Re: [asterisk-users] twenty thousands (20, 000) users, which asterisk and how many servers
you can find more details @AsteriskSCF project. On Wed, May 23, 2012 at 11:16 PM, bilal ghayyad bilmar...@yahoo.com wrote: Dear; So it is a hardware issue and not software? I am afraid that asterisk software it self is not able to support 20 000 users and 2000 concurrent calls. About the high availability: is there a method that if the first asterisk server down, then the call will stay connected and failover to second asterisk server? Regards Bilal -- 20.000 users is really a big number, as big as 2000 concurrent calls. As previously stated on this list, it depends... it depends by the type of calls for example. If all media is offloaded from the server letting the phones to reinvite each other, than your server CAN support the call volume. If instead even a tiny portion of the call volume uses service on the pbx, like IVR, music on hold, conferences, queues or even worst, transcoding, then the server is obviously underpowered. From my point of view, servicing 20.000 users with a single piece of hardware is highly risky. It can broke in the middle of the day, leaving all your users without service. I think a better approach will be to have more less powered servers working all together to serving your users. If a day one or two of them broke, you have not to worry because the other will continue to serve your users and nobody notice the little decrease in power. There are a lots of way to achieve the high availability, load sharing, each with its pros and cons. Right now I am building a pbx with high availability and load sharing in mind, for a client who wants to achieve numbers you have just said. Let's see how it works in few months. Leandro 2012/5/23 bilal ghayyad bilmar...@yahoo.com Hi All; I need to use Asterisk for 20 000 users, so which asterisk version to be used? Is there asterisk version that supports 20,000 users on one hardware machine? Can I use one strong hardware server i7 with 64 GB RAM and fast hard desk to handle 20 000 users, and concurrent calls 2000? Or I need multiple servers, how much? If I am going to use multiple servers (until now I do not know how much, and I do not know if the barrier will be the asterisk software or the hardware), then do I have to use special SIP proxy or I have to use load balancer)? In this case, I have to use asterisk Database (so all the servers will read/write from the database)? What about AsteriskNow, can it support? Regards Bilal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Regards, Ali R. Taleghani -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Playing Multiple File ( simultaneously ) on Channel
Yep, But I think that is can be done, independent to Queue app. If we have an application which call MusicOnHold inside! maybe we can control simultaneously playing back of files... if we have such ability then it can be replaced by what exist inside the Queue. or maybe an application which written from scratch... can be help full. On Fri, Apr 27, 2012 at 7:39 PM, Kevin P. Fleming kpflem...@digium.comwrote: On 04/27/2012 09:01 AM, shayne.al...@gmail.com wrote: I am interested to know if there is any application or way to help me for this Scenario: When we put Callers in Q, the MOH will stop for announcements! how if we able to increase the MOH (RT/TX) and then play any announce with greater RX/TX ( and there so louder ) on the channel! without stoping MOH? This is an interesting idea, but at this time Asterisk's app_queue has no ability to do what you are asking for. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- __**__**_ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- Regards, Ali R. Taleghani -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Playing Multiple File ( simultaneously ) on Channel
so nice! it's a good idea. i will try it... tnx On Fri, Apr 27, 2012 at 8:35 PM, Leif Madsen leif.mad...@asteriskdocs.orgwrote: On 27/04/12 11:57 AM, shayne.al...@gmail.com wrote: Yep, But I think that is can be done, independent to Queue app. If we have an application which call MusicOnHold inside! maybe we can control simultaneously playing back of files... if we have such ability then it can be replaced by what exist inside the Queue. or maybe an application which written from scratch... can be help full. I guess you could get crazy and start injecting audio onto the channel through the use of ChanSpy() in whisper mode. It could probably be triggered through AMI to execute a Local channel which then connects to the channel you want to play the audio back onto, and then whisper that audio over top of whatever playback is happening. Check the Audio Manipulation section of the Asterisk Cookbook for some simple examples here: http://ofps.oreilly.com/**titles/9781449303822/c03-* *AudioManipulation_id302347.**htmlhttp://ofps.oreilly.com/titles/9781449303822/c03-AudioManipulation_id302347.html I also talked a bit about injecting audio onto a channel at AstriCon 2011 in my Cooking With Asterisk talk. It's the last recipe I talk about in this video: http://www.astricon.net/**videos/Cooking-with-Asterisk.**htmlhttp://www.astricon.net/videos/Cooking-with-Asterisk.html -- Leif Madsen http://www.oreilly.com/**catalog/asteriskhttp://www.oreilly.com/catalog/asterisk -- __**__**_ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- Regards, Ali R. Taleghani -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Playing Multiple File ( simultaneously ) on Channel
:) sorry , cos i am not native english... On Fri, Apr 27, 2012 at 8:38 PM, Kevin P. Fleming kpflem...@digium.comwrote: On 04/27/2012 10:57 AM, shayne.al...@gmail.com wrote: Yep, But I think that is can be done, independent to Queue app. If we have an application which call MusicOnHold inside! maybe we can control simultaneously playing back of files... if we have such ability then it can be replaced by what exist inside the Queue. or maybe an application which written from scratch... can be help full. It's hard to parse what you are saying, but yes... it would be possible for someone to write code to do what you want to do. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- __**__**_ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- Regards, Ali R. Taleghani -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call recovery feature
this will be a wonderful feature, if be possible, with asterisk... as i looked back to this, i think you will find this on Asterisk-SCF. but if this be possible with Core-Asterisk! then this what i am looking back for a long.. On Thu, Apr 26, 2012 at 6:59 PM, Kristijan Vrban vrban.l...@googlemail.comwrote: Hello, what about: This feature means you can restart Asterisk after a failure (or asterisk restart itself with safe_asterisk), and keep existing calls up with only a few seconds of audio dropped. That would be a feature! there is a other pbx that has this feature... Anyone else would like to see that feature? just want to start some brainstorming, Kristijan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Regards, Ali R. Taleghani -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] update CDR fields after Queue
Dears;
I have been faced with a problem that I am not sure about how can I solve
it...
I my scenario there is a variable which will be ready just after the callee
had hanged up and the caller, which coming throw a Queue.
But the CDR fields are logged into DB just after the Queue application. so
the '*userfield' *field will remained Blank.
is there any way to suspend CDR write INTO DB untill the end of the whole
dialplan priorities?
*exten = research,1,Answer()*
*exten = research,n,Wait(3)*
*exten = research,n,Queue(research,tkc)*
*exten = research,n,Noop(${BRIDGEPEER:4:4})*
*exten = research,n,SayNumber(${BRIDGEPEER:4:4})*
*exten = research,n,read(dtmf,,1,,,10)*
*exten = research,n,Set(CDR(userfield)=${**dtmf}**)*
*exten = research,n,Hangup*
--
Regards,
Ali R. Taleghani
0936 322 4069
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[asterisk-users] Detect #,* DTMF in dialplan
Dear Mr,Ms; I am planing for a custom IVR, for example to act as a simple installer! I mean there is some choice via 0-9 and # as *Next* and * as *Back* button. is there any way for me to detect if the caller pressed # vs * on Dialplan ? -- Regards, Ali R. Taleghani 0936 322 4069 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Hardphone that works well with asterisk
zyxel On Sat, Feb 12, 2011 at 4:01 PM, ast guy ast...@gmail.com wrote: Hi, I have been out of touch with asterisk for quit some time and needed some recommendations. I am looking for SIP hardphone that works well with asterisk server. Pls suggest. cheers /ag -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Regards, Ali R. Taleghani 0936 322 4069 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] what are QUEUE_MAX_PENALTY and QUEUE_MIN_PENALTY veriables?
Dears; I am looking for a way to handle callers via queuerules, but am not able to exactly understand the meaning and affect of this two variables on Queue Application, and how it change the priority of a caller to be answered sooner. QUEUE_MAX_PENALTY QUEUE_MIN_PENALTY tnx -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to get Current Calls details
I think you are looking for a way to have such a report, on console: CallCenter*CLI sip show channelstats Peer Call ID Duration Recv: Pack Lost ( %) Jitter Send: Pack Lost ( %) Jitter 2.168.11.731ea659 00:03:39 00236 ( 0.00%) 0.000 004685 00 ( 0.00%) 0.0011 2.168.21.556897bd 00:38:11 01153 0105 ( 0.09%) 0.000 000113 000231 ( 0.20%) 0.0052 it's possible, as u see i have it on my console, but i didn't remember in which configuration part, I had enabled it... I hope someone remember it... On Thu, Feb 3, 2011 at 3:54 PM, Daniel Tryba dan...@tryba.nl wrote: On Wed, Feb 02, 2011 at 03:29:37PM +0530, Nikhil wrote: Thanks of reply. The command core show verbose is working. but the problem is, for one call we can see 2 results,there is no common field on these two. Take a closer look at the output. The link between the 2 can be found by matching channel==dstchannel for all channels. -- Daniel Tryba -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Regards, Ali R. Taleghani 0936 322 4069 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to Change The Caller Position in Queue
Dear Mr/Ms; web have some Queues and our Call Center and put caller in Queue Based on some regional decisions. by the way, after the Caller placed on Queues, we like to be able to reorder them on our rules. as an example: there is a queue which have 10 caller in waiting stage right now, one with the no:7 is VIP! so we need to change her place to no:2. ** again: i don't need to just make decision on incoming calls! I am about the callers which has been Queue before now! they are some where in the middle of the Queue. *what is the best way to do such a things, or alike...* -- Regards, Ali R. Taleghani 0936 322 4069 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
