On 6/13/13 16:20 , Matthew J. Roth wrote:
It's hard to be certain without seeing a full SIP trace, but I think the
INVITE
with the internal IP is actually a re-INVITE that Asterisk is sending to
establish a native bridge between the SIP friend and the SIP gateway to PSTN
converter.
It's
Andreas Sikkema wrote:
On 6/13/13 16:20 , Matthew J. Roth wrote:
It's hard to be certain without seeing a full SIP trace, but I think the
INVITE
with the internal IP is actually a re-INVITE that Asterisk is sending to
establish a native bridge between the SIP friend and the SIP gateway
Hello Matthew,
My version is Asterisk 1.6.2.9.
Or have you seen NAT? I have no NAT on my network. Have you seen my little
diagram above?
Here it is:
SIP friends (phones) - Asterisk - SIP gateway to PSTN converter
80.236.215.61109.69.217.6 internal IP (
On Thursday 13 June 2013, Mickael MONSIEUR wrote:
Hello Matthew,
My version is Asterisk 1.6.2.9.
Or have you seen NAT? I have no NAT on my network. Have you seen my little
diagram above?
Here it is:
SIP friends (phones) - Asterisk - SIP gateway to PSTN converter
Mickael MONSIEUR wrote:
My version is Asterisk 1.6.2.9.
Or have you seen NAT ? I have no NAT on my network . Have you seen my little
diagram above ?
Here it is:
SIP friends (phones) - Asterisk - SIP gateway to PSTN converter
80.236.215.61109.69.217.6 internal
You mean the SDP payload? You kind of need that
c= is used for RTP transmission. o= always confuses
me so I will just say it's important at well.
You can put a proxy in the middle and do topology
hiding I guess however, that is beyond the scope of
this list?
Kind Regards,
Nick.
On
Mickael MONSIEUR wrote:
I have a standard Asterisk configuration:
SIP friends (phones) - Asterisk - SIP gateway to PSTN converter
80.236.215.61109.69.217.6 internal IP (
10.4.0.10/255.255.255.0 )
When analyzing traffic on a SIP friend/phone I see this:
INVITE
