Hi all,
i have a beginners question. How are SIP calls and IAX2 calls processed by
Asterisk over the network?
What i mean is, is there a permanent connection required between the Asterisk
Server and the clients or is the Asterisk Server only involved for lets call it
the routing?
From my
The variable is *canreinvite.*
*Please check on voipinfo. If canreinvite is enabled then only SIP signaling
is passed through Asterisk and the media is not passed through Asterisk
resulting in less bandwidth usage and probably less jitter buffer, etcif
you are two phones are closer to each
Subject: Re: [asterisk-users] General network question regarding SIP and IAX2
The variable is canreinvite.
Please check on voipinfo. If canreinvite is enabled then only SIP signaling is
passed through Asterisk and the media is not passed through Asterisk resulting
in less bandwidth usage
Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Fri, Jul 9, 2010 8:13 pm
Subject: Re: [asterisk-users] General network question regarding SIP and
IAX2
The variable is *canreinvite.*
*Please check on voipinfo. If canreinvite is enabled then only SIP
