On Wed, Sep 17, 2014 at 6:39 PM, aawaise wrote:
> How can we extract domain from the subscriber table to logs ??
> Its beyond my control at the moment to shift to kamailio-4.1. So have to
> persist with kamailio-3.1 ?
>
Look at the pseudovariable documentation. It sounds like $rd is what you
wan
How can we extract domain from the subscriber table to logs ??
Its beyond my control at the moment to shift to kamailio-4.1. So have to
persist with kamailio-3.1 ?
--
View this message in context:
http://sip-router.1086192.n5.nabble.com/Print-callee-domain-in-logs-tp130597p130733.html
Sent fro
On 17/09/14 17:19, Paweł Sternal wrote:
Thanks Daniel, you save my day :-)
You're welcome! Great to see you ended up with more than expected :-)
Cheers,
Daniel
Two rtpengine solves my problem and works perfect. This solution also
adds me the possibility to record calls, when between two rtpe
Hello,
the value to insert in db is invalid, now to see what was supposed to be
there. Can you send the parameters you set for acc module?
Cheers,
Daniel
On 17/09/14 19:48, Igor Potjevlesch wrote:
Hello,
(gdb) frame 2
#2 0x7fb984c0b089 in db_mysql_val2str (_c=0x7fb985033d18,
_v=0x7fb
Richard Fuchs writes:
> Just as a follow-up, do you have the NO_FALLBACK option set in the
> config?
Yes, I have
NO_FALLBACK=yes
in the default file, which I think is the default.
-- Juha
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-user
On 08/21/14 03:09, Juha Heinanen wrote:
> Now I made another dist-upgrade upgrading rtpengine from
> 3.3.0.0+0~mr3.4.1.0 to 3.3.0.0+0~mr3.5.0. Indeed dist-upgrade tries to
> setup daemon after the kernel module was removed but before new kernel
> module was installed.
...
> Setting up ngcp-rtpengi
Hello,
(gdb) frame 2
#2 0x7fb984c0b089 in db_mysql_val2str (_c=0x7fb985033d18,
_v=0x7fb981921f60, _s=0x1c51bf0 "0669','0123456789','')",
_len=0x7fffab695f74) at km_val.c:79
79 _s +=
mysql_real_escape_string(CON_CONNECTION(_c), _s, VAL_STR(_v).s,
VAL_STR(_v).l
Hello,
Thats clear. Thank you!
Regards,
Igor.
De : Daniel-Constantin Mierla [mailto:mico...@gmail.com]
Envoyé : mercredi 17 septembre 2014 11:56
À : Igor Potjevlesch; 'Kamailio (SER) - Users Mailing List'
Objet : Re: [SR-Users] kamctl alias add issue
Hello,
the default q valu
On Sep 17, 2014, at 8:13 AM, Bruno Emer wrote:
> Hello!
>
> I am relatively new to Kamailio and I'm trying to create a new enviroment
> using it in my company. I am thinking about use Amazon to host the servers
> and use OpsWorks to automatically escalate then if necessary. To accomplish
> t
Thanks Daniel, you save my day :-) Two rtpengine solves my problem and
works perfect. This solution also adds me the possibility to record calls,
when between two rtpengine instances I will put rtpproxy.
Regards,
Pawel
2014-09-17 9:35 GMT+02:00 Daniel-Constantin Mierla :
> Hello,
>
> I would add
The source tree includes an example of how to do this:
https://github.com/kamailio/kamailio/blob/master/examples/vm_proxy.cfg
After fr_inv_timer expires, Kamailio will generate a 408 and call your
failure_route, which is where you can redirect to your voicemail server.
On Wed, Sep 17, 2014 at 3:
Hi Daniel,
Here is something i traced in the log:
ip-172-31-47-138 rtpengine[4879]: Unknown flag encountered: 'force'
ip-172-31-47-138 kernel: [4155571.651074] traps: rtpengine[4884] general
protection ip:41e313 sp:7f2bf1934418 error:0 in rtpengine[40+3]
What's the cause of this error? i
Hello!
I am relatively new to Kamailio and I'm trying to create a new enviroment
using it in my company. I am thinking about use Amazon to host the servers
and use OpsWorks to automatically escalate then if necessary. To accomplish
this, my idea is to separate the servers, using one dedicated serv
On Tue, 2014-09-16 at 19:46 +, Mike Hancock wrote:
> Hello,
>
> [SNIP] I would like to purchase support to help to get this up and
> running as well as configured to get us going. [/SNIP]
>
>
> Mike Hancock
Hi Mike,
This is the link for the business directory:
http://www.kamailio.org/w
Hello,
On 12/09/14 14:49, [PRE s.r.l.] - Alex wrote:
Hello,
the situation is:
client -> kamailio + rtproxy -> asterisk -> rtpproxy + kamailio ->
other client
the idea is that rtpproxy has to proxy the whole rtp traffic between
asterisk or other media proxy and the client.
the strange prob
I would like to send a call to voicemail server after simple timeout.
Called client replies with prov response (183 with SDP) but doesn't answer, so
no 200OK message.
Is it possible to route call to voicemail server? How?
Many thanks___
SIP Express Route
Hello,
the default q value initialization was on wrong place -- so you had to
explicitly do it. It should get fixed with the next patch release tomorrow.
Cheers,
Daniel
On 11/09/14 23:12, Igor Potjevlesch wrote:
Hello Daniel,
It works! Thank you.
What went wrong ?
Regards,
Igor.
*De :*
Hello,
your remarks seems legit -- if you send a patch, I will review and
commit it. Otherwise, open an issue on bugh tracker and will be fixed soon.
Cheers,
Daniel
On 17/09/14 10:06, dongwf wrote:
Hi kamailio dev:
I am writing a module to implement some features, my version is 4.1.2,
I am
Hi kamailio dev:
I am writing a module to implement some features, my version is 4.1.2, I am not
sure if it is a known issue, I found:
1. When I get a INVITE request with from/to like from:"display",
then I call tm's new_dlg_uas with the sip req, now the dlg's rem_uri set as
value: "display";
2.
Hi Daniel,
As you instructed, i installed kamailio from the master branch (which has
rtpengine module). Along with this, i installed the rtpengine package from
sipwise, as instructed by them.
I also updated this param : modparam("nathelper", "sipping_from", "
sip:pin...@abc.com") to my domain
No
Hello,
a reminder about packaging 4.1.6 -- it will be done tomorrow. Therefore
if you are aware of patches to be backported or issues to be fixed,
report them to sr-dev.
Cheers,
Daniel
On 12/09/14 10:48, Daniel-Constantin Mierla wrote:
Hello,
I am considering releasing minor version 4.1.6
Hello,
I would add the RTP-WebRTC gateway between SIP Kamailio and SIP UAC,
from resources point of view, it is the only leg that needs
encryption/decryption.
Otherwise, you can try to work with two rtpengine instances (sets) in WS
Kamailio, one to use for ws client to proxy and the other on
Hello,
I haven't played with such setup yet, but it's somewhere on my list for
Christmas wishes (well, the time for it)... so I am interested in seeing
this discussion elaborated more.
I noticed various results when searching web for kamailio+docker. A
dockfile by Victor Seva is available at
Hello,
On 16/09/14 16:49, parcerit...@gmail.com wrote:
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 e
Hello,
the corex has a function to tell if the message is received by kamailio
or sent out:
- http://kamailio.org/docs/modules/devel/modules/corex.html#idp29928
If you interconnect with other servers/gateways when you don't what to
do special encoding, then you need to test src ip or look ar
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