[asterisk-users] Configuring sip.conf to allow guest access
Hi, I am configuring a conferencing server and need to allow SIP clients guest access. In iax.conf, I can allow guest access to the [conference] context with this entry === iax.conf == [guest] type=user host=dynamic context=conference So anyone connecting without username/password will be logged in as guest and restricted to the conference context. How can I do the same in sip.conf? How do I also configure my xlite client to login as guest? How to configure IAX is explained here http://www.voip-info.org/wiki/index.php?page=Asterisk+config+iax.conf but not for SIP. http://www.voip-info.org/wiki-Asterisk+config+sip.conf Thanks in advance. ___ --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] Configuring sip.conf to allow guest access
Tried this...it worked...but is this the best way? == sip.conf == [general] context=conference allowguest=yes [guest] type=friend nat=yes host=dynamic canreinvite=no context=conference --- Richard OSS [EMAIL PROTECTED] wrote: Hi, I am configuring a conferencing server and need to allow SIP clients guest access. In iax.conf, I can allow guest access to the [conference] context with this entry === iax.conf == [guest] type=user host=dynamic context=conference So anyone connecting without username/password will be logged in as guest and restricted to the conference context. How can I do the same in sip.conf? How do I also configure my xlite client to login as guest? How to configure IAX is explained here http://www.voip-info.org/wiki/index.php?page=Asterisk+config+iax.conf but not for SIP. http://www.voip-info.org/wiki-Asterisk+config+sip.conf Thanks in advance. ___ --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 ___ --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-Java] SipShowPeerAction
I think you have to implement the ManagerEventHandler interface to handle the PeerEntryEvent.ManagerEventHandler mgrHdlr = new ManagerEventHandler { public void handleEvent(ManagerEvent event) { PeerEntryEvent peerEntryEvent = (PeerEntryEvent) event;System.out.println(peerEntryEvent.getStatus()); } } And you have to register to receive your eventHandler managerConnection.addEventHandle(mgrHdlr);HTH. richard Coco [EMAIL PROTECTED] wrote: Hi all,first of all sorry for the question. I know there isan asterisk-java mailinglist but i am not subscribedto this list and i am sure there
are asterisk-javaguru on this list who can help me.I am trying to get the status of a peer using"SipShowPeerAction". Unfortunately the getStatusmethod gives me everytime "null".SipShowPeerAction sipShowPeerAction = newSipShowPeerAction("2001");managerConnection.sendAction(sipShowPeerAction);PeerEntryEvent peerEntryEvent = newPeerEntryEvent(sipShowPeerAction);System.out.println(peerEntryEvent.getStatus());What wrong with this example? Maybe someone can giveme a working example.hope someone can help...thx in advance ___
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Re: [Asterisk-Users] Help with JIAXClient
I got most of it working. The sendDtmf() does not seem to work. However I can register and make calls using the call(...) method.I cannot compile the sources (native) so I just got the jar files from http://www.hem.za.org/jiaxclient/and played around with the java code to fit with my app.RichardScott Gifford [EMAIL PROTECTED] wrote: "Enrique Sanchez" <[EMAIL PROTECTED]>writes: I'm trying to make a little example program for register to an Asterisk PBX and dial a softphone, but i just can't register to the PBX.Buenos noches Enrique,I've been trying to get JIAXClient working for over a month with nosuccess. It fails mysteriously at different places with noexceptions, and my efforts to figure out why have so far failed.So I can't offer helpful advice, only sympathy. :)Does this work for other people? If so, what's your build environmentlike?---Scott.___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users___ --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] Help with JIAXClient
I think you have to set where to get the libraries (jiaxc*.jar files).Setup a webserver somewhere and put the jar files there. Then in your code before initialize client.setCodeBase("your URL to the jar files");HTH,richard Enrique Sanchez [EMAIL PROTECTED] wrote:I'm trying to make a little example programfor register to an Asterisk PBX and dial a softphone, but i just can't register to the PBX.package iax; import net.sourceforge.iaxclient.Call;import net.sourceforge.iaxclient.JIAXClient;import net.sourceforge.iaxclient.Registration; public class TestIAX
{ public static void main(String[] args) { Registration registration; JIAXClient client = JIAXClient.getInstance(); client.initialize (1, 10); registration = client.register("kike", "elkike", "10.32.81.31:4569"); client.setCallerID("Kike", "1001"); client.call("1002"); System.out.println(registration); }} I'm frustrated because JIAX doesn't throw any exception, but the code is not working properly. Greetings, -- Enrique Sanchez ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users___
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Re: [Asterisk-Users] Meetme from MySQL
try http://sourceforge.net/projects/web-meetmeChris Blunt [EMAIL PROTECTED] wrote:Hi List, Is it possible to store meetme config in a MySQL table?If so, any pointers would be appreciated.ThanksChris --Chris Blunt Entropy IT Ltd ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users___ --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] WebMeetme defines.php?
I think you should put your php files (the ones you downloaded) in /var/www/html where Apache looks for files to serve. The defines.php I think is in the lib directory (/var/www/html/WebMeetme/lib).Here is my defines.php?php define ("WEBROOT", "http://192.168.0.101/"); // Change this to your serverdefine ("FSROOT", "/var/www/html/Web-MeetMe/"); // Where Meetme is located //GUI titledefine("GUI_TITLE", "Web-MeetMe");define("GUI_ICON", "asterisk.gif"); define ("LIBDIR", FSROOT."lib/"); define ("HOST", "localhost"); // I think your MySQL serverdefine ("PORT", "3306");define ("USER", "root"); // user to DBdefine ("PASS", "mysqlpass"); // password to DBdefine ("DBNAME", "DB_CDR");define ("DB_TYPE", "mysql"); // mysql or postgresI am using though the one available at www.fitawi.com/Asterisk not from areski. Not sure how big a difference the conf will be.richard Jordan Novak [EMAIL PROTECTED] wrote:
I am looking at some directions on how to install and it is asking me to edit defines.php, it states that the file should be located in the source directory, but I cant seem to find it anywhere on my machine. Anyone been thru this? Jordan Novak Communications Technician ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users___
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RE: [Asterisk-Users] web meetme instructions
Do you have an idea when this new submission will be available?Thanks. Dan Austin [EMAIL PROTECTED] wrote: Sorry for the late reply, I was away on vacation.Version 1.2 was created by Areski and I extended it to include the scheduling functions. I guess I should get an account on the Wiki an d make some updates.If all you need is a tool for monitoring conferences, version 1.2 is the way to go. If you want scheduling features with optional granular access controls, then the latest version from www.fitawi.com/Asterisk is the correct choice. I am expecting a fairly large code contribution soon, from a nice gent with much more PHP experience than myself. Once I get a chance to review it I will post an update to 2.1. If you'd like I can notify you before I release it and we can work on getting your translations in. DanFrom: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ben QSent: Thursday, March 23, 2006 7:35 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] web meetme instructions Hi all,I also installed web meet-me and found it quite difficult (the app_cbmysql, with the makefile changes). But I got it work with some help from this list. :)I'm a little bit confused with the versioning shame. What's the current version 2.0 ( www.fitawi.com/Asterisk/ ) or 1.2 ( http://www.voip-info.org/wiki/index.php?page=MeetMe-Web-Control ) ?I did an internationa lization / translation (french and german) based on the 2.0 version that I could contribute if I know how.Dan, thanks for your work! benq ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users___ --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] Mitel 3300 PRI problems
I had this same problem with SX2000. I think you have to configure the Mitel to "Speed Dial" to your Asterisk server. What I think is happening is that the PBX grabs a trunk but does not "dial" into Asterisk.Ask the person configuring the Mitel PBX to setup an extension (e.g. 1234) when callers connect to the PBX. That's what they had to do to get data to my Asterisk.Goodluck.richardJohn Fulton [EMAIL PROTECTED] wrote: We are attempting to interface Asterisk with a Mitel 3300.Initially we tried FXO Ground start T1 and had many problems with that. Even spend 5 hours on the phone with a Digium engineer and were unable to get it working right.Next we turned it up as an EM T1 and that actually worked great. No problems at all.Now we are testing a T1 PRI configuration and have been unable to get it to work right. We tried different switchtype configurations. We are back to "national" (NI-2). I am setup as pri_net. And I know that is working right because if I set the PRI for pri_cpe, I get error messages in the log that the remote end thinks it's a cpe and to change the signalling.When a call it placed to the PRI, I get nothing. No answer, no data across the D channel, yet the Mitel says it is successfully grabbing a trunk in it's debugs.I turned up pri intense debug span and the ONLY thing I get is the following over and over again. It seems to be at regular intervals and I see nothing else even when I place a call from the Mitel to the Asterisk server. Any ideas? Here's the debug output:T203 counter expired, sending RR and scheduling T203 againSending Receiver Ready (115) [ 02 01 01 e7 ] Supervisory frame: SAPI: 00 C/R : 1 EA: 0 TEI: 000 EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 115 P/F: 1 0 bytes of data-- Restarting T203 counter [ 02 01 01 e7 ] Supervisory frame: SAPI: 00 C/R: 1 EA: 0 TEI: 000 EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 115 P/F: 1 0 bytes of data-- ACKing all packets from 114 to (but not including) 115-- Since there was nothing left, stopping T200 counter-- Stopping T203 counter since we got an ACK-- Nothing left, starting T203 counter-- Got RR response to our frame-- Restarting T203 counter http://www.telalaska.com -TelAlaska, Inc. John Fulton201 E 56th Ave. Senior Network ArchitectAnchorage, Alaska Cisco Certified CCNA/CCDA99518 Microsoft Certified MCP907 563 2003 A+ Certified TechnicianFax 550 1646 [EMAIL PROTECTED]-- http://www.arctic.net --Interior Telephone Company Mukluk Telephone CompanyEyecom Cable TelAlaska Long DistanceTelAlaska Networks Seward Net and Arctic Net--- ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users___ --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] Failed to read gains: Invalid argument
Hello,When I start Asterisk, I get the following in my log (/var/log/asterisk/full):Mar 15 17:16:55 VERBOSE[4242] logger.c: == Parsing '/etc/asterisk/zapata.conf': Mar 15 17:16:55 VERBOSE[4242] logger.c: == Parsing '/etc/asterisk/zapata.conf': FoundMar 15 17:16:55 DEBUG[4242] chan_zap.c: Failed to read gains: Invalid argumentMar 15 17:16:55 DEBUG[4242] chan_zap.c: Failed to read gains: Invalid argumentMar 15 17:16:55 DEBUG[4242] chan_zap.c: Updated conferencing on 1, with 0 conference usersMar 15 17:16:55 VERBOSE[4242] logger.c: -- Registered channel 1, PRI Signalling signallingMar 15 17:16:55 DEBUG[4242] chan_zap.c: Failed to read gains: Invalid argumentMar 15 17:16:55 DEBUG[4242] chan_zap.c: Failed to read gains: Invalid argumentMar 15 17:16:55 DEBUG[4242] chan_zap.c: Updated conferencing on 2, with 0 conference usersMar 15 17: 16:55 VERBOSE[4242] logger.c: -- Registered channel 2, PRI Signalling signallingMar 15 17:16:55 DEBUG[4242] chan_zap.c: Failed to read gains: Invalid argumentMar 15 17:16:55 DEBUG[4242] chan_zap.c: Failed to read gains: Invalid argumentMar 15 17:16:55 DEBUG[4242] chan_zap.c: Updated conferencing on 3, with 0 conference usersMar 15 17:16:55 VERBOSE[4242] logger.c: -- Registered channel 3, PRI Signalling signallingMar 15 17:16:55 DEBUG[4242] chan_zap.c: Failed to read gains: Invalid argumentMar 15 17:16:55 DEBUG[4242] chan_zap.c: Failed to read gains: Invalid argumentMar 15 17:16:55 DEBUG[4242] chan_zap.c: Updated conferencing on 4, with 0 conference usersMar 15 17:16:55 VERBOSE[4242] logger.c: -- Registered channel 4, PRI Signalling signallingMar 15 17:16:55 DEBUG[4242] chan_zap.c: Failed to read gains: Invalid argumentMar 15 17:16:55 DEBUG[4242] chan_zap.c: Failed to rea d gains: Invalid argumentMar 15 17:16:55 DEBUG[4242] chan_zap.c: Updated conferencing on 5, with 0 conference usersMar 15 17:16:55 VERBOSE[4242] logger.c: -- Registered channel 5, PRI Signalling signallingMar 15 17:16:55 DEBUG[4242] chan_zap.c: Failed to read gains: Invalid argumentMar 15 17:16:55 DEBUG[4242] chan_zap.c: Failed to read gains: Invalid argumentMar 15 17:16:55 DEBUG[4242] chan_zap.c: Updated conferencing on 6, with 0 conference usersMar 15 17:16:55 VERBOSE[4242] logger.c: -- Registered channel 6, PRI Signalling signallingMar 15 17:16:55 DEBUG[4242] chan_zap.c: Failed to read gains: Invalid argumentMar 15 17:16:55 DEBUG[4242] chan_zap.c: Failed to read gains: Invalid argumentMar 15 17:16:55 DEBUG[4242] chan_zap.c: Updated conferencing on 7, with 0 conference usersMar 15 17:16:55 VERBOSE[4242] logger.c: -- Registered channel 7, PRI Signalling signallingMar 1 5 17:16:55 DEBUG[4242] chan_zap.c: Failed to read gains: Invalid argumentMar 15 17:16:55 DEBUG[4242] chan_zap.c: Failed to read gains: Invalid argumentMar 15 17:16:55 DEBUG[4242] chan_zap.c: Updated conferencing on 8, with 0 conference usersMar 15 17:16:55 VERBOSE[4242] logger.c: -- Registered channel 8, PRI Signalling signallingMar 15 17:16:55 DEBUG[4242] chan_zap.c: Failed to read gains: Invalid argumentMar 15 17:16:55 DEBUG[4242] chan_zap.c: Failed to read gains: Invalid argumentMar 15 17:16:55 DEBUG[4242] chan_zap.c: Updated conferencing on 9, with 0 conference usersMar 15 17:16:55 VERBOSE[4242] logger.c: -- Registered channel 9, PRI Signalling signallingMar 15 17:16:55 DEBUG[4242] chan_zap.c: Failed to read gains: Invalid argumentMar 15 17:16:55 DEBUG[4242] chan_zap.c: Failed to read gains: Invalid argumentMar 15 17:16:55 DEBUG[4242] chan_zap.c: Updated conferencing on 10, with 0 confe rence usersMar 15 17:16:55 VERBOSE[4242] logger.c: -- Registered channel 10, PRI Signalling signalling What does this log mean?My zapata.conf is [channels]language = encontext = incominggroup = 1signalling = pri_netswitchtype = dms100callerid = asreceived underrxwink = 300usecallerid = yeshidecallerid = nocallwaiting = yesusecallingpres = yesthreewaycalling = yestransfer = yescancallforward = yescallreturn = yesechocancel = yesechocancelwhenbridged = yesechotraining = 400rxgain = 0.0txgain = 0.0callgroup = 1pickupgroup = 1immediate = nochannel = 1-23 Thanks in advance.richard ___ --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] Failed to read gains: Invalid argument
Ooops...I thought I'd clean up my config files and check for typos. Turns out I was making my problems worse. grinAm surprised As terisk does not complain. Another thing happened to me also. I typed [channel] instead of [channels]. ztcfg looked ok. Asterisk did not complain when started. It just did not create the channels.Learning a lot. It's painful but fun.Thanks.richard"Kevin P. Fleming" [EMAIL PROTECTED] wrote: Richard OSS wrote: rxgain = 0.0 txgain = 0.0 callgroup = 1 pickupgroup = 1 immediate = no channel = 1-23Where did you find any example that suggested using '=' for everysetting was appropriate? It 's not. Use '=' for normal settings, and '='for assigning channel numbers.___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users___ --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] problem configuring a digium quad E1 card
If you are using connecting the card to a smart jack (incoming line from Telco), then you need a straight T1/E1 Cable, which is identical to a straight Ethernet cable. If you are doing a back-to-back configuration, or connecting the card to another PBX or channel bank, then you need a cross-T1/E1 cable with the following pin-outs: http://sangoma.editme.com/wanpipe-linux-asterisk-appendix#sampleZaptel http://www.ldfacts.com/faq_files/How-to-Make-a-T1-Crossover-cable.htmHope this helps.David Masure [EMAIL PROTECTED] wrote:Hi again,Can you specify the pin order for each end ?thanks -Message d'origine-De: JOSE MANUEL CORTES DAVID [mailto:[EMAIL PROTECTED]De la part de JOSE MANUEL CORTES DAVIDEnvoyé: mercredi 15 mars 2006 16:28À: Asterisk Users Mailing List - Non-Commercial DiscussionObjet: RE: [Asterisk-Users] problem configuring a digium quad E1 cardHi Youneed touse a cross-over E1 cable(not an ethernet cross-over one)Good luckJose Manuel Cortes David XSemestre Ingenieria Electronica PONTIFICIA UNIVERSIDAD JAVERIANA De: [EMAIL PROTECTED] en nombre de David MasureEnviado el: Mié 15/03/2006 9:41Para: asterisk-users@lists.digium.comAsunto: [Asterisk-Users] problem configuring a digium quad E1 card Hi,I bought a Digium Quad E1 card model TE406P. Till now, I can't make it work... p; I mean, I have red alarm when I configure one E1. The provider is in France (France Télécom) and I use the following zaptel config :span=1,1,0,ccs,hdb3,crc4bchan=1-15,17-31dchan=16I'm using Linux 2.6.15 and when I run ztcfg -, it seems that all channels are configured...So can someone give me an advice on that matter... maybe someone in France who already configured that type of access.Also, I would like that you confirm the type of cable which can be used to connect the card to the Telco : can I use a straight cable or use a crossed cable with pair 1-2 connect to 4-5 ?ThanksBest regards!David ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users___ --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] Mitel SX-2000 -- TE210P Red Alarm
Hello,I am trying to connect a TE210P to a SX-2000 but zttool shows RED ALARM. The SX-2000 provides the internal T1 to the * server.Telco - SX-2000 TE210PMy config file is below which I got from http://www.voip-info.org/wiki/view/NFAS /etc/zaptel.conf span=1,1,0,esf,b8zs span=2,2,0,esf,b8zs bchan=1-23,25-48 dchan=24 loadzone = us defaultzone = us /etc/asterisk/zapata.conf [trunkgroups]trunkgroup=1,24 spanmap = 1,1,0 spanmap = 2,1,2 [channels]language=en context=incoming group=1 signalling=pri_cpe switchtype=dms100callerid=asreceived rxwink=300 usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=ye s callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=400 rxgain=0.0 txgain=0.0 callgroup=1 pickupgroup=1 immediate=no channel = 1-23,25-48 ztcfg -vv looks fine. Could not see errors in the /var/log/messages and /etc/asterisk/messagesThe LEDs on the TE210P just blinks (red/orange).I read somewhere that RED ALARM means that there is no connection available (i.e. bad cable, etc.)My question is: Do I need a T1 crossover cable for my setup?Any tips on debugging this problem? Any suggestion on how to test if the SX-2000 is configured properly and that the problem is on the TE210P configuration?Thank you in advance. richard___ --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] Mitel SX-2000 and Asterisk integration
Hello,Somebody has managed to make Mitel SX-2000 and Asterisk integration work. http://www.voip-info.org/wiki-Asterisk+legacy+integrationCan you please post your zaptel.conf and zapata.conf for T1/PRI config?I will be configuring a TE210P to connect to an SX-2000 PBX.Thanks.richard ___ --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] Conference bridge dimensioning
Hi Jordan,We are planning on building the same thing. We are still waiting for the hardware. We are using a Dell PE 2850 3GHz with 2G of RAM and a TE210P.I asked Digium support if this can suupport 50 users in one conference and the tech support guy said yes. Here's also a response from this list http://lists.digium.com/pipermail/asterisk-users/2006-February/147956.htmlLet's share our experiences.Goodluck.richardJordan Novak [EMAIL PROTECTED] wrote:We are using an Asterisk box to do conferencing right now. I have had about sixteen active lines in conference and the quality was acceptable. We now have a need for 50 people to conference at one time. Does anyone have enough experience doing this to give me some pointers. Will it even be reasonable to try this? Is the mixing done on the the hardware, I plan on using a quad span t-1 card from Digium? The server is a fedora box with a dual core xeon at 2.0 Ghz and 2 gigs of Ram. Is there a rule of thumb to go by as far as conferencing resources?Jordan Novak Communications Technician Logistics Health Inc. 1319 Saint Andrews Street La Crosse WI 54603 1-800-666-2833 x299 (608) 783-7560 x299 ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] Voice conferencing server capacity
Hello,We are building a conference server using a Dell PE 2850 3GHz with 2G memory. This conference server will be used to hold a large conference with 30-50 simultaneous users in a conference room. This large conference will take place two days per week for 3 hours each day. When a large conference is going on, no other conference room will be created.The rest of the week, several small conferences will take place (5-10 simultaneous users) using 5-10 conference rooms.Is the Dell PE 2850 3GHz with 2G memory up to the task? If not, will adding another processor solve it?These entries from the dimensioning portion of the Wiki gives me hope that id does http://www.voip-info.org/wiki/view/Asterisk+dimensioning Capacity of MeetMe: With 28 persons on a Pentium II, 300MHz, 128 MB vmstat shows 70% idle. "I have anywhere from 15, to a peak max of 30 traders all using the same meetme conf during the day. My * is running on a old 4U 500Mhz machine (dual board, one processor installed now). With the exception of of a few problems from software sip phones, our implementation has been relatively problem free.I just want to get more opinions from others especially those who have built conferencing servers recently.Thank you very much.richard ___ --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] Voice conferencing server capacity
Thank you very much. Will go ahead and build the system. Hope everything goes smoothly.richardBJ Weschke [EMAIL PROTECTED] wrote: On 2/22/06, Richard OSS <[EMAIL PROTECTED]>wrote: Hello, We are building a conference server using a Dell PE 2850 3GHz with 2G memory. This conference server will be used to hold a large conference with 30-50 simultaneous users in a conference room. This large conference will take place two days per week for 3 hours each day. When a large conference is going on, no other conference room will be created. The rest of the week, several small conferences will take place (5-10 simultaneous users) using 5-10 conference rooms. Is the Dell PE 2850 3GHz with 2G memory u p to the task? If not, will adding another processor solve it? These entries from the dimensioning portion of the Wiki gives me hope that id does http://www.voip-info.org/wiki/view/Asterisk+dimensioning Capacity of MeetMe: With 28 persons on a Pentium II, 300MHz, 128 MB vmstat shows 70% idle. "I have anywhere from 15, to a peak max of 30 traders all using the same meetme conf during the day. My * is running on a old 4U 500Mhz machine (dual board, one processor installed now). With the exception of of a few problems from software sip phones, our implementation has been relatively problem free. I just want to get more opinions from others especially those who have built conferencing servers recently.Speaking from experience, I'd expect that this setup is more thanadequate. You may want to disable hyperthreading though on the procsif you can. There really isn't an advantage to having it enabled asthe "additional CPU" has no FPU.--Bird's The Word Technologies, Inc.http://www.btwtech.com/___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] Dell PowerEdge 2850
Thank you very much Darren.I did look at Dell's website for the info but was not able to find the PCI voltage info. Perhaps I looked at the wrong place or missed it. Googling also did not give me answers.I called Dell myself and the tech support person was very helpful. He confirmed that Dell PE 2850 indeed has 3.3V for PCI X.Digium's support also confrimed this. They suggested I exchange my TE205P card with the TE210P card which works with PCI-X 3.3V. I am a newbie at Asterisk and am learning a lot thanks to the responses of the members of this list.richardDarren Reilly [EMAIL PROTECTED] wrote: Dell website Useguide has the info and its:Expansion Bus Bus type PCI-X, PCI Express Expansion slots via riser card cage:PCI-Xone 3.3-V, 64-bit, 100-MHz or three 3.3-V, 64 bit, 133MHz PCI Expressone x4 lane width one x8 lane widthI would have thought the dell website would have been the first place to look.Took less than 2 mins to get the relevant info I cannot believe tech support couldn't give you that information.-Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Klaus DarilionSent: 20 February 2006 23:23To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Dell PowerEdge 2850Ryan Amos wrote: I use a PE2850 with CentOS 4.2 on it (as parent says, it is essentially RHEL 4 without the support contract.) Extremely stable; no problems with asterisk at all. Dell makes 2 PCI riser cards for this serve r, I believe one of them has 5v slots. I have a 3.3v card so I can't tell you on that.Der PE2850 bietet eine Auswahl aus zwei E/A-Riser-Karten:o E/A-PCI-Riser-Karte (3 PCI-X-Steckplätze: 3 x 64 Bit/133 Mhz) odero E/A-PCI-Riser-Karte (2 PCI Express-Steckplätze: 1 x8-Lane und 1 x4-Lane, beide mit x8-Anschlüssen, und 1 PCI-X-Steckplatz: 1x 64 Bit/100 MHz)Both riser cards only have 64 Bit PCI slots. I think 64 bit is always 3.3 Volt - isn't it?regardsklaus -Ryan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dovid Bender Sent: Monday, February 20, 2006 3:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Dell PowerEdge 2850 Don't know about the Dell. I personaly use Cent OS (www.centos.org) which is RHEL ES without paying for it. I have it on my server and it seems to be holding up just fine. --- Richard OSS <[EMAIL PROTECTED]>wrote: Hello, Digium uses the Dell PE 2850 for their testing. This site says that 3.3V PCI slot. http://www.voip-info.org/wiki/view/Asterisk+hardware We are planning on purchasing a Dell PE 2850 and putting a TE205P card on it. However, the needs a 5V PCI slot. Does Dell PE 2850 has a 5V PCI slot? A person in our group tried to call Dell's customer support but they do not seem to know. We will also be using RHEL ES 4 as the OS. Anybody have experience (good/bad) for this type of configuration? We are going to use it primarily as a conferencing server serving 30-50 simultaneous users. Can anybody recommend an alternative server that works well with TE205P and RHEL ES 4? This is our first time using Asterisk so we would like to have it pain free as much as possible. Thank you very much. richard ___ --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] co-location providers in Ottawa, Canada
Thank you very much.I will contact Sprint, Magma, and Unlimitel about their service.For those who want to participate in Ottawa Asterisk Users Group, please send me email off-list at oss_richard at rogers dot comso I can update you and share ideas on activities.I can ask Carleton University to use one of their facilities on weekends (free parking) for group meetings.richardVirTERM [EMAIL PROTECTED] wrote: You can use Sprint (Group Telecom) and/or Magma. Keep us posted about the group meetings.. Thanks,Wojtek- Original Message - From: Richard OSS To: asterisk-users@lists.digium.com Sent: Sunday, February 19, 2006 12:03 AM Subject: [Asterisk-Users] co-location providers in Ottawa, CanadaAnybody know ifthere are co-location providers in Ottawa, Canada? We are planning on co-locating our Asterisk conferencing server.< DIV>One more thing, is there an interest in reviving the Ottawa Asterisk User Group? Seems like the original group has been inactive for quite awhile. I will volunteer to organize it.Thanks.richard___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] Dell PowerEdge 2850
Hello,Digium uses the Dell PE 2850 for their testing. This site says that 3.3V PCI slot. http://www.voip-info.org/wiki/view/Asterisk+hardwareWe are planning on purchasing a Dell PE 2850 and putting a TE205P card on it. However, the needs a 5V PCI slot. Does Dell PE 2850 has a 5V PCI slot? A person in our group tried to call Dell's customer support but they do not seem to know.We will also be using RHEL ES 4 as the OS.Anybody have experience (good/bad) for this type of configuration? We are going to use it primarily as a conferencing server serving 30-50 simultaneous users.Can anybody recommend an alternative server that works well with TE205P and RHEL ES 4?This is our fi rst time using Asterisk so we would like to have it pain free as much as possible.Thank you very much.richard___ --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] co-location providers in Ottawa, Canada
Anybody know ifthere are co-location providers in Ottawa, Canada? We are planning on co-locating our Asterisk conferencing server.One more thing, is there an interest in reviving the Ottawa Asterisk User Group? Seems like the original group has been inactive for quite awhile. I will volunteer to organize it.Thanks.richard___ --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] Ottawa Asterisk Users Group
Hello,I am new to the world of Asterisk but I am excited.Do you still meet first Saturday afternoon each month, usually at the Royal Oak on Wellington at Holland?richard___ --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
