Hi Daniel,
Thanks for this.
I took the entire config files and configured it as per my ips and ports,
after doing that, still no call establishment(webrtc to classic sip phones
and vice-versa). Following is what i get in kamailio.log:
rtpp_test(): rtp proxy found, support for it enabled
ERROR:
Hello,
first, if you have modules_s and modules_k with folders from modules,
then you are using an old, no longer maintained at this time, kamailio
version (respective 3.x). You should be using 4.x, especially when
starting a new deployment or upgrading old one. In 4.x there is no
longer a mo
On 16/09/14 06:26, aawaise wrote:
What if in INVITE packet, callee's domain is domain1.
For example user1@domain1 > calls > user2@domain1
but user2 is registered with domain2. its aor is user2@domain2 in location
table. Now I want to somehow route my Invite packet to domain2. How
Hi Daniel,
I was able to solve a fraction of my problem, Actually, the github link had
used rtpengine.so and i was using rptproxy-ng.so, there is a difference in
the flag conventions between the two; i modified that to achieve a little
progress.
Now, i am able to call on webrtc(firefox) from sip
Hello,
maybe you should play with kamailio master branch (which is in testing
phase before becoming 4.2) -- there you have the rtpengine -- and see
if you get it working. Once that, you can look at using an older
version, knowing you have it working and be able to compare. As I needed
latest
Hello,
A crash just occurred.
I use the patch for the PAI issue. I had a look to the core dump and it
looks to be another issue:
(gdb) bt full
#0 0x0030f2230f30 in escape_string_for_mysql () from
/usr/lib64/mysql/libmysqlclient.so.16
No symbol table info available.
#1 0x0030f2226
Hello,
can you get the output in gdb for:
frame 2
p *_v
Cheers,
Daniel
On 16/09/14 17:34, Igor Potjevlesch wrote:
Hello,
A crash just occurred.
I use the patch for the PAI issue. I had a look to the core dump and
it looks to be another issue:
(gdb) bt full
#0 0x0030f2230f30 in esc
Hello,
I don't know if perl embedded api has the option to set the dst_uri (the
equivalent of $du config variable).
If not, set the new uri in an avp and in kamailio.cfg, after executing
your perl script, do:
$du = $avp(urifromperl);
Cheers,
Daniel
On 15/09/14 19:17, Vik Killa wrote:
I li
Hello,
if you want to rely on kamailio to memorize where to send the following
register, the hash table is the right tool.
Cheers,
Daniel
On 16/09/14 03:37, Vik Killa wrote:
Hello,
I'm learning Kamailio. My ultimate goal is to be able to route SIP
messages from User Agents to specific FS bo
Is there any way that i can create a list of ips for kamailio to compare
and do something, for example create an authorize list of ips on txt that
it can read for each invite that i receive than check a variable against
that list.
for example do a search $si on the external text file and if match
Thanks Daniel,
I want to remove the htable entry after the register goes through, and I
also want to check if the htable entry exists later.
I see
sht_rm_value_re("ha=>.*");
to remove the htable entry but how do i later check if the key exists?
Thanks
/V
On Tue, Sep 16, 2014 at 11:52 AM, Danie
Hello,
I was going through the new features and stumbled upon this new one -
developed by Mohd. Shahzad Shafi.
As already mentioned on the wiki about this module, I intend to use it for
my custom security layer between UACs and SIP Proxy (Kamailio) but the
issue is - the custom security layer (enc
Hello,
On 16/09/14 18:37, Vik Killa wrote:
Thanks Daniel,
I want to remove the htable entry after the register goes through, and
I also want to check if the htable entry exists later.
I see
sht_rm_value_re("ha=>.*");
this is going to remove all the items from the hash table -- perhaps you
d
Hello,
I would recommend to develop (or extend) a module for it if you have C
code -- this should be trivial if you have C knowledge and the other
code is already in C -- especially if the performance is a demand.
For a proof of concept, expecting that the encryption/description can be
done
Thanks for replying !
But how to check whether a particular message received by Kamailio was sent
by UAC or SIP Server ?
Also, on the same lines - how to know whether a particular message about to
be send from Kamailio is bound to UAC or SIP Server ?
On Tue, Sep 16, 2014 at 10:51 PM, Muhammad Sha
Hi,
Did you get some free cycles to look at it ?
On Wed, Sep 17, 2014 at 12:12 AM, Rahul MathuR
wrote:
> Thanks for replying !
>
> But how to check whether a particular message received by Kamailio was
> sent by UAC or SIP Server ?
> Also, on the same lines - how to know whether a particular me
Hello,
We have recently installed the 4.0 Siremis/Kamailio 4.1 on Centos 6.5, to use
as our SBC for our VoIP/hosted PBX. We struggled with the installation and
aren't sure if it is in fact complete. We weren't able to find documentation
that explained the correct setup and implementation of th
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