Hello,
We are using Kamailio's perl module and want our custom library to set the
body on a SIP message.
Is it possible to set a SIP message body from within Perl, or can we return
a string to the Kamailio configuration to then use with set_body()?
Thanks for any advice.
--
David Cunningham
/2012 07:36 AM, David Cunningham wrote:
Hi Vlad,
This was on OpenSIPS 1.4.5, so pretty old. It was a once-in-a-year event
so I'm not sure if we're going to be able to do any useful debugging on
this system unfortunately.
Does 1.8 have a known fix for this error?
Thank you.
On 17 October
,
Vlad Paiu
OpenSIPS Developerhttp://www.opensips-solutions.com
On 10/15/2012 02:46 AM, David Cunningham wrote:
Hello,
We recently experienced a database error Commands out of sync; you can't
run this command now. I understand this is a known issue with presence.
Can anyone advise what
: driver error on query: Commands out
of sync; you can't run this command now
Oct 12 10:34:01 myhost /sbin/opensips[10735]: ERROR:core:db_do_query: error
while submitting query
Oct 12 10:34:01 myhost /sbin/opensips[10735]: ERROR:auth_db:get_ha1: failed
to query database
--
David Cunningham
Hello,
I believe there was a fix for the bug which causes a crash with:
/sbin/opensips[27869]: CRITICAL:core:anchor_lump: offset exceeds message
size (902 900) aborting...
Was this fixed in the 1.4 branch and which version would we need to have the
fix?
Thank you!
--
David Cunningham
.
.
--
David Cunningham, Voisonics
http://voisonics.com/
US toll-free: +1 888 842 2720
UK: +44 (0) 20 3298 1642
Australia: +61 (0) 2 8063 9019
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
.
Anyone got any advice?
Thanks,
--
David Cunningham, Voisonics
http://voisonics.com/
US toll-free: +1 888 842 2720
UK: +44 (0) 20 3298 1642
Australia: +61 (0) 2 8063 9019
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin
of the docs, this is the opposite
of fix_nated_contact.
Thanks for any advice.
--
David Cunningham, Voisonics
http://voisonics.com/
US toll-free: +1 888 842 2720
UK: +44 (0) 20 3298 1642
Australia: +61 (0) 2 8063 9019
___
Users mailing list
Users
and Foo is
still mentioned. Can anyone help?
Thanks,
2011/3/31 Stanisław Pitucha virap...@gmail.com
On 31 March 2011 12:11, David Cunningham dcunning...@voisonics.com
wrote:
The problem is that the destination phone is showing the call with the
Asterisk IP address in it's history, and so
Thanks for the ideas.
On Thu, Aug 5, 2010 at 12:18 AM, mayamatakeshi mayamatake...@gmail.com wrote:
On Thu, Aug 5, 2010 at 2:25 AM, David Cunningham dcunning...@voisonics.com
wrote:
Hello,
We have a system using the usrloc module (with dbmode=3) and for some
reason a few phones have
how to fix or work around this?
Someone advised it may be possible to tell usrloc to only use the most
recent contact entry for that phone, can anyone advise how it can be
done?
Thanks in advance!
--
David Cunningham, Voisonics
http://voisonics.com/
US toll-free: +1 888 842 2720
UK: +44 (0) 20
;transport=TCP
Server: Telviva SIP proxy
Content-Length: 0
Warning: 392 xx.xx.xx.xx:5060 Noisy feedback tells: pid=5271
req_src_ip=yy.yy.yy.yy req_src_port=3780 in_uri=sip:foo.com
out_uri=sip:foo.com via_cnt==1
--
David Cunningham, Voisonics
http://voisonics.com/
US toll-free: +1 888 842 2720
UK
you can set opensipsctlrc
to the pid file and fifo or socket for each. Makes things like dispatcher
reload work nicer.
Richard
On May 27, 2010, at 5:00 PM, Saúl Ibarra Corretgé wrote:
Hi,
On 27/05/10 22:57, David Cunningham wrote:
Hello,
I've seen mentioned that you can run
Bogdan,
Thank you for your help again! We found the source of the problem.
On Thu, May 13, 2010 at 7:05 PM, Bogdan-Andrei Iancu
bog...@voice-system.ro wrote:
Sorry David,
never used the perl stuff and have no idea how it works :(...
Regards,
Bogdan
David Cunningham wrote:
Thanks Bogdan
to ignore the RR indication - according to the RR
received in 200 OK, the UAC should sent the ACK to sip01 and not to
ast01 .
Regards,
Bogdan
David Cunningham wrote:
Hi Bogdan,
That sounds reasonable, but an ngrep trace on the opensips server is
showing the ACK from the phone is being
- this does not happen only when
using the exec module (which you do not have) - the only alternative is
that the perl scripts you are using are doing the fork (maybe some perl
function?) and do not properly terminate the extra procs...
Regards,
Bogdan
David Cunningham wrote:
Hello
Hi Bogdan,
That's correct, the ACK for the 200OK.
On Wed, May 12, 2010 at 8:46 AM, Bogdan-Andrei Iancu
bog...@voice-system.ro wrote:
Hi David,
David Cunningham wrote:
Hello,
We are using OpenSIPS with a Perl module which needs to branch SIP
messages to multiple Asterisk servers
.
Content-Length: 0.
On Wed, May 12, 2010 at 9:59 AM, Bogdan-Andrei Iancu
bog...@voice-system.ro wrote:
In such a case, from SIP point of view, there is a single ACK
corresponding to the winning branch (which sent the 200 OK).
Regards,
Bogdan
David Cunningham wrote:
Hi Bogdan,
That's correct
for your help!
--
David Cunningham, Voisonics
http://voisonics.com/
US toll-free: +1 888 842 2720
UK: +44 (0) 20 3298 1642
Australia: +61 (0) 2 9037 2180
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo
, nat_bflag, 1 )
modparam( usrloc, timer_interval, 5 )
Thank you!
On Wed, Apr 28, 2010 at 4:53 PM, Bogdan-Andrei Iancu
bog...@voice-system.ro wrote:
Hi David,
by chance, using the exec module ?
Or, can you list the modules you are using ?
Regards,
Bogdan
David Cunningham wrote:
Hello
the parent process of the zombie procs - check
with ps and correlate (for the name) with opensipsctl fifo ps
I guess the parent of the zombies should be the attendant proc . BTW,
are you running with the respawn patch ?
Regards,
Bogdan
David Cunningham wrote:
Hello,
Thanks again for your
.
Much appreciate your help.
On Thu, Apr 22, 2010 at 6:07 PM, Bogdan-Andrei Iancu
bog...@voice-system.ro wrote:
Hi David,
David Cunningham wrote:
Hello,
Thank you for the reply!
The log doesn't say anything useful, just Listening on and then the
UDP and TCP IP address and port, and Aliases
?
Regards,
Bogdan
David Cunningham wrote:
Hello,
We have a server which is creating a lot of defunct OpenSIPS
processes. An example process tree is below (from ps -ef --forest).
I have no idea where to start looking for the cause of this. Any
suggestions very welcome!
user 7811 1
03:19 ?00:00:00 \_ /sbin/opensips -m 256
-P /var/run/user/opensips.pid
Cheers,
--
David Cunningham, Voisonics
http://voisonics.com/
US toll-free: +1 888 842 2720
UK: +44 (0) 20 3298 1642
Australia: +61 (0) 2 9037 2180
___
Users mailing list
24 matches
Mail list logo