I have kamailio behind a TLS termination proxy so the sockets are correctly
deduced to be TCP. However the clients only talk TLS to the proxy and are
confused when the top Via header added by Kamailio is TCP. Is there a way
for Kamailio to forcibly pretend its protocol is TLS? Like
advertised_address but "advertised_protocol" instead.
(With pjsip testing: it has a flag use_tls which ignores TCP from Kamailio
and continues to use the persistent TLS transport to proxy. Linphone fails
because it tries to honor TCP in Via and is unable to establish TCP
transport).
BTW I am using t_relay_to_tcp so Kamailio will return traffic to the proxy
as TCP even though the contact addresses specify transport=TLS.
Hi everybody,
I'm just testing Kamailio 5.4.1 with dialog replication over DMQ. This
seems to work very good. Dialogs are replicated without problems.
When I'm restarting one node I would have expected, that all dialogs are
synced again, just like in dmq_usrloc.
But this does not happen. After a restart the nodes dialog-list is empty.
Did I miss somethin? Is there a special parameter that I have to set?
BR, Björn
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Hi All,
I am facing an issue in understanding how the min_se should be working in
kamailio. As per the SST documentation, it seems like if the min_se is
configured as 500, then any value of Session-Expires OR MIN-SE if lower
than 500, can be responded to by a 422.
However, I strangely see the reverse happening. To investigate further, I
looked in to the ki_sst_check_min() code in the master, and these seems
like a potential issue.
Ref Code: Inside ki_sst_check_min(), there is an if condition like below:
if (sst_min_se < MIN(minse, se.interval)) {
However, shouldn't it be the other way around? ie
if (sst_min_se > MIN(minse, se.interval)) {
because we need to send 422 if the received value(in INVITE etc) is
smaller than the sst configure min_se value?
I also found a different documentation, at
https://git.sgu.ru/oldssu/ex-opensips/blob/cb9df8d59dbb254a9d862569fd5d11f6…
where
the check is as below?
if (sst_min_se > MIN(minse, se.interval)) {
Can someone confirm if this is broken, or my understanding is incorrect?
Regards,
Harneet
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"Once you eliminate the impossible, whatever remains, no matter how
improbable, must be the truth" - Sir Arthur Conan Doyle
On Mon, May 07, 2018 at 04:44:14PM +0200, Daniel Tryba wrote:
> Sure. Attached. Problem appears to be that the topos query can't find
> callid-totag (from the response).
>
> I'll try the same scenario with the mysql backend to see if it behaves
> different.
Config works fine with mysql as topos backend. So the bug is restricted
to topos-redis.
Hello ,
im using kamailio with two interfaces external and internal.
i need a way either to :
* enable topoh hiding only when outgoing interface is external ( mask contact and Via ip only when ougoing interface is external
* or if it is not possible to enable it only in one direction.i want to know how to configure dynamic(for example in xavp or avp) ip to put in Contact and Via when topoh is enabled.
i see that the 'mask_ip' parameter of topoh module is a string. so we can not set a dynamic value here unfortunately.
Thanks
Hello,
is there an rsync endpoint available or is there a possiblity of setting
this up? Creating a mirror via HTTP is a rather quick and dirty solution
and while the deb-repo can be mirrored using debmirror, the rpm repo is
hard to sync to a non-CentOS-based machine due to missing dependencies
such as yum and reposync in latest Debian-based systems.
Would be great to get some input in regards to this topic.
Cheers
Hello,
the formal notification that the development for the next major version
5.6.0 is now frozen. The focus has to be on testing the master branch.
Also, the master branch should not get commits with new features till
the branch 5.6 is created, expected to happen in 2-4 weeks, a matter of
how testing goes on. Meanwhile, the commits with new features in the C
code can be pushed to personal branches, new pull requests can still be
done, but they will be merged after branching 5.6.
Can still be done commits with documentation improvements, enhancements
to related tools (e.g., kamctl, kamcmd), merging exiting pull requests
at this moment, exporting missing KEMI functions and completing the
functionality
of the new modules added for 5.6.
Once the branch 5.6 is created, new features can be pushed again to
master branch as usual. From that moment, the v5.6.0 should be out very
soon, time used for further testing but also preparing the release of
packages.
If someone is not sure if a commit brings a new feature, just make a
pull request and it can be discussed there on github portal or via
sr-dev mailing list.
A summary of what is new in upcoming 5.6 is going to be built at:
* https://www.kamailio.org/wiki/features/new-in-5.6.x
Upgrade guidelines will be collected at:
* https://www.kamailio.org/wiki/install/upgrade/5.5.x-to-5.6.0
Everyone is more than welcome to contribute to the above wiki pages,
especially to the upgrade guidelines, to help everyone else during the
migration process from v5.5.x to 5.6.x.
Cheers,
Daniel
--
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Hello! I wish everyone who reads this message a good day.
I would appreciate some guidance with the configuration of my Kamailio.
Initially, I have the following architecture:
Endpoints <----Public IP----> Kamailio (listen PrivateIP advertise
PublicIP) <----Private IP----> Asterisk
CONFIG:
https://gist.github.com/Carlos-Escalona94/3a289af05b05c69ae563ab29c9ec4710
In general, calls initiated from outside the system work perfectly, but I
have a problem with calls initiated from Asterisk.
According to what I was able to investigate and understand, the problem
arises from the fact that Kamailio uses the public IP for the entire
routing system that involves the Record-Route and Route Headers. This in
turn causes Asterisk to get lost trying to send messages outside of the
INITIAL INVITE transaction, for example, the ACK response to the 200 Ok
received from the Endpoint since it tries to use Kamailio's public IP to
which Asterisk doesn't have access.
I tried to force Asterisk to send all messages to Kamailio's private IP
regardless of their nature, but it doesn't seem like an appropriate
solution.
On the other hand, I tried to modify the architecture a bit so that it was
something similar to this:
Endpoints <---- Public IP----> Kamailio (listen PrivateIP:P1 advertise
PublicIP) <---- PrivateIP-----> Kamailio (listen PrivateIP:P2) <----
PrivateIP ----> Asterisk
CONFIG:
https://gist.github.com/Carlos-Escalona94/4d681bb189c6190941d291965e123889
It seems to me that this would solve the problem, but I have two doubts
about this architecture, the first is that I would like to know if there is
an easier way to solve the problem that I am not considering, and on the
other hand, I have not found a way to identify from which interface is
receiving the message to perform the routing properly.
Thanks for the attention.
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Hello,
I am trying to make UAC register with a SIP telco. But when I run...
shell$ kamctl rpc uac.reg_add mydomain.com 0123456789 myteldomain.com 0123456789 sip6.telco.nl telcoreseller.nl 999999 secretpw . udp:sip6.telco.nl 360 0 10 'udp:[fd8:9::10]:5090'
...it gives me the cryptic response...
ERROR: jsonrpcs [jsonrpcs_mod.c:666]: jsonrpc_scan(): field is not a string - type 3
The module docs only give "..." as example of attributes. And, according to...
http://www.kamailio.net/docs/modules/5.4.x/modules/uac.html#uac.r.uac.reg_a…https://github.com/kamailio/kamailio/blob/5.4.4/src/modules/uac/uac_reg.c#1…
...it also seems to be behind on diff_expires, timer_expires, reg_init.
What I am missing is the format, and in some cases the purpose of attributes.
Could you help me to an example, perhaps based on the command I am trying?
Thanks,
-Rick
P.S. It's been years since I worked with OpenSIPS. It feels like Kamailio is
friendlier, and easier to use.