You need canreinvite=no in the config for your sip phone and the
veracity connection, otherwise Asterisk will just mediate the call setup
then try to allow the sip phone and veracity to talk directly to one
another.
Jim Dickenson wrote:
I have a SIP phone at home behind a NAT router
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Dropping RTP packets
You need canreinvite=no in the config for your sip phone and the
veracity connection, otherwise Asterisk will just mediate the call setup
then try
I have a SIP phone at home behind a NAT router registered with an * box at
my office with a routable static IP address running version
SVN-branch-1.6.0-r175638M.
If I make a call from my SIP phone out a PRI circuit to my cell phone
everything works as expected. I hear audio in both directions and
