> 2009/7/21 Matt Florell <[email protected]> > > On 7/21/09, Jose Arias <[email protected]> wrote: > > > Hi all, > > > > > > I'm planning to develop a custom autodialer application which will be > > > dealing with its own model for agents and queues, therefore it won't use > > > neither asterisk agents nor asterisk queues, nor asterisk cdr. The > > > application will supply the whole reporting and agent managing features > by > > > itself. > > > > > > The application will command asterisk through an AMI telnet connection > using > > > only the originate, redirect and hangup AMI commands plus the stream > file > > > AGI command (AsyncAGI patch will be required). > > > > > > The application will make outbound calls, then they will be redirected > on > > > the fly to dynamically defined meetme rooms, then the application will > call > > > extensions (registered endpoints) where it will know there are available > > > agents in order to redirect them to the previous meetme rooms. If the > > > application launched more calls than available agents it would play > prompts > > > while waiting for agents to become available. > > > > > > Since the planned features set from asterisk to be used by the > application > > > will be very short, but the figures can be very large (in terms of > > > concurrent calls, registered endpoints, traffic on the AMI port, etc..) > I > > > would appreciate if anybody can help me to find out what's the more > suitable > > > asterisk version to use in terms of scalability and stability: > > > > > > - concurrent registered endpoints (SIP and IAX) > > > - concurrent two and tree party meetme rooms (whatever codec can be > used) > > > - concurrent mixmonitor recordings > > > - concurrent playings for prompts > > > - commands and events rate on the AMI port > > > > > > It's important to notice the advanced features from asterisk aren't a > > > priority. > > > > > > I already looked over some links like > > > > http://www.voip-info.org/wiki/view/Asterisk+dimensioning > > > and others but I found more questions than answers there. > > > > > > Thanks in advance > > > Jose > > > > > > > This sounds a lot like ViciDial, which does use meetme instead of > > Asterisk Queues/Agents, is already engineered to be multi-server, is > > capable of placing 200,000+ outbound calls per server per day, has a > > web-based GUI for configuring the system and a web-based agent > > interface. > > > > > > > > - concurrent registered endpoints (SIP and IAX) > > > > Doesn't really matter, we've done 500+ on a single server before and > > it didn't really affect load much. As for number of agents, we are > > usually conservative on that front, usually we keep it under 50 agents > > per outbound server, but we have done 100 before. > > > > > > - concurrent two and tree party meetme rooms (whatever codec can be used) > > > > Everything is transcoded in a meetme room to slin. ViciDial does > > everything in Meetme, and while it does use slightly more resources > > than Asterisk Queues, it is more stable and offers more flexibility > > > > - concurrent mixmonitor recordings > > > > We do not recommend using mxmonitor. It is better to have a custom > > recording handling script. And if you are using Meetme for everything > > you don't have to bother mixing recordings anyway. > > > > - concurrent playings for prompts > > > > This depends on a lot of different things, if load or playback quality > > becomes an issue then you should put prompts on a RAM drive or tmpfs > > > > > > - commands and events rate on the AMI port > > > > Use a single point(or a few limited points) of entry to the AMI to > > keep it working well. You should not have an AMI connection for each > > agent. > > > > > > We currently use a version of 1.4.21.2 that has about 8 patches > > applied to it, and we have found it to be very stable in production. > > > > MATT--- On 7/21/09, Jose Arias <[email protected]> wrote: > Many thanks Matt, > > I heard asterisk had some problems with registering over 100 SIP endpoints > and I was worried about how much the transcoding load could be for over 100 > concurrents calls too. I expect to be over these figures. Regarding the AMI > connection, yes, there will be only one, like any third-party cti-link but > my concern was about how many commands an events asterisk is able to handle > without becoming in a bottleneck. > > You said you're using about 8 patches. Are all of them to make sure the > stability and scalability of the system? Well, one of them is the AsyncAGI > patch, isn't? Is there anyone to mach originate commands with new_channel > events? > > I'm planning to use asterisk 1.4.18 > > Regards > Jose > Hello,
We don't use AsyncAGI at all, the patches to Asterisk are mostly for issues with waitforsilence, chan_sip, AGI defunct channels and gsm codec. We also add patches for changed meetme enter/leave sounds and Sangoma CPD SIP message processing. We match Originates to new channel using code in our AMI listener app. MATT--- _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
