[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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