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Mike at Trest.COM
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 Posted: Sat May 03, 2008 2:02 pm Post subject: [asterisk-users] RTP and Sip Provider |
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HI,
In sip.conf you need only the call SETUP ip address.
The RTP may come from anywhere . It is NOT assured that
it is just another port on the same IP address.
Therefore, be careful you do not block the RTP port ranges in a firewall.
Google voip-info for more information about RTP port ranges and Asterisk.
Exactly where the RTP is will be established by details passed on
the INVITE when a call is setup (either direction).
..mike..
At 02:48 PM 5/3/2008, you wrote:
| Quote: | Hello all,
I need to configure a new provider to complete
calls to us, the provider gave to me 2 different
ip address, one is the default host and another
to RTP server, so far as i knew the rtp server
should be the same address but different ports,
anyway i think i?m completelly wrong about it..
someone could tell me how can i configure in
asterisk this connection in sip.conf?
Thanks,
Chet
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miles.chet at gmail.com
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| Posted: Sat May 03, 2008 5:26 pm Post subject: [asterisk-users] RTP and Sip Provider |
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Mike,
Thanks....
On Sat, May 3, 2008 at 4:02 PM, Mike Trest - On Travel <Mike at trest.com>
wrote:
| Quote: | HI,
In sip.conf you need only the call SETUP ip address.
The RTP may come from anywhere . It is NOT assured that
it is just another port on the same IP address.
Therefore, be careful you do not block the RTP port ranges in a firewall.
Google voip-info for more information about RTP port ranges and Asterisk.
Exactly where the RTP is will be established by details passed on
the INVITE when a call is setup (either direction).
..mike..
At 02:48 PM 5/3/2008, you wrote:
| Quote: | Hello all,
I need to configure a new provider to complete
calls to us, the provider gave to me 2 different
ip address, one is the default host and another
to RTP server, so far as i knew the rtp server
should be the same address but different ports,
anyway i think i?m completelly wrong about it..
someone could tell me how can i configure in
asterisk this connection in sip.conf?
Thanks,
Chet
_______________________________________________
-- 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
|
_______________________________________________
-- 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
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