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Has anyone done any large scale intercom deployments with Asterisk? I've been asked about building a system to one-way page 500 phones simultaneously from a single server. My concerns are: - My limited math capabilities suggest 41 Mbps of RTP traffic, which seems like a lot, plus asterisk would be taking a single input stream and exploding it out to 500 endpoints. - There are 500

I have some lines outside the USA that I can't get to for up-close testing. The issue is that when I make an outbound call on these lines, I don't get answer indication until 20 seconds into the call. Asterisk always thinks the line is answered, even if it's still ringing or busy. On incoming calls the opposite is true, Asterisk doesn't know the line has been hung up by the

Two asterisk servers, one running a recent HEAD, the other 1.0.9. I have both ends set up with trunk=yes, notransfer=yes, type=friend. I notice that the trunking works from HEAD to 1.0.9 only (the direction in which calls are originated). I know this by bandwidth usage and by iax2 trunk debug. I did have to use trunktimestamps=no on the HEAD end to keep it quiet. I assume this is the new

Has anyone done any installations using the Xorcom TS-1 and multiple T1's? I'm looking for a reliable box to put in a closet and route calls between T1's. No voicemail, IVR, etc. Any other suggestions? Mark Willis Cartama

Has anyone done SLA with Polycom phones? I've got a large project coming up where the customer is keen on SLA for trunks and extensions. Trunks will be on a PRI. We may do this with Cisco phones if they work better. Mark Willis

Is it possible to do SIP<->Asterisk<->TDM in a single step with FFA? Or does FFA always use TIFF files? I'm using Free FFA on 1.6.2.15 and I want to be able to use SPA 2102 ATA's at the fax machines and send faxes directly over a PRI. Mark -- Mark Willis Star One Telecom Office: 1-800-889-7001 Cell: 210 880 5097 http://staronetel.com

Need some help understanding codec preferences: I have 2 asterisk servers. Server 1 sends calls to the PSTN and has allow=g729 allow=gsm and allow=ulaw in iax.conf Server 2 receives calls and routes them to server 1. It has the same allow lines. We receive calls from a phone co and route them via server 2 to server 1. The calls originate in g729 and everything works fine. Now I want to take