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
Re: [asterisk-users] Call someone to instantly join conference using MeetMe
Arpit Use Auto dial. http://www.voip-info.org/wiki-Asterisk+auto-dial+out Create a .call file as mentioned by Dave. Dave Miller wrote: Arpit Mehta wrote on 5/19/07 10:18 PM: I was just wondering how would the application be where the Asterisk calls a number and that number joins the conference as soon as the call connects. There would be only one conference already defined in meetme.conf and there is one person already joined the conference. Currently MeetMe requires a person dialing into it and the joining the conference. How could this be done using MeetMe or any other conference application? Any suggestions/hints/links are welcome. Set up an extension that dials directly into the conference in question, then use that extension via the Local channel as the source of a call to the number you want to dial, triggered via the Management API or a call file. [meetme-dialin] exten = 1234,1,Answer() exten = 1234,n,MeetMe(4321) Pipe the following into the Manager API with an extra blank line at the end: Action: Originate Channel: Local/[EMAIL PROTECTED] Context: from-inside (or whatever context is appropriate) Exten: (the number you want to call) Priority: 1 I'm going from memory, so you may have to play with it a little bit but that's the basic idea. ___ --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 On Solaris 10
Hi List Whats the best way to run * on Solaris 10 with x86 architecture. I am following solarisvoip.com using svn, but came across issues like 1. app_lookupcnam compilation issue - Wrong format of ELF. Is this the correct way. ___ --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] Please help me finding good A-Z provider
We do that. www.direct-internet.co.in. BTW whats your location and where is your major calling. Lee Jenkins wrote: VoIP User wrote: Hi Everyone, can you please recommend me a good VoIP provider as I am not satisfied by my current provider. Does not matter what protocol it uses. I'm looking for good rates, stable quality and not so big prepayment required. Thanks to all. I've had a great experience with www.axvoice.com. Good quality, great rates...been using them for about 7-8 months. ___ --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: Asterisk On Solaris 10
Any help is appreciated. Kapil Dhawan wrote: Hi List Whats the best way to run * on Solaris 10 with x86 architecture. I am following solarisvoip.com using svn, but came across issues like 1. app_lookupcnam compilation issue - Wrong format of ELF. Is this the correct way. ___ --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] RTP Mixer
Anybody I am still waiting. Kapil Dhawan wrote: Hi Can somebody brief me the working of RTP mixer from MeetMe perspective. ___ --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] RTP Mixer
I was reading an article on RTP Mixer so started studying about the mixing done by Asterisk in MeetMe. Read that CC should contain the no of participants ifupto 15 and CSRC should come, but not getting any by asterisk. Tzafrir Cohen wrote: On Sat, May 12, 2007 at 10:44:19AM +0530, Kapil Dhawan wrote: Hi Can somebody brief me the working of RTP mixer from MeetMe perspective. (RTP? in MeetMe? I don't really know this well but AFAIK RTP doesn't get mixed directly in MeetMe. MeetMe is a generic Asterisk channels mixer) Aparantly people either don't know enough or don't have the time. Try rephrasing your question so it will be more specific and thus also hopefully take shorter time to answer. Do you have a working system? Do you need to set up one? What version of Asterisk? What types of channels do you try to mix? ___ --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] Simultaneous Capacity
Hi List I want to try Asterisk with 10 PRI on a single Xeon machine with g711. Is it feasible. ___ --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] Simultaneous Capacity
Just a quick brief I have a requirement of running 10 PRI's (300 Channels). I still have to decide on hardware and cards. Can you suggest some. As per my understanding it will be tough to go beyond 150. Alex Balashov wrote: On Mon, 14 May 2007, Kapil Dhawan said something to this effect: I want to try Asterisk with 10 PRI on a single Xeon machine with g711. Is it feasible. In truth, it is very unlikely. How are you planning to pick up the PRIs, anyway? 3 quad-span T1 cards? -- Alex Balashov [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 ___ --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] RTP Mixer
Perfect Josh...but if i am running an application which has a capability of showing number or participants depending upon CC value, that doesn't work. Secondly, Asterisk can show on CLI about current talking users where it is maintaining talking status but not sending it down the line to be used by other apps. Anyways, i will go with your statement and leave it on core developers to comment. Joshua Colp wrote: Kapil Dhawan wrote: I was reading an article on RTP Mixer so started studying about the mixing done by Asterisk in MeetMe. Read that CC should contain the no of participants ifupto 15 and CSRC should come, but not getting any by asterisk. I'll just leave it at this so we can all move on with our lives: Asterisk isn't totally an RTP Mixer in the sense you are reading about. It is an audio mixer. Frame of audio comes in over RTP, gets sent in (only the audio portion) to be mixed, frame comes out, gets turned into RTP again. The RTP part has no idea that multiple sources were mixed together, 'nor should it care. The sources could have been Zaptel channels for example in which case they couldn't be added to the list. Joshua Colp Software Developer Digium, Inc. ___ --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] RTP Mixer
Hi Can somebody brief me the working of RTP mixer from MeetMe perspective. ___ --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] RTP Mixer
Hi Just an assumption. After packets reach Asterisk, it does the conversion into the required format and forwards it to Zaptel driver, which in turn combines and sends one RTP stream back to Asterisk. How can a client check about number of participants etc. ___ --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 PBX
Hi List I am very new to Asterisk but have been alloted a job to replace my traditional PBX with it. Kindly provide me some useful info (PDF's etc) to setup Asterisk with FXO and FXS both. I have to cater some 60 users with 10 simultaneous calls. Regards _ Biography of Shah Rukh. His profile, awards, films. http://server1.msn.co.in/Profile/shahrukh.asp Find more here! ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
