Hi,
I hope you can help me. My login details are correct but it keeps saying
timeout. It just suddenly stopped working. I have made no changes?
https://url.odesk.com/_01tJ3v5df4REETK4PV-B3hQtMyc7vTyOQO
Best,
Don
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Hello,
check this parameter of nathelper if suits your needs:
http://kamailio.org/docs/modules/stable/modules/nathelper.html#idp83280
It is not marking the user as unreachable, it marks it as expired.
For TCP (and friends) contacts, there is no keepalive, thus similar
behavior is achieved by u
It turned out to be the NAT handling process that screwed the gruu
treatment. Kamailio modified Contact from the OK (because this user is
marked as natted) and calling fix_nated_contact modified the Req-URI of
further in-dialog requests.
I have to look at the details but, using the standard config
Hello,
for iptel.org sip service, address your questions to mailing list:
- http://lists.iptel.org/mailman/listinfo/services
Cheers,
Daniel
On 01/09/14 17:53, Donovan wrote:
Hi,
I hope you can help me. My login details are correct but it keeps
saying timeout. It just suddenly stopped worki
Indeed it makes sense to skip contact mangling if gruu is present.
Cheers,
Daniel
On 02/09/14 11:45, samuel wrote:
It turned out to be the NAT handling process that screwed the gruu
treatment. Kamailio modified Contact from the OK (because this user is
marked as natted) and calling fix_nated_c
If you get signling routed ok but no audio, then you have problems
bridging rtp stream.
Most probably you need to use rtpproxy (eventually with advertise
address (there is a patch or use second parameter for
rtpproxy_manage())) to bridge.
I never used sip-natting in kernel, so I am not aware
As a complete "guide" to set up gruu handling, I've added below is_gruu
treatment in WITHINDLG, NATMANAGE, and NATDETECT routes.
# Handle requests within SIP dialogs
route[WITHINDLG] {
if (has_totag()) {
(...)
if(is_gruu()){
Hi All,
Kindly check my issue and please provide me your suggestions.
In my requirement, I want to add a route header to the INVITE message which is
sent out from the Kamailio server,
Basically I will be making a call and that INVITE will reach the Kamailio, then
the Kamailio will send the
Hello,
You can add headers using the insert_hf() / append_hf() functions.
However, you should be aware that UACs generate the Route set. Kamailio
is a proxy. You do not need to add a Route header in order to have
Kamailio send a request to another Kamailio instance or proxy, or to a
UAS. You
Hello Kamailio Team,
I am using kamailio server successfully for some time now. I have a
specific set of problem. We have a Kamailio Server setup in Amazon EC2
Cloud, the users are able to register on the server and make use of this.
*Now the problem:*
If I am trying to register multiple SIP cli
Hello,
sending a short reminder that the 4.2.0 feature freezing day is planned
in about one week, respectively Wednesday, September 10, 2014.
Cheers,
Daniel
On 25/08/14 18:04, Daniel-Constantin Mierla wrote:
Hello,
during the last devel meeting on IRC, done before the summer, we set
the 4.
Hello,
you should watch the traffic on kamailio server to see the source ip and
port of packets and where the SIP traffic is sent back to phones. You
can use:
ngrep -d any -qt -W byline "sip" port 5060
I guess that the nat router assigns the same public port for all phones,
but it forwards
Indeed, currently Kamailio does not support SNI (was dropped with ser merge)
Klaus
On 29.08.2014 16:11, Daniel-Constantin Mierla wrote:
> Hello,
>
> starting with 3.0 we got the implementation from SER at that time (being
> more flexible with config and later getting asynchronous support).
>
>
On 29.08.2014 11:12, Lukas Wygasch wrote:
> Is there a Module i could use to do this Job? (configurable through
> Siremis e.g. Dispatcher List?)
Kamailio is a proxy, thus it can not authenticate itself to some other
SIP server. (Actually it can using the uac module but this is not nice).
> How
Not sure what you trying to do, but the Via header is for transactions.
It may be different for every transaction. Thus, if you need transaction
matching (requests to responses) then you are fine and should use purely
the branch id.
If you want to match messages from one transaction to messages fr
Hi Klaus,
thanks for updating on the status.
Do you remember what implied to add support for SNI?
It should be brought back if we lost it. Maybe you can adapt the old
patch if it not something that complex and you have time to look at it.
Otherwise, any further details about what you had to d
Adding SNI was rather easy. I used the original SNI patch for Apache and
copy-pasted this patch into Kamailio. We could do this again, but this
patch does not have any license details, thus I would recommend to not
do it. Unfortunately I haven't found proper SNI API desription of
libssl. Maybe we c
El 02/09/14 05:17, Daniel-Constantin Mierla escribió:
If you get signling routed ok but no audio, then you have problems bridging rtp
stream.
Most probably you need to use rtpproxy (eventually with advertise address
(there is a patch or use second parameter for rtpproxy_manage())) to bridge.
> "KD" == Klaus Darilion writes:
KD> Maybe we can find some software with SNI support and BSD license
KD> and then copy/paste the code.
nginx is a possibility.
-JimC
--
James Cloos OpenPGP: 0x997A9F17ED7DAEA6
___
SIP Express Router (SER
Thank you so much for your informative response.
Yes the "peer" may be correct term in this sense as i am trying to identify
"devices" (SIP UAs or Proxy) that are directly connected to Kamailio via
SIP signalling (i.e. there is no other intermediate SIP device [SIP UA or
Proxy] in the middle). Tha
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