Hi Carsten,
Thanks, that solved the problem.
/Morten
On Tue, Sep 20, 2011 at 10:03 PM, Carsten Bock wrote:
> 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
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:
http://git.sip-router.org/cgi-bin/gitweb.cgi?p=sip-router;a=blo
Hi,
Sorry for the slightly off topic question, but I am trying to debug a
buggy PBX and want to check if the digest information is correct.
I have this perl script.
#!/usr/bin/perl
use strict;
use Digest::MD5 qw(md5 md5_hex md5_base64);
my $uri = 'sip:sip1.uni-tel.dk';
my $method = "SIP/2.0";
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.
>
>
>
># MANA
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
#===
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:
>
> http://git.sip-router.org/cgi-bin/
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.
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
I didn't get any syslog messages by defining WITH_DEBUG. I've setÂ
log_stderror = NO and now I get DEBUG messages. I'll keep
monitoring...
Thanks.
El 20/09/2011 10:54, Daniel-Constantin Mierla escribió:
Hello,
On 9/20/11 10:13 AM, Al
> 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
___
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 re
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)
{
LM_DBG(
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 9/16/11 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)
force_send_socket(localho
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
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 that
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 tru
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.
stefa
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
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 fix_n
Hello,
something happened while initialization of the proxy. Do you get a core
file on the disk?
What is route table and what module is using it? AFAIK is not a default
table for existing modules. Note that dbtext engine is very limited in
supporting various data types or working with large
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