[Asterisk-Users] Several questions. Please help

[Irakli Natsvlishvili]

Hello,

Question #1:

I have * with g729 installed, and two phones - Cisco 7960 and Cisco 7905.

If g729 is the only available codec for 7905's configuration, then call from
7960 to 7905 goes without any problem and both phones use g729.
But if I call from 7905 to 7960 the following is displayed on * console:

WARNING[5220]: rtp.c:1545 ast_rtp_bridge: codec0 = 256 is not codec1 = 4,
cannot native bridge.

And * does transcoding from g729 to g711.

Both phones have reinvite turned on.

Why everything works only way and does not work other way?

Question #2:

What approach should be used to have an * as a MoH server?

For example, I want to have 100 simultaneous SIP calls. The only destination
of SIP calls are MoH. In a hypotactic scenario could be a case when each
call requests deferent file, or 50 calls request the same file, 20 calls -
another file and the rest - each individual files.

So question is following - if I want to use Asterisk for this purpose, on
what should I focus to? If all files are on the same server where * is
installed, then in which format they should be stored, if a) only g711 codec
is used and b) speech is 90% of each individual file? What player should I
use for this purpose?
Where do you see a resource bottleneck - CPU, disk system or something
different?

If having sound files on Asterisk server is a bad idea, where should be they
stored?

Question #3

How do I see ongoing transcoding session done by Asterisk from CLI?

Question #4

How do I configure the following situation:

Call comes in extension 555. While extension 555 is ringing extension 444
picks up. 555 continues ringing until someone picks up and in this moment
call is automatically transferred from 444 to 555.

Thanks.