What distro are u using? This sounds more like a problem in the init
script, where it tries to restart opensips before the last process has
existed, which then fails to start. Try adding - - retry option to
start-stop-daemon.
On May 7, 2015 4:01 AM, Max Mühlbronner m...@42com.com wrote:
Hello Everyone;
I want to be able to forward calls from Asterisk to a phone registered with
OpenSIPs, can anyone give me some information on how this can be done? Or a
link to the info I would need to do this.
Thanks for any help
Juls
___
Users
Comparing opensipsctl ps and the pid of the remaining process shows
it's the attendant process.
root@opensips1:/etc/opensips# opensipsctl ps
Process:: ID=0 PID=25935 Type=attendant
root@opensips1:/etc/opensips# /etc/init.d/opensips restart
Restarting opensips:
Just define a peer in sip.conf
then Dial(SIP/peer_name/OpensipsAccount) on extensions.conf. Very simple.
regards,
s
Il 07/05/2015 08:32, Julian Kay ha scritto:
Hello Everyone;
I want to be able to forward calls from Asterisk to a phone registered
with OpenSIPs, can anyone give me some
Hi,
Debian. testing now. Could this have any other undesired side-effects?
BR
Max M.
On 07.05.2015 10:12, Podrigal, Aron wrote:
What distro are u using? This sounds more like a problem in the init
script, where it tries to restart opensips before the last process
has existed, which then
Kneeoh has already started an earlier thread related to this problem [1]
This should be moved to the GitHub tracker [2]. We need either a
relevant SIP trace, or a way of explaining the NULL socket behaviour /
replicating the errors ourselves.
[1]
Hello again!
And the winner of the last OpenSIPS 2.1 Bug Hunt edition[1] is Dan
Bogos(@danbogos)[2] for reporting cases where ACC events were not
properly raised. Congratulations!
Since the OpenSIPS 2.1 Final Release due date is today, there won't be
any Bug Hunt editions for OpenSIPS 2.1.
Hi,
I have installed 2.1 but i didn't understand use of Async mode. I was
reading article http://www.opensips.org/Documentation/Script-Async-2-1
But don't understand how and where i can use in my script because in my
script i am doing some SQL operation but i don't know how it can fit and
how
Hi, all!
Good news! We have just released three new minor releases for OpenSIPS:
1.11.5, 1.10.5 and 1.8.8. All three versions are stable and ready for
production use. Since they contain the latest bug fixes, we strongly
recommend you to upgrade your current instances!
Thank you all for your
Hi Satish,
Do you do DB queries using avp_db_query() ? If yes, see the example:
http://www.opensips.org/Documentation/Script-Async-2-1#toc3
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 07.05.2015 18:39, Satish Patel wrote:
Hi,
I have
Hi Max,
The attendant is responsible for doing the cleanup for all modules and
core - this is why it may take longer to complete. Do you have memory
logging or some expensive DB flushing on shutdown ?
BTW: see you in Amsterdam for the OpenSIPS Summit !
Regards,
Bogdan-Andrei Iancu
OpenSIPS
Hi Rahul,
As per RFC3261, the stateful CANCELs are hop-by-hop - which means each
hop consumes the incoming CANCEL and generates a new one for the next hop.
This is why the headers do not propagate.
What kind of headers are looking to be passed further ?
Regards,
Bogdan-Andrei Iancu
OpenSIPS
Hi Dan,
As in ACC you cannot do START/STOP and CDRs in the same time, I suggest
to use the CDR event for accounting purposes and for monitoring the
dialog status take a look at the E_DLG_STATE_CHANGED event:
http://www.opensips.org/html/docs/modules/2.1.x/dialog.html#id297207
Regards,
Hi Mickael,
No, you cannot force it - it is against the TCP philosophy :) .
Which entity closes the TCP conn - OpenSIPS or the SBC ?
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 06.05.2015 11:02, Mickael Marrache wrote:
Forget the
Hi Bogdan,
I want to use opensips as stateless proxy that's why I tried forward() which is
causing the branch issue as shown below, the branch in both the Via headers
are same.
CANCEL sip:@IP:PORT SIP/2.0
From: Alan
Hi Staish,
If you do IP base auth, I guess you do not use the subscriber table at
all, right ? SO you can set the RPID directly in the address table (I
assume you use it for the IP auth) in the attr fields, so the RPID will
be directly available for you
Regards,
Bogdan-Andrei Iancu
Hi Kneeoh,
The dialog replication is done assuming that both opensips servers do
share the listening interface (via vrrp, heartbeat, etc). Do you
different listening IPs on the 2 opensips instances ?
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
Rahul,
As per RFC, in stateless mode there is no branch (for the newly added
VIA) - as branch is transaction oriented. And the RFC3261 recommends to
reuse the previous VIA hdr (if exists).
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On
Hi Aleksey,
Try do the a manual comparison:
if ($au != $fU)
see http://www.opensips.org/Documentation/Script-CoreVar-1-11#toc7
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 24.04.2015 17:35, Aleksey Fedorov wrote:
Hi!
How to check
Hi all,
The day is here - 2.1 RC becomes 2.1.0 stable.
After 1.5 months of a collective effort of testing, reporting and
fixing, 2.1 because stable, production ready. See our bug-hunting
contest (with many thanks to everybody involved):
http://www.opensips.org/Community/BugHuntContest
Hi Bogdan, Both Opensips hosts are set to use corosync/heartbeat to failover
the two IPs in our config. Both hosts are set to non-localbind and opensips is
explicitly listening on both of the VIPs. This is why I'm confused. It seems
that everything is configured correctly yet I'm getting these
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