashing of the last rehash of a specific subject, just hit the delete key instead of clogging up the mailing list with yet another thread on whether a GUI or a CLI is better. (for example). In Paul's defense, it looked to me like his original post was simply a joke that was misunderstood. (I thought it was funny, anyway) I suppose I should take my own advice on this one, but sometimes I guess we all just can't resist. grin Tom Tom Thanks for your prompt and excellent response... Regards, Senad ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk GUI
Troy Ayers wrote: I would have been convinced if you had not top-posted! heh Rob Schall wrote: Tom, I disagree with your argument for a number of reasons. Each of these reasons should be more than enough to convince you I'm correct and you should do it my way and only my way. And for the record, VI and CLI. Rob Ability to listen is a gift. People who have it apply data received into prosperity and greater good personally and collectively. Senad ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk GUI
Tom Rymes wrote: On Jun 16, 2007, at 4:37 PM, Tzafrir Cohen wrote: On Sat, Jun 16, 2007 at 08:55:24PM +0100, Senad Jordanovic wrote: Brett Crapser wrote: On Saturday 16 June 2007 02:24 am, Senad Jordanovic wrote: Paul Hales wrote: GUI bad! CLI good! PaulH Really...? So explain why every major PBX manufacturer has GUI of some sort? Surely they would have had CLI only if GUI is bad!!! Senad Senad - it is really to cover the inability of 'average' people to understand CLI. CLI is useful for small/simple dial tone installations. Anything above that even very competent administrator will make syntax/logical errors. Hence automation is required. Automation does not imply GUI. Bad GUIs get in the way of automation. How many times does it have to be said? Don't feed the trolls! Tom Tom...Who in your opinion is a troll? Senad ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk GUI
Tzafrir Cohen wrote: On Sat, Jun 16, 2007 at 08:55:24PM +0100, Senad Jordanovic wrote: Brett Crapser wrote: On Saturday 16 June 2007 02:24 am, Senad Jordanovic wrote: Paul Hales wrote: GUI bad! CLI good! PaulH Really...? So explain why every major PBX manufacturer has GUI of some sort? Surely they would have had CLI only if GUI is bad!!! Senad Senad - it is really to cover the inability of 'average' people to understand CLI. CLI is useful for small/simple dial tone installations. Anything above that even very competent administrator will make syntax/logical errors. Hence automation is required. Automation does not imply GUI. Bad GUIs get in the way of automation. Automation is another subject/scope. However, GUI is collection of knowledge and experience. If applied correctly it can only improve the company offerings. I have personally spent years learning CLI in order to apply it to initial design of our GUI- PBXware. Thousands installation after, I have no full knowledge of CLI any more and I do not need to. It is embedded into PBXware and our team has collective knowledge of the whole solution. That is something CLI can NOT offer since detailed knowledge/training is required individually from the vary basics. That translates into: GUI - team/company knowledge, less training, faster time to market CLI - knowledge of individual / unnecessary dependency/training /longer time to market Senad ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk GUI
Paul Hales wrote: GUI bad! CLI good! PaulH Really...? So explain why every major PBX manufacturer has GUI of some sort? Surely they would have had CLI only if GUI is bad!!! Senad www.bicomsystems.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk GUI
Brett Crapser wrote: On Saturday 16 June 2007 02:24 am, Senad Jordanovic wrote: Paul Hales wrote: GUI bad! CLI good! PaulH Really...? So explain why every major PBX manufacturer has GUI of some sort? Surely they would have had CLI only if GUI is bad!!! Senad Senad - it is really to cover the inability of 'average' people to understand CLI. CLI is useful for small/simple dial tone installations. Anything above that even very competent administrator will make syntax/logical errors. GUIs do not make such mistakes and in addition do allow TRAINED average person to make changes by them selves. Any 4 year old can run a GUI and that is why the skill level of people programming phone systems has gone down hill so much. I have never heard of 4 year old been allowed to play with any companies phone systems !!! Remember no dial tone, no customers. As for skill level, I agree a lot training and patience needs to be invested into end users/resellers using GUI administration let alone CLI. Senad ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Call Center Application
bilal ghayyad wrote: 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 Try PBXware call centre edition. Full call centre stats, real time monitoring, unlimited agents etc. http://www.bicomsystems.com/products/C/P/319/154_2573/ Regards, Senad www.bicomsystems.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asterisk Business Edition Question
_ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: 26 April 2007 08:15 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk Business Edition Question Hi all, Thanks for the posts. We i think i should have elaborated on myself a bit more. Even though im nore a pure Linux guy i do have knowledge on linux. I had been Using TRIXBOX right from ver 2.4. I do know what exactly asterisk is and i am currently hosting 2 asterisk servers at remote locations and managing them on SSH. Why i was asking the question was because, i had a query from a customer who wants to have a PBX. As they are a corporate and they want their own staff to manage the asterisk remotely, i wanted to give him a hosted solution with GUI so that they can manage on their own with some initial training on managing it. they i bumped into Asterisk Business Edition and i thought of going ahead on it. All my questions were based on Asterisk Business Edition and not on my choices of other parallel solutions. So if the answers to that point that would be appreciates. Ofcourse i had a look at the comparison chart and if i got it right, there is no GUI inbuilt. Which means that i can give the customer a standard product. 2. Is there a GUI to manage asterisk? 3. Can it be compared with Asterisk NOW? 4. Is the CD a complete installation package? 5. If im looking for hiring a server on a remote location how will i be able to install it? So if anyone can talk about the above questions on Business Editions Features that would be great. Thanks On 25/04/07, shadowym [EMAIL PROTECTED] wrote: If you have an interest in learning a bit of Linux I would suggest looking at Trixbox. I would not have said that 1 year ago but it has come a long ways since then. Eventually as you learn more you can install your own Linux/Asterisk/FreePBX from scratch just for the sake of being able to learn more and have more control and less unnecessary bloat. Trixbox seems to be working fine for most people in production install these days though. There is the option to install AsteriskNow as part of the new TrixBox 2.2 release which is in final beta set to be production released very soon. Asterisk 1.4 and AsteriskNow are still immature so I would not consider either of them for a production install yet. AsteriskNOW is not nearly as full featured as FreePBX yet. They included it as an optional install in the new Trixbox for testing. -Original Message- From: [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] Sent: Wednesday, April 25, 2007 2:45 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk Business Edition Question Hi, Can anyone in the list help me with these queries on Asterisk Business Edition. 1. Why would anyone choose the Business Editon when the whole thing is avalable as GPL? 2. Is there a GUI to manage asterisk? 3. Can it be compared with Asterisk NOW? 4. Is the CD a complete installation package? 5. If im looking for hiring a server on a remote location how will i be able to install it? If someone can guide me on it, that would be great. Also i would like the users to share a bit on their experiences with it. Thanks Danny. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Sunil Charly Business Development Executive OrbitTel - KolTelecom ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Softphone that supports central provisioning?
Andrew Furey wrote: On 24/04/07, Senad Jordanovic [EMAIL PROTECTED] wrote: Tzafrir Cohen wrote: Dear Senad, The setup program for your soft phone can be downloaded from here: a href=http://malwareserver.com/malware.exe;http://LINK/a During the setup you will be asked for configuration file. Please use attached file. Tzafrir is referring to possible link that user can receive from someone... Since I was referring to SYSTEM email message generated from within PBXware, above is not possible without some serious hacking of the network, the box, the chroot etc... If one is at that level it then becomes a criminal issue. Not denying the criminal aspect, but who says the email has to really come from that box? If there's one thing SMTP is good at, it's allowing forged emails... it wouldn't take a decent phisher 10 minutes to craft an email that has all the same content including From addresses. Sure, the full headers would give up the game - but how many of your users would (a) check them, and (b) understand what they're seeing? I'd be surprised if it's more than 5% - and in many cases it only takes one person to fall for it... Andrew Hi Yeah, all valid points. Thanks for bringing this up. In order to eliminate above the setup program is actually in user self care on the local box. That is where the link refers to. The user self care is password protected. In addition, all of the above is on LAN. For someone to know there is installation going on at some LAN is very private matter so anyone wanting send these emails will have to be psychic. Regards, Senad ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Softphone that supports central provisioning?
Tzafrir Cohen wrote: On Sat, Apr 21, 2007 at 08:59:27AM +0100, Senad Jordanovic wrote: What about creating a configuration file on server for each soft phone extension automatically and then importing that file into the soft phone? In another words, user receives a link to the setup program and the configuration file in an email. That is how we are configuring our soft phones :-) This requires much education. Not to mention educating the users to run a certain program that that they have recieved by email, but not any other. Oh, and you may want to occasionally change configuation without going through the whole complex setup process. Please separate program from data. Right...so u mean this is difficult: -- Dear $USER, The setup program for your soft phone can be downloaded from here: http://LINK During the setup you will be asked for configuration file. Please use attached file. -- Senad ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Softphone that supports central provisioning?
Stephen Bosch wrote: Hi, Tzafrir: Tzafrir Cohen wrote: Dear Senad, The setup program for your soft phone can be downloaded from here: a href=http://malwareserver.com/malware.exe;http://LINK/a During the setup you will be asked for configuration file. Please use attached file. I tried this link, but it's broken. What gives? -Stephen- Stephen, Tzafrir is referring to possible link that user can receive from someone... Since I was referring to SYSTEM email message generated from within PBXware, above is not possible without some serious hacking of the network, the box, the chroot etc... If one is at that level it then becomes a criminal issue. Senad ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Softphone that supports central provisioning?
Stephen Bosch wrote: Hi, Tzafrir: Tzafrir Cohen wrote: Dear Senad, The setup program for your soft phone can be downloaded from here: a href=http://malwareserver.com/malware.exe;http://LINK/a During the setup you will be asked for configuration file. Please use attached file. I tried this link, but it's broken. What gives? -Stephen- Stephen, Tzafrir is referring to possible link that user can receive from someone... Since I was referring to SYSTEM email message generated from within PBXware, above is not possible without some serious hacking of the network, the box, the chroot etc... If one is at that level it then becomes a criminal issue. Senad -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Softphone that supports central provisioning?
Philipp Kempgen wrote: Tzafrir Cohen wrote: On Fri, Apr 20, 2007 at 11:48:20AM -0400, James FitzGibbon wrote: Has anyone found a softphone that supports pulling it's configuration from a central server via TFTP/FTP/HTTP, much like hard desk phones use? Why would you want to do that? Because you could provision softphones the way you provision hard phones. Dynamic configuration through HTTP or even SIP messages. That would really be great. I think it's a valid question and I've been searching for such softphones as myself. They should be usable (so most of them fail) and should work on a real OS (tm). And no Java please :) Regards, Philipp What about creating a configuration file on server for each soft phone extension automatically and then importing that file into the soft phone? In another words, user receives a link to the setup program and the configuration file in an email. That is how we are configuring our soft phones :-) Senad www.bicomsystems.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] outCALL- the open source Asterisk integration applicaiton for Microsoft Outlook
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 calls integrated with users Outlook contacts. The open source PBX market needed integration with Microsoft Outlook which works with Asterisk (www.asterisk.org) . After developing and offering outCALL as a proprietary application, we decided to release outCALL as an open source application licensed under BSD license in order to further stimulate development and use of Asterisk said Senad Jordanovic, the systems architect at Bicom Systems Ltd. We released OutCALL as open source in the wish that the application would to be of good use to everyone and to be enjoyed it for free. Some things are not just about money, we are pleased to contribute to the wider community said Sergej Kasumovic, Chief Developer at Bicom System Ltd. OutCALL is written in C++ . It is a stable and robust application. It took many months of hard work to get it into current state. At the end, I would just say that I will be very happy if OutCALL will make difference to someone. said Denis Komaradic, OutCALL developer at Bicom Systems Ltd. There is ever growing demand to see existing CRM style packages integrated with Telephony Platforms. There are many CRM programs both proprietary and open-source that could benefit from this code. We chose the BSD licence as in our opinion it allows for the broadest possible promotion of the software. We look forward to seeing this open up many more possible integrations with other existing software both by in-house and other commercial vendors, said Stephen Wingfield at Bicom Systems Ltd. For full details on Bicom Systems products please www.bicomsystems.com. To download a copy of OutCall, please visit http://outcall.sf.net/ .Documentation is available at www.bicomsystems.com/docs/outcall/ . About Bicom Systems Bicom Systems is a provider of PBX and soft switch turn key solutions with a presence in the United States and the European Union and supported by a network of resellers across the world. Its solutions allow easy deployment, maintenance and control of a wide range of telephony solutions. The company leads the industry in providing the most integrated, ready to deploy, feature packed Telephony Solutions for creating PBXs and BROADBAND PHONE COMPANIES. For more information about Bicom Systems, please visit www.bicomsystems.com. Outlook is a registered TradeMark of Microsoft Corporation. In no manner should this press release be understood to represent any relationship between Bicom Systems and Microsoft or any endorsement of either company or the products of either company. For more information, please contact: Stephen Wingfield 44-20-7043-3489 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] outCALL- the open source Asterisk integrationapplicaiton for Microsoft Outlook
To download a copy of OutCall, please visit http://outcall.sf.net/ .Documentation is available at www.bicomsystems.com/docs/outcall/ . I forgot to include in my original post: Please use http://sf.net for any further communications in regards to outCALL Thanks Senad ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Re: Two or More Bri Cards
Edoardo Serra wrote: I Always had very bad experiences with 2 HFC cards in the same box I strongly suggest you to use a dual port card Regards Edoardo Interesting... I mean one would think that is the case all the time. In another words, that is logical, and I though the same but recently we have installed: 4 x one port and 1 X 4 port cards into a same box running PBXware. That is 5 cards in total... No complaints for 3 months running 2-3 thousands calls daily. Regards, Senad www.bicomsystems.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Re: Two or More Bri Cards
Hi Senad, Could you elaborate ? Which type BRI cards did you mix ? Which driver and channel ? -- Hi HFC based chip based one port cards and Jugnhaans quadBRI using bristuff drivers. We have spent log time making sure the bristuff drivers are handled correctly for our customers needs. Apart from that, all standard configurations as per drivers docs. Above of course using PBXware which uses internal logic handling bristuff drivers. Senad Www.bicomsystems.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Need comparison between PBXtra, Trixbox, Thirdlane, Druid, Aheeva etc.
Hi, For a customer, I am looking for a good and reliable Asterisk based system. Five servers will be installed at different locations and will be linked together with each other. This system will work as a call center as well. It has to be a stable and reliable. Customer also needs GUIs for system administration and agents call activities. He also wants video conferencing Please help me select a good system. Try here: http://www.bicomsystems.com/products/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asterisk and outlook
Hi list. Has anyone used any commercial or open source application to integrate Asterisk into MS Outlook 2003 which can be used to place calls directly to contacts from Outlook? And if so how well does it work? Here you go... Enjoy:) http://www.bicomsystems.com/products/C/P/319/288/ Senad ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Management GUI
Hi Scott... http://www.bicomsystems.com/products/ Senad From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Scott Pinhorne Sent: 08 December 2006 11:00 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Management GUI Hi All Can anyone suggest a comprehensive GUI manager for Asterisk. It doesn't matter if it is open source or commercial. We currently have 100's of users currently managed via the real time database. Groups of users belong to their own contexts. We would like a system that is able to integrate with our current real time setup and then allow us the ability to customise every feature of a user account from an interface as well as allowing other users to login an only manage people within their context. The GUI needs to have a distinction between configuring phones to act as terminals and then configuring agents who can roam around these phones. I look forward to hearing from anyone that can suggest a good GUI or maybe from someone who has a GUI they can customise for us. Many Thanks in Advance SP ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] A question on ISDN cards... (in the UK)
Would anyone like to recommend a good and reasonable quality ISDN card for use in the UK, as after a lot of good results with TDM400P cards with several systems installed now, I need to look at a few ISDN BRI (old business highway about to move to ISDN2) and possibly a single-line PRI (ISDN-30) system. Many thanks, For low cost 1 port: http://www.bicomsystems.com/products/C/P/319/286_2875/ For 4 ports, try: http://www.bicomsystems.com/products/C/P/319/282/ We also have just done some tests with new Digium BRI card which is promising since it has echo cancellation on board. Some other people use CAPI cards so you may want to look at that too... Senad ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Desktop integration
Tim Panton wrote: On 13 Nov 2006, at 13:15, Ondrej Valousek wrote: Hi Dean, I will check that site - thanks for the hint. The biggest problem I see with authentication and I do not think mexuar could help me here (and I am definitely going to pay $2000 for it :-) But it is another story... Hi I have not had time to read this post from beginning. However maybe a FREE or partner copy of outCALL may suit your needs. More details at: www.bicomsystems.com Senad ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] OutCall Release
Tzafrir Cohen wrote: On Tue, Nov 14, 2006 at 03:41:08PM +0100, Stephen Wingfield wrote: LONDON, UK (14th November 2006) - Bicom Systems announced today it has released its first freeware software to the Asterisk Community, OutCall. Tzarif, Thanks for your contribution in clarification of outCALL . To avoid any confusion: free here means a limited license for one copy per user. As matter of fact we do not limit how many copies one can use. Knock your self out If you wish :) And for those who are wondering what this OutCall is (as this press release tell you nothing), it seems to be some sort of MS-Outlook integration designed for either Bicom's prorietary PBXWare, or for a standard Asterisk installation. Correct it works with either PBXware or vanilla asterisk. We had a great success with outCALL deployments. It is not our main line of product, we are not setup to collect 30-50$ per copy so what a heck, let asterisk community Enjoy it. Have a fun :) Senad ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Reception Console
Viktor Tatianin wrote: Hello Paul Yes, I very interesting Hi We have MS Windows based operator consol/ panel available :) http://www.bicomsystems.com/products/C/P/319/154_2571/# Senad ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Multiple asterisk same GUI
Michiel van Baak wrote: On 15:52, Thu 28 Sep 06, Sharon Lim wrote: Hi there, Im wondering, is it possible to have single GUI on same DB but write to different asterisk server? Means assuming you have 3 asterisk server with same configurations. Therefore with the same DB but it write to different asterisk server conf files. where is the connection that we should focus? What GUI ? PBXware can do above. In fact it was designed to do that from the day one. http://www.bicomsystems.com/products/ PLEASE NOTE: We are having some ISP routing issues, some parts of the world are not able to access the site hence please try later. Regards, Senad ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] High Availability with PRI failover
[EMAIL PROTECTED] wrote: Hi After a month or so using Asterisk we've had or first downtime period due to a faulty RAM chip on the server, so we're starting to think about the possible high-availability solutions. Hi If you can afford it, below will give you total fault tolerant solution. http://www.bicomsystems.com/products/C/P/319/255_2797/ Senad www.bicomsystems.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] RE VoipNow 1.2.0 Beta
[EMAIL PROTECTED] wrote: Yes it is an addon of Plesk, thats stating the obvious. But while your complaining about people writing stuff to use what are you doing. If your not a developer don't critisize the developers. I see nothing more than you displaying that you are the Vice President of a 2 man consulting firm. Which means you have to sell other peoples developed products. Not to mention you are being critical of plesk, yet you use to host you websites for your business. Dude, we all have opinions, like crapholes they all stink, your's just stood out. I will stand up here and join Greg... (and we are not 2 man business, nor is Greg's to my knowledge) One can not build the GUI administration for asterisk from a base that was designed for another purpose without reaching the limits. And guess what then...? a complete re-write will required. :) In addition... supporting your customers is not trivial with asterisk. You HAVE to know your telephony/VoIP stuff very very well specially since you are giving them the interface to which they have no knowledge of internals... Regards, Senad www.bicomsystems.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] DNS lookups failing for SIP register
[EMAIL PROTECTED] wrote: Joshua Colp wrote: Putting an entry in /etc/hosts, or using an IP address in the register line, works. That's really really weird, it should use the system to lookup the hostname... did restarting Asterisk not help? Joshua, That was exactly my reaction! Restarting Asterisk did not help. I had a look at the Asterisk source code, and it just uses gethostbyname() so I can't imagine why it would be any different from host or ping. Asterisk is not able to resolve on the open socket hence why one needs to to the reload! We found this months ago when we were implementing the dynamic IP support for PBXware. So... reload should do the job:) as why it did not do it on the restart is another question. Senad www.bicomsystems.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] VoipNow 1.2.0 Beta
[EMAIL PROTECTED] wrote: Tom Vile wrote: Did you look on the site? http://www.4psa.com/products/voipnow/demo.php Does above means that the license for voipnow need to be paid to packet 8 as well? http://biz.yahoo.com/prnews/060613/sftu062.html Senad Hate replying on my post but what a heck!!! My understanding is that ANY hosted IP PBX coded in any object oriented programming language is falling under the above mentioned patent. Anyone has any thoughts on this? Senad ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] VoipNow 1.2.0 Beta
Another reason not to do business in the USA! Any good suggestions on where to buy rack space in a country that is not honoring stupid US patent law and has great and secure Internet connections? Tom Ehrm... Russia, China... You could also try several European countries, such as the Netherlands, Luxembourg, Switzerland... Well if it was that easy :) IE... just because you host in another country does not mean that you are not obliged with laws and regulations where your business trades from or to... Senad ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] VoipNow 1.2.0 Beta
Tom Vile wrote: Did you look on the site? http://www.4psa.com/products/voipnow/demo.php Does above means that the license for voipnow need to be paid to packet 8 as well? http://biz.yahoo.com/prnews/060613/sftu062.html Senad ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Server redundancy
[EMAIL PROTECTED] wrote: On Tue, 2006-07-11 at 14:34 -0400, Mike Lynchfield wrote: #3b.. we need multiple listening addies.. since asterisk can only listen to one ip its sucks for now That was case for asterisk 1.x ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Fonality vs TrixBox UI
This is all based on information from several (8) months ago. Things could have changed. That is exactly my understanding of Fonality product as it stands. PBXware on the other hand offers: DELIVERY METHOD in CD, tarball, appliance or hosted (VPS) formats. An appliance for wide range of installation types, performance or reliability requirements INSTALLATION of PBXware system in minutes PBXware tarball onto existing Linux server PBXware CD onto any Intel/Amd/Via based server or commodity PC PBXware hosted into a virtual server(s) SETUP of Fully functional PBX system in minutes of User Extensions using special auto configuration of Trunks to PSTN/VoIP networks automatically CREATION of Unlimited number of voicemail boxes Unlimited number of ACD Queues Unlimited number of queue agents Unlimited number of DIDs Unlimited conference bridges Unlimited IVR (Interactive voice response) Unlimited call reports Unlimited number of queue statistics USE of The system with web browser Phones from Cisco, Linksys, Snom, Grandstream and Aastra PSTN network using E1/T1, Analog or ISDN BRI cards VoIP using SIP or IAX protocols. Unlimited number of operator using included OPCOM (Operator Console Module) NETWORKING of PBXware into a single global/national voice network PBXware into a network with no limit of number of extensions. OFFER Each user their own self care with access to personal data Integration with Microsoft Outlook for each user Remote access for travelling or home working users SETTING of Dialling permissions to system, network, national, long distance and international calls Employees privileges to the system administration System auto update mode System auto backup RECORDING of All incoming/outgoing calls for one or more extensions Current call instantly just by pressing a button MONITOR of Calls in real time for one or more extensions Status of extensions, trunks, queues, conferences and live calls Real Time queue and agent data System operation status Full info at: www.bicomsystems.com Senad ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Fonality vs TrixBox UI
[EMAIL PROTECTED] wrote: On Fri, Jul 07, 2006 at 09:35:18PM +0100, Senad Jordanovic wrote: This is all based on information from several (8) months ago. Things could have changed. That is exactly my understanding of Fonality product as it stands. PBXware on the other hand offers: [ snip sales pitch. Wrong list. ] And it is still a properietary app, that locks you to its vendor. No it does not and since you brought up the subjectWe NEVER so far have refused a customers request... EVER... Customer can choose to wait for regular queue development cycle or escalate it by purchasing the appropriate development. Either way will WILL implement it. Senad ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] TE420P/TE415P?
[EMAIL PROTECTED] wrote: - C F [EMAIL PROTECTED] wrote: I like the TC400P card, how many T1s will that take? or is it just a Daughter card on the TE4xx ? How many channels can it transcode? Neither. It's a separate device, entirely unrelated to any TDM cards (which means it can be used for any type of channel, not just TDM). The final specs for the number of channels are not yet determined, but we expect to do at least 100 channels of G.729 and/or G.723.1 per board. Kevin, What about the dimensions... i.e. half or full lenght? Senad ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Anyone see this?
[EMAIL PROTECTED] wrote: Dunno if anyone else has seen this yet: http://www.scmagazine.com/us/news/article/563800/vulnerabilities+put+asteris k+telephone+systems+risk/ we do not run asterisk (or any other critical services including PBXware) as root on the host as normal process. we are using chroot with very limited set of tools and non-root operation. so ... it is much more secure. Senad ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Anyone see this?
[EMAIL PROTECTED] wrote: On Fri, Jun 16, 2006 at 08:57:02AM +0100, Senad Jordanovic wrote: [EMAIL PROTECTED] wrote: Dunno if anyone else has seen this yet: http://www.scmagazine.com/us/news/article/563800/vulnerabilities+put+asteris k+telephone+systems+risk/ we do not run asterisk (or any other critical services including PBXware) as root on the host as normal process. we are using chroot with very limited set of tools and non-root operation. so ... it is much more secure. Well, that protects the rest of the system from a potential problem with Asterisk. But not the rest of the network. Not to mention that it does not protect the PBX itself. A good practice, nontheless. of course, but other than that one cannot do much more without going into IDS services. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Turning AAAH into a call-center
[EMAIL PROTECTED] wrote: I believe there are quite different levels for the Asterisk market, so most people who run call centers wont feel confident in downloading a couple of ISOs from the internet and setting things up themselves. l. Absolutely correct. I could not agree more. Senad ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] asterisk management interface
[EMAIL PROTECTED] wrote: 2006/5/8, Senad Jordanovic [EMAIL PROTECTED]: Why would you need to create your own? Many reasons: 1. not relying on already busy open source developers Free projects are getting better that closed ones because the programmers don't need to re-invent the weel. Yo don't need to relay on another developper. You can become one of them and introduce improvements like them instead of starting from the beginning. Do not agree with you because the base of the software can be very very difficult (if not impossible) to change hence creating more problems. If they don't accept your improvements, you can start a fork. It always will be better than starting from nothing and dealing with problems that are already solved. Yap... 2. creating something that you can possibly offer as your own commercial offering You can make commercial offerings with open source/gpl software. If you don't beleave this, look at Digium and many others. Absolutely... 3. have it designed exactly they way you want it from ground up If you think you have a better idea than the other projects that do the same thing (amp in example), and you think they can't be modified for implementing your idea, I think it may be a good reasong from starting a new project. I wish I had time for a hobby like that :) 4. have a lot fun with it (and headaches :) ) You also can have fun colaborating with other people. Sure thing... Senad ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] asterisk management interface
[EMAIL PROTECTED] wrote: http://www.freepbx.org Why would you need to create your own? Many reasons: 1. not relying on already busy open source developers 2. creating something that you can possibly offer as your own commercial offering 3. have it designed exactly they way you want it from ground up 4. have a lot fun with it (and headaches :) ) etc... It is a long road though. We started PBXware in 2003 and there are still many features we wish to implement. Senad ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] asterisk management interface
[EMAIL PROTECTED] wrote: Those are reasons for WANTING to create your own, he specifically said I HAVE to make my own and I wanted to know why he HAS TO create his own when there are fantastics products already available. There is a huge difference in saying I would like to create my own and I have to create my own. I totally understand the 'want', I want something that is different and don't the way I want but I don't need to right now. -Kerry upps.. u got me :) I suppose my desire to WANT to create damn very good product has just shown so much. In past I have tried with open source web solutions and they are generally fine until one comes to the point where the fundamental design of the product just stops one getting what they NEED. Senad ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] *.conf utilities for Asterisk
[EMAIL PROTECTED] wrote: Hi all, I was wondering if anyone has any recommendations for *.conf generators for Asterisk. Creating *.conf files manually for Asterisk requires too much effort for what I do other than minor tweaking. I run Asterisk as a network appliance (Astlinux on CF) so something like FreePBX that needs to run on the Asterisk server itself is not an option. I have been using IPmanager up until now which worked great but development has been discontinued on that product. PBXware will do that... Run a central web interface, and manage as many PBXware servers as you require. More info here: www.bicomsystems.com Senad ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] billing realtime
[EMAIL PROTECTED] wrote:
Hi Senad
i looking for same thing, that is consider absolutetimeout as a
timer, everytime is near t zero, 3 secs for example, renew it,
reacalculate real credit, and start again until some of the parties
hangup.
The problem is how to iterate in asterisk config, or in deadagi,
you will need some time values from asterisk anyway, CDR{billsec} and
CDR{duration}, because i think we have to give this control to
asterisk, he really knows the timing of calls. Now the problem number
two. Asterisk set those values above, when the call is completely
finished, i have tried with deadagi in php whit sleep function,
nothing, the values of the varialbles are set after hangup extension,
after deadagi final execution.
If I understood well, when each call is made u give him duration time
based on the billing.
Its wrong direction at start. The only possible solution is in the
asterisk. You need global variable with total time for all channels,
then you need the timer.
Timer can be one by each channel, and each channel timer decrements same
global time variable when it becomes a zero or less terminate all active
channels for that account.
The other way would be to have one timer who decrements global time
variable based on number of active channels. Timer is inactive when
there is no active channels for account.
To explain this, if timer decrement cycle is n second then he should
decrement global remained time variable ACCOUNT_TIME = ACCOUNT_TIME- (n
active channels at the moment) x (timer cycle in seconds).
Then check condition ACCOUNT_TIME = 0 if true hangup all active
channels for that account.
Then check condition (n active channels for account == 0) if true stop
the timer.
The n active channels should be checked on asterisk.
If you create account time variable when first channel of account
becomes active like AV_{some id} and timer who will process this
remaining time.
Then on each new channel for that account you just increment other
variable NAC_{some id} or decrement.
The best is that this variables be asterisk variables (global).
We have not tried above, so be my guest if you have free time :)
Senad
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RE: [Asterisk-Users] billing realtime
[EMAIL PROTECTED] wrote: On Wed April 26 2006 16:31, Jon Farmer [EMAIL PROTECTED] wrote: JP Carballo wrote: Yes, certainly, through deadagi. I just have one question though, why reinvent the wheel? There are prepaid systems that work with asterisk. I have yet to find a prepaid system that allows multiple concurrent calls per account. Most seem to be based on a pin number also which I don't want. Anyone know of a system that allows concurrent calls? -- Jon Farmer Telford, Shropshire, UK Hi Jon. If a customer has 10 minutes of call credit left and he makes 2 concurrent calls, how do you know to cut off the 2 calls at the 5 minute mark rather than cut off both calls after 10 minutes? The way we solved this is: 1/ Each account has incoming/outgoing channels 2/ Once call is started then the total balance is divided by number of outgoing channels for that account. This sets the time limit. 3/ If more calls are made then each new call has same absolute timeout. Above is not perfect, since we are limiting each call to less talk time then total balance allows, hence why we are currently looking into possibility in changing the value of absolute timeout in memory for each of the calls. Senad ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: billing realtime
[EMAIL PROTECTED] wrote: In article [EMAIL PROTECTED], Senad Jordanovic [EMAIL PROTECTED] wrote: The way we solved this is: 1/ Each account has incoming/outgoing channels 2/ Once call is started then the total balance is divided by number of outgoing channels for that account. This sets the time limit. 3/ If more calls are made then each new call has same absolute timeout. Above is not perfect, since we are limiting each call to less talk time then total balance allows, hence why we are currently looking into possibility in changing the value of absolute timeout in memory for each of the calls. The other situation to take account of is when the caller somehow adds to his prepaid balance while he has one or more calls in progress, in order to avoid being cut off during the call. Noted!!! Thanks :) Senad ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Digium cards for sale
Hi, We got following surplus for sale: TE210P $700 TE410P $1100 TE411P $1950 Bundle (All 3 cards) please make an offer :) Cards not used except for development testing. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] asterisk management interface
[EMAIL PROTECTED] wrote: Looks very nice.. Is it GPL, GNU? PBXware interface is not GPL/GNU currently. Some time in the future we may release is it under GPL/GNU license :)... ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] asterisk management interface
Anton Krall wrote: For most of my everyday needs I ended up coding my own small one in MySQL and PHP, does the job I need but its far from complete.. to me, AMP is still the king :) Hi everyone, i am face with an asterisk use management interface, at the pressent, i am using AMP (asterisk Management Portal: http://coalescentsystems.ca/index.php?option=com_contenttask=viewid=31Ite mid=57 ). Does anyone know a better and more documented management interface for * ? Thanks Try this: www.bicomsystems.com/docs/pbxware/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] call-limit kills hints
The outgoinglimit never worked, so we haven't had that part working for a long time. It's been disabled in the code since 1.0 I think. Sure... I was just informed that was the case :)... hence why we never used it but implemented an alternative method instead! Thanks for your reply! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] call-limit kills hints
Since the device status system relies on it, I rewrote the incominglimit and outgoinglimit into the combined call-limit. The keywords incominglimit and outgoinglimit will be removed, but call-limit will stay. /O Olle/// What happens when it not a simple phone/ATA but a providers trunk which sometimes need different values for IN/OUT channels? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Recommendations on web interface for IT staff
[EMAIL PROTECTED] wrote: I am proposing an Asterisk system of many servers to service multiple departments in a number of locatations to a large client. They have an IT department but their Linux skills are weak and they are likely to face a high churn rate in staff so it would not be wise to expect a high level of Linux expertise to be maintained. I am thinking it would be best to do the nitty gritty glue work at the config file level myself but have a web based interface to common tasks such as managing extensions, adding trucks, voicemail etc. They are anxious for obvious reasons that they are able to manage the system without having to call me every time they need changes. As it will be a multi-server system there will be some fairly detailed configs to put together, so I would think a [EMAIL PROTECTED] installation would not be suitable, but I haven't tested that theory so I am not against trying it. What recommendations for web management can you make from experience of larger systems? It doesn't have to be limited to free systems. I am also interested in opinions on whether you would implement one monster server to do everything and have parts to maintain it, or would your preference be to have one server per department and interlink them, keeping the hardware the same and having a standby system ready to fill in for failed systems. On one hand there is only one server to monitor, on the other there is redundancy but also complexity. I can see advantages in both approaches. Chris, PBXware comes as standard with the features your client requires: http://www.bicomsystems.com/popup/319/C/features/P_2571/#a1597 If you need more info please contact me! Senad ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Meetme and Sipura SPA-941 - bad jitter/distortion
I'd greatly appreciate any help or thoughts! try: RTP Packet size on SIP tab ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
