Hello,
hmm, they are multiple of 8, so they should be aligned to 64bits.
Or maybe the 'block' variable value is not aligned to 8bytes ...
Have you done any sip traffic via this kamailio instance. Is all ok at
runtime?
Cheers,
Daniel
On 14/01/16 08:01, Spencer Thomason wrote:
> Hi Daniel,
> See
Hi Daniel,
See below:
(gdb) p group->var_offset
$1 = 64
(gdb) p mapping[i].offset
$2 = 56
Thanks!
Spencer
> On Jan 13, 2016, at 10:18 PM, Daniel-Constantin Mierla
> wrote:
>
> Hello,
>
> can you get the values for group->var_offset and mapping[i].offset in
> frame 0?
>
> Cheers,
> Daniel
>
Hello,
can you get the values for group->var_offset and mapping[i].offset in
frame 0?
Cheers,
Daniel
On 14/01/16 05:41, Spencer Thomason wrote:
> Hello,
> I’m trying to get Kamailio running on Solaris 11 SPARC64 and I’m receiving a
> bus error on shutdown. If needed, we can make SPARC hardwar
Hello,
I’m trying to get Kamailio running on Solaris 11 SPARC64 and I’m receiving a
bus error on shutdown. If needed, we can make SPARC hardware available for
testing.
Thanks,
Spencer
Core was generated by `/opt/kamailio/sbin/kamailio -f
/opt/kamailio/etc/kamailio/kamailio.cfg -P /syst'.
Pr
Hello,
it looks like you have a symmetric nat router, so the allocated port is
randomly selected.
If you don't control the nat router to set a static forwarding rule or
it doesn't provide the option to set static forwarding, then you are
pretty much left with sniffing the traffic to discover the
Kamailio is not generating another reply if freeswitch is sending one
(unless enforced in config file).
Are you sure the 404 is sent by Kamailio? What do you mean by
"freeswitch is generated UN-allocated number"? Isn't free switch sending
a sip reply in this case?
Cheers,
Daniel
On 13/01/16 22:
Kamailio SIP is sending 404 not found since freeswitch is generated
UN-allocated number, the call got rejected and goes to voicemail.
Thank you again for your help
Abdul
From: Daniel-Constantin Mierla
Sent: Wednesday, January 13, 2016 9:06 PM
To: malik sherif;
Hi Sergey,
thanks for testing and the feedback. The patch is also useful to have
around for those than want to get the feature in 4.3 by one step, not
cherry picking each commit.
One more thing: are the local message properly captured? I mean the
replies generated with sl_send_reply()/t_reply() a
Hello,
to complete: if you want destination to not to be pinged anymore and not
used for routing, you have to disable (deactivate) it, not to set it in
state inactive. There are three states:
- active
- inactive (at any time can become active based on ping activity)
- disabled (it is going to be
What application is sending the 404?
Cheers,
Daniel
On 13/01/16 21:43, malik sherif wrote:
>
> Any hint as to how to correct this issue?
>
>
> 1 404 /UNALLOCATED_NUMBER/ Unallocated (unassigned) number [Q.850
> value 1] This cause indicates that the called party cannot be reached
> because, altho
Perhaps this is a security message embedded in freeswitch code to
instruct the admin to change default values coming in configs, otherwise
you expose yourself to being hacked straight away.
So open that file and change the default password. I have no clue about
what is it, I am just interpreting t
I am not familiar with your freeswitch config and what freeswitch should
do. Also, I don't deal much with freeswitch in order to assist you with
it, maybe other people here can help, if not, you can eventually ask on
freeswitch mailing list.
Cheers,
Daniel
On 13/01/16 18:15, malik sherif wrote:
>
Hello,
if you work with large data records that you want to get via kamcmd,
then you have to increase the buffer sizes of ctl module -- see the
readme of that module to identify the parameters to adjust.
Cheers,
Daniel
On 11/01/16 20:22, Vik Killa wrote:
> It seems dialplan module does not work
Hello,
realoading is done in a separate structure, which is then swapped with
the old one and the old one is freed -- like you said, a change of the
pointer to the new structure.
Cheers,
Daniel
On 11/01/16 19:49, Vik Killa wrote:
> Hi Daniel,
> We are experimenting with the dialplan module.
> I
Hello,
as an extra hint on top of Carsten's remarks: search your configuration
for sl_send_reply("500", "Service Unavailable") or t_reply/send_reply
with same parameters. Then you can identify better why that reply is
sent. As Carsten pointed, such reply code and reason text is not coming
from sou
Any hint as to how to correct this issue?
1 404 UNALLOCATED_NUMBER Unallocated (unassigned) number [Q.850 value 1] This
cause indicates that the called party cannot be reached because, although the
called party number is in a valid format, it is not currently allocated
(assigned).
___
is there a new to edit vars.xml file? I haven't touched this file but one of
the warning about default password
016-01-13 05:37:29.572184 [CRIT] mod_dptools.c:1638 Open
/usr/local/freeswitch/conf/vars.xml and change the default_password.
From: sr-users on beha
Hello,
I finally were able to run my Kamailio behind NAT but in order to
accomplish that I included:
listen=udp:SOURCE-IP:5060 advertise PUBLIC-IP:52548
52548 is the port my internet router change when doing NAT (5060->52548). I
found this port sniffing traffic
Conclusions at this point are:
-
Thanks again Daniel for replying.
Now the call is failing with 404 not found and goes to voicemail. I was calling
between 7632689991 and 7632689993, I looked the extensions on freeswitch, and
look OK but it is possible I might have missed something. Freeswitch issues the
following errors. Thank
hi. Sven.
1. Have you set modparam("dispatcher", "ds_probing_mode", 1)? If so, it
will be moved to PROBING mode after ds_ping_interval
2. 'X' means "not any set". just a cross ;)
2016-01-13 16:17 GMT+03:00 Sven Neuhaus :
> Hello,
>
> when trying to set a destination to inactive using dispatch
On 01/13/2016 04:26 AM, riko nir wrote:
Hello,
A call from a remote webrtc client is coming to (opensips+rtpengine).
The media streams from the webrtc client is multiplexed. Can I use
rtpengine to demultiplex the multiplexed streams and send it to other
end as de-multiplexed SRTP traffic . This
On 01/13/2016 02:37 AM, riko nir wrote:
Hi, Thanks for the answer.
Do you have any options for sending this keys to opensips somehow, by
modifying the code in rtpengine and in opesips script file?
I don't know much about Opensips and so can't provide guidance about how
to pass these values ba
Hello,
when trying to set a destination to inactive using dispatcher.set_state
after setting the state to "i", when I do a "dispatcher.list" I see the
state as "ix". The "x" state appears not to be documented.
After a short time (seconds) the flags return to state "ip" (probing).
According to the
When you are running Kamailio behind a NAT you should use advertise
parameter of listen address. According to documentation " A typical use
case for advertise address is when running SIP server behind a
NAT/Firewall, when the local IP address (to be used for bind) is different
than the public IP ad
Yes, the Graphite server is openly available: http://graphite.wikidot.com/
Greetings,
Paul
Am 13.01.2016 um 03:53 schrieb Juha Heinanen:
rtpengine README.md mentions "graphite statistics server". is that
server openly available somewhere?
-- juha
_
Hello,
A call from a remote webrtc client is coming to (opensips+rtpengine). The
media streams from the webrtc client is multiplexed. Can I use rtpengine to
demultiplex the multiplexed streams and send it to other end as
de-multiplexed SRTP traffic . This is because, the other end handle the
srtp
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