o Juha Heinanen on 09/19/2011 08:39 PM:
when sems receives voice call invite from ua behind nat that has
direction:active in its sdp, like this:
Session Description Protocol
Session Description Protocol Version (v): 0
Owner/Creator, Session Id (o): foo.bar 0 0
Stefan Sayer writes:
The latter. The direction attribute from comedia draft is a media
level attribute. If it is set in the video stream only, it applies
only to the video stream. fix_nated_sdp should put it in the audio m
section as well.
stefan,
thanks for your reply. i fixed
Hello,
On 9/19/11 9:47 PM, Stefan Sayer wrote:
o Juha Heinanen on 09/19/2011 08:39 PM:
when sems receives voice call invite from ua behind nat that has
direction:active in its sdp, like this:
Session Description Protocol
Session Description Protocol Version (v): 0
Hello,
On 9/20/11 9:19 AM, Juha Heinanen wrote:
Stefan Sayer writes:
The latter. The direction attribute from comedia draft is a media
level attribute. If it is set in the video stream only, it applies
only to the video stream. fix_nated_sdp should put it in the audio m
section as well.
On 9/16/11 3:40 PM, Klaus Darilion wrote:
IIRC (there was a similar thread years ago): yes
this is true indeed -- just wanted to add that the other logical
operators don't evaluate right side of the expression if the overall
result is known from the left side, for example, if left side is
Hi,
Since we installed last version of Kamailio (3.1.4) we have been
experiencing a big problem with REGISTER retransmissions. When the
server receives a retransmitted REGISTER it removes the binding and the
UAC remains unregistered until next refreshing period. I include a PCAP
capture
Hello,
On 9/16/11 2:46 PM, Andreas Granig wrote:
Hi,
On 09/16/2011 09:10 AM, Daniel-Constantin Mierla wrote:
It's actually an educated guess that this could be related to wt_timer,
but I don't know what else it could be.
what happens is that when transaction is active and tm is handling the
Hello,
On 9/16/11 3:19 PM, Andreas Granig wrote:
Me again,
On 09/16/2011 02:46 PM, Andreas Granig wrote:
And this is what I'd need to add if I got you right:
# the default reply route used when transaction is already gone
onreply_route
{
if(reply from inside)
Hi everyone,
I'd like to check that a client certificat is revoked or not against a crl.
Actually, opensips use context SSL_CTX. How can I do with this context?
I do this change to load the crl :
load_crl(SSL_CTX * ctx, char *filename)
{
LM_DBG(entered load crl\n);
X509_STORE
Hello,
On 9/19/11 5:54 PM, tomsc wrote:
Hi everyone,
I'd like to check that a client certificat is revoked or not against a crl.
Actually, opensips use context SSL_CTX. How can I do with this context?
I do this change to load the crl :
load_crl(SSL_CTX * ctx, char *filename)
{
Hello,
On 9/20/11 10:13 AM, Alejandro Mingo wrote:
Hi,
Since we installed last version of Kamailio (3.1.4)
the last version is now 3.1.5, but this is not really the most important
aspect. However, it is recommended to upgrade, there were some fixed to
registration handling when register
we have been experiencing a big problem with REGISTER retransmissions.
When the server receives a retransmitted REGISTER it removes the
binding and the UAC remains unregistered until next refreshing
period.
have you tried calling t_newtran() on register request?
-- juha
I didn't get any syslog messages by defining WITH_DEBUG. I've set
log_stderror = NO and now I get DEBUG messages. I'll keep
monitoring...
Thanks.
El 20/09/2011 10:54, Daniel-Constantin Mierla escribi:
Hello,
On 9/20/11 10:13 AM,
Hey Timo
Thanks for your email.
Yes dlg_manage(); has to now be called on INVITE and BYE/CANCEL messages.
Where would i have to call loose_route()? Only on INVITE?
My configuration did not change between 3.1.2 and 3.1.5.
Call flow example:
==
Cisco PGW === Kamailio 3.1.5 === VOIP
Hey Phillip,
On 20.09.2011 13:48, Phillman25 Kyriacou wrote:
Thanks for your email.
Yes dlg_manage(); has to now be called on INVITE and BYE/CANCEL messages.
Where would i have to call loose_route()? Only on INVITE?
On *all* in-dialog requests, i.e., all requests which contain a To tag.
Hi,
On 09/20/2011 10:16 AM, Daniel-Constantin Mierla wrote:
So is there actually a limitation on when I'm allowed to call
force_send_socket()? Any way to force force_send_socket()? :)
are you using a recent version of 3.1 branch? This was fixed with:
Hey Timo
Thanks for your email.
I apologise i never copied the config properly. I missed a } to close the
if statement.
You can see that the route(WITHINDLG); is called for all requests from this
config.
# MANAGE ALL DIALOGS
#===
Hey,
On 20.09.2011 15:23, Phillman25 Kyriacou wrote:
Hey Timo
Thanks for your email.
I apologise i never copied the config properly. I missed a } to close
the if statement.
You can see that the route(WITHINDLG); is called for all requests from
this config.
# MANAGE ALL
Hi Morten,
The method is not SIP/2.0. You should try the used method (e.g.
REGISTER or INVITE).
Hope that helps, everything else looks allright; if it is still not
okay, you could compare it to the rfc2617.c of modules/auth-Module:
19 matches
Mail list logo
