Hi Daniel,
I have just create pull request for this change.
https://github.com/kamailio/kamailio/pull/842
--
Best regards,
Sergey Basov e-mail: sergey.v.ba...@gmail.com
tel: (+38067) 403-62-54
2016-11-01 12:09 GMT+02:00 Daniel-Constantin Mierla :
> Hello,
>
> can you open a
Than is because of your db_mode. If you want everything to be written in
the database use db_mode 3. For more information read this:
http://kamailio.org/docs/modules/stable/modules/usrloc.html#usrloc.p.db_mode
On Wed, Nov 2, 2016 at 6:38 AM, Никитенко Виталий wrote:
> Hi all!
> I have kamailio
Hi all!
I have kamailio 4.2.3 with options
modparam ( "usrloc", "db_url", DBURL)
modparam ( "usrloc", "db_mode", 2)
But not all registred users displayed in table "location". When viewed "kamctl
ul show" displays
Domain :: location table = 1024 records = 1172 max_slot = 5
But "location" table ha
Might be you are doing fix_nated_sdp, multiple times in configuration.
On Nov 1, 2016 5:03 PM, "Yuriy Gorlichenko" wrote:
> I trying to use
> fix_nated_sdp with 0x02 and 0x08 flags for changing ip addres and the SDP
> body part
>
> like
>
> fix_nated_sdp(10,"1.2.3.4")
>
> But for now it beaks SD
Hi Daniel,
Thanks for your quick response. Makes totally sense, I'll need to debug why I'm
getting that malformed header.
Do you think my second question is related? I cannot see the same malformed
data in the example where the to-header is rewritten, it just looks like
Kamailio adds ";user=
Thanks for the recommendations Alberto. I'll definitely try it out and
hopefully will be able to call a softphone from webrtc client.
Cheers,
Serhat
On 1 November 2016 at 15:17, Alberto Llamas
wrote:
> Hi Serhat,
>
> If you take a look of SDP body of your INVITES you will note that you are
> of
Hi Serhat,
If you take a look of SDP body of your INVITES you will note that you are
offering SRTP.
What you should do from my point of view is detect when an INVITE from the
sipml5 softphone goes to the ims-softphone or other end-point which you are
aware doesn't support SRTP and use the RTPEngi
Hi Alberto,
Thanks for looking into this. In the expert settings of sipml5 it says that
disabling RTCWeb Breaker should make it compatible with softphones which
are not implementing SRTP, that's how I have been testing it though. May I
ask what attribute you looked at to get to the conclusion you
Hello Serhat,
When you are using the webphone (sipml5) by WebRTC the media is secured
with SRTP. So if the other end-point supports SRTP usually you don't have
major issues. It is like when you communicate between two sipml5 web phones
A and B.
But when you are trying to communicate to the IMS so
I trying to use
fix_nated_sdp with 0x02 and 0x08 flags for changing ip addres and the SDP
body part
like
fix_nated_sdp(10,"1.2.3.4")
But for now it beaks SDP
At the output i see next
o=- 7300689428214760503 2 IN IP4 1.1.1.1
c=IN IP4 1.1.1.1
1.2.3.41.2.3.4 <- is just a line that added after fi
Thank you so much. I will test it!
On Tue, Nov 1, 2016 at 1:55 PM, Daniel-Constantin Mierla
wrote:
> An alternative, which I haven't tried it to be 100% sure, but it should
> work as the accounting should happen when a reply is sent out -- anyhow,
> the idea is:
>
> - set an onreply_route for th
Hi Daniel, hi Alberto,
Thanks for your prompt replies. I have put 2 pcap files in dropbox (
https://www.dropbox.com/sh/fzclmbpniebrvx1/AAAOOv4h2ci7bJuuJvSbs3poa?dl=0 )
. trace.mercuro.pcap is the one where the session is set up, but there is
no audio flow and trace.boghe.pcap is the one with 488 e
Hello,
can you get the SIP INVITE content that was received by the endpoint
returning 488? Maybe we can spot if there is something wrong in the sip
message content or an issue in the endpoint software. Maybe it doesn't
like headers with random string instead of ip addresses (e.g., in via,
contact
HI Serhat,
Is it possible to have a packet capture for the cases you mention.
Regards,
On Tue, Nov 1, 2016 at 12:15 PM, Serhat Guler wrote:
> Hi,
>
> I have a setup as follows:
>
> IMS enabled on Kamailio and whereas websockets are enabled for PCSCF for
> webrtc calls.
>
> Calls(both audio and
Hi,
I have a setup as follows:
IMS enabled on Kamailio and whereas websockets are enabled for PCSCF for
webrtc calls.
Calls(both audio and video) between to sipml5 clients using firefox web
browser is possible. The session is setup for the calls from sipml5 to
Mercuro, but then there isn't audio
Hi,
I have a setup as follows:
IMS enabled on Kamailio and whereas websockets are enabled for PCSCF for
webrtc calls.
Calls(both audio and video) between to sipml5 clients using firefox web
browser is possible. The session is setup for the calls from sipml5 to
Mercuro, but then there isn't audio
Hello,
I am proposing a new IRC meeting to discuss the current major issues and
the plans for next Kamailio release, on Wednesday, Nov 09, 2016, with
alternatives for previous or next day. If many developers are not
available, we can postpone it to another date in the near future (make
proposals i
Hello,
I am considering to package a new release out of latest stable branch
4.4, respectively v4.4.4. The target date is one of the days between Nov
8-10, a matter of work load that needs to be done for getting it out.
As usual, if you are aware of any issue not yet reported on github bug
tracke
An alternative, which I haven't tried it to be 100% sure, but it should
work as the accounting should happen when a reply is sent out -- anyhow,
the idea is:
- set an onreply_route for the BYE requests and inside it, if the code
is 481, the reset the accounting flag(ie., resetflag(FLT_ACC)).
If s
I replied to the other email with similar topic, sent few days before
this message.
Cheers,
Daniel
On 22/10/16 06:21, Ginti Saurabh wrote:
> Hi,
>
> Please help to sort the problem.
>
> Getting "Unresolvable destination" error in /var/log/messages.
--
Daniel-Constantin Mierla
http://twitter.co
Hello,
the unresolvable destination should be triggered when the dns server
failed to resolve the hostname in r-uri and in this case it's clear it's
an invalid one being a contact for webrtc. You are not handling
correctly the devices connected over websocket, you have to deal with
them as with de
Hello,
this looks like a loop back to the server, causing the sip message size
to grow too much. Be sure that the domain you use in the SIP headers is
listed as an alias in kamailio.cfg:
alias=mydomain.com
Cheers,
Daniel
On 20/10/16 16:39, Hermann Norpois wrote:
> Hello,
>
> if my SIP client t
Hello,
can you open a pull request on github with the patch you propose for the
fix?
- https://github.com/kamailio/kamailio
It is easier to review and travis-ci will take care to compile the patch
and be sure it doesn't break latest master version.
Cheers,
Daniel
On 26/10/16 08:59, Sergey B
Hello,
when you define the hash table, there is an option to write back to
database, but I think it's happening only at shutdown.
The location_attrs table is used by usrloc module, you just need to set
some xavp in order to get them stored, respectively:
https://www.kamailio.org/docs/modules/sta
On 11/01/2016 02:58 AM, Gholamreza Sabery wrote:
Is there an option in the ACC module for this, or I should handle
dialog tracking using dialog module?
No, there's no option for it in the ACC module.
Using dialog tracking is one option. Adding a database constraint or
trigger that would prev
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