Here is an example, payload taken from Identity header.
Identity was added with secsipid_add_identity
Payload test:
$var(test) =
That’s the same way I am doing it, I was just trying to do a verification that
the identity header/payload was correct before activating new changes.
I will do further testing and share results. Just found it odd that the header
would decode but payload wouldn’t.
Daniel W. Graham, CTO
I DO IT WITH:
# Break JWT
$var(jwt1) = $(hdr(Identity){s.select,0,.}{s.decode.base64t});
$var(jwt2) = $(hdr(Identity){s.select,1,.}{s.decode.base64t});
Regards,
David Villasmil
email: david.villasmil.w...@gmail.com
phone: +34669448337
On Wed, Jun 30, 2021 at 8:48 PM
Shahid Hussain writes:
> Following are the REGISTER and response messages. Is it possible to
> confirm the JSSIP client has full implementation of SIP outbound?
Looks like it if you define two sockets.
-- Juha
__
Kamailio - Users Mailing
Hello,
not all the modules are packaged on a distro because they may depend on
libraries not available in that distro.
It is also the case here, the stirshaken module depends on
libstirshaken, which is not available on debian.
Same is with secsipid module, which also offers STIR/SHAKEN
Hello,
for the sake of completion: the autoexpire should clean the items if
they are not used during the expiration interval. If you want to get
them deleted after first expiration interval always, see the
updateexpire attribute for htable modparam.
Also, note that replication should be done
Hi,
I know that it's possible to compile manually but what about apt method to
install stirshaken module?
I am checking the list of modules using:
apt search kamailio
But... Nothing looks like stirshaken.
Thus, only several modules available and for others we should compile
manually ?
Please
Hello Daniel,
Thanks for the feedback. I think I might have been too quick to blame
htable for this behaviour. In fact, version 5.4 seems to consume more
memory than 5.5 (175129776 bytes vs 20581096), which makes sense since it
has been running for longer (I missed the extra digit previously).
Hello,
do you replicate items in the htable via dmq? Does the htable have
autoexpire value set?
Cheers,
Daniel
On 30.06.21 13:54, George Diamantopoulos wrote:
> Forwarding my reply to the list, using gmail's reply button set
> Henning as the sole recipient :-\
>
> -- Forwarded message
Due to other tests, I had missed baresip account's ;outbound paramater.
Once I added it, also reg-id was added.
-- Juha
WSS 192.168.43.160:50442 -> 192.168.43.160:5061
REGISTER sip:test.tutpro.com SIP/2.0
Via: SIP/2.0/WSS 127.0.0.1:9;branch=z9hG4bK5a4ad01f9164d358;rport
Contact:
Hi everyone,
I'm using Kamailio (Ver 5.5.1, Public IP X.X.X.11, Private IP 172.18.0.20) +
RTPEngine (Ver 9.4.1.1, Public IP X.X.X.7, Private IP 172.18.0.50) alongside
Freeswitch (Private IP 172.18.0.40) in my backend and delivering calls to a
mobile application (call flow is Freeswitch ->
Hi everyone,
I'm using Kamailio (Ver 5.5.1, Public IP X.X.X.11, Private IP 172.18.0.20) +
RTPEngine (Ver 9.4.1.1, Public IP X.X.X.7, Private IP 172.18.0.50) alongside
Freeswitch (Private IP 172.18.0.40) in my backend and delivering calls to a
mobile application (call flow is Freeswitch ->
Olle E. Johansson writes:
> Full support for SIP outbound (using REG-id when registering etc).
> Last time I looked we did not have all nuts and bolts for it, but
> let’s give it a try.
Yes, reg-id is missing from contact. It would be good to add so that
sip proxy can detect if registration is
Olle E. Johansson writes:
> > Have you checked baresip?
>
> I don’t recall baresip having a full SIP outbound implementation.
baresip is able to register with two outbound proxies and supports gruu
(below). What else is needed?
-- Juha
#
TLS 192.168.43.160:49556 -> 192.168.43.160:5061
Forwarding my reply to the list, using gmail's reply button set Henning as
the sole recipient :-\
-- Forwarded message -
From: George Diamantopoulos
Date: Sat, 26 Jun 2021 at 02:25
Subject: Re: [SR-Users] Possible memory leak on 5.5.x (new)?
To: Henning Westerholt
Hello
Olle E. Johansson writes:
> Full implementation of SIP outbound is the only solution close to
> solving this problem in the IETF standards.
> However, I have seen no single SIP client that have implemented this,
> even though Kamailio supports
> it on the server side. The idea is that you
Shahid Hussain writes:
> Would like to know what is the recommended solution for this problem using
> alias or is it a limitation of using alias?
Maybe a limitation. Try with SIP User Agents that support gruu and thus
identify themselves using sip.instance.
-- Juha
I am using a JSSIP client and they claim to be implemented RFC-5626.
Following are the REGISTER and response messages. Is it possible to
confirm the JSSIP client has full implementation of SIP outbound?
If it supports fully then I can debug outbound and gruu functionality at
Kamailio(I have it
On 30.06.21 09:28, Juha Heinanen wrote:
> Shahid Hussain writes:
>
>> Would like to know what is the recommended solution for this problem using
>> alias or is it a limitation of using alias?
> Maybe a limitation. Try with SIP User Agents that support gruu and thus
> identify themselves using
> On 30 Jun 2021, at 10:14, Juha Heinanen wrote:
>
> Olle E. Johansson writes:
>
>>> Have you checked baresip?
>>
>> I don’t recall baresip having a full SIP outbound implementation.
>
> baresip is able to register with two outbound proxies and supports gruu
> (below). What else is needed?
> On 30 Jun 2021, at 09:49, Juha Heinanen wrote:
>
> Olle E. Johansson writes:
>
>> Full implementation of SIP outbound is the only solution close to
>> solving this problem in the IETF standards.
>> However, I have seen no single SIP client that have implemented this,
>> even though
> On 30 Jun 2021, at 09:10, Shahid Hussain wrote:
>
> Hi,
> Websocket module documentation has a code reference to use aliases for SIP
> routing. However, aliases will not work in the following setup and situation.
> 1. Kamailio is configured with active and standby node
> 2. Ping is
Hi,
Websocket module documentation has a code reference to use aliases for SIP
routing. However, aliases will not work in the following setup and
situation.
1. Kamailio is configured with active and standby node
2. Ping is implemented from webclient and kamailio responds with pong.
3. Two clients
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