On 2012-05-22 at 12:40, Daniel-Constantin Mierla wrote:
> ok, done few time ago -- I kind of forgot about it.
Thanks!
--
Øyvind Kolbu
pgp27x2xYmzrC.pgp
Description: PGP signature
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
On 5/22/12 11:20 AM, Øyvind Kolbu wrote:
On 17.05.2012 18:16, Spencer Thomason wrote:
I just tested the patch and it works perfectly.
Daniel, can you backport to 3.2?
Hello,
ok, done few time ago -- I kind of forgot about it.
Cheers,
Daniel
--
Daniel-Constantin Mierla - http://www.asipto.
On 17.05.2012 18:16, Spencer Thomason wrote:
I just tested the patch and it works perfectly.
Daniel, can you backport to 3.2?
--
Øyvind
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://
Hi Daniel,
I just tested the patch and it works perfectly.
Thanks!
Spencer
On May 16, 2012, at 11:37 PM, Daniel-Constantin Mierla wrote:
> Hello,
>
> indeed, rtpproxy_manage() didn't handle UPDATE requests. I just pushed a
> patch in git master branch:
>
> http://git.sip-router.org/cgi-bin/
Hello,
indeed, rtpproxy_manage() didn't handle UPDATE requests. I just pushed a
patch in git master branch:
http://git.sip-router.org/cgi-bin/gitweb.cgi/sip-router/?a=commit;h=24ff0d9aa060d183fbe40b1fcb5910d60940585b
If you can test the patch and report the results, I will backport to 3.2
if
Hi Daniel,
I have updated my script to ensure these UPDATEs call route(NATMANAGE) but it
seems the problem is that rtpproxy_manage() does not handle UPDATEs. Since the
call is already passing through rtpproxy is there any way I can force these
UPDATEs to keep it there?
Thanks,
Spencer
On May
Hello,
be sure you call route(NATMANAGE) for UPDATE request and set an
onreply_route where the reply will be handled and you have to call there
route(NATMANAGE) as well.
Cheers,
Daniel
On 5/16/12 12:45 AM, Spencer Thomason wrote:
Hello,
I'm working on a residential type application where we
Hello,
I'm working on a residential type application where we are using Kamailio for
NAT traversal and Freeswitch as a voicemail and media server. When a UA that
is behind NAT sends an INVITE to check voicemail everything works correctly
until the user listens to the message. The sdp in the in
