Re: [asterisk-users] Call Recordings...
So basically, He wants all calls recorded, but he wants a sequence that he can push, so that when he rants and raves at a customer, there won't be evidence to say that he did that... :) Just a hunch on that. :) I don't know. Eugen On 7/22/08, Gregory Malsack [EMAIL PROTECTED] wrote: Hello, My boss is asking me to setup the asterisk server to record all calls. (Simple). However, he wants to be able to enter a key sequence during the call to stop the recording. Any ideas on how I would do that? Thanks, Greg No virus found in this outgoing message. Checked by AVG. Version: 7.5.524 / Virus Database: 270.5.4/1566 - Release Date: 7/22/2008 6:00 AM ___ -- 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 ___ -- 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] G729 license count...
Silly newbie question. Is the license only required for the Digium product? Or is it also required for the "unofficial, unsupported (other than yourself, and those who want to help)"? Thank you for clarification! es Moises Silva wrote: http://store.digium.com/productview.php?product_code=G729CODEC http://www.digium.com/en/docs/G729/g729policy.php http://www.voip-info.org/wiki-Asterisk+G.729+Licensing On Thu, Apr 17, 2008 at 11:14 AM, Carlos Chavez [EMAIL PROTECTED] wrote: I need a refresher course on how many licenses I need to buy. I have an Asterisk server that receives calls by SIP (G729) and then sends them to the PSTN via 32 Zap interfaces on an Astribank. I cannot remember if the license is per channel or per call so I do not know if I need 32 or 64 licenses for this application. Could anyone please remind me? -- Telecomunicaciones Abiertas de Mxico S.A. de C.V. Carlos Chvez Prats Director de Tecnologa +52-55-91169161 ext 2001 ___ -- 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] G729 license count...
Where you calling my question silly? :) Well here is another one..question that is. I was not aware of the licensing of G.729. Is it that way with other codecs? Some that I have heard are G.711 and SPEEKS/SPEAKS... Perhaps pointing towards a resource that would have what ones require a license and what doesn't. Maybe someone has that handy. Thanks! Es Raúl Gómez C. wrote: Another silly question, In the first Digium link posted before there is a line that said "The G.729 codec works with all Digium cards", but this license will work with a Sangoma Remora Card??? Or do I need to buy it from Sangoma??? (I don't know if the are selling G729 licenses) Thanks... -- Raul Gomez Linux Counter #156439 On Sat, Apr 19, 2008 at 9:41 AM, Steve Totaro [EMAIL PROTECTED] wrote: Required by all (with the exception of academic work possibly?) Thanks, Steve Totaro On Fri, Apr 18, 2008 at 9:55 AM, Eugen Soare [EMAIL PROTECTED] wrote: Silly newbie question. Is the license only required for the Digium product? Or is it also required for the "unofficial, unsupported (other than yourself, and those who want to help)"? Thank you for clarification! es Moises Silva wrote: http://store.digium.com/productview.php?product_code=G729CODEC http://www.digium.com/en/docs/G729/g729policy.php http://www.voip-info.org/wiki-Asterisk+G.729+Licensing On Thu, Apr 17, 2008 at 11:14 AM, Carlos Chavez [EMAIL PROTECTED] wrote: I need a refresher course on how many licenses I need to buy. I have an Asterisk server that receives calls by SIP (G729) and then sends them to the PSTN via 32 Zap interfaces on an Astribank. I cannot remember if the license is per channel or per call so I do not know if I need 32 or 64 licenses for this application. Could anyone please remind me? -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 ___ -- 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] polycom 501 stopped working
You mentioned that you removed sip.cfg. Do you have the file with the way it was before you made any changes? If so, try placing it back where it's supposed to be, then reboot the server, then reboot the phone. (p.s. I don't have an *, I don't know anything about your problem! this is just suggestions, based upon what you have written, please and don't yell at me if it doesn't work. :) es Jerry Geis wrote: try rebotting the phone. at boot up. hold down 4,8,6,* - that should reset the local config for the phone On Tue, Apr 15, 2008 at 5:58 PM, Jerry Geis geisj at pagestation.com http://lists.digium.com/mailman/listinfo/asterisk-users wrote: / Hi all, // // I have a polycom 501 phone that I rebooted today. It stopped working... // Normally the screen shows New call, Forward and that is all... // Now the screen shows New call, Forward, MyStat, Buddies. // It no longer accepts incoming calls nor can I make outgoing calls. // // I have reloaded factory defaults, and rebooted, reset and rebooted // and it wont go back to normal... // // What happened and how do I get it back to normal? // // All my phones boot my TFTP, This is working as I see the requests // in the log file. // // I was playing with sip.cfg (which I never had used at all before). // I have now removed the sip.cfg I created and rebooted the phone // but it still isnt working... // // The sip.cfg file is the only file I created/changed and now I have // removed it. // // What gives? // // Jerry/ Still coming up with the Mystat menu after doing that... Jerry ___ -- 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 really good?? (added, What is ViciDial??)
I'm glad so much has been sent about on the thread I create (bloated ego head :) ) It has gotten my curiosity up. What is VICIDIAL? Is it Public Domain? Pay for Software? What's it all about? (not looking for all the features, maybe I should put my understanding of it's functions and people can correct me.) It seems to be a software product that can handle call centers, be they in coming our out going calls. Has modules to take credit cards / and is customizable so that added functionality can be written. This is been very interesting! es Matt Florell wrote: On 4/14/08, Jay R. Ashworth [EMAIL PROTECTED] wrote: On Sun, Apr 13, 2008 at 04:39:39PM -0700, Steve Edwards wrote: The "shell script" approach has the advantage of "light weight." I do a "minimal" Centos 5 install and wget a single script which does everything -- configures the network, installs packages (OpenSER, Asterisk, Zaptel, Libpri, MySQL), adds users, and configures everything from services to timezone. I may stick with it, but it's getting a bit combersome and am interested in what has worked for others. Noted. Our solution may not help you all that much; I gather that with the exception of one small chunk of one file, all our boxen are configured exactly the same. It is actually two small chunks of two small files in Asterisk and one line in the vicidial conf file, and that's about it for unique server configurations, everything else is pretty much the same. We did recently add a custom backup utility to our SVN for VICIDIAL(AST_backup.pl) that will backup all conf files, agi, sound and other files(optionally web files and mysql DB and my.cnf backup) and tar/gz them then send to FTP server. This has worked well for multi-server backups for a couple of our clients so far and it will be included with the next release of VICIDIAL. The idea behind the script is to create a very simple hot-spare solution where all you have to do to replace a running machine is change the IP address of the spare server and un-tar/gz the file on a base-installed system and it will take the place of the failed machine within minutes. We haven't had to use it in production in this capacity yet, but it has worked in testing. MATT--- ___ -- 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 really good?? (added, What is ViciDial??)
Matt. Thanks for the reply and Link. That should get me started looking at that. Unfortunately, coming from the Nortel world. It may take some time to get up to speed on things. The hardest part (as I see it) is getting hardware/software instructions on setting up and then maybe connecting to someone elses box to play around with the integration of different sites. This looks like a good Fall/Winter project. Need to remodel the basement now. Anyway, I think that's a little off list. :) oops. It looks like there is a link on the web-page of the link that you sent, that provides a "startup from scratch! COOL! Thanks again. Eugen Matt Florell wrote: On 4/14/08, Eugen Soare [EMAIL PROTECTED] wrote: I'm glad so much has been sent about on the thread I create (bloated ego head :) ) It has gotten my curiosity up. What is VICIDIAL? Is it Public Domain? Pay for Software? What's it all about? (not looking for all the features, maybe I should put my understanding of it's functions and people can correct me.) It seems to be a software product that can handle call centers, be they in coming our out going calls. Has modules to take credit cards / and is customizable so that added functionality can be written. This is been very interesting! es Hello, VICIDIAL is call center software for Asterisk. It is designed around Asterisk, not compiled into Asterisk. VICIDIAL takes a different approach to the call center application from how Asterisk inbound Queues/Agents does it, since it uses Meetme rooms to house the agents allowing for more consistency across versions of Asterisk as well as a lot more flexibility in terms of features. The agent web interface is an AJAX application that will run well in most modern web browsers on computers with a PIII 500MHz or higher. With VICIDIAL you can do inbound/outbound/blended call handling and there are all sorts of features for call handling and agent functions. The latest VICIDIAL release is GPLv2, but for future major releases we are moving to the AGPLv2. VICIDIAL is free as in cost and speech. There are currently well over 400 companies using VICIDIAL in over 40 countries(unconfirmed survey results show over 700 company users, with over 17,000 seats total) and the agent interface is available in 9 languages. Hope that helps. For more info go to: http://astguiclient.sourceforge.net/vicidial.html MATT--- ___ -- 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] Loosing SIP registration.
How about a change in IP from the IP provider? es (just a calculated guess, but it was a 286 calculator. :) ) Klaverstyn, David C wrote: Hi All, I am having problems with some SIP peers. I seem to loose registration. If I reload SIP the registration comes back. They usually stay registered for about 2 days before they drop. The problem is not all of them drop usually just the list 2 in the list. The other strange thing is that the 2 the do drop their registration do not occur at the exact same time. It could be many hours between them. I am using Asterisk 1.4.18.1 Any help would be greatly appreciated. My parents server is having the problems. My server does not exhibit this problem. I just took my router/firewall down to them as I have just purchased a new one and they are still experiencing the problem. sip show registry Host Username Refresh State Reg.Time 202.168.56.133:5060 61990xx 105 Registered Fri, 11 Apr 2008 15:15:58 sip.pennytel.com:5060 61289xx 105 Request Sent Thu, 10 Apr 2008 21:38:54 sip2.bbpglobal.com:5060 617000xxx 105 Request Sent Thu, 10 Apr 2008 20:43:20 sip reload Reloading SIP == Parsing '/etc/asterisk/sip.conf': Found == Parsing '/etc/asterisk/sip-register.conf': Found == Parsing '/etc/asterisk/sip-klavo.conf': Found == Parsing '/etc/asterisk/users.conf': Found == Parsing '/etc/asterisk/sip_notify.conf': Found sip show registry Host Username Refresh State Reg.Time 202.168.56.133:5060 61990xx 105 Registered Fri, 11 Apr 2008 15:16:15 sip.pennytel.com:5060 61289xx 105 Registered Fri, 11 Apr 2008 15:16:16 sip2.bbpglobal.com:5060 617000xxx 105 Registered Fri, 11 Apr 2008 15:16:16 ___ -- 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] Connecting Asterisk to Nortel Succession 4.0 sip...
Succession 1000SG running 4.0. Using SIP trunks. es CunningPike wrote: What type of Nortel? How are you connected to the Nortel? CP Eugen Soare wrote: Well I am entering into a realm that I don't know. 3 sites with Asterisk 1 site with Nortel Asterisk/Sip calls working fine between the 3 sites. Asterisk to Nortel set calls working fine. (call comes from asterisk to nortel and rings telephone, people answer and talk happens, hangup call clears) Nortel to Asterisk. Set on Nortel gets a busy signal. Any suggestions on what to look for? Much appreciated! Eugen ___ -- 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 really good??
I am looking at that. hmm... what to do... don't want any regrets you know! :) thanks, es Vincent wrote: On Thu, 10 Apr 2008 11:46:48 -0700, Eugen Soare [EMAIL PROTECTED] wrote: So this is just a general question, Is Asterisk really good? Yes, but you should also look at an alternative that used Asterisk as a reference (www.freeswitch.org), and make an informed decision. ___ -- 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 really good??
wow! That was cool! thanks for the pdf. es Matt Florell wrote: On 4/10/08, Jay R. Ashworth [EMAIL PROTECTED] wrote: On Thu, Apr 10, 2008 at 05:49:26PM -0400, Al Baker wrote: Please share more about this. What/How are you "clustering" the boxes ? Is this all VOIP or TDMF front and VOIP for agents in back ? What kind of Boxes ? What O/S What tools are you using to monitor this big-azz mother ? What, Matt? You haven't already talked about this here? :-) My new job is Matt Florell's old job, where VICIdial got started. I haven't counted the boxes lately, but I think there are 14 with quad-T cards in them, separate boxes for MySQL and Apache. Our architecture is FXS T-1 channel banks for the agent phones, usually 1 + 3 IXC spans per box, though we turned up a box a couple weeks ago with 3 channel banks, and no spans. All TDM; the only VoIP is the IAX trunks hauling load-balance calls around. And just the usual VICIdial tools, mostly, though I'm fixin to deploy either Big Sister or Nagios. Of course I have talked about it here, 3 years ago:) I even gave a presentation about it at Astricon in 2005: http://eflo.net/presentations/Astricon2005_matt_florell_PDF.pdf It is a bit dated(as are some of the servers there) but it is a good description of how that system was originally set up. MATT--- ___ -- 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
[asterisk-users] Is Asterisk really good??
So this is just a general question, Is Asterisk really good? Reliability? Functionality? Customization's? I am coming from a Nortel world, were you pay for everything, and you can't delve into the software. But it seems that customization would be a great thing. Like, setting up a war-dialer to customer lists, incoming/outgoing faxes (that's possible with Asterisk, right?) and making your own cool voice mail stuff. But before I delve into it, I thought a question to the community would be in order. 2 more questions. 1 - Can you really make free outgoing calls from let's say Portland OR, to Frankfurt Germany? 2 - What would it take to set one up? cards / computer power / pricing on software? What has your experience been? Thank you for taking time to look at this post! Eugen ___ -- 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] Connecting Asterisk to Nortel Succession 4.0 sip...
Well I am entering into a realm that I don't know. 3 sites with Asterisk 1 site with Nortel Asterisk/Sip calls working fine between the 3 sites. Asterisk to Nortel set calls working fine. (call comes from asterisk to nortel and rings telephone, people answer and talk happens, hangup call clears) Nortel to Asterisk. Set on Nortel gets a busy signal. Any suggestions on what to look for? Much appreciated! Eugen ___ -- 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
