On Tue, Jan 15, 2019, at 1:17 PM, Brian J. Murrell wrote:
> On Tue, 2019-01-15 at 12:01 -0500, Joshua C. Colp wrote:
> >
> > The chan_sip module has this implemented under the "nat" option using
> > "comedia" as I recall.
>
> Yeah. The help for which reads:
>
> Send media to the port Asterisk
On Tue, 2019-01-15 at 12:01 -0500, Joshua C. Colp wrote:
>
> The chan_sip module has this implemented under the "nat" option using
> "comedia" as I recall.
Yeah. The help for which reads:
Send media to the port Asterisk received it from regardless
of where the SDP says to send it.
> It causes
On Tue, Jan 15, 2019, at 12:18 PM, Brian J. Murrell wrote:
> On Tue, 2019-01-15 at 09:00 -0700, John Kiniston wrote:
> > How is your endpoint currently configured in asterisk?
>
> It's configured as a chan_sip peer.
>
> > Have you tried
> > rtp_symmetric to see if the endpoint sends audio to
On Tue, 2019-01-15 at 09:00 -0700, John Kiniston wrote:
> How is your endpoint currently configured in asterisk?
It's configured as a chan_sip peer.
> Have you tried
> rtp_symmetric to see if the endpoint sends audio to asterisk if
> asterisk
> can send audio back to the client?
That would
How is your endpoint currently configured in asterisk? Have you tried
rtp_symmetric to see if the endpoint sends audio to asterisk if asterisk
can send audio back to the client?
Alternatively if your SIP Proxy is also a Media proxy you could set the
media_address on the endpoint to be your proxy
This is going to be a bit of an odd situation, but perhaps might become
more and more common (as mobile phone SIP clients utilize PUSH proxies
instead of the battery draining direct registering with SIP servers).
I have a SIP client which can be on the same RFC-1918 LAN as my
Asterisk server.
