Additionally, there's no other way than implementing dialog module to keep a
variable between the beginning and the end of a call?
Regards,
Igor.
De : Igor Potjevlesch [mailto:igor.potjevle...@gmail.com]
Envoyé : lundi 16 février 2015 18:36
À : 'Kamailio (SER) - Users Mailing
Hello,
if you know that the contact address is valid and should be used for
opening connections towards UA, then do not call fix_nated_register()
for REGISTER request.
Unfortunately UA behind NAT using STUN can lead to public address in
Via/Contact/etc... but with a wrong port, therefore we have
On 16/02/15 12:39 PM, Muhammad Shahzad wrote:
I haven't done something like that myself but i think if you use
RTPEngine with media-address set correctly in offer and answer
functions, you can easily achieve this. Simply check if request/reply is
coming from FS or the end-user and adjust the
dear Kamailians,
I have Kamailio+rtpproxy in front of FreeSWITCH.
Kamailio and FreeSWITCH are on the same private network.
Public Internet IP address ports are redirected to Kamailio and
rtpproxy (same situation as in Amazon EC2).
Clients comes from Internet, and make calls to Internet, SIP
Hello,
I'm looking for a way to track a call by using basic AVP like this:
Into a route that treats INVITE:
$avp(s:state)=call_start;
Then, if I test this AVP into WITHINDLG route:
if($avp(s:state)!=call_start) ; the test fails.
Did I miss something?
The goal is to update this
Hello,
avps are lasting for the duration of the transaction. In route withindlg
you handle already another transaction than the initial invite, so the
avp is gone. Try to use $dlg_var(...) for this case -- check also if
there is no $dlg(...) var that returns the state of the dialog and you
can
Hello,
rtpproxy doing bridging requires two network interfaces to work with.
You can try one of the following:
- let freeswitch advertise the public ip for media and skip rtpproxy
completely
- use the second parameter of rtpproxy_manage() to set the advertised ip
address for media and don't
I haven't done something like that myself but i think if you use RTPEngine
with media-address set correctly in offer and answer functions, you can
easily achieve this. Simply check if request/reply is coming from FS or the
end-user and adjust the media appropriately without even invoking
Hello,
afaik, the pua_reginfo is for publishing details of location records to
another sip server node, main purpose being location replication. I
don't think it is something for an end UA.
If you want publishing online/offline states for an user based on its
registration state, look at
BTW, if nothing works, you can always use network:msg event route to find
/ replace any part of the SIP request and response, including media IP in
SDP. ;-)
http://kamailio.org/docs/modules/4.2.x/modules/corex.html#async.evr.network_io
Thank you.
On Mon, Feb 16, 2015 at 6:39 PM, Muhammad
Hello,
you should be able to extract it with {line} and {subst} transformations
applied to sdp body.
Cheers,
Daniel
On 10/02/15 23:42, Ryan Brindley wrote:
Hey community,
What's the best way to pull out the media ip from the SIP INVITE body
(for logging)?
Ryan Brindley
As far as i know AVPs are transaction specific only. So they will be
deleted as soon as transaction is over, i.e. 200 OK for INVITE is received
for example. They will not be available in in-dialog transactions such as
ACK, or BYE etc. What you need is to set dialog variable instead, see more
info
Thank you guys, I will try this.
I misunderstood the notion of transaction. I was thinking that it was the
whole call-flow.
Regards,
Igor.
De : sr-users [mailto:sr-users-boun...@lists.sip-router.org] De la part de
Muhammad Shahzad
Envoyé : lundi 16 février 2015 18:27
À : Kamailio
Well, you can also put them in some storage backend e.g. MySQL, PGSQL using
AVPOPS or memory caches such as Redis etc.
Another way is to set it as record-route parameter using RR module. (not
recommended)
http://kamailio.org/docs/modules/4.2.x/modules/rr.html#add-rr-param-id
Thank you.
On
On 16/02/15 01:00 PM, Virmantas Variakojis wrote:
Hi,
There pathch with -A can be found or it is allready implemented into
specific rtpengine version?
Latest master from git. The command line syntax is a bit different from
rtpproxy, but the basic idea is the same.
Cheers
Why not an RR parameter? It's probably the most reliable way to store some dialog-persistent data, since it doesn't depend on any in-memory/runtime state to be kept by the proxy itself, instead using the SIP
Indeed, RR could do the job. But it will not be easy to get the value after. It
could be possible with regex I guess.
I will look at htable too. It's looks to be easier than dialog.
For AVPOPS, why not. I'm just afraid with the delay.
Many thanks for all these suggestions.
Regards,
I tried with $sht(myhash=$ci::state) = call_start.
It works fine!! Many thanks.
Is that could work too: $sht(myhash=$ci::$ft::state) = call_start?
To delete this, can I do sht_rm_name_re(myhash=$ci);? I want to be sure that
after the call ends, everything is cleared.
Regards,
Could you provide us a little example? For examlple i have kamailio with
three interfaces: two interfaces (vlan's look at two different providers)
and third interface looks at sip clients.
Thank's in advance!
2015 vas. 16 20:04 Richard Fuchs rfu...@sipwise.com rašė:
On 16/02/15 01:00 PM,
It's pretty straightforward using the right transformations on $hdr(Record-Route). Have a look at the transformations docs.
On 16/02/15 01:12 PM, Virmantas Variakojis wrote:
Could you provide us a little example? For examlple i have kamailio with
three interfaces: two interfaces (vlan's look at two different
providers) and third interface looks at sip clients.
You would define two interfaces with different names,
Yes for the same reasons as you mentioned, it adds dependency on external
entities in your setup and may not be suitable for any sensitive data (e.g.
related to billing etc.).
Thank you.
On Mon, Feb 16, 2015 at 7:05 PM, Alex Balashov abalas...@evaristesys.com
wrote:
Why not an RR parameter?
Yes, and yes.
Thank you Alex.
I'm not sure to understand the parameter size associated to the hashtable.
I have setup 4. So, I understand that I can have 2^4 entries. Does it mean
that, if the table is composed with $ci+$ft, I can have 16 concurrent calls
store into the table?
Regards,
Igor.
You could simply let the RTP traffic to flow directly between FS and
endpoints (no need for rtpproxy).
All you need to do is:
- forward the appropriate RTP ports to FS;
- fix the private IP in SDP by replacing it with the public IP for
the inbound rtp streams (to FS).
-ovidiu
On Mon, Feb 16,
Hello,
the SNI (server name indication) support was available in kamailio v1.5
and then lost when the code was integrated with ser. It was on my to-do
to re-add it but no time for it in the past. I just pushed a partial
patch that allows to set a server_name for each TLS server domain
(context)
Hello,
I followed the documentation from
http://kamailio.org/docs/modules/4.2.x/modules/debugger.html#idp84752. I have
the global debug flag at 9.
modparam(debugger, cfgtrace, 1)
modparam(debugger, mod_level_mode, 1)
modparam(debugger, mod_level, core=3)
My Kamailio complains with a parsing
Could you show the revelant codes in rtpproxy and kamailio.cfg? I am unable
to get the audio pass through from extranet to intranet as private IP
address is used after rtpproxy.
I use Kamailio 4.2 and rtpproxy in Debian wheezy. Both are installed from
repository.
On Tue, Feb 17, 2015 at 8:27 AM,
Just to add that the Contact address used for bridging can be changed
via module parameter:
- http://kamailio.org/docs/modules/stable/modules/dialog.html#idp1855568
Should be changed to reflect local IP of the server.
Cheers,
Daniel
On 15/02/15 18:02, Ben Langfeld wrote:
The REFER's contact
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