On Friday 04 September 2015 10:49:57 Michael Nielsen wrote:
> Yes.
Than you should make packetcaptures on the rptproxy to see where the missing
rtp leg is going to. If it is send to the correct ip/port without any feedback
that that combo is unreachable the problem is at the receiving endpoint.
Yes.
On Fri, Sep 4, 2015 at 10:19 AM, Daniel Tryba wrote:
> On Thursday 03 September 2015 19:19:44 Michael Nielsen wrote:
> > Is it something to do with codec or could it be something else?
>
> Are you using an rtp proxy?
>
> ___
> SIP Express Router (
On Thursday 03 September 2015 19:19:44 Michael Nielsen wrote:
> Is it something to do with codec or could it be something else?
Are you using an rtp proxy?
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-ro
This is my setup:
Kamailio -> FreeSWITCH (for voicemail)
...SIP gateway for connecting GSM
So every call is routed directly via Kamailio - between SIP clients and
in/out from the GSM space.
Testing with a cellular phone as one client and X-Lite on Mac OS X as the
other - everything works.
Testi
