yes, i am referring to ubuntu 18.04 sorry not being clear about it
it's an LTS so it would be good to support this
Kelvin Chua
On Sat, May 19, 2018 at 2:09 AM Henning Westerholt wrote:
> Am Freitag, 18. Mai 2018, 15:00:03 CEST schrieb Daniel-Constantin Mierla:
>
> > Hello,
>
> >
>
> > what is
I think the reason for the second issue listed below is that Kamailio never
receives the ‘200 OK’ from back from the INVITE to Asterisk. I have noticed
that starting sometime after version 14.6, Asterisk puts its Public IP-Address
in the Contact field where as in 14.6 it puts it Private IP-Addr
Am Dienstag, 15. Mai 2018, 19:47:54 CEST schrieb Wilkins, Steve:
> Where do I set this at?
>
> Also, I noticed that when adding $ru=$ru + ";transport=tcp", tcp does get
> appended to $ru, it just refuses to send over TCP connection.
> [..]
Hello,
have a look to the kamailio startup parameter,
So what actually happens when you do kamcmd cfg.set ? I assumed the variable
is set in shared memory somewhere and all the processes reference the same
shared memory variable.
What I'm seeing is that I set the value to 1 using cfg.set and then some time
later I do
sbin/kamctl kamcmd cfg.get f
Am Donnerstag, 10. Mai 2018, 12:33:19 CEST schrieb Amar Tinawi:
> recently i faced strange behaviour when dealing with ng-voice version of
> kamailio ims
> specially in P-CSCF node, the database "pcscf" dropped suddenly (deleted)
> without any command or action from user
>
> this happened many tim
Am Mittwoch, 16. Mai 2018, 18:06:25 CEST schrieb KamDev Essa:
> So I finally got to the docs on hunting under the TM Modules and there was
> an awesome write up and it is making a lot more sense now. Basic logic is
> to append_branch with your extensions and and manage serial and/or parallel
> for
Am Freitag, 18. Mai 2018, 15:00:03 CEST schrieb Daniel-Constantin Mierla:
> Hello,
>
> what is bionic in this case?
>
Hello,
he is probably referring to Ubuntu 18.04 LTS Bionic Beaver.
Best regards,
Henning
___
Kamailio (SER) - Users Mailing List
sr
Am Freitag, 18. Mai 2018, 09:19:05 CEST schrieb Andreas Granig:
> I'd like to inform you about Alcatel-Lucent Enterprise (ALE) acquiring a
> majority stake in Sipwise. Over the last 10 years, we at Sipwise were
> committed to contributing significant efforts to the open source telecom
> environment
I sent the log but I just wanted to clear up the issue.
This is a WebRTC to WebRTC call
* If I use Kamailio 5.2 (UDP) and Asterisk 14.6, the call works perfect!
* If I use Kamailio 5.2 (UDP) and Asterisk 15.3, the call connects but
disconnects after 30 seconds (Asterisk never received AC
I don't understand, are you able to send an INVITE to Asterisk via TCP?
Could you share a trace of the calls, masking the sensible information?
Cheers,
Federico
On Fri, May 18, 2018 at 4:56 PM, Wilkins, Steve wrote:
> I have tried using t_relay_to_tcp() and t_realy, both with and without
> set
I have tried using t_relay_to_tcp() and t_realy, both with and without setting
$ru = $ru + ";transport=tcp" right before the call.
I get different errors and failures depending on which way I make the call.
Which log would you want to see?
* I also wanted to point out that if I use UDP, th
Hi Aqs,
rtpproxy should start relaying immediately to the port specified in the
SDP. If the UA sends UDP packets from another source:IP port, it should
update internal session data, so that all subsequent packets will be
relayed to the proper IP:port. Otherwise it will continue to user the
sa
Hi,
if a new connection is created and you don't specify $fs or use
force_send_socket, again is the OS choosing the port.
The point is understanding here why a new connection is created instead of
reusing the one created for the INVITE.
Could please paste the logs, possibly at DEBUG level, when kam
The INVITE has no port assigned in ruri, so I assume it is using the default
port of 5060; the ACK does have 5060 in the ruri. Even if a new connection is
created shouldn’t it be using the listen tcp port since I have
tcp_reuse_port=yes?
Thank you!
From: sr-users [mailto:sr-users-boun...@list
Hello,
what is bionic in this case?
Cheers,
Daniel
On 13.05.18 11:26, Kelvin Chua wrote:
> any plans of adding support for bionic?
>
> Kelvin Chua
>
>
> ___
> Kamailio (SER) - Users Mailing List
> sr-users@lists.kamailio.org
> https://lists.kamailio.o
Hello,
can you provide more details about what you think is going wrong?
The value of this variable is updated by each process when that process
does some particular operations (e.g., receiving a sip message), it is
not propagated automatically to every kamailio process when you do the
update v
Hello,
are you doing lookup(location) for the BYE that is attempted to be
routed on the old connection?
Cheers,
Daniel
On 17.05.18 17:45, Idris AVCI wrote:
> Hi,
>
> We use kamailio as a backend for JSSIP based WebRTC UAs.
> We use SIP over Weboscket as transport and everythings work fine
> gen
Hello,
this sounds like the body for reply was already set. Can you enable
cfgtrace in debugger module and see what actions are executed in that
situation?
Cheers,
Daniel
On 17.05.18 18:22, Asgaroth wrote:
> Hi All,
>
> We are trying to query the json rpc server on our registrar for the
> conta
Can you print the logs when it tries to send the ACK?
Check also that the ACK ruri contains the same destination port which was
used for the INVITE, otherwise a new connection will be created.
Best regards,
Federico
On Fri, May 18, 2018 at 2:31 PM, Wilkins, Steve wrote:
> Thank you Alex and Fe
Thank you Alex and Federico,
I verified, and SO_REUSEPORT is defined on my OS. I am using Kamailio 5.2 and
I set ‘tcp_reuse_port=yes;’ and $fs; this has been to no avail as ‘ACK’s’ are
still using the high port number randomly assigned by Kamailio.
Thank you all for sharing your knowledge!
It looks a simple question but whats outgoing RTP packets destination? Is
CALLEE IP present on that NATed destination?
Confirm that RTP Proxy determined that NATed IPs properly.
What did you receive in 200OK from CALLEE?
Check if engaged RTP ports are open in the flow?
--
regards,
abdul basit
O
Hi Andreas,
I am glad to hear that your efforts over all these years materialized in
a stronger business venture for you and your partners. Congratulations!
I am confident that the open source components released and supported by
Sipwise are continuing to progress in terms of innovation and new
f
Corrigendum:
We are getting packets from CALLER*
Regards
Abbasi
On Fri, 18 May 2018 at 12:46 PM, Bilal Abbasi wrote:
> Actually they dont even get out of rtpproxy, we cant see that in the in
> sngrep/tcpdump.
> We are getting packets from callee but nothing going out of
> rtpproxy(talking about
Actually they dont even get out of rtpproxy, we cant see that in the in
sngrep/tcpdump.
We are getting packets from callee but nothing going out of
rtpproxy(talking about local dump)
Regards
Abbasi
On Fri, May 18, 2018, 12:13 Giovanni Tommasini - evosip
wrote:
> Hi Younas,
>
> when you have a c
Dear community,
I'd like to inform you about Alcatel-Lucent Enterprise (ALE) acquiring a
majority stake in Sipwise. Over the last 10 years, we at Sipwise were committed
to contributing significant efforts to the open source telecom environment,
resulting amongst others in the free and open sour
Hi Younas,
when you have a call if you make a trace in your sip server with sngrep or
tshark can you see the RTP packet ?
you says "I have kamailio server behind nat with rptproxy.", so is it
possible that the router in front your SIP server blocks the traffic? could
you have a trace there?
I mean
Am Donnerstag, 17. Mai 2018, 15:02:13 CEST schrieb Daniel Tryba:
> [..]
> > I wonder why these particular module parameters would break the
> > functionality though, I thought the idea was that each proxy would be able
> > to generate the nonce with a shared secret if it recieved a challenge
> > re
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