Closed GoogleCodeExporter closed 9 years ago
With "registration uri" empty the app will not register at all to the sip
server. So by removing the registration uri you might have hidden some initial
problem.
Then about the 408 it's usually about a connectivity problem. So my first idea
is to suspect that what's returned by the dns srv resolution points to an
invalid ip/port that cannot be reached from the network you are connected to.
Without more details about your sip server domain so that dns config can be
checked it will be hard to help, but I can confirm that dns srv feature is
working properly with properly configured dns srv domains.
Original comment by r3gis...@gmail.com
on 22 Oct 2013 at 7:48
Hello, I have two opensips proxy servers. I tested with Bria and Zoiper and all
is working whitout problems. I see that csipsimple add to sip message "Route"
option...
I use proxy uri in this case
Thanks
Original comment by ultrab...@gmail.com
on 22 Oct 2013 at 8:41
If you don't want pjsip (the stack behind csipsimple) to add the route header,
add ";hide" at the end of the proxy uri (without the double quotes).
If your sip server fails when sip client indicates the route it's probably
because your sip server does not resolved dns-srv and realize it's himself.
(and probably as consequence it fails to re-route the reply to the sip client,
and that's why you get a timeout). This header is in standard and sip server
properly configured should not be lost if the route header is added.
Original comment by r3gis...@gmail.com
on 22 Oct 2013 at 10:32
Hello again, all is working fine with ";hide" at the end of the proxy uri
Thanks
Original comment by ultrab...@gmail.com
on 22 Oct 2013 at 4:32
great :-)
I sill advise you to check your config on server side however. CSipSimple
relies on pjsip stack which is one reused in many other projects. some of the
other projects does not gives as many tweak options than CSipSimple. So you
might find this problem impossible to solve with other sip clients.
Also pjsip stack was announced to be the sip stack of next version of Asterisk
servers, so in future you might find also problems if you do trunking with
other servers (pjsip is a very serious, robust and rfc strict implem of sip)
Original comment by r3gis...@gmail.com
on 22 Oct 2013 at 9:35
Original issue reported on code.google.com by
ultrab...@gmail.com
on 21 Oct 2013 at 11:08