Hi!
Once again I want to discuss the problem of avpops in reply_routes (e.g.
http://www.openser.org/pipermail/users/2005-October/001281.html)
I can bypass the syntax parser by putting the AVP-commands into a normal
route block which is executed from the reply_route.
As far as I understand this may cause problems as the AVP-ops are done
without locking. But what may happen in worst case? Bad AVPs or a crash?
thanks
klaus
btw: is there any other way to handle data from a reply_route ?
Hello,
I'm using openser 1.1.0
I'd like to make a dinamic routing, the plan is radius set an avp, which
contains the routeid,
but my config is bad, and i don't know what can be the problem
route($avp(i:20)); don't work so i tryied this way:
switch($avp(i:20)) {
case 0:
route(0);
break;
case 1:
route(1);
break;
case 2:
route(2);
break;
}
but this is bad too, can anybody give me an advice?
the problem is in the line "switch($avp(i:20)) {" i guess,
but switch can accept pseudo vars, so i don't kow what the problem
caould be?
Thanks any help,
Tamas
I have a primary and secondary dns server setup in resolv.conf, with a timeout option of 1 second.The idea is when the primary fails, my secondary server picks up the workload.When openser is running, and primary dns is down, I am seeing several re-transmissions of 200 OK (invite) to my originator *after* openser has finished processing ACK. I was thinking the final responses would stop reTX after the ACK was t_relayed. Have you seen such behavior? And better yet- Is there a solution for this?Thanks, Will
hello,
using my fritzbox fon with an account from Carpo, ser
0.10.99, to call a cisco 7960 behind the same fritzbox
fon, that uses as well an Carpo account, an hang up,
the called cisco 7960 continues ringing for a minute.
(fritzbox - ser - (same) fritzbox - 7960)
Vice versa the same. (7960 - fritzbox - ser -
fritzbox)
Not so when I use the 7960 with a sipgate account
(7960 - sipgate - telefonica gateway - Carpo ser -
fritzbox) to call the fritzbox.
Is this a ser problem? Perhaps a problem, when the
call-phone and the called-phone use the same IP?
Or is this a configuration problem of ser?
What can I debug to find where the problem is?
___________________________________________________________
Telefonate ohne weitere Kosten vom PC zum PC: http://messenger.yahoo.de
Hi,
I was having a few issues with mediaproxy when it came to re-invites and
call setup where the SDP does not get sent with the first INVITE. I
noticed this was due a couple couple of assumptions in mediaproxy which
don't always hold:
1) That SDP is always set up first by the caller (i.e. in the INVITE)
then updated by the called party (i.e. in the 200). This is not the
case. The first SDP can be sent in the 200 response by the called party
and the caller send their SDP in the ACK.
2) That RTP streams don't change once set up. This is wrong for reINVITEs.
This would cause problems when a user agent sent a reINVITE to redirect
RTP to on hold music for example - the RTP change would be ignored. I
think this had an impact on some fax setups as well, but I haven't
tested it. I found a patch to fix this by storing the streams in a hash
by media type, but that is a limiting assumption also (maybe I want 2
audi streams?).
The attached patch fixes this by changing the following:
1) This pathc allows a mediaproxy RTPSession to be set up by either the
caller or called party. A 'caller' argument is passed to the RTPSession
functions telling it if the caller or called party is doing the setup or
update.
2) The official mediaproxy does not even look at SDP details for streams
which are already set up. This patch lets every SDP packet update all of
the RTP streams for that party (either caller or called).
If you have issues with one-way-audio with media proxy after reINVITEs,
have calls the first SDP from the called party or are just plain
adventurous, please give it a go. It has been used in a production
environment for a bit over a month now. The patch was made against
mediaproxy 1.7.2, but will work against 1.8.0 as well since none of the
rtphander.py code changed.
Regards,
Jeff
Hi,
I am having this problem with the mysql module on ser.
0(31212) ERROR: load_module: could not open module
</opt/ser/modules/mysql/mysql.so>: /opt/ser/modules/mysql/mysql.so:
undefined symbol: log
0(31212) parse error (30,13-14): failed to load module
ERROR: bad config file (1 errors)
I can confirm that mysql.so exists in that directory.
I have mysql 5.0.27 installed.
I have already tried to clean and make all from scratch but the problem
remained.
Any ideas how to solve this?
Thanks a lot,
Josef
Hi!
I want to use the dispatcher module to use load-balancing/failover to
gateways. Thus, I also want to change the domainpart of the RURI to be the
IP address of the gateway - but dispatcher only sets the DURI.
Is there an easy way to set the RURI/domain too or do I have to use:
avp_write("$rU","$avp(i:666)");
ds_select_dst(...);
avp_pushto("$ru","$du");
avp_pushto("$ru/username","$avp(i:666)");
Is there a more easy way?
btw - I think above workaround would not work in failure_route as
ds_next_dst adds a new branch but avp_pushto works against branches[0] -
correct?
thanks
klaus
