[asterisk-users] Newbie Planning Help

Gleim, Jason jgleim at ats-ohio.com
Wed Feb 28 15:50:28 MST 2007


Answers in-line...

Hope this helps!
Jason

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Alan
Chandler
Sent: Wednesday, February 28, 2007 3:46 PM
To: asterisk-users at lists.digium.com
Subject: [asterisk-users] Newbie Planning Help

<snip>
            ------                             -------

a) to what extent Asterisk can manage everything necessary to allow 
machines A and B to communicate if they were SIP phones.  Is it 
possible to go for a setup with the firewalls/NAT devices as shown

	- Asterisk can register and manage both A & B even though they
are behind NAT devices. NAT=yes is required, of course, for Asterisk and
the endpoint to properly communicate. You probably know but just in
case, SIP endpoints maintain a signaling channel through port 5060. When
a call comes in, they open a RTP media stream somewhere between port
10000 and port 20000. NAT can sometimes mess this up and it usually
shows itself as one-way audio. IAX endpoints send signaling and media
over the same port so there is less risk in NAT problems.

b) if I go with IAX softphones, does communication between A and B have 
to go through S, or can Asterisk "hand-off" the IAX conversation so 
that A and B talk directly.

	- I do not believe IAX allows for a hand-off between the two
endpoints. Most people don't want the hand-off anyway as it prevents the
parties from using in-call feature codes. This is why most everyone sets
canreinvite=no for SIP endpoints.

c) the example documentation shows seperate entries in iax.conf for 
incoming and outgoing calls.  In my case (assuming IAX softphones) 
would I just have entries for A and B of type friend?

	- yes. 'friend' is you friend for IAX softphones!

Can someone give me some advice about how to proceed.

Thanks


-- 
Alan Chandler
http://www.chandlerfamily.org.uk
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