Re: [asterisk-users] Please advise: Asterisk on Dell PowerEdge 1750 w/ hyperthreading
Redhat Enterprise Zeeshan Zakaria wrote: I've to setup Asterisk on a Dell PowerEdge 1750 server. Its dual Xeon 3GHz with Hyperthreading. People on this list who have experience with this server please advise me how is the performance of Asterisk on this server, what flavour of linux is good on it etc. Is Hyperthreading going to be a problem or not. I once read somewhere that hyperthreading caused some voice quality problems in Asterisk. Is it fixed in or not yet? Any other suggestions will also be helpful. Thanks -- Zeeshan A Zakaria ___ --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
[asterisk-users] ZAPTEL problem
Hi I have 100XP Digium clone card Installed in my pc and compiled zaptel and asterisk again after installing the card but after i rebooted i can load zaptel and wcfxo modprobe with out any problem but when i intiated ztcfg - i get the following error Zaptel Configuration == 1 channels configured. ZT_CHANCONFIG failed on channel 1: No such device or address (6) - dmesg errors Zapata Telephony Interface Unloaded Zapata Telephony Interface Registered on major 196 Zaptel Version: SVN-branch-1.2-r1725 Echo Canceller: KB1 Failed to initailize DAA, giving up... wcfxo: probe of :00:10.0 failed with error -5 -- lspci 00:0c.1 SCSI storage controller: Adaptec AIC-7896U2/7897U2 00:10.0 Communication controller: Motorola Unknown device 5608 any suggestions ram ___ --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] Progress passing problem.
Hi, i have Asterisk 1.2.7.1 and outgoing trunk connected via SIP (this is Cisco AS5350)and user is connected via sip too. When user calling out (via AS5350) he receives progress tone generated by voip-phone not that passing from telco line. I turned on debug and see that the AS send: 183 Session Progreess but to user is sent Ringing, not progress. I have progressinband=never in sip.conf so shouold be transferred. Where can be a problem? Regards, Adam Rybak ___ --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 Time Card
On 5/26/07, Nitesh Divecha [EMAIL PROTECTED] wrote: Thanks Shanon and everyones input... Finally, got the application working as planned with PHPAGI... Now the only draw back is the voice... I am using text2wav to prompt all the questions, but the voice is creepy... Is their any easier way to replace the text2wav voice with proper recorded female voice? Please advice... what codec are you using ram ___ --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] Meet me
I am using asterisk 1.4.4 now and facing a problem with meetme,the code I was using with asterisk 1.2 is not functioning with 1.4 ,my code is conf = 222| at meetme.conf at meet_me_additional like this exten = 21,1,MeetMe(21,dq) exten = 21,2,Playback(beep) or this exten = 222,1,GotoIfTime(*|mon-sun|08-08|may-may?223,1) exten = 222,n,Playback(vm-goodbye) exten = 222,n,Hangup exten = STARTMEETME,1,MeetMe(${MEETME_ROOMNUM},${MEETME_OPTS},${PIN}) exten = STARTMEETME,n,Hangup exten = h,1,Hangup exten = 223,1,Set(MEETME_ROOMNUM=222) exten = 223,n,GotoIf($[${DIALSTATUS} = ANSWER]?READPIN) exten = 223,n,Answer exten = 223,n,Wait(1) exten = 223,n(READPIN),Read(PIN,enter-conf-pin-number,,) exten = 223,n,GotoIf($[foo${PIN} = foo]?USER) exten = 223,n,GotoIf($[${PIN} = ]?ADMIN) exten = 223,n,Playback(conf-invalidpin) exten = 223,n,Goto(READPIN) exten = 223,n(ADMIN),Set(MEETME_OPTS=aAwciMs) exten = 223,n,Goto(STARTMEETME,1) exten = 223,n(USER),Set(MEETME_OPTS=ciMs) exten = 223,n,Goto(STARTMEETME,1) please guide me * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --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] Zonbu
Hi, On Sun, 2007-05-27 at 17:35 -0400, Nabeel Jafferali wrote: Looks like a rebadged Patton 6075 to me: http://www.patton.com/products/pe_products.asp?category=337 also patton rebrands that unit. At Cebit there was plenty of these boxes from .tw manufacturers. Matteo -- Matteo Brancaleoni RD Director Tel :+39.02.70633354 Voip :sip:[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
[asterisk-users] Queues with announce
Hello *, do queues allow me to set an announce like the A() option of the Dial() cmd? The announce that I've found is a message that is heard by the caller. I'd like to send a message to the member of the queue that picks up the call. Thanks in advance, -- Dott. Andrea Spadaccini Multimedia Technologies Institute - MTI S.r.l. ___ --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] include=context in REALTIME!!!!!!
hi list, im setting up a realtime , and while entering datas in my extensions_table one question came in my mind how should insert the different includes ligns (includes=context)in my database knowing that im not going to use the extensions.conf file anymore should i add thoses lines in the pbx_config table ??? any clue??? thanks for sharing BR -- Cheikhou DIAW intern iTG Software Engineering India Pvt.Ltd 331-332, Nemi Sagar Colony, Queens Road, Vaishali Nagar, Jaipur, 302021 Rajasthan, India Tel: +91-141-2357582 Tel: +91-141-2357583 Mobile:+91 9828825224 msn: [EMAIL PROTECTED] e-mail:[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
[asterisk-users] Limit outgoing call for sip peer
Hi All, I need to limit outgoing calls in my sip peers... I tried to use call-limit=1 in these peers in the sip.conf, but it didn't work... Here is my peer configuration in the sip.conf: [sip.broadvoice.com] accountcode=broadvoice type=peer dynamic=yes username=MYUSERNAME fromuser=MYUSERNAME authname=MYUSERNAME user=MYUSERNAME secret= host=sip.broadvoice.com fromdomain=sip.broadvoice.com outboundproxy=proxy.nyc.broadvoice.com insecure=port disallow=all allow=gsm allow=g729 dtmfmode=rfc2833 dtmf=inband canreinvite=no context=from-broadvoice ntp_ip=ntp.broadvoice.com call-limt=1 Here is my configuration in the extensions.conf: exten = _9XX[2-5].,1,dial(SIP/01155${EXTEN:[EMAIL PROTECTED],40) exten = _9XX[2-5].,2,congestion() exten = _9XX[2-5].,102,busy() How can I limit calls in my sip peers?? Everton Goularth Uberlandia - MG - Brazil ___ Yahoo! Mail - Sempre a melhor opção para você! Experimente já e veja as novidades. http://br.yahoo.com/mailbeta/tudonovo/ ___ --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 Time Card
What does codec has to do...? I am using G729a Cheers, Nitesh ram wrote: On 5/26/07, *Nitesh Divecha* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Thanks Shanon and everyones input... Finally, got the application working as planned with PHPAGI... Now the only draw back is the voice... I am using text2wav to prompt all the questions, but the voice is creepy... Is their any easier way to replace the text2wav voice with proper recorded female voice? Please advice... what codec are you using ram ___ --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] vmoutcall
On May 24, 2007, at 3:28 PM, Doug Lytle wrote: Paul Aviles wrote: Hello guys, I have been looking for a way to call a cell phone after someone has left a This can easily be done with database lookups and .call files to accomplishing this? Most analog pbx's have this feature and I am amazed Asterisk does not natively. It can be done natively; within the dial plan. Doug An aside, this can also be done using the externnotify option in the voicemail.conf file. That option allows you to specify an external script that will run when VoiceMailMain() exits. Watch out, because (as I just found out yesterday, the hard way), this script is run both when a voicemail message is left and when a user logs out of voicemail after checking their messages. From there, create a script in your favorite language (a simple shell script ought to work for your purposes) and have it create a call file to call out to the user and drop them into a context that plays an announcement (You have new voicemail) and then asks them to confirm with a keypress (Dial 2 for voicemail). At that point, set the IVR up so dialing 2 drops them into Voicemail() and you are good to go. Tom ___ --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] help on asterisk sipp
Good morningI was wondering whether you could help me. I installed sipp on my Asterisk server but I don't really understand how does it fonction! Has someone ever tried it?If you can explain to me the principle, I would be extremely grateful.Thank you very much in advance. _ Lancez des recherches en toute sécurité depuis n'importe quelle page Web. Téléchargez GRATUITEMENT Windows Live Toolbar aujourd'hui ! http://toolbar.live.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] help on asterisk sipp
Hello! If ou mean SIPp, the testing tool for the SIP protocol or kind of a call generator for Asterisk PBX, have a look at http://sipp.sourceforge.net/doc/reference.html cheers - Original Message - From: khawla khawla To: asterisk-users@lists.digium.com Sent: Monday, May 28, 2007 3:09 PM Subject: [asterisk-users] help on asterisk sipp Good morning I was wondering whether you could help me. I installed sipp on my Asterisk server but I don't really understand how does it fonction! Has someone ever tried it? If you can explain to me the principle, I would be extremely grateful. Thank you very much in advance. -- Lancez des recherches en toute sécurité depuis n'importe quelle page Web. Téléchargez GRATUITEMENT Windows Live Toolbar aujourd'hui ! Essayez-le maintenant ! -- ___ --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] CDR not recording accountcode on SIP Response 302 Call Forward From Phone
On Fri, 2007-05-25 at 17:17 -0500, JR Richardson wrote: Hi All, Call comes into Asterisk Asterisk answers and Dials SIP Phone SIP phone has call forward enabled to a long distance number Asterisk receives a SIP response 302 Moved Temporarily back from phone Asterisk then forwards inbound call to 'Local/[EMAIL PROTECTED]' thanks to phone 2 problems with the CDR: 1. intermittent 'bill sec' accuracy, sometimes 0 even when the call was answered and many minutes usage on call. 2. no accountcode is recorded. So the implication here is that if a phone user forwards their phone, I have no way of tracking the usage or what account the call should be billed to. I have a feeling this is normal behavior for Asterisk as no real channel gets invoked with an accountcode parameter, but there has got to be something that accounts for this situation. Does anyone have a work around or remedy? I'm running 1.2.9. Thanks. JR JR-- Good news and bad news. To be truthful, I personally have doubts that 1.2 will ever be fixed as far as transfers are concerned. The changes that will be necessary to correct the situation will result in non-backwards compatible changes to the behavior of CDR's. It'll simply muck up everyone who has built CDR systems to date. That said, I'm beginning a final solution to the problems in 1.4 and trunk; I see what needs to be done, and how to do it. I think I have a grip on what to do with Local channels, and masqueraded channels. But the devil's in the details, and I'll find out how good my guesses are when I start testing. I'll try to make sure that channel goodies don't get smudged off the CDR's, like acctcode, etc. So, hang on, and I'll post notices as to my progress. murf -- Steve Murphy [EMAIL PROTECTED] Digium ___ --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] Advanced Billing System for Asterisk - MOR v0.4 released
Hello, We are proudly to present new version of our billing system MOR v0.4 What's new in MOR FREE v0.4 * Extended stability and reliability * Extended configuration options for clients and providers * User blocking * Prepaid support * Increased security * New tariff/rating engine * Registration * PayPal integration * Authorization by IP * Extended calls view What's new in MOR PRO v0.4 * Extended Calling Card engine * CDR import from CSV file * Device grouping * Country Stats * Providers' Stats * Auto-Dialer * Callback * Click2Call * Invoice generation * Custom Rates * Localization More info in: http://www.kolmisoft.com/mor Regards/Pagarbiai, Mindaugas Kezys http://www.kolmisoft.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] DTMFToText Installation process
Hi, I was looking for a way to pass alphanumeric variables to asterisk via the keypad, found this application app_dtmftotext.c , And I already tried with 'spandsp' application for this. But I am getting errors.[I followed the instructions at http://www.soft-switch.org/installing-spandsp.html]. Can anybody be of help Me on this getting DTMFToText() application on asterisk with the help of app_dtmftotext.c and/or spandsp application is appreciated. Regards K.Rajesh. _ Voice your questions and our experts will answer them http://content.msn.co.in/Lifestyle/AskExpert/Default01.htm ___ --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] Cisco remote reboot
I've fired a script from an AGI-BIN to accomplish that. Try this one: #!/usr/bin/perl # mk 2004 feel free to distribute # [EMAIL PROTECTED], _Vile # perl script to reboot phones # try telnetting to your phone, first. # use Net::Telnet (); $phone_ip = shift; # Your Cisco 79xx prompt $prompt = Enter Your Prompt Here; # Your Password $password = xx; # Reset Command $command = reset; if ($phone_ip eq all) { reboot(xxx.xx.x.xx,$password,$command,$prompt); reboot(xxx.xx.x.xx,$password,$command,$prompt); reboot(xxx.xx.x.xx,$password,$command,$prompt); reboot(xxx.xx.x.xx,$password,$command,$prompt); } elsif ($phone_ip eq ) { print Enter an IP or 'all' for All.; } else { reboot($phone_ip,$password,$command,$prompt); } exit; sub reboot{ my ($ip,$password,$command,$prompt) = @_; $t = new Net::Telnet; $t-open($ip); $t-waitfor('/Password :.*$/'); $t-print($password); $t-waitfor('/'.$prompt.'.*$/'); $t-print($command); } ___ --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] Zonbu
How much does a Patton NanoServ 607x cost? Their page has no price, an inactive Ordering tab, Google doesn't have (nanoserv 6070 price) in its index (except a couple unresponsive del.ic.ious pages). PingTel announce a SIPxNano based on it, for under $1000 in 2006Q3: http://www.pingtel.com/page.php?id=70view=117 . Is there pricing for just the HW without whatever bundled SW or service these telcos are bundling/subsidizing it with? On Sun, 2007-05-27 at 19:51 -0700, [EMAIL PROTECTED] wrote: Date: Sun, 27 May 2007 23:18:26 -0300 From: Gustavo Cordeiro [EMAIL PROTECTED] Subject: RE: [asterisk-users] Zonbu To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1; format=flowed $99,00 for one box, but you need a subscription plan... Zonbu is $99 with a two-year subscription plan. With month to month plan, Zonbu is $249. Sds, Gustavo From: Nabeel Jafferali [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED],Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: 'Asterisk Users Mailing List - Non-Commercial Discussion'asterisk-users@lists.digium.com Subject: RE: [asterisk-users] Zonbu Date: Sun, 27 May 2007 17:35:20 -0400 Looks like a rebadged Patton 6075 to me: http://www.patton.com/products/pe_products.asp?category=337 Nabeel -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Dean Collins Sent: May 27, 2007 11:53 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Zonbu I just came across www.Zonbu.com http://www.zonbu.com/ it's a fanless box about the size of a paperback book. It has no hard drive but runs it's Linux OS on a flash card - relying on document storage from an online service (rebadged Amazon S3). http://deancollinsblog.blogspot.com/2007/05/zonbu-net-pc.html I wonder who's going to be the first to hack an asterisk server onto this thing? At $99 it's a hell of an option for a fanless Asterisk server. Regards, Dean Collins [EMAIL PROTECTED] +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). Call Button http://click.mexuar.com/webuser/click/7/userurl/Cognation http://click.mexuar.com/webuser/nojs/7/userurl/Cognation -- (C) Matthew Rubenstein ___ --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] Urgent: DTMF does not work with rtpmap:101 telephone-event/8000
25 maj 2007 kl. 06.40 skrev JK: Hello asterisk-users list. I have been scratching my head for almost a week. We are trying to set a service with a company (ip=XXX.XXX.XXX.XXX) and dtmf is not working. In our scenario the SP is sending call to our ser server and ser is forwarding the call to asterisk. In the asterisk debug I can see the DTMF keys are coming but ivr does not recognice those keys at all. I can see this in the debug. We are using ulaw and alaw for codec. May 24 20:14:00 DEBUG[26803] rtp.c: Sending dtmf: 49 (1), at XXX.XXX.XXX.XXX May 24 20:14:01 DEBUG[26803] rtp.c: Sending dtmf: 49 (1), at XXX.XXX.XXX.XXX May 24 20:14:01 DEBUG[26803] rtp.c: Sending dtmf: 49 (1), at XXX.XXX.XXX.XXX Voice part works great. I mean if I forward that call to asterisk sip user we can talk. Every thing is working great with other SP. The only difference I can see is the rtpmap:101 telephone-event/8000. With the working SP the rtpmap is rtpmap:100 telephone-event/8000. Your debug did not have any SIP messages. I need to see the INVITE and the 200 OK. Thanks. /O ___ --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] Blindside Web Conferencing
Hello, We are creating a web-based conferencing application using Asterisk as the voice conferencing server. This as an open source project. We are trying to determine if there is interest of the community and perhaps work together to improve the application. Using the web application, you can upload your powerpoint presentation, manage the participants in the conference thru the web interface (mute/unmute/kick), chat, participants can raise/lower hand to indicate she wants to talk when muted. Please visit http://present.blindsideproject.org ( http://present.sce.carleton.ca if that doesn't work) to try it out. Click on the requirements link for instructions on how to setup your Idefisk. Project website is at http://www.blindsideproject.org ( http://www.blindsideproject.org:8080 if that doesn't work). We are requesting the University permissions to configure DNS/firewall, so please bear with us. Please send feedback/suggestions to this mailing list. We're also trying to setup our own. Hope the moderators of this list don't mind. Hoping for your feedbacks. Thanks. Blindside Project Team ___ --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] Blindside Web Conferencing
I think this is a great potential application for Asterisk - I couldn't actually determine if/where you had a downloadable POC or if it was still just in development conceptualization at the moment. Either way keep up the good work and put a paypal tip jar up once you have something people can actually use. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). http://click.mexuar.com/webuser/click/7/userurl/Cognation http://click.mexuar.com/webuser/nojs/7/userurl/Cognation From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Richard Alam Sent: Monday, 28 May 2007 10:18 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Blindside Web Conferencing Hello, We are creating a web-based conferencing application using Asterisk as the voice conferencing server. This as an open source project. We are trying to determine if there is interest of the community and perhaps work together to improve the application. Using the web application, you can upload your powerpoint presentation, manage the participants in the conference thru the web interface (mute/unmute/kick), chat, participants can raise/lower hand to indicate she wants to talk when muted. Please visit http://present.blindsideproject.org ( http://present.sce.carleton.ca if that doesn't work) to try it out. Click on the requirements link for instructions on how to setup your Idefisk. Project website is at http://www.blindsideproject.org ( http://www.blindsideproject.org:8080 if that doesn't work). We are requesting the University permissions to configure DNS/firewall, so please bear with us. Please send feedback/suggestions to this mailing list. We're also trying to setup our own. Hope the moderators of this list don't mind. Hoping for your feedbacks. Thanks. Blindside Project Team image001.gif___ --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] Blindside Web Conferencing
Yes, we have some downloadable code. We are in the process of completing the instructions (build/deploy/etc.). Code is located here. http://www.blindsideproject.org:8080/cgi-bin/trac.cgi/browser/conference/trunk Partial docs is located here. http://www.blindsideproject.org:8080/cgi-bin/trac.cgi/wiki/BlindsideDevelopment Richard On 5/28/07, Dean Collins [EMAIL PROTECTED] wrote: I think this is a great potential application for Asterisk – I couldn't actually determine if/where you had a downloadable POC or if it was still just in development conceptualization at the moment. Either way keep up the good work and put a paypal tip jar up once you have something people can actually use. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). [image: Call Button]http://click.mexuar.com/webuser/click/7/userurl/Cognationhttp://click.mexuar.com/webuser/nojs/7/userurl/Cognation -- *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *Richard Alam *Sent:* Monday, 28 May 2007 10:18 AM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] Blindside Web Conferencing Hello, We are creating a web-based conferencing application using Asterisk as the voice conferencing server. This as an open source project. We are trying to determine if there is interest of the community and perhaps work together to improve the application. Using the web application, you can upload your powerpoint presentation, manage the participants in the conference thru the web interface (mute/unmute/kick), chat, participants can raise/lower hand to indicate she wants to talk when muted. Please visit http://present.blindsideproject.org ( http://present.sce.carleton.ca if that doesn't work) to try it out. Click on the requirements link for instructions on how to setup your Idefisk. Project website is at http://www.blindsideproject.org ( http://www.blindsideproject.org:8080 if that doesn't work). We are requesting the University permissions to configure DNS/firewall, so please bear with us. Please send feedback/suggestions to this mailing list. We're also trying to setup our own. Hope the moderators of this list don't mind. Hoping for your feedbacks. Thanks. Blindside Project Team ___ --Bandwidth and Colocation provided by Easynews.com http://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] Bottom line on fax reception
Anybody?? -Original Message- From: shadowym [mailto:[EMAIL PROTECTED] Sent: Thursday, May 24, 2007 9:35 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Bottom line on fax reception So what is the bottom line? Does it work or not. I've heard stories it works, it doesn't work, it kinda sorta works when it's not raining out side. Everything under the rainbow. What's the bottom line with recent updates on 1.2.x? Is it production ready for fax? By production ready I mean that it just works all the time and doesn't need any babysitting. Do I have to worry about dropped lines, sometimes not detecting incoming fax toneyada yada. I know I don't have to use fax on Asterisk but I really want to for various reasons. Mostly incoming but outgoing is a nice to have. Should I use an addon package and if so which one? Any help would be appreciated. ___ --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] Blindside Web Conferencing
At 17.09 28/05/2007, Richard Alam wrote: Yes, we have some downloadable code. We are in the process of completing the instructions (build/deploy/etc.). Code is located here. http://www.blindsideproject.org:8080/cgi-bin/trac.cgi/browser/conference/trunkhttp://www.blindsideproject.org:8080/cgi-bin/trac.cgi/browser/conference/trunk How can I get it? Partial docs is located here. http://www.blindsideproject.org:8080/cgi-bin/trac.cgi/wiki/BlindsideDevelopmenthttp://www.blindsideproject.org:8080/cgi-bin/trac.cgi/wiki/BlindsideDevelopment Richard On 5/28/07, Dean Collins mailto:[EMAIL PROTECTED][EMAIL PROTECTED] wrote: I think this is a great potential application for Asterisk I couldn't actually determine if/where you had a downloadable POC or if it was still just in development conceptualization at the moment. Either way keep up the good work and put a paypal tip jar up once you have something people can actually use. Regards, Dean Collins Cognation Pty Ltd mailto:[EMAIL PROTECTED][EMAIL PROTECTED] +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). -- From: mailto:[EMAIL PROTECTED][EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] ] On Behalf Of Richard Alam Sent: Monday, 28 May 2007 10:18 AM To: mailto:asterisk-users@lists.digium.comasterisk-users@lists.digium.com Subject: [asterisk-users] Blindside Web Conferencing Hello, We are creating a web-based conferencing application using Asterisk as the voice conferencing server. This as an open source project. We are trying to determine if there is interest of the community and perhaps work together to improve the application. Using the web application, you can upload your powerpoint presentation, manage the participants in the conference thru the web interface (mute/unmute/kick), chat, participants can raise/lower hand to indicate she wants to talk when muted. Please visit http://present.blindsideproject.org/http://present.blindsideproject.org (http://present.sce.carleton.ca if that doesn't work) to try it out. Click on the requirements link for instructions on how to setup your Idefisk. Project website is at http://www.blindsideproject.org/http://www.blindsideproject.org (http://www.blindsideproject.org:8080 if that doesn't work). We are requesting the University permissions to configure DNS/firewall, so please bear with us. Please send feedback/suggestions to this mailing list. We're also trying to setup our own. Hope the moderators of this list don't mind. Hoping for your feedbacks. Thanks. Blindside Project Team ___ --Bandwidth and Colocation provided by http://easynews.com/Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-usershttp://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 Roberto Fichera. ___ --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] Blindside Web Conferencing
Sounds cool. You could probably use some code from the various open source jabber clients that allow for shared whiteboard and pushing URLs too. Thanks, Steve Totaro http://www.asteriskhelpdesk.com KB3OPB _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Richard Alam Sent: Monday, May 28, 2007 10:18 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Blindside Web Conferencing Hello, We are creating a web-based conferencing application using Asterisk as the voice conferencing server. This as an open source project. We are trying to determine if there is interest of the community and perhaps work together to improve the application. Using the web application, you can upload your powerpoint presentation, manage the participants in the conference thru the web interface (mute/unmute/kick), chat, participants can raise/lower hand to indicate she wants to talk when muted. Please visit http://present.blindsideproject.org (http://present.sce.carleton.ca if that doesn't work) to try it out. Click on the requirements link for instructions on how to setup your Idefisk. Project website is at http://www.blindsideproject.org (http://www.blindsideproject.org:8080 if that doesn't work). We are requesting the University permissions to configure DNS/firewall, so please bear with us. Please send feedback/suggestions to this mailing list. We're also trying to setup our own. Hope the moderators of this list don't mind. Hoping for your feedbacks. Thanks. Blindside Project Team ___ --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] Bottom line on fax reception
Someone already answered this question. The answer is no, it does not work by your definition of production ready. Thanks, Steve Totaro http://www.asteriskhelpdesk.com KB3OPB -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of shadowym Sent: Monday, May 28, 2007 11:20 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Bottom line on fax reception Anybody?? -Original Message- From: shadowym [mailto:[EMAIL PROTECTED] Sent: Thursday, May 24, 2007 9:35 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Bottom line on fax reception So what is the bottom line? Does it work or not. I've heard stories it works, it doesn't work, it kinda sorta works when it's not raining out side. Everything under the rainbow. What's the bottom line with recent updates on 1.2.x? Is it production ready for fax? By production ready I mean that it just works all the time and doesn't need any babysitting. Do I have to worry about dropped lines, sometimes not detecting incoming fax toneyada yada. I know I don't have to use fax on Asterisk but I really want to for various reasons. Mostly incoming but outgoing is a nice to have. Should I use an addon package and if so which one? Any help would be appreciated. ___ --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] Meet me
change conf = 222 to conf = 222 ( remove | ) I had same problem as freepbx always put | removing it fixed the problem On 29/05/07, Khaled Chehab [EMAIL PROTECTED] wrote: I am using asterisk 1.4.4 now and facing a problem with meetme,the code I was using with asterisk 1.2 is not functioning with 1.4 ,my code is conf = 222| at meetme.conf at meet_me_additional like this exten = 21,1,MeetMe(21,dq) exten = 21,2,Playback(beep) or this exten = 222,1,GotoIfTime(*|mon-sun|08-08|may-may?223,1) exten = 222,n,Playback(vm-goodbye) exten = 222,n,Hangup exten = STARTMEETME,1,MeetMe(${MEETME_ROOMNUM},${MEETME_OPTS},${PIN}) exten = STARTMEETME,n,Hangup exten = h,1,Hangup exten = 223,1,Set(MEETME_ROOMNUM=222) exten = 223,n,GotoIf($[${DIALSTATUS} = ANSWER]?READPIN) exten = 223,n,Answer exten = 223,n,Wait(1) exten = 223,n(READPIN),Read(PIN,enter-conf-pin-number,,) exten = 223,n,GotoIf($[foo${PIN} = foo]?USER) exten = 223,n,GotoIf($[${PIN} = ]?ADMIN) exten = 223,n,Playback(conf-invalidpin) exten = 223,n,Goto(READPIN) exten = 223,n(ADMIN),Set(MEETME_OPTS=aAwciMs) exten = 223,n,Goto(STARTMEETME,1) exten = 223,n(USER),Set(MEETME_OPTS=ciMs) exten = 223,n,Goto(STARTMEETME,1) please guide me * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --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
[asterisk-users] Astmanproxy
Hello all, Some of you are using astmanproxy with asttapi or activa TSP? How does you make to work? Thanks VoipCrazy ___ --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] Multiple TDM400p cards in one machine -- no longer an issue?
Hi all, Years ago, I was pretty sure attempting to use two TDM400p cards in one machine was recommended against by Digium ... probably because the cards couldn't hack it, and/or interrupt problems etc I have seen some posts recently that seem to indicate it is in fact possible these days thanks to some updated firmware perhaps? . I just need to have two in the server because the 4 ports aren't enough ... I'd rather just expand by one card rather than get a TDM2400 (or TDM800??) Anyone had recent success/failure with this sort of thing? -- Chris Earle ___ --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] Blindside Web Conferencing
On 5/28/07, Roberto Fichera [EMAIL PROTECTED] wrote: At 17.09 28/05/2007, Richard Alam wrote: Yes, we have some downloadable code. We are in the process of completing the instructions (build/deploy/etc.). Code is located here. http://www.blindsideproject.org:8080/cgi-bin/trac.cgi/browser/conference/trunk http://www.blindsideproject.org:8080/cgi-bin/trac.cgi/browser/conference/trunk How can I get it? Hi, Here is our first attempt at writing the doc. http://www.blindsideproject.org:8080/cgi-bin/trac.cgi/wiki/BlindsideWebApp Alternatively, you can c:\source svn co http://www.blindsideproject.org:8080/svn/repos/blindside/conference/trunk/blindside-core c:\source svn co http://www.blindsideproject.org:8080/svn/repos/blindside/conference/trunk/blindside-main c:\source svn co http://www.blindsideproject.org:8080/svn/repos/blindside/conference/trunk/blindside-web Then c:\sourcecd blindside-main c:\source\blindside-main mvn install (you must have maven installed/configured) c:\source\blindside-main cd ../blindside-web c:\source\blindside-web mvn jetty:run From your browser, visit http://localhost:8080/conference Please let me know if you run into problems. 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] Blindside Web Conferencing
Hi Steve, Yes, we are looking for that. Do you know of any projects that provides those? I know one written in TCL/TK. Thanks. Richard On 5/28/07, Steve Totaro [EMAIL PROTECTED] wrote: Sounds cool. You could probably use some code from the various open source jabber clients that allow for shared whiteboard and pushing URLs too. Thanks, Steve Totaro http://www.asteriskhelpdesk.com KB3OPB -- *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *Richard Alam *Sent:* Monday, May 28, 2007 10:18 AM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] Blindside Web Conferencing Hello, We are creating a web-based conferencing application using Asterisk as the voice conferencing server. This as an open source project. We are trying to determine if there is interest of the community and perhaps work together to improve the application. Using the web application, you can upload your powerpoint presentation, manage the participants in the conference thru the web interface (mute/unmute/kick), chat, participants can raise/lower hand to indicate she wants to talk when muted. Please visit http://present.blindsideproject.org ( http://present.sce.carleton.ca if that doesn't work) to try it out. Click on the requirements link for instructions on how to setup your Idefisk. Project website is at http://www.blindsideproject.org ( http://www.blindsideproject.org:8080 if that doesn't work). We are requesting the University permissions to configure DNS/firewall, so please bear with us. Please send feedback/suggestions to this mailing list. We're also trying to setup our own. Hope the moderators of this list don't mind. Hoping for your feedbacks. Thanks. Blindside Project Team ___ --Bandwidth and Colocation provided by Easynews.com http://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
[asterisk-users] Octasic echo cancellation
Hi, I'm currently testing SoftEcho, an echo cancellation software for Asterisk from Octasic. I noticed an important increase of the quality of my coms, but I still have a few echo problems. There is an ERL parameter which corresponds to an initial ERL value probably to optimize the echo training or something in that kind. Is there a way to monitor, using one of the zttools, the instant ERL value, to be able to set this parameter correctly? In that purpose, I would like to know where do the echo cancellation take place in the communication chain, is it after or before the amplification by the Rx/Tx gain parameters? Will this gain apply on the echo cancelled signal or on the gross signal? Anybody with experience on this product could give me some advice in the aim of removing all trace of echo? Thanks by advance Sebastien ___ --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] Blindside Web Conferencing
At 19.19 28/05/2007, you wrote: On 5/28/07, Roberto Fichera mailto:[EMAIL PROTECTED][EMAIL PROTECTED] wrote: At 17.09 28/05/2007, Richard Alam wrote: Yes, we have some downloadable code. We are in the process of completing the instructions (build/deploy/etc.). Code is located here. http://www.blindsideproject.org:8080/cgi-bin/trac.cgi/browser/conference/trunkhttp://www.blindsideproject.org:8080/cgi-bin/trac.cgi/browser/conference/trunk http://www.blindsideproject.org:8080/cgi-bin/trac.cgi/browser/conference/trunk How can I get it? Hi, Here is our first attempt at writing the doc. http://www.blindsideproject.org:8080/cgi-bin/trac.cgi/wiki/BlindsideWebApphttp://www.blindsideproject.org:8080/cgi-bin/trac.cgi/wiki/BlindsideWebApp Alternatively, you can c:\source svn co http://www.blindsideproject.org:8080/svn/repos/blindside/conference/trunk/blindside-corehttp://www.blindsideproject.org:8080/svn/repos/blindside/conference/trunk/blindside-core c:\source svn co http://www.blindsideproject.org:8080/svn/repos/blindside/conference/trunk/blindside-mainhttp://www.blindsideproject.org:8080/svn/repos/blindside/conference/trunk/blindside-main c:\source svn co http://www.blindsideproject.org:8080/svn/repos/blindside/conference/trunk/blindside-webhttp://www.blindsideproject.org:8080/svn/repos/blindside/conference/trunk/blindside-web Then c:\sourcecd blindside-main c:\source\blindside-main mvn install (you must have maven installed/configured) c:\source\blindside-main cd ../blindside-web c:\source\blindside-web mvn jetty:run From your browser, visit http://localhost:8080/conferencehttp://localhost:8080/conference Does the Blindside run in linux full installation? Please let me know if you run into problems. Ok! I'll have a look on it, maybe in the middle/end of this week. 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 Roberto Fichera. ___ --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] Blindside Web Conferencing
On 5/28/07, Roberto Fichera [EMAIL PROTECTED] wrote: At 19.19 28/05/2007, you wrote: On 5/28/07, Roberto Fichera mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] wrote: At 17.09 28/05/2007, Richard Alam wrote: Yes, we have some downloadable code. We are in the process of completing the instructions (build/deploy/etc.). Code is located here. http://www.blindsideproject.org:8080/cgi-bin/trac.cgi/browser/conference/trunk http://www.blindsideproject.org:8080/cgi-bin/trac.cgi/browser/conference/trunk http://www.blindsideproject.org:8080/cgi-bin/trac.cgi/browser/conference/trunk How can I get it? Hi, Here is our first attempt at writing the doc. http://www.blindsideproject.org:8080/cgi-bin/trac.cgi/wiki/BlindsideWebApp http://www.blindsideproject.org:8080/cgi-bin/trac.cgi/wiki/BlindsideWebApp Alternatively, you can c:\source svn co http://www.blindsideproject.org:8080/svn/repos/blindside/conference/trunk/blindside-core http://www.blindsideproject.org:8080/svn/repos/blindside/conference/trunk/blindside-core c:\source svn co http://www.blindsideproject.org:8080/svn/repos/blindside/conference/trunk/blindside-main http://www.blindsideproject.org:8080/svn/repos/blindside/conference/trunk/blindside-main c:\source svn co http://www.blindsideproject.org:8080/svn/repos/blindside/conference/trunk/blindside-web http://www.blindsideproject.org:8080/svn/repos/blindside/conference/trunk/blindside-web Then c:\sourcecd blindside-main c:\source\blindside-main mvn install (you must have maven installed/configured) c:\source\blindside-main cd ../blindside-web c:\source\blindside-web mvn jetty:run From your browser, visit http://localhost:8080/conference http://localhost:8080/conference Does the Blindside run in linux full installation? Not sure exactly what you mean. It is a J2EE/Ajax web application. So if you've got Tomcat 5.x running on Linux, you can deploy the war file into it. I'll upload a WAR file (blindside.war) into our server tonight or tomorrow. That way you can download it, put it into your Tomcat's webapps directory, modify the blindside.properties file and off you go. Richard Please let me know if you run into problems. Ok! I'll have a look on it, maybe in the middle/end of this week. 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] Divitas
From: EdPimentl [EMAIL PROTECTED] Date: Sun, 27 May 2007 16:12:09 -0400 There will be a number of companies set to offer similar services. In 3 months we will have a 24 port SIP-GSM-SKYPE gateway -E On 5/27/07, Dean Collins [EMAIL PROTECTED] wrote: I was cleaning through some old IT magazines this long weekend when I came across a company called Divitas in the April 30th edition of Network Computing. I've never heard of them but has anyone else heard of them? Basically they have a call control appliance that can deliver centrally held up calls between not only GSM but also redirect the call to a wifi hotspot if you are in range. It seems like a neat concept that shouldn't necessarily be beyond the capabilities of Asterisk (apart from the fact that the end Win Mobile 5 / Symbian handset would need some type of client). Any thoughts? At $550 per seat looks an expensive way to transfer calls between networks but I've never seen another CPE piece of equipment that can do this. According to another IT magzaine, Divitas indeed uses Asterisk. But Divitas does not seem to be a pure CPE solution. That may be why they could charge a premium. Yuan Liu http://www.divitas.com/products Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). [image: Call Button]http://click.mexuar.com/webuser/click/7/userurl/Cognationhttp://click.mexuar.com/webuser/nojs/7/userurl/Cognation ___ --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] Scaling Asterisk: Dual-Core CPUs not yielding gains at high call volumes - Low volume benchmarks - Correction
Luki wrote: Perhaps a naive question, but how does 0.137% CPU utilization per call equal 1735 MHz per call? If 1735 MHz / 0.137% = 1735 MHz / 0.00137 = 1266423 MHz at 100% utilization ??! Even with 4 CPUs, those would be 316 GHz CPUs. I think you meant: Average CPU utilization per call: 0.137% (~17 MHz) Luki, You are absolutely right. Thank you for pointing out and correcting my mistake. The corrected statistics are below. Note that the MHz per call statistic is calculated with the following formula: MHzPerCall = (numCPUs * CPUspeed) * (avgCPUperCall * .01) Thank you, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer The Numbers (Corrected) --- DC - Incoming SIP to the Playback() application === calls %user %system %iowait %idle 00.00 0.01 0.01 99.98 10.02 0.04 0.00 99.94 20.02 0.06 0.00 99.92 30.03 0.11 0.00 99.86 40.04 0.13 0.00 99.83 50.05 0.16 0.00 99.80 60.05 0.20 0.00 99.75 70.07 0.24 0.00 99.70 80.07 0.25 0.00 99.67 90.08 0.27 0.00 99.65 100.09 0.33 0.00 99.58 Average CPU utilization per call: 0.040% (~9.60 MHz) SC - Incoming SIP to the Playback() application === calls %user %system %iowait %idle 00.01 0.02 0.00 99.98 10.02 0.10 0.00 99.88 20.03 0.17 0.00 99.80 30.06 0.21 0.00 99.73 40.08 0.28 0.00 99.63 50.10 0.34 0.01 99.55 60.11 0.48 0.00 99.41 70.14 0.49 0.00 99.37 80.16 0.57 0.00 99.28 90.17 0.63 0.01 99.19 100.18 0.75 0.00 99.07 Average CPU utilization per call: 0.091% (~11.52 MHz) DC - Incoming SIP to the Queue() application - In queue === calls %user %system %iowait %idle 00.00 0.01 0.00 99.99 10.01 0.03 0.00 99.96 20.01 0.05 0.00 99.94 30.01 0.08 0.00 99.91 40.02 0.10 0.00 99.88 50.03 0.12 0.00 99.84 60.04 0.16 0.00 99.80 70.03 0.17 0.00 99.80 80.04 0.20 0.00 99.76 90.03 0.22 0.00 99.75 100.05 0.27 0.00 99.68 Average CPU utilization per call: 0.031% (~7.44 MHz) SC - Incoming SIP to the Queue() application - In queue === calls %user %system %iowait %idle 00.02 0.02 0.00 99.96 10.03 0.07 0.00 99.91 20.03 0.13 0.00 99.83 30.04 0.18 0.00 99.78 40.05 0.23 0.00 99.72 50.06 0.27 0.00 99.67 60.07 0.33 0.00 99.60 70.09 0.38 0.00 99.53 80.09 0.40 0.00 99.51 90.11 0.46 0.01 99.43 100.11 0.48 0.00 99.41 Average CPU utilization per call: 0.055% (~6.97 MHz) DC - Incoming SIP to the Queue() application - Bridged to an agent == calls %user %system %iowait %idle 00.00 0.01 0.00 99.99 10.01 0.06 0.00 99.93 20.02 0.14 0.00 99.84 30.03 0.16 0.00 99.81 Average CPU utilization per call: 0.060% (~14.40 MHz) SC - Incoming SIP to the Queue() application - Bridged to an agent == calls %user %system %iowait %idle 00.01 0.02 0.00 99.98 10.02 0.16 0.00 99.82 20.04 0.28 0.00 99.68 30.07 0.36 0.00 99.57 Average CPU utilization per call: 0.137% (~17.35 MHz) ___ --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] Asterisk and cell phones
Hello, PBX vendors used to sell software extensions providing enterprise services to cell phones. Main features were : - 4 digit dialing or directory access, - call forwarding, - unified messaging. Now that WiFi and dual mode cell phones get more popular, these Java-based software should be more successful on the market as bandwidth costs decrease. My question are : 1. How Asterisk would compare on cell phone integration with PBX or IPBX vendors ? 2. Is there now a way to enhance high end cell phones with third party software to gain access to Asterisk features ? 3. More specifically, has anyone tested Nokia EXX dialing capabilities ? Would it be easy to customise some callback features with such phones For example : you dial a number and hit a home-made Callback key : your phone then dials a callback server number instead, hangs up, waits for this callback server to call and lastly replies the real number you would like to be connected with (this could be useful when receiving charges are (much) cheaper than emitting charges). Best regards ___ --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: Scaling Asterisk: Dual-Core CPUs not yielding gains at high call volumes
JR Richardson wrote: Do you get any errors at max call capacity about too many open files? You may try increasing your file descriptors. JR, Thanks for the response, but I have the maximum number of open files available to Asterisk set to 65536. Thank you, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer ___ --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] Blindside Web Conferencing
At 19.56 28/05/2007, you wrote: On 5/28/07, Roberto Fichera mailto:[EMAIL PROTECTED][EMAIL PROTECTED] wrote: At 19.19 28/05/2007, you wrote: On 5/28/07, Roberto Fichera mailto: [EMAIL PROTECTED]mailto:[EMAIL PROTECTED][EMAIL PROTECTED] wrote: At 17.09 28/05/2007, Richard Alam wrote: Yes, we have some downloadable code. We are in the process of completing the instructions (build/deploy/etc.). Code is located here. http://www.blindsideproject.org:8080/cgi-bin/trac.cgi/browser/conference/trunkhttp://www.blindsideproject.org:8080/cgi-bin/trac.cgi/browser/conference/trunk http://www.blindsideproject.org:8080/cgi-bin/trac.cgi/browser/conference/trunk http://www.blindsideproject.org:8080/cgi-bin/trac.cgi/browser/conference/trunkhttp://www.blindsideproject.org:8080/cgi-bin/trac.cgi/browser/conference/trunk How can I get it? Hi, Here is our first attempt at writing the doc. http://www.blindsideproject.org:8080/cgi-bin/trac.cgi/wiki/BlindsideWebApphttp://www.blindsideproject.org:8080/cgi-bin/trac.cgi/wiki/BlindsideWebApp Alternatively, you can c:\source svn co http://www.blindsideproject.org:8080/svn/repos/blindside/conference/trunk/blindside-corehttp://www.blindsideproject.org:8080/svn/repos/blindside/conference/trunk/blindside-core http://www.blindsideproject.org:8080/svn/repos/blindside/conference/trunk/blindside-core c:\source svn co http://www.blindsideproject.org:8080/svn/repos/blindside/conference/trunk/blindside-mainhttp://www.blindsideproject.org:8080/svn/repos/blindside/conference/trunk/blindside-main http://www.blindsideproject.org:8080/svn/repos/blindside/conference/trunk/blindside-main c:\source svn co http://www.blindsideproject.org:8080/svn/repos/blindside/conference/trunk/blindside-webhttp://www.blindsideproject.org:8080/svn/repos/blindside/conference/trunk/blindside-web Then c:\sourcecd blindside-main c:\source\blindside-main mvn install (you must have maven installed/configured) c:\source\blindside-main cd ../blindside-web c:\source\blindside-web mvn jetty:run From your browser, visit http://localhost:8080/conferencehttp://localhost:8080/conferencehttp://localhost:8080/conference Does the Blindside run in linux full installation? Not sure exactly what you mean. Since I had a fast look over the installation docs which are talking about cygwin and other things running on windows, I was supposing that it run on Windows. It is a J2EE/Ajax web application. So if you've got Tomcat 5.x running on Linux, you can deploy the war file into it. Yep ;-)! I'll upload a WAR file (blindside.war) into our server tonight or tomorrow. That way you can download it, put it into your Tomcat's webapps directory, modify the blindside.properties file and off you go. That's will be really great and fast to test :-)! Richard Please let me know if you run into problems. Ok! I'll have a look on it, maybe in the middle/end of this week. 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 Roberto Fichera. ___ --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] Scaling Asterisk: Dual-Core CPUs not yielding gains at high call volumes - Low volume benchmarks
William Moore wrote: Are you recording memory figures as well and have you checked the total used memory? Or did I miss it somewhere? Thanks for doing this, scalability testing is always good. William, This round of benchmarking is heavily focused on CPU utilization, because it is causing an immediate problem for me. However, I am tracking some other statistics on a daily basis including memory utilization, swap utilization, load averages, and active channels and calls. One of my colleagues takes the text file I produce and creates graphs using Cacti and rrdtool. You'll be interested in these two (sorry for the format of the URLS, but otherwise the list was eating my posts): - Percent CPU Used With No. Calls and No. Channels img509DOTimageshackDOTusSLASHimg509SLASH3927SLASHastcpuandcallsbf4DOTpng - Asterisk Memory Used (KB) img47DOTimageshackDOTusSLASHimg47SLASH7615SLASHastmemusedgq9DOTpng Note that even with a peak call volume of approximately 400 active calls and 550 active SIP channels, the memory utilization never surpasses 600 KB. I'd estimate that most Asterisk installations would avoid swapping with 1 GB of RAM. A 2nd GB might be useful to provide plenty of room for file caching so that your hard disk doesn't become a bottleneck. We also record all of our calls to a 6 GB RAM disk, so our server has a total of 8 GB of RAM but that isn't necessary in most circumstances. Overall, Asterisk seems to be very efficiently coded as far as memory is concerned. Note that for other reasons we perform a nightly reboot, so I don't know if there are any memory leaks that would surface over time. Thank you, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer ___ --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 cell phones
Hi Olivier, Do a search on my blog www.collins.net.pr/blog for Orative as a suggested application. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). http://click.mexuar.com/webuser/click/7/userurl/Cognation http://click.mexuar.com/webuser/nojs/7/userurl/Cognation From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Olivier Sent: Monday, 28 May 2007 2:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk and cell phones Hello, PBX vendors used to sell software extensions providing enterprise services to cell phones. Main features were : - 4 digit dialing or directory access, - call forwarding, - unified messaging. Now that WiFi and dual mode cell phones get more popular, these Java-based software should be more successful on the market as bandwidth costs decrease. My question are : 1. How Asterisk would compare on cell phone integration with PBX or IPBX vendors ? 2. Is there now a way to enhance high end cell phones with third party software to gain access to Asterisk features ? 3. More specifically, has anyone tested Nokia EXX dialing capabilities ? Would it be easy to customise some callback features with such phones For example : you dial a number and hit a home-made Callback key : your phone then dials a callback server number instead, hangs up, waits for this callback server to call and lastly replies the real number you would like to be connected with (this could be useful when receiving charges are (much) cheaper than emitting charges). Best regards image001.gif___ --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: Scaling Asterisk: Dual-Core CPUs not yielding gains at high call volumes
JR Richardson wrote: The Dual-Core system you are working with must have cost a bundle, several thousand. My approach has been to stick with single cpu, single core servers and add more servers to the cluster, versus building bigger, faster Proc servers. With sub $1000 servers, I can achieve 150-200 calls per server, cluster several servers together and for the same price as a quad proc dual-core server have 700-1000 call capacity. Now, with that said, a cluster becomes harder to build and operate than a 1 server Asterisk implementation and does not work well in some environments, such as with large call queues. But when you are talking straight call capacity, multiple servers will usually dominate singe servers in relation to cost. At the start of our Asterisk project, scaling vertically seemed like the simpler approach and to a certain extent it is necessary because of our call queues. Now that it seems like we've pushed this approach as far as it will go, my eagerness to study and implement Asterisk clusters is growing. I know that horizontal scaling will have its own set of problems, but at this point I think they will be more manageable than what we're currently dealing with. Could you please give me a rough overview of your clustering architecture? I don't need too many details, but a list of the technologies/programs you are using would be a great basis for my research. I currently planning to look into SER and DUNDi, and another poster suggested looking at OpenSSI. Nice discussion, and thanks for posting your benchmark results and feedback. You're welcome. I'll post some large scale numbers off of our production server soon. I'm also going to start looking at SIPp, which may give me a way to gather large scale statistics in a more controlled environment. Thank you for your responses, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer ___ --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] Scaling Asterisk: Dual-Core CPUs not yielding gains at high call volumes
Mark Coccimiglio wrote: Sounds like you are running into the hardware limitations of your systems PCI or Front Side Bus (FSB) and not necessarily an issue of asterisk. In short there is a limited amount of bandwidth on the computer's PCI Bus (33 MHz) and the FSB (100-800MHz). One thing to remember is that ALL cores and data streams need to share the PCI and FSB.Asterisk is very processor and memory intensive. At the extreme level of usage more cores won't help if data is stuck in the pipe. So the performance planing you described would be expected. Mark, That is a great theory and I'd like to follow up on it. Do you know if the PCI or FSB buses are instrumented by Linux? If not, are you aware of any way to gather statistics about their utilization? I'd like to see if the numbers support your idea and, if so, which bus is saturated. Let me add a little bit of extra information to this discussion. The CPU utilization does not flatten out at 50%. In fact, as more calls are added, Asterisk will eventually drive the idle percentage down to single digits with surprisingly few problems. If PCI or FSB bandwidth were the limiting factor, wouldn't the CPU utilization top out at the point that the available bandwidth was used? Thank you, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer ___ --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] Octasic echo cancellation
Hi Sebastien, I'm just a lowly user but I will tell you what I think I understand about it. There is nothing in the Octasic documentation that suggests you can have continuosly updated statistics but I agree that would be a nice to have feature. Have you tried contacting Octasic about that? You can set the ERL and tail length in the octveqd.conf file which you need to manually copy into the /etc/ directory as per the Octasic documentation and octveqd.conf file included with softecho. By default they are set to 9db and 64ms respectively. It is my understanding EC takes place AFTER the tx/rx gain as part of the Zaptel driver function. Therefore your tx/rx gain WILL have an effect on it's operation and (I think) the ERL values. If your using Digium cards you should run fxotune and set your gains. With Sangoma cards you only need to set your gains. If that is all set up ok softecho should get rid of all your echo. If not I would suggest you try increase the tail length to 128ms in octveqd.conf. I don't think you should need to change any of the other settings. -Original Message- From: Sebastien Leclere [mailto:[EMAIL PROTECTED] Sent: Monday, May 28, 2007 10:34 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Octasic echo cancellation Hi, I'm currently testing SoftEcho, an echo cancellation software for Asterisk from Octasic. I noticed an important increase of the quality of my coms, but I still have a few echo problems. There is an ERL parameter which corresponds to an initial ERL value probably to optimize the echo training or something in that kind. Is there a way to monitor, using one of the zttools, the instant ERL value, to be able to set this parameter correctly? In that purpose, I would like to know where do the echo cancellation take place in the communication chain, is it after or before the amplification by the Rx/Tx gain parameters? Will this gain apply on the echo cancelled signal or on the gross signal? Anybody with experience on this product could give me some advice in the aim of removing all trace of echo? Thanks by advance Sebastien ___ --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] Bottom line on fax reception
Sorry but I must have missed it if someone else responded. If the built in fax reception doesn't work very well what about the 3rd party stuff mentioned on the Asterisk Wiki? -Original Message- From: Steve Totaro [mailto:[EMAIL PROTECTED] Sent: Monday, May 28, 2007 8:38 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Bottom line on fax reception Someone already answered this question. The answer is no, it does not work by your definition of production ready. Thanks, Steve Totaro http://www.asteriskhelpdesk.com KB3OPB -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of shadowym Sent: Monday, May 28, 2007 11:20 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Bottom line on fax reception Anybody?? -Original Message- From: shadowym [mailto:[EMAIL PROTECTED] Sent: Thursday, May 24, 2007 9:35 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Bottom line on fax reception So what is the bottom line? Does it work or not. I've heard stories it works, it doesn't work, it kinda sorta works when it's not raining out side. Everything under the rainbow. What's the bottom line with recent updates on 1.2.x? Is it production ready for fax? By production ready I mean that it just works all the time and doesn't need any babysitting. Do I have to worry about dropped lines, sometimes not detecting incoming fax toneyada yada. I know I don't have to use fax on Asterisk but I really want to for various reasons. Mostly incoming but outgoing is a nice to have. Should I use an addon package and if so which one? Any help would be appreciated. ___ --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
[asterisk-users] help on asterisk sipp
I was wondering whether someone could help me. I installed sipp on my Asterisk server but I don't really understand how does it fonction! Has someone ever tried it?If you can explain to me the principle, I would be extremely grateful.Thank you very much in advance. Lancez des recherches en toute sécurité depuis n'importe quelle page Web. Téléchargez GRATUITEMENT Windows Live Toolbar aujourd'hui ! Essayez-le maintenant ! _ Appelez vos amis de PC à PC -- C'EST GRATUIT http://get.live.com/messenger/overview___ --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] Polycom Static IP
I am still having issues with my Polycom 301 phones when I disable DHCP. I give the phone a static address and I keep getting the error 'could not contact boot server using existing config'. As soon as I set it back to DHCP enabled the phone can see the boot server and I'm online. Steve ___ --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] Bottom line on fax reception
Doug Lytle wrote: shadowym wrote: So what is the bottom line? Does it work or not. I've heard stories it As it has been said many many times before, Fax detection is an art and most of the time is not reliable. Faxing on the other hand, using iaxmodem along with HylaFAX+ works very well. Search the archives. Doug Reposting Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --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] RE: Bottom line on fax reception
I gave up on the rxfax business as it never worked for me. I use iaxmodem and hylafax and it works perfectly, every single time i use it. inbound or outbound doesnt matter. I have not read about anyone using iaxmodem and hylafax having any issues. and its fairly easy to setup. Took me about 1 hour total to get everything installed and configured. Message: 3 Date: Mon, 28 May 2007 08:20:22 -0700 From: shadowym [EMAIL PROTECTED] Subject: RE: [asterisk-users] Bottom line on fax reception To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii Anybody?? -Original Message- From: shadowym [mailto:[EMAIL PROTECTED] Sent: Thursday, May 24, 2007 9:35 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Bottom line on fax receptionSo what is the bottom line? Does it work or not. I've heard stories it works, it doesn't work, it kinda sorta works when it's not raining out side. Everything under the rainbow. What's the bottom line with recent updates on 1.2.x? Is it production ready for fax? By production ready I mean that it just works all the time and doesn't need any babysitting. Do I have to worry about dropped lines, sometimes not detecting incoming fax toneyada yada. I know I don't have to use fax on Asterisk but I really want to for various reasons. Mostly incoming but outgoing is a nice to have. Should I use an addon package and if so which one? Any help would be appreciated. Message: 6 Date: Mon, 28 May 2007 11:38:01 -0400 From: Steve Totaro [EMAIL PROTECTED] Subject: RE: [asterisk-users] Bottom line on fax reception To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=US-ASCII Someone already answered this question. The answer is no, it does not work by your definition of production ready. Thanks, Steve Totaro http://www.asteriskhelpdesk.com KB3OPB ___ --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] Bottom line on fax reception
On 5/24/07, shadowym [EMAIL PROTECTED] wrote: So what is the bottom line? Does it work or not. I've heard stories it works, it doesn't work, it kinda sorta works when it's not raining out side. Everything under the rainbow. What's the bottom line with recent updates on 1.2.x? Is it production ready for fax? By production ready I mean that it just works all the time and doesn't need any babysitting. Do I have to worry about dropped lines, sometimes not detecting incoming fax toneyada yada. I know I don't have to use fax on Asterisk but I really want to for various reasons. Mostly incoming but outgoing is a nice to have. Should I use an addon package and if so which one? Any help would be appreciated. No. Digium still refuses to include proper faxing support in Asterisk. OpenPBX is still unstable and developers really aren't too excited about getting it to work... to much other stuff to fix before they worry about fax. FWIW OPBX has seemed sort of dead recently. If you want proper (T.38) fax support then pick up a Cisco AS53xx, AS54xx, AS58xx, 2600, 3600, 7200. You need IOS 12.1 or above, double check Cisco for specific WIC, RAM and other misc. requirements of course! ___ --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] Blindside Web Conferencing
http://www.thecoccinella.org/ looks pretty nice I have not tried this one. It has been a couple of years since I played around with IM clients and I cannot remember what I was using. Thanks, Steve Totaro http://www.asteriskhelpdesk.com/ KB3OPB _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Richard Alam Sent: Monday, May 28, 2007 1:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Blindside Web Conferencing Hi Steve, Yes, we are looking for that. Do you know of any projects that provides those? I know one written in TCL/TK. Thanks. Richard On 5/28/07, Steve Totaro [EMAIL PROTECTED] wrote: Sounds cool. You could probably use some code from the various open source jabber clients that allow for shared whiteboard and pushing URLs too. Thanks, Steve Totaro http://www.asteriskhelpdesk.com http://www.asteriskhelpdesk.com/ KB3OPB _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] ] On Behalf Of Richard Alam Sent: Monday, May 28, 2007 10:18 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Blindside Web Conferencing Hello, We are creating a web-based conferencing application using Asterisk as the voice conferencing server. This as an open source project. We are trying to determine if there is interest of the community and perhaps work together to improve the application. Using the web application, you can upload your powerpoint presentation, manage the participants in the conference thru the web interface (mute/unmute/kick), chat, participants can raise/lower hand to indicate she wants to talk when muted. Please visit http://present.blindsideproject.org http://present.blindsideproject.org/ (http://present.sce.carleton.ca http://present.sce.carleton.ca/ if that doesn't work) to try it out. Click on the requirements link for instructions on how to setup your Idefisk. Project website is at http://www.blindsideproject.org http://www.blindsideproject.org/ (http://www.blindsideproject.org:8080 http://www.blindsideproject.org:8080/ if that doesn't work). We are requesting the University permissions to configure DNS/firewall, so please bear with us. Please send feedback/suggestions to this mailing list. We're also trying to setup our own. Hope the moderators of this list don't mind. Hoping for your feedbacks. Thanks. Blindside Project Team ___ --Bandwidth and Colocation provided by Easynews.com http://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] Polycom Static IP
Sounds like a firmware bug, VLAN or other network configuration bug in the phone (subnet perhaps?) Thanks, Steve Totaro http://www.asteriskhelpdesk.com KB3OPB _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Forum Sent: Monday, May 28, 2007 3:56 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Polycom Static IP I am still having issues with my Polycom 301 phones when I disable DHCP. I give the phone a static address and I keep getting the error 'could not contact boot server using existing config'. As soon as I set it back to DHCP enabled the phone can see the boot server and I'm online. Steve ___ --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: Bottom line on fax reception
Please qualify your usage. A couple faxes a day, a couple hundred, a couple thousand, or a couple hundred thousand? Are you running asterisk and hylafax on the same machine? What is your TDM connectivity? Hylafax uses quite a lot of CPU juice. Anyone ever scale up a quad T1/E1 server for faxing using asterisk and hylafax? Must be a heck of a server! Thanks, Steve Totaro http://www.asteriskhelpdesk.com http://www.asteriskhelpdesk.com/component/option,com_wrapper/Itemid,37/ KB3OPB _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Greg Kennedy Sent: Monday, May 28, 2007 4:02 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] RE: Bottom line on fax reception I gave up on the rxfax business as it never worked for me. I use iaxmodem and hylafax and it works perfectly, every single time i use it. inbound or outbound doesnt matter. I have not read about anyone using iaxmodem and hylafax having any issues. and its fairly easy to setup. Took me about 1 hour total to get everything installed and configured. Message: 3 Date: Mon, 28 May 2007 08:20:22 -0700 From: shadowym [EMAIL PROTECTED] Subject: RE: [asterisk-users] Bottom line on fax reception To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii Anybody?? -Original Message- From: shadowym [mailto:[EMAIL PROTECTED] Sent: Thursday, May 24, 2007 9:35 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Bottom line on fax reception So what is the bottom line? Does it work or not. I've heard stories it works, it doesn't work, it kinda sorta works when it's not raining out side. Everything under the rainbow. What's the bottom line with recent updates on 1.2.x? Is it production ready for fax? By production ready I mean that it just works all the time and doesn't need any babysitting. Do I have to worry about dropped lines, sometimes not detecting incoming fax toneyada yada. I know I don't have to use fax on Asterisk but I really want to for various reasons. Mostly incoming but outgoing is a nice to have. Should I use an addon package and if so which one? Any help would be appreciated. Message: 6 Date: Mon, 28 May 2007 11:38:01 -0400 From: Steve Totaro [EMAIL PROTECTED] Subject: RE: [asterisk-users] Bottom line on fax reception To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] .com Content-Type: text/plain; charset=US-ASCII Someone already answered this question. The answer is no, it does not work by your definition of production ready. Thanks, Steve Totaro http://www.asteriskhelpdesk.com KB3OPB ___ --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] Send parked call extension to set
Anyone know how to send a post transfer (and possibly post hangup) message to an aastra set with the pickup extension Dave Dave Bour Desktop Solution Center 905.381.0077 [EMAIL PROTECTED] For those who just want it to work... Giving you complete IT peace of mind. (Sent via Blackberry - hence message may be shorter than my usual verbose responses) PIN 4cc364db (as of March 24, 2007) ___ --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] Blindside Web Conferencing
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Yes, we are looking for that. Do you know of any projects that provides those? I know one written in TCL/TK. You might also want to have a look at http://www.version2software.com/v2whiteboard.html - its a plugin for the Java based Jabber client Spark (from igniterealtime.org) =Stefan -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.6 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFGW0DZcVCZDfrn+pMRAq4KAJ961ZBIsSNhn7p4+SQI4RPPe1gsHwCdG4dv pQOw6ugERcCUKy7pjDHf/qs= =JI7J -END PGP SIGNATURE- ___ --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: Bottom line on fax reception
Quoting Steve Totaro [EMAIL PROTECTED]: Please qualify your usage. A couple faxes a day, a couple hundred, a couple thousand, or a couple hundred thousand? well what is your usage where it doesn't work ? I would like to know where it does and doesn't work as well but so far various groups have conflicting opinions. Are you running asterisk and hylafax on the same machine? What is your TDM connectivity? Hylafax uses quite a lot of CPU juice. Anyone ever scale up a quad T1/E1 server for faxing using asterisk and hylafax? Must be a heck of a server! Thanks, Steve Totaro http://www.asteriskhelpdesk.com http://www.asteriskhelpdesk.com/component/option,com_wrapper/Itemid,37/ KB3OPB _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Greg Kennedy Sent: Monday, May 28, 2007 4:02 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] RE: Bottom line on fax reception I gave up on the rxfax business as it never worked for me. I use iaxmodem and hylafax and it works perfectly, every single time i use it. inbound or outbound doesnt matter. I have not read about anyone using iaxmodem and hylafax having any issues. and its fairly easy to setup. Took me about 1 hour total to get everything installed and configured. Message: 3 Date: Mon, 28 May 2007 08:20:22 -0700 From: shadowym [EMAIL PROTECTED] Subject: RE: [asterisk-users] Bottom line on fax reception To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii Anybody?? -Original Message- From: shadowym [mailto:[EMAIL PROTECTED] Sent: Thursday, May 24, 2007 9:35 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Bottom line on fax reception So what is the bottom line? Does it work or not. I've heard stories it works, it doesn't work, it kinda sorta works when it's not raining out side. Everything under the rainbow. What's the bottom line with recent updates on 1.2.x? Is it production ready for fax? By production ready I mean that it just works all the time and doesn't need any babysitting. Do I have to worry about dropped lines, sometimes not detecting incoming fax toneyada yada. I know I don't have to use fax on Asterisk but I really want to for various reasons. Mostly incoming but outgoing is a nice to have. Should I use an addon package and if so which one? Any help would be appreciated. Message: 6 Date: Mon, 28 May 2007 11:38:01 -0400 From: Steve Totaro [EMAIL PROTECTED] Subject: RE: [asterisk-users] Bottom line on fax reception To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] .com Content-Type: text/plain; charset=US-ASCII Someone already answered this question. The answer is no, it does not work by your definition of production ready. Thanks, Steve Totaro http://www.asteriskhelpdesk.com KB3OPB Jon Pounder _/_/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/ _/ _/_/_/ _/ _/_/ _/_/ _/_/ _/ _/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/_/ _/_/_/_/ Inline Internet Systems Inc. Thorold, Ontario, Canada Tools to Power Your e-Business Solutions www.inline.net www.ihtml.com www.ihtmlmerchant.com www.opayc.com This message was sent using IMP, the Internet Messaging Program. ___ --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] Language in Zaptel.conf
I am having a problem setting the default language for Zap interfaces. I have an Asterisk 1.4.4 server on CentOS 5 with two Astribank 8 units for analog devices. Here is a sample configuration on one of the ports: language=es context=oficina callerid=Miriam Perez Vite100 mailbox=100 usecallerid=yes callwaiting=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echotraining=800 rxgain=0.0 txgain=0.0 busydetect=yes busycount=6 callprogress=no accountcode=General amaflags=default signalling=fxo_ks pickupgroup=1 callgroup=1 faxdetect=no group=0 channel = 32 As you can see I set the language=es parameter (and do this for all interfaces). I installed the spanish sound set for Asterisk in /var/lib/asterisk/sounds/es (with links to the appropriate directories for letter, digits, phonetic, etc). All Zap interfaces still play all sounds in English. Only if I do a Set(CHANNEL(language)=es) do I get the sounds in Spanish. When I do a zap show channels the language column is blank. The server also has an E1 with Unicall where I have also set the default language to Spanish and all calls coming through the E1 do play the sounds in Spanish. Any ideas why the Zap channels do not want to set the default language? -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part ___ --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] ZAPTEL problem
On Mon, May 28, 2007 at 02:22:34PM +0530, ram wrote: Hi I have 100XP Digium clone card Installed in my pc and compiled zaptel and asterisk again after installing the card but after i rebooted i can load zaptel and wcfxo modprobe with out any problem but when i intiated ztcfg - i get the following error Zaptel Configuration == 1 channels configured. ZT_CHANCONFIG failed on channel 1: No such device or address (6) No device to talk to. Why? - dmesg errors Zapata Telephony Interface Unloaded Zapata Telephony Interface Registered on major 196 Zaptel Version: SVN-branch-1.2-r1725 Echo Canceller: KB1 Failed to initailize DAA, giving up... wcfxo: probe of :00:10.0 failed with error -5 The problem is here. Defective card? PCI-level problems? An example of PCI-level issues is the following magic that worked for me with several cards with some computers (at least with some specific kernels) - adding the boot parameter 'pci=noacpi' to the kernel command-line. -- lspci 00:0c.1 SCSI storage controller: Adaptec AIC-7896U2/7897U2 00:10.0 Communication controller: Motorola Unknown device 5608 I assume that the second line is the X100P card. -- 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] Bottom line on fax reception
I use Asterisk and Hylafax and iaxmodem to send 600+ faxes per week with no baby sitting, I receive about 20 and it requires no baby sitting Hylafax and iaxmodem with Asterisk works very well - check out iaxmodem and hylafax lists for much bigger examples Cheers Duncan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of shadowym Sent: Tuesday, 29 May 2007 7:34 a.m. To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Bottom line on fax reception Sorry but I must have missed it if someone else responded. If the built in fax reception doesn't work very well what about the 3rd party stuff mentioned on the Asterisk Wiki? -Original Message- From: Steve Totaro [mailto:[EMAIL PROTECTED] Sent: Monday, May 28, 2007 8:38 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Bottom line on fax reception Someone already answered this question. The answer is no, it does not work by your definition of production ready. Thanks, Steve Totaro http://www.asteriskhelpdesk.com KB3OPB -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of shadowym Sent: Monday, May 28, 2007 11:20 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Bottom line on fax reception Anybody?? -Original Message- From: shadowym [mailto:[EMAIL PROTECTED] Sent: Thursday, May 24, 2007 9:35 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Bottom line on fax reception So what is the bottom line? Does it work or not. I've heard stories it works, it doesn't work, it kinda sorta works when it's not raining out side. Everything under the rainbow. What's the bottom line with recent updates on 1.2.x? Is it production ready for fax? By production ready I mean that it just works all the time and doesn't need any babysitting. Do I have to worry about dropped lines, sometimes not detecting incoming fax toneyada yada. I know I don't have to use fax on Asterisk but I really want to for various reasons. Mostly incoming but outgoing is a nice to have. Should I use an addon package and if so which one? Any help would be appreciated. ___ --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 ___ --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] language in zapata.conf [was: Language in Zaptel.conf]
On Mon, May 28, 2007 at 04:08:35PM -0500, Carlos Chavez wrote: As you can see I set the language=es parameter (and do this for all interfaces). I installed the spanish sound set for Asterisk in /var/lib/asterisk/sounds/es (with links to the appropriate directories for letter, digits, phonetic, etc). All Zap interfaces still play all sounds in English. Only if I do a Set(CHANNEL(language)=es) do I get the sounds in Spanish. When I do a zap show channels the language column is blank. Duh. It seems that the value of language in zapata.conf is indeed ignored. My bad. Writing a fix. -- 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] RE: Bottom line on fax reception
Steve Totaro wrote: Please qualify your usage. A couple faxes a day, a couple hundred, a couple thousand, or a couple hundred thousand? Couple hundred thousand per month - at least on one installation. Are you running asterisk and hylafax on the same machine? What is your TDM connectivity? Yes, same machine, TDM is PRI, usually... at least it is on the installation I am mentioning. Hylafax uses quite a lot of CPU juice. Huh? Certainly much, much less than Asterisk. Anyone ever scale up a quad T1/E1 server for faxing using asterisk and hylafax? Must be a heck of a server! It's okay. :-) Lee. ___ --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] language in zapata.conf [was: Language in Zaptel.conf]
On Tue, May 29, 2007 at 12:34:33AM +0300, Tzafrir Cohen wrote: On Mon, May 28, 2007 at 04:08:35PM -0500, Carlos Chavez wrote: As you can see I set the language=es parameter (and do this for all interfaces). I installed the spanish sound set for Asterisk in /var/lib/asterisk/sounds/es (with links to the appropriate directories for letter, digits, phonetic, etc). All Zap interfaces still play all sounds in English. Only if I do a Set(CHANNEL(language)=es) do I get the sounds in Spanish. When I do a zap show channels the language column is blank. Duh. It seems that the value of language in zapata.conf is indeed ignored. My bad. Writing a fix. And sadly I was looking at an obsolete copy of the SVN. That has already been fixed in the SVN after 1.4.4 was released: http://bugs.digium.com/view.php?id=9626 http://svn.digium.com/view/asterisk?rev=62331view=rev (reported by sergee and fixed by russel) Well. At least I fixed the title of the thread... -- 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] Bottom line on fax reception
Thanks for all the replies. Seems there are at least 2 or 3 people giving strong recommendations to iaxmodem/hylafax as a reliable (ie. business grade production) solution. That is just the sort of feedback I was looking for. My application is just standard business reception of faxes. Right now they use WinFax and probably receive about 30 to 50 faxes a day. I want to wean them off Winfax as it's not really supported anymore and I dislike all things Symantec in general. Receiving faxes on Asterisk has the added benefit of being able to use the fax line as an extra outgoing line when the rest are in use. That is what they are doing now on their key system and they don't want to lose that ability. I don't blame them. -Original Message- From: Duncan Turnbull [mailto:[EMAIL PROTECTED] Sent: Monday, May 28, 2007 2:18 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Bottom line on fax reception I use Asterisk and Hylafax and iaxmodem to send 600+ faxes per week with no baby sitting, I receive about 20 and it requires no baby sitting Hylafax and iaxmodem with Asterisk works very well - check out iaxmodem and hylafax lists for much bigger examples Cheers Duncan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of shadowym Sent: Tuesday, 29 May 2007 7:34 a.m. To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Bottom line on fax reception Sorry but I must have missed it if someone else responded. If the built in fax reception doesn't work very well what about the 3rd party stuff mentioned on the Asterisk Wiki? -Original Message- From: Steve Totaro [mailto:[EMAIL PROTECTED] Sent: Monday, May 28, 2007 8:38 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Bottom line on fax reception Someone already answered this question. The answer is no, it does not work by your definition of production ready. Thanks, Steve Totaro http://www.asteriskhelpdesk.com KB3OPB -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of shadowym Sent: Monday, May 28, 2007 11:20 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Bottom line on fax reception Anybody?? -Original Message- From: shadowym [mailto:[EMAIL PROTECTED] Sent: Thursday, May 24, 2007 9:35 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Bottom line on fax reception So what is the bottom line? Does it work or not. I've heard stories it works, it doesn't work, it kinda sorta works when it's not raining out side. Everything under the rainbow. What's the bottom line with recent updates on 1.2.x? Is it production ready for fax? By production ready I mean that it just works all the time and doesn't need any babysitting. Do I have to worry about dropped lines, sometimes not detecting incoming fax toneyada yada. I know I don't have to use fax on Asterisk but I really want to for various reasons. Mostly incoming but outgoing is a nice to have. Should I use an addon package and if so which one? Any help would be appreciated. ___ --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 ___ --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] Blindside Web Conferencing
Thanks Stefan! I was just thinking the other day that it would be great if I could whiteboard in Spark. Back on topic, I'm definitely interested in this web conferencing app. I'll have to check it out once a .war is made available and I have a few spare moments. - Brad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stefan Reuter Sent: Monday, May 28, 2007 4:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Blindside Web Conferencing -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Yes, we are looking for that. Do you know of any projects that provides those? I know one written in TCL/TK. You might also want to have a look at http://www.version2software.com/v2whiteboard.html - its a plugin for the Java based Jabber client Spark (from igniterealtime.org) =Stefan -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.6 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFGW0DZcVCZDfrn+pMRAq4KAJ961ZBIsSNhn7p4+SQI4RPPe1gsHwCdG4dv pQOw6ugERcCUKy7pjDHf/qs= =JI7J -END PGP SIGNATURE- ___ --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 The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. ___ --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] DTMFToText Installation process
On Mon, May 28, 2007 at 07:18:59PM +0530, rajesh koniki wrote: Hi, I was looking for a way to pass alphanumeric variables to asterisk via the keypad, found this application app_dtmftotext.c , And I already tried with 'spandsp' application for this. But I am getting errors.[I followed the instructions at http://www.soft-switch.org/installing-spandsp.html]. Can anybody be of help Me on this getting DTMFToText() application on asterisk with the help of app_dtmftotext.c and/or spandsp application is appreciated. app_dtmftotext.c is an asterisk module that uses the library spandsp to provide the dialplan application DTMFToText . What version of Asterisk do you have? What version of spandsp did you try to install? Which linux distribution do you use? -- 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] Blindside Web Conferencing
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hey Brad, I am not sure if you know about the Asterisk-IM plugin for Openfire. Basically it supports dialing contacts and arbitrary numbers through Spark and updates presence based on being on call or not. One of our next steps would be to integrate conferencing so you could setup (and control) a voice conference much the same way you can do with Jabber groupchat. We also have a web conferencing app in a pre beta state sitting around for some time now (based on Asterisk-Java, DWR and Tomcat) with the original intend to use it for a commercial service which never got really started though. I am not sure if we could come together in some way but if you are interested feel free to contact me off-list. =Stefan -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.6 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFGW1V9cVCZDfrn+pMRAt0TAJ4n0BPLDu1EBqqZg5RtIy4tEsLsJgCeJQFW yePaEzQ9FX65+SoTGxs8B6M= =TUw5 -END PGP SIGNATURE- ___ --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] TE205P, E1, Panasonic PBX and hang-up issues
Yes that makes more sense. Now to the problem, please post your zapata.conf as well as your zaptel.conf. Also if you don't mind downloading the config file from the Panasonic TD1232 and email to me off list so I can take a look at it and make sure the settings are ok on the panasonic side. Thank you On 5/28/07, Barry O'Donovan [EMAIL PROTECTED] wrote: On Fri 25 May 2007, C F wrote: Are you sure the panasonic is TVP 100? I have installed over 50 Panasonic systems in my life, and service many more, I have never heard of that system, and a quick google shows it's just a VoiceMail system and not a PBX. Thanks for the reply. Does D1232 Digital Super Hybrid System make more sense? Thanks, Barry On 5/23/07, Barry O'Donovan [EMAIL PROTECTED] wrote: Hey folks, I have a Digium TE205P working as a man in the middle: PRI line Asterisk/TE205P PBX The PBX is a Panasonic KX - TVP 100. Everything is working great except for one little issue. Asterisk isn't hanging up the PRI B channel when the PBX channel is hung up. I don't want to overload you with information but please ask if more is needed. I suspect I'm really hoping someone who had a similar problem with just say ah, I know what that is!. Versions in use for Zaptel, LibPRI and Asterisk are all the SVN 1.4 branch. To replicate: 1. dial a mobile (say) from one of the PBX phones; 2. when you here a ring tone, hang up the PBX phone; 3. the mobile continues to ring. The verbose output is: -- Accepting overlap call from '' to 'unspecified' on channel 0/17, span 2 -- Starting simple switch on 'Zap/48-1' -- Executing [EMAIL PROTECTED]:1] Set(Zap/48-1, RECORDFILE=/srv/recordings/live/1179858572.0) in new stack -- Executing [EMAIL PROTECTED]:2] MixMonitor(Zap/48-1, /srv/recordings/live/1179858572.0.wav|b) in new stack -- Executing [EMAIL PROTECTED]:3] SetCallerPres(Zap/48-1, allowed) in new stack -- Executing [EMAIL PROTECTED]:4] SetCallerID(Zap/48-1, 5400) in new stack -- Executing [EMAIL PROTECTED]:5] Dial(Zap/48-1, Zap/g0/0868017669) in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called g0/0868017669 == Begin MixMonitor Recording Zap/48-1 -- Zap/1-1 is ringing -- Channel 0/17, span 2 got hangup request, cause 16 -- Zap/1-1 answered Zap/48-1 -- Channel 0/1, span 1 got hangup request, cause 0 -- Hungup 'Zap/1-1' == Spawn extension (pbx, 0868017669, 5) exited non-zero on 'Zap/48-1' == End MixMonitor Recording Zap/48-1 -- Hungup 'Zap/48-1' asterisk1*CLI Any suggestions or fixes that you might have from prior instances would be greatly appreciated. Thanks a million, Barry O'Donovan http://www.barryodonovan.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 ___ --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 -- Kind regards, Barry O'Donovan +353 86 801 7669 http://www.barryodonovan.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] language in zapata.conf [was: Language in Zaptel.conf]
On Tue, 2007-05-29 at 00:53 +0300, Tzafrir Cohen wrote: On Tue, May 29, 2007 at 12:34:33AM +0300, Tzafrir Cohen wrote: On Mon, May 28, 2007 at 04:08:35PM -0500, Carlos Chavez wrote: As you can see I set the language=es parameter (and do this for all interfaces). I installed the spanish sound set for Asterisk in /var/lib/asterisk/sounds/es (with links to the appropriate directories for letter, digits, phonetic, etc). All Zap interfaces still play all sounds in English. Only if I do a Set(CHANNEL(language)=es) do I get the sounds in Spanish. When I do a zap show channels the language column is blank. Duh. It seems that the value of language in zapata.conf is indeed ignored. My bad. Writing a fix. And sadly I was looking at an obsolete copy of the SVN. That has already been fixed in the SVN after 1.4.4 was released: http://bugs.digium.com/view.php?id=9626 http://svn.digium.com/view/asterisk?rev=62331view=rev (reported by sergee and fixed by russel) Well. At least I fixed the title of the thread... So is this fixed by using Asterisk SVN or Zaptel SVN? -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part ___ --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: Bottom line on fax reception
Greg Kennedy wrote: I gave up on the rxfax business as it never worked for me. I use iaxmodem and hylafax and it works perfectly, every single time i use it. inbound or outbound doesnt matter. I have not read about anyone using iaxmodem and hylafax having any issues. and its fairly easy to setup. Took me about 1 hour total to get everything installed and configured. Where is a how to on this and does it pass thru to a fax machine? I've been fighting with getting faxing to work on my home asterisk machine and have given up. But if you say it works well and reliably on large volume I'd be willing to try again on my home machine. We get a quite low volume of faxes 10 or less per week. But they are how my wifes family communicates and when our fax does work we start to loose contact with them, so she would argue that we should go back to plain old telephones. ___ --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] Bottom line on fax reception
A colleague of mine did some testing the other day, with a digium TDM400 with FXS modules hooked up to fax machines and a TE120P hooked up to our testing E1 line. It seems to work pretty well, and she said it was easy to configure. PaulH On Mon, 2007-05-28 at 14:55 -0700, shadowym wrote: Thanks for all the replies. Seems there are at least 2 or 3 people giving strong recommendations to iaxmodem/hylafax as a reliable (ie. business grade production) solution. That is just the sort of feedback I was looking for. My application is just standard business reception of faxes. Right now they use WinFax and probably receive about 30 to 50 faxes a day. I want to wean them off Winfax as it's not really supported anymore and I dislike all things Symantec in general. Receiving faxes on Asterisk has the added benefit of being able to use the fax line as an extra outgoing line when the rest are in use. That is what they are doing now on their key system and they don't want to lose that ability. I don't blame them. -Original Message- From: Duncan Turnbull [mailto:[EMAIL PROTECTED] Sent: Monday, May 28, 2007 2:18 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Bottom line on fax reception I use Asterisk and Hylafax and iaxmodem to send 600+ faxes per week with no baby sitting, I receive about 20 and it requires no baby sitting Hylafax and iaxmodem with Asterisk works very well - check out iaxmodem and hylafax lists for much bigger examples Cheers Duncan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of shadowym Sent: Tuesday, 29 May 2007 7:34 a.m. To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Bottom line on fax reception Sorry but I must have missed it if someone else responded. If the built in fax reception doesn't work very well what about the 3rd party stuff mentioned on the Asterisk Wiki? -Original Message- From: Steve Totaro [mailto:[EMAIL PROTECTED] Sent: Monday, May 28, 2007 8:38 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Bottom line on fax reception Someone already answered this question. The answer is no, it does not work by your definition of production ready. Thanks, Steve Totaro http://www.asteriskhelpdesk.com KB3OPB -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of shadowym Sent: Monday, May 28, 2007 11:20 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Bottom line on fax reception Anybody?? -Original Message- From: shadowym [mailto:[EMAIL PROTECTED] Sent: Thursday, May 24, 2007 9:35 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Bottom line on fax reception So what is the bottom line? Does it work or not. I've heard stories it works, it doesn't work, it kinda sorta works when it's not raining out side. Everything under the rainbow. What's the bottom line with recent updates on 1.2.x? Is it production ready for fax? By production ready I mean that it just works all the time and doesn't need any babysitting. Do I have to worry about dropped lines, sometimes not detecting incoming fax toneyada yada. I know I don't have to use fax on Asterisk but I really want to for various reasons. Mostly incoming but outgoing is a nice to have. Should I use an addon package and if so which one? Any help would be appreciated. ___ --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 ___ --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] RE: Bottom line on fax reception
Tim Litwiller wrote: Greg Kennedy wrote: I gave up on the rxfax business as it never worked for me. I use iaxmodem and hylafax and it works perfectly, every single time i use it. inbound or outbound doesnt matter. I have not read about anyone using iaxmodem and hylafax having any issues. and its fairly easy to setup. Took me about 1 hour total to get everything installed and configured. Where is a how to on this and does it pass thru to a fax machine? I've been fighting with getting faxing to work on my home asterisk machine and have given up. But if you say it works well and reliably on large volume I'd be willing to try again on my home machine. We get a quite low volume of faxes 10 or less per week. But they are how my wifes family communicates and when our fax does work we start to loose contact with them, so she would argue that we should go back to plain old telephones. Fax reception does work reliably IF (big IF) you are faxing using a transport media that is conducive to faxing. The internet is not a transport method that will result in 100% reliable connectivity. If you're on a reasonably good internet connection with low-latency and jitter between you and your voip service AND you are using ulaw, you should get acceptable results for residential purposes. I've connected my Brother MFC to a Digium TDM400 card and successfully sent and received faxes over the internet in the past. I've also had my share of failures with the same connection when a large file was being downloaded, even with traffic shaping enabled. People need to be VERY clear about this when they say they are faxing successfully through Asterisk. Lee has all sorts of ammunition why you shouldn't even try it over IP, at least not in a business setting. Darrick -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.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: Bottom line on fax reception
If you are a junk spam faxer then it should suit your needs. If you occasionally send faxes and if you do not receive one or the other party does not receive one or it spits out junk but that is OK, then it should fit your needs. If you are faxing contracts or other important documents that are worth something, then go for a more reliable solution. On a 3ghz HP DL320 with a gig of RAM, each fax took about 5% indicated by top. I would not want to go above ten simultaneous faxes so I setup ten IAX Modems (50% in top). Even at that rate, there were a lot of failures. I did not bother to figure out why because these were legal contracts, in bulk, amounting to big dollars. The variables are very simple for any of these kind of decisions. Don't think about savings, think about costs. Costs of equipment Costs of time (resources) implementing Costs of maintenance Costs of losing data (faxes in this case) Costs of going back and doing it the right way if you find the above costs are higher than another solution. Thanks, Steve Totaro http://www.asteriskhelpdesk.com KB3OPB -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Jon Pounder Sent: Monday, May 28, 2007 4:54 PM To: asterisk-users@lists.digium.com Subject: RE: [asterisk-users] RE: Bottom line on fax reception Quoting Steve Totaro [EMAIL PROTECTED]: Please qualify your usage. A couple faxes a day, a couple hundred, a couple thousand, or a couple hundred thousand? well what is your usage where it doesn't work ? I would like to know where it does and doesn't work as well but so far various groups have conflicting opinions. Are you running asterisk and hylafax on the same machine? What is your TDM connectivity? Hylafax uses quite a lot of CPU juice. Anyone ever scale up a quad T1/E1 server for faxing using asterisk and hylafax? Must be a heck of a server! Thanks, Steve Totaro http://www.asteriskhelpdesk.com http://www.asteriskhelpdesk.com/component/option,com_wrapper/Itemid,37/ KB3OPB _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Greg Kennedy Sent: Monday, May 28, 2007 4:02 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] RE: Bottom line on fax reception I gave up on the rxfax business as it never worked for me. I use iaxmodem and hylafax and it works perfectly, every single time i use it. inbound or outbound doesnt matter. I have not read about anyone using iaxmodem and hylafax having any issues. and its fairly easy to setup. Took me about 1 hour total to get everything installed and configured. Message: 3 Date: Mon, 28 May 2007 08:20:22 -0700 From: shadowym [EMAIL PROTECTED] Subject: RE: [asterisk-users] Bottom line on fax reception To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii Anybody?? -Original Message- From: shadowym [mailto:[EMAIL PROTECTED] Sent: Thursday, May 24, 2007 9:35 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Bottom line on fax reception So what is the bottom line? Does it work or not. I've heard stories it works, it doesn't work, it kinda sorta works when it's not raining out side. Everything under the rainbow. What's the bottom line with recent updates on 1.2.x? Is it production ready for fax? By production ready I mean that it just works all the time and doesn't need any babysitting. Do I have to worry about dropped lines, sometimes not detecting incoming fax toneyada yada. I know I don't have to use fax on Asterisk but I really want to for various reasons. Mostly incoming but outgoing is a nice to have. Should I use an addon package and if so which one? Any help would be appreciated. Message: 6 Date: Mon, 28 May 2007 11:38:01 -0400 From: Steve Totaro [EMAIL PROTECTED] Subject: RE: [asterisk-users] Bottom line on fax reception To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] .com Content-Type: text/plain; charset=US-ASCII Someone already answered this question. The answer is no, it does not work by your definition of production ready. Thanks, Steve Totaro http://www.asteriskhelpdesk.com KB3OPB Jon Pounder _/_/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/ _/ _/_/_/ _/ _/_/ _/_/ _/_/ _/ _/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/_/ _/_/_/_/ Inline Internet Systems Inc. Thorold, Ontario, Canada Tools to Power Your e-Business Solutions www.inline.net www.ihtml.com www.ihtmlmerchant.com www.opayc.com This
Re: [asterisk-users] Multiple TDM400p cards in one machine -- no longer an issue?
Chris Earle wrote: Hi all, Years ago, I was pretty sure attempting to use two TDM400p cards in one machine was recommended against by Digium ... probably because the cards couldn't hack it, and/or interrupt problems etc I have seen some posts recently that seem to indicate it is in fact possible these days thanks to some updated firmware perhaps? . I just need to have two in the server because the 4 ports aren't enough ... I'd rather just expand by one card rather than get a TDM2400 (or TDM800??) Anyone had recent success/failure with this sort of thing? Sangoma Remora Card may be an option? http://www.sangoma.com/datasheets/p_a200-specs -- Warm Regards, Lee ___ --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 TDM400p cards in one machine -- nolonger an issue?
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Lee Jenkins Sent: Monday, May 28, 2007 8:57 PM To: Chris Earle; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Multiple TDM400p cards in one machine -- nolonger an issue? Chris Earle wrote: Hi all, Years ago, I was pretty sure attempting to use two TDM400p cards in one machine was recommended against by Digium ... probably because the cards couldn't hack it, and/or interrupt problems etc I have seen some posts recently that seem to indicate it is in fact possible these days thanks to some updated firmware perhaps? . I just need to have two in the server because the 4 ports aren't enough ... I'd rather just expand by one card rather than get a TDM2400 (or TDM800??) Anyone had recent success/failure with this sort of thing? Sangoma Remora Card may be an option? http://www.sangoma.com/datasheets/p_a200-specs -- Warm Regards, Lee Maybe it is time to look at a fractional T1? I recently used www.shopfort1.com for realtime pricing. Thanks, Steve ___ --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: Bottom line on fax reception
Quoting Steve Totaro [EMAIL PROTECTED]: If you are a junk spam faxer then it should suit your needs. If you occasionally send faxes and if you do not receive one or the other party does not receive one or it spits out junk but that is OK, then it should fit your needs. If you are faxing contracts or other important documents that are worth something, then go for a more reliable solution. On a 3ghz HP DL320 with a gig of RAM, each fax took about 5% indicated by top. I would not want to go above ten simultaneous faxes so I setup ten IAX Modems (50% in top). Even at that rate, there were a lot of failures. I did not bother to figure out why because these were legal contracts, in bulk, amounting to big dollars. anyone have a comparison with a multicpu machine with the same or lower clock rate ? The variables are very simple for any of these kind of decisions. Don't think about savings, think about costs. Costs of equipment Costs of time (resources) implementing Costs of maintenance Costs of losing data (faxes in this case) Costs of going back and doing it the right way if you find the above costs are higher than another solution. Thanks, Steve Totaro http://www.asteriskhelpdesk.com KB3OPB -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Jon Pounder Sent: Monday, May 28, 2007 4:54 PM To: asterisk-users@lists.digium.com Subject: RE: [asterisk-users] RE: Bottom line on fax reception Quoting Steve Totaro [EMAIL PROTECTED]: Please qualify your usage. A couple faxes a day, a couple hundred, a couple thousand, or a couple hundred thousand? well what is your usage where it doesn't work ? I would like to know where it does and doesn't work as well but so far various groups have conflicting opinions. Are you running asterisk and hylafax on the same machine? What is your TDM connectivity? Hylafax uses quite a lot of CPU juice. Anyone ever scale up a quad T1/E1 server for faxing using asterisk and hylafax? Must be a heck of a server! Thanks, Steve Totaro http://www.asteriskhelpdesk.com http://www.asteriskhelpdesk.com/component/option,com_wrapper/Itemid,37/ KB3OPB _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Greg Kennedy Sent: Monday, May 28, 2007 4:02 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] RE: Bottom line on fax reception I gave up on the rxfax business as it never worked for me. I use iaxmodem and hylafax and it works perfectly, every single time i use it. inbound or outbound doesnt matter. I have not read about anyone using iaxmodem and hylafax having any issues. and its fairly easy to setup. Took me about 1 hour total to get everything installed and configured. Message: 3 Date: Mon, 28 May 2007 08:20:22 -0700 From: shadowym [EMAIL PROTECTED] Subject: RE: [asterisk-users] Bottom line on fax reception To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii Anybody?? -Original Message- From: shadowym [mailto:[EMAIL PROTECTED] Sent: Thursday, May 24, 2007 9:35 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Bottom line on fax reception So what is the bottom line? Does it work or not. I've heard stories it works, it doesn't work, it kinda sorta works when it's not raining out side. Everything under the rainbow. What's the bottom line with recent updates on 1.2.x? Is it production ready for fax? By production ready I mean that it just works all the time and doesn't need any babysitting. Do I have to worry about dropped lines, sometimes not detecting incoming fax toneyada yada. I know I don't have to use fax on Asterisk but I really want to for various reasons. Mostly incoming but outgoing is a nice to have. Should I use an addon package and if so which one? Any help would be appreciated. Message: 6 Date: Mon, 28 May 2007 11:38:01 -0400 From: Steve Totaro [EMAIL PROTECTED] Subject: RE: [asterisk-users] Bottom line on fax reception To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] .com Content-Type: text/plain; charset=US-ASCII Someone already answered this question. The answer is no, it does not work by your definition of production ready. Thanks, Steve Totaro http://www.asteriskhelpdesk.com KB3OPB Jon Pounder _/_/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/ _/ _/_/_/ _/ _/_/ _/_/ _/_/ _/ _/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/_/ _/_/_/_/ Inline Internet Systems Inc. Thorold, Ontario, Canada Tools to Power Your e-Business Solutions www.inline.net www.ihtml.com www.ihtmlmerchant.com www.opayc.com
RE: [asterisk-users] TE205P, E1, Panasonic PBX and hang-up issues
According to your CLI output, the channel is being torn down. Is there a lag on the CLI between the inside channel and the outfacing channel getting the hangup request? Does this only happen on mobile phones? I know if I call my cell and hangup, it will continue to ring a couple or even a few more times. Thanks, Steve Totaro http://www.asteriskhelpdesk.com KB3OPB -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of C F Sent: Monday, May 28, 2007 6:30 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] TE205P, E1, Panasonic PBX and hang-up issues Yes that makes more sense. Now to the problem, please post your zapata.conf as well as your zaptel.conf. Also if you don't mind downloading the config file from the Panasonic TD1232 and email to me off list so I can take a look at it and make sure the settings are ok on the panasonic side. Thank you On 5/28/07, Barry O'Donovan [EMAIL PROTECTED] wrote: On Fri 25 May 2007, C F wrote: Are you sure the panasonic is TVP 100? I have installed over 50 Panasonic systems in my life, and service many more, I have never heard of that system, and a quick google shows it's just a VoiceMail system and not a PBX. Thanks for the reply. Does D1232 Digital Super Hybrid System make more sense? Thanks, Barry On 5/23/07, Barry O'Donovan [EMAIL PROTECTED] wrote: Hey folks, I have a Digium TE205P working as a man in the middle: PRI line Asterisk/TE205P PBX The PBX is a Panasonic KX - TVP 100. Everything is working great except for one little issue. Asterisk isn't hanging up the PRI B channel when the PBX channel is hung up. I don't want to overload you with information but please ask if more is needed. I suspect I'm really hoping someone who had a similar problem with just say ah, I know what that is!. Versions in use for Zaptel, LibPRI and Asterisk are all the SVN 1.4 branch. To replicate: 1. dial a mobile (say) from one of the PBX phones; 2. when you here a ring tone, hang up the PBX phone; 3. the mobile continues to ring. The verbose output is: -- Accepting overlap call from '' to 'unspecified' on channel 0/17, span 2 -- Starting simple switch on 'Zap/48-1' -- Executing [EMAIL PROTECTED]:1] Set(Zap/48-1, RECORDFILE=/srv/recordings/live/1179858572.0) in new stack -- Executing [EMAIL PROTECTED]:2] MixMonitor(Zap/48-1, /srv/recordings/live/1179858572.0.wav|b) in new stack -- Executing [EMAIL PROTECTED]:3] SetCallerPres(Zap/48-1, allowed) in new stack -- Executing [EMAIL PROTECTED]:4] SetCallerID(Zap/48-1, 5400) in new stack -- Executing [EMAIL PROTECTED]:5] Dial(Zap/48-1, Zap/g0/0868017669) in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called g0/0868017669 == Begin MixMonitor Recording Zap/48-1 -- Zap/1-1 is ringing -- Channel 0/17, span 2 got hangup request, cause 16 -- Zap/1-1 answered Zap/48-1 -- Channel 0/1, span 1 got hangup request, cause 0 -- Hungup 'Zap/1-1' == Spawn extension (pbx, 0868017669, 5) exited non-zero on 'Zap/48-1' == End MixMonitor Recording Zap/48-1 -- Hungup 'Zap/48-1' asterisk1*CLI Any suggestions or fixes that you might have from prior instances would be greatly appreciated. Thanks a million, Barry O'Donovan http://www.barryodonovan.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 ___ --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 -- Kind regards, Barry O'Donovan +353 86 801 7669 http://www.barryodonovan.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 ___ --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: Bottom line on fax reception
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Jon Pounder Sent: Monday, May 28, 2007 9:10 PM To: asterisk-users@lists.digium.com Subject: RE: [asterisk-users] RE: Bottom line on fax reception Quoting Steve Totaro [EMAIL PROTECTED]: If you are a junk spam faxer then it should suit your needs. If you occasionally send faxes and if you do not receive one or the other party does not receive one or it spits out junk but that is OK, then it should fit your needs. If you are faxing contracts or other important documents that are worth something, then go for a more reliable solution. On a 3ghz HP DL320 with a gig of RAM, each fax took about 5% indicated by top. I would not want to go above ten simultaneous faxes so I setup ten IAX Modems (50% in top). Even at that rate, there were a lot of failures. I did not bother to figure out why because these were legal contracts, in bulk, amounting to big dollars. anyone have a comparison with a multicpu machine with the same or lower clock rate ? Let me further qualify my results. This was done with whatever the current stable versions of Asterisk, Hylafax, and IAXmodem were available in January of this year. The faxes were outbound. PDFs put into a Samba share and a cron job moving them over to the Hylafax monitored directory. Thanks, Steve Totaro www.asteriskhelpdesk.com The variables are very simple for any of these kind of decisions. Don't think about savings, think about costs. Costs of equipment Costs of time (resources) implementing Costs of maintenance Costs of losing data (faxes in this case) Costs of going back and doing it the right way if you find the above costs are higher than another solution. Thanks, Steve Totaro http://www.asteriskhelpdesk.com KB3OPB ___ --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: Bottom line on fax reception
Quoting Steve Totaro [EMAIL PROTECTED]: -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Jon Pounder Sent: Monday, May 28, 2007 9:10 PM To: asterisk-users@lists.digium.com Subject: RE: [asterisk-users] RE: Bottom line on fax reception Quoting Steve Totaro [EMAIL PROTECTED]: If you are a junk spam faxer then it should suit your needs. If you occasionally send faxes and if you do not receive one or the other party does not receive one or it spits out junk but that is OK, then it should fit your needs. If you are faxing contracts or other important documents that are worth something, then go for a more reliable solution. On a 3ghz HP DL320 with a gig of RAM, each fax took about 5% indicated by top. I would not want to go above ten simultaneous faxes so I setup ten IAX Modems (50% in top). Even at that rate, there were a lot of failures. I did not bother to figure out why because these were legal contracts, in bulk, amounting to big dollars. anyone have a comparison with a multicpu machine with the same or lower clock rate ? Let me further qualify my results. This was done with whatever the current stable versions of Asterisk, Hylafax, and IAXmodem were available in January of this year. The faxes were outbound. PDFs put into a Samba share and a cron job moving them over to the Hylafax monitored directory. for my application I am more concerned with inbound working, outbound is just a bonus if it works. one of the big points is when you have a distributed workforce conventional fax machines don't work out since you get a paper result in one place and the recipient in another. Hylafax output can easily be redirected from a general delivery mailbox, or people can have their own fax extensions or DID to automate delivery even more. In my application voip itself really doesn't factor in either, the fax setup is on the same box the analog lines physically terminate at. I have had pretty good luck with an old slow machine, ancient asterisk, low quality channel bank, and a physical fax modem on the same box as asterisk running hylafax, analog line in - pbx - analog line out - faxmodem, occasionally I get errors on faxes, and rarely someone can't get a fax through, but giving them the extension of a physical fax machine always works. So I am not convinced that problem is purely to blame on anything other than the far end station. What I would like to eliminate is the fxs port and physical faxmodem from the setup and use iaxmodem instead (frees up a port, plus doesn't need faxmodem at all, and less complicated) it sounds like this sort of configuration works pretty well according to most of the posters. I know there are some issues with fax autodetection, but normally the sender fax is programmed to retry a few times, and failing that, your answer message could include a message to hit start on the fax machine if it does not start automatically, or dial an extension manually to start it. another thing I like to do is if I scribble something down on a piece of paper, I just drop it in the fax machine and send it to the fax modem by calling its extension, I get a nicely scanned pdf in the mail that I can then forward to anyone without knowing their fax number or paying for a fax call, great for emailing diagrams of things without taking the time to draw them on the computer. Thanks, Steve Totaro www.asteriskhelpdesk.com The variables are very simple for any of these kind of decisions. Don't think about savings, think about costs. Costs of equipment Costs of time (resources) implementing Costs of maintenance Costs of losing data (faxes in this case) Costs of going back and doing it the right way if you find the above costs are higher than another solution. Thanks, Steve Totaro http://www.asteriskhelpdesk.com KB3OPB ___ --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 Jon Pounder _/_/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/ _/ _/_/_/ _/ _/_/ _/_/ _/_/ _/ _/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/_/ _/_/_/_/ Inline Internet Systems Inc. Thorold, Ontario, Canada Tools to Power Your e-Business Solutions www.inline.net www.ihtml.com www.ihtmlmerchant.com www.opayc.com This message was sent using IMP, the Internet Messaging Program. ___ --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: Bottom line on fax reception
Let me further qualify my results. This was done with whatever the current stable versions of Asterisk, Hylafax, and IAXmodem were available in January of this year. The faxes were outbound. PDFs put into a Samba share and a cron job moving them over to the Hylafax monitored directory. for my application I am more concerned with inbound working, outbound is just a bonus if it works. one of the big points is when you have a distributed workforce conventional fax machines don't work out since you get a paper result in one place and the recipient in another. Hylafax output can easily be redirected from a general delivery mailbox, or people can have their own fax extensions or DID to automate delivery even more. In my application voip itself really doesn't factor in either, the fax setup is on the same box the analog lines physically terminate at. I have had pretty good luck with an old slow machine, ancient asterisk, low quality channel bank, and a physical fax modem on the same box as asterisk running hylafax, analog line in - pbx - analog line out - faxmodem, occasionally I get errors on faxes, and rarely someone can't get a fax through, but giving them the extension of a physical fax machine always works. So I am not convinced that problem is purely to blame on anything other than the far end station. What I would like to eliminate is the fxs port and physical faxmodem from the setup and use iaxmodem instead (frees up a port, plus doesn't need faxmodem at all, and less complicated) it sounds like this sort of configuration works pretty well according to most of the posters. I know there are some issues with fax autodetection, but normally the sender fax is programmed to retry a few times, and failing that, your answer message could include a message to hit start on the fax machine if it does not start automatically, or dial an extension manually to start it. another thing I like to do is if I scribble something down on a piece of paper, I just drop it in the fax machine and send it to the fax modem by calling its extension, I get a nicely scanned pdf in the mail that I can then forward to anyone without knowing their fax number or paying for a fax call, great for emailing diagrams of things without taking the time to draw them on the computer. Yes, I suppose the thread title is reception. I am pretty sure the PDF decoding or encoding is what eats up the processor cycles, tiff would probably be much less processing. The poor man's scanner option is also pretty nice. Depending on the fax machine, it could be on par with a Panafax which is a costly little and awesome piece of equipment. Thanks, Steve ___ --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] [1.2.18] Wrong steps in extensions.conf?
Hello, Sometimes, when a call comes in from the PSTN through our VoIP gateway, the information that is sent to our web page that logs calls includes the original CID name instead of the one that is we expect to be rewritten on the fly using Asterisk's LookupCIDName: = ;extensions.conf [internal] exten = group,1,LookupCIDName exten = group,n,AGI(web.agi|${CALLERID(num)}|${CALLERID(name)}) exten = group,n,Dial(${EXT204}) = ;/var/lib/asterisk/agi-bin/web.agi #!/usr/bin/perl #use LWP::Simple; use URI::Escape; use LWP 5.64; open STDOUT, '/dev/null'; #Causes double entry? fork and exit; my $cidnum = $ARGV[0]; my $cidname = $ARGV[1]; $safe_cidname = uri_escape($cidname); my $browser = LWP::UserAgent-new; my $url = http://www.acme.com/input.php?;; $url .= name= . $safe_cidname . ; $url .= number= . $cidnum . ; ($min, $hrs, $day, $month, $year) = (localtime) [1,2,3,4,5]; $currentdate = sprintf(%02d/%02d/%02d, $day, $month+1, $year % 100); $currenttime = sprintf(%02d:%02d, $hrs,$min); $url .= date= . $currentdate . ; $url .= time= . $currenttime; #print $url . \n; my $response = $browser-get( $url ); die Can't get $url -- , $response-status_line unless $response-is_success; print $response-content; = Could it be that, sometimes, Asterisk doesn't wait for the previous step to be completed before moving on to the next? Thank you. ___ --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] ekiga register problems
returning newbie. Trying to register ekiga for the first time to my asterisk server only. [204] user=204 context=internal type=friend secret=xxx insecure=very canreinvite=no host=dynamic disallow=all allow=ulaw allow=alaw nat=no Can anyone tell me what I am missing? I am not behind NAT or a firewall ___ --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] Alcatel - Asterisk setup
Hi all: We are looking for someone with experience in Alcatel PBX - PRI - Asterisk integration Please get in touch off list.. We're wanting to hire a professional subcontractor, developer or company to get around some issues like these: Asterisk shows PRI to Alcatel is up, but when trying to dial from Alcatel to Asterisk results in a disc tone (Asterisk do send calls properly into Alcatel) If / when we manage to get anything from Alcatel, we get just the first digit of the number the user is intending to call.. Asterisk expects the whole number at once, so it fails.. Most of the time we get nothing at all from Alcatel, we think something is missing, so Alcatel sees the link is down. Please let me know if you have done this type of work before. We are not wanting to involve the Alcatel people, unless really required. Is there any special way to set up zaptel/zapata so Alcatel detects the PRI to be operational? Is there any special way to receive the calls once the PRI is up? Right now asterisk is set with: pri_net Any information or hints will be greatly appreciated Thank you, Carlos NZ ___ --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] Alcatel - Asterisk setup
Hi, You need to enable overlapdial. Regards, Sahil Gupta Chief Executive Officer VoiceValley Group of Companies Phone: +61-7-30188403 Fax: +61-7-30188499 On Tue, 29 May 2007, Carlos Hernandez wrote: Hi all: We are looking for someone with experience in Alcatel PBX - PRI - Asterisk integration Please get in touch off list.. We're wanting to hire a professional subcontractor, developer or company to get around some issues like these: Asterisk shows PRI to Alcatel is up, but when trying to dial from Alcatel to Asterisk results in a disc tone (Asterisk do send calls properly into Alcatel) If / when we manage to get anything from Alcatel, we get just the first digit of the number the user is intending to call.. Asterisk expects the whole number at once, so it fails.. Most of the time we get nothing at all from Alcatel, we think something is missing, so Alcatel sees the link is down. Please let me know if you have done this type of work before. We are not wanting to involve the Alcatel people, unless really required. Is there any special way to set up zaptel/zapata so Alcatel detects the PRI to be operational? Is there any special way to receive the calls once the PRI is up? Right now asterisk is set with: pri_net Any information or hints will be greatly appreciated Thank you, Carlos NZ ___ --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] language in zapata.conf [was: Language in Zaptel.conf]
On Mon, May 28, 2007 at 06:08:38PM -0500, Carlos Chavez wrote: On Tue, 2007-05-29 at 00:53 +0300, Tzafrir Cohen wrote: On Tue, May 29, 2007 at 12:34:33AM +0300, Tzafrir Cohen wrote: On Mon, May 28, 2007 at 04:08:35PM -0500, Carlos Chavez wrote: As you can see I set the language=es parameter (and do this for all interfaces). I installed the spanish sound set for Asterisk in /var/lib/asterisk/sounds/es (with links to the appropriate directories for letter, digits, phonetic, etc). All Zap interfaces still play all sounds in English. Only if I do a Set(CHANNEL(language)=es) do I get the sounds in Spanish. When I do a zap show channels the language column is blank. Duh. It seems that the value of language in zapata.conf is indeed ignored. My bad. Writing a fix. And sadly I was looking at an obsolete copy of the SVN. That has already been fixed in the SVN after 1.4.4 was released: http://bugs.digium.com/view.php?id=9626 http://svn.digium.com/view/asterisk?rev=62331view=rev (reported by sergee and fixed by russel) Well. At least I fixed the title of the thread... So is this fixed by using Asterisk SVN or Zaptel SVN? Asterisk SVN. Or by applying the above small fix to 1.4.4 . -- 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] ekiga register problems
On Mon, May 28, 2007 at 10:45:25PM -0400, Brad Sumrall wrote: returning newbie. Trying to register ekiga for the first time to my asterisk server only. [204] user=204 context=internal type=friend secret=xxx insecure=very canreinvite=no host=dynamic disallow=all allow=ulaw allow=alaw nat=no Can anyone tell me what I am missing? I am not behind NAT or a firewall What exactly is the problem you get? What is the line for 204 in 'sip show peers'? -- 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] ekiga register problems
204/20466.176.193.46D 5063 Unmonitored It just came up after a reboot on its own??? Go figure, windows problem! Thank you -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: Tuesday, May 29, 2007 12:17 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] ekiga register problems On Mon, May 28, 2007 at 10:45:25PM -0400, Brad Sumrall wrote: returning newbie. Trying to register ekiga for the first time to my asterisk server only. [204] user=204 context=internal type=friend secret=xxx insecure=very canreinvite=no host=dynamic disallow=all allow=ulaw allow=alaw nat=no Can anyone tell me what I am missing? I am not behind NAT or a firewall What exactly is the problem you get? What is the line for 204 in 'sip show peers'? -- 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 ___ --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