Hello,
I am using Kamailio as registration server and FreeSwitch for signalling
(RTP packet handling). Can I use GStreamer instead of Freeswitch or
Asterisk?
Thank you.
Regards,
CM
Hi there,
unfortunately I am personally not a complete IT geek, I am more in to Pro Audio. Anyway, I have to maintain a SIP Server for high quality Audio transmission with special UA's.
Also we use Software UA's running as Apps (Luci Live) on mobile devices such as I Phones and I pads which are by nature situated in mobile networks. Due to "lack of IP address issues" most providers use private ranges separated by CGNAT from the real Internet. This causes a lot of trouble. In some cases stun server entries on the clients helps, in other cases don't!
But these days I read in announcement of a german company (Mayah Communications), which is in the ProAudio biz since decades, that they offer a SIP Service where no stun is recommended whether the clients are behind NAT or not.
After I acknowledged that IT business is not witchcraft (almost :-) ) , I thought that there might exist a solution for my Kamailio SIP server to make it work like the "Mayah" Sip Server.
What settings have to be done that Kamailio uses IP Header Information for SIP signalling instead of the content of SIP packages themselves. Without stun you find the Local Address in the SIP packages, and with Stun the external Address but mostly with the wrong portnumber, but in the IP header you'll find the correct IP:port.
I already read a lot of the Kamailio documentation but due to the fact that I am not too deep in IT I am not sure if I got it all. If someone could give me a useful hint on that, that would make my day.
Hope this description is unterstandable.
Nice Regards
Gerhard Pinter
Dear all,
I would need some help setting up Kamailio with RTP engine.
I'm trying to make use of RTPEngine recording daemon's forward-to option,
to have the calls streams sent to 3rd party application.
Have someone managed to have this forward-to option work?
If I understood well, when RTPEngine's forward-to option is turned on, the
call metainfo + streams can be sent to a socket server?
Thanks in advance!
Regards,
Balazs
Greetings,
I'm getting a query result in a $xavp. I'm trying to copy the entire
structure to a new $xavp like this : $xavp(Valid_Acd_Results) =
$xavp(ACD_Query_Result[$var(iterator)]);
However, i can't retrieve the value with : xnotice(" Mode =
$xavp(Valid_Acd_Results=>TBK_AcdMode)"); since it returns NULL. Am i doing
something wrong?
Thanks
Hi!
Tell me why when using the save ('') function in the response generated by the server, there are the Record-route headers that were in the request. And when using the following functions:
- proxy_challenge (),
- www_challenge (),
- send_reply (),
- sl_send_reply
there is no Record-route header.
How can you add them?
--
Evgeniy
I'm having problems with my To field.
I'm doing an async_route, and afterwards sends the call with auth to a
SIP server.
My problem is that in the first invite I see the expected To field
sip:004520202020@isp.com
But when the Auto is send the To field is changed to the, To field for
the PBX
sip:1234@pbx.local
I tried to fix this with uac_replace_to("$ru"), but that ends up in this
To field:
sip:004520202020@isp.comsip:004520202020@isp.com
It seems like uac_replace_to sometimes appends rather that replaces.
If I run it twice, then it for sure appends. Is this expected behaviour.
--
-------------------- Med Liberalistiske Hilsner ----------------------
Civilingeniør, Kjeld Flarup - Mit sind er mere åbent end min tegnebog
Sofienlundvej 6B, 7560 Hjerm, Tlf: 40 29 41 49
Den ikke akademiske hjemmeside for liberalismen - www.liberalismen.dk
I have cr_route working well. Only problem is that the username is sent off to the carrier and the 3 or 4 digit username appears as the callerid.
Is there a module or scripts that can replace the username with a db table defined callerid. Say 202 maps to a 10 digit TN.
KD
Hello,
Kamailio SIP Server v5.1.3 stable release is out.
This is a maintenance release of the latest stable branch, 5.1, that
includes fixes since the release of v5.1.2. There is no change to
database schema or configuration language structure that you have to do
on previous installations of v5.1.x. Deployments running previous v5.1.x
versions are strongly recommended to be upgraded to v5.1.3.
For more details about version 5.1.3 (including links and guidelines to
download the tarball or from GIT repository), visit:
* https://www.kamailio.org/w/2018/04/kamailio-v5-1-3-released/
RPM, Debian/Ubuntu packages will be available soon as well.
Many thanks to all contributing and using Kamailio!
Cheers,
Daniel
--
Daniel-Constantin Mierla
www.twitter.com/miconda -- www.linkedin.com/in/miconda
Kamailio Advanced Training - www.asipto.com
Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com
Hello all,
I've been trying to figure out a cleaner way to determine the next hop for
a SIP message, mainly for use within the NATMANAGE route for multi-homed
kamailio instances (with three or more network interfaces on the kamailio
host).
So far I have achieved this with a series of nested ifs, depending on
whether the message is a request or a response and by calculating the next
hop based on the various headers (R-URI, Route) and variables ($T_req($Ri),
$dd, ) involved in SIP routing.
A simpler way to do it, of course, would be to use the onsend_route, but
that would most likely introduce an unnecessary overhead for all routed
messages.
I recently noticed there a pseudovariable called $nh(key), and I believe I
can use $nh(d) to the same effect. I understand however, that this works
for requests only. Also, the description of this PV in the documentation
reads as follows:
Return attributes of next hop for the SIP request. Address is taken from
> dst_uri, if set, if not from new r-uri or original r-uri.
>
What is not clear to me is if this covers in-dialog requests with the Route
header set as well. Does the inclusion of a route header set the dst_uri
PV? And if yes, is it safe to rely on dst_uri during request processing or
is it set only after completion of script processing?
Lastly, is there an analogue for SIP responses?
If not, is it safe to rely on first Via header to determine next hop for
responses, or are there any other corner cases I need to heed?
Thanks in advance,
BR,
George
