[OpenSIPS-Users] Return body/string from Perl library

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

Re: [OpenSIPS-Users] Error Commands out of sync; you can't run this command now

/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

Re: [OpenSIPS-Users] Error Commands out of sync; you can't run this command now

, 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

[OpenSIPS-Users] Error Commands out of sync; you can't run this command now

: 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

[OpenSIPS-Users] What version for anchor_lump: offset exceeds message size fix?

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

[OpenSIPS-Users] Why do we get 481 bad request, no route tag?

. . -- 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

[OpenSIPS-Users] OpenSIPS temporarily loses connection to MySQL?

. 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

[OpenSIPS-Users] Opposite of fix_nated_contact?

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

Re: [OpenSIPS-Users] Opposite of fix_nated_contact?

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

Re: [OpenSIPS-Users] Multiple contact entries

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

[OpenSIPS-Users] Multiple contact entries

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

[OpenSIPS-Users] Registrations from different ports?

;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

Re: [OpenSIPS-Users] Multiple opensips instances

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

Re: [OpenSIPS-Users] OpenSIPS Perl module branching and logging

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

Re: [OpenSIPS-Users] OpenSIPS Perl module branching and logging

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

Re: [OpenSIPS-Users] OpenSIPS defunct processes

- 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

Re: [OpenSIPS-Users] OpenSIPS Perl module branching and logging

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

Re: [OpenSIPS-Users] OpenSIPS Perl module branching and logging

. 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

[OpenSIPS-Users] OpenSIPS Perl module branching and logging

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

Re: [OpenSIPS-Users] OpenSIPS defunct processes

, 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

Re: [OpenSIPS-Users] OpenSIPS defunct processes

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

Re: [OpenSIPS-Users] OpenSIPS defunct processes

. 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

Re: [OpenSIPS-Users] OpenSIPS defunct processes

? 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

[OpenSIPS-Users] OpenSIPS defunct processes

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