Hi all,
I'm trying to use the keepalive / ping option in OpenSIPS Loadbalancer
module, but asterisk always responds with a 404 not found. Has anybody
managed to get this working with Asterisk 1.6.1.X + OpenSIPS 1.6.2 ?
Thanks!
Alex
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Yes, you are right. They're the same.
Had this bug been fixed yet? It seems still open now.
Best Regards,
Liu Fan
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Hi Rajib,
each SIP message (request or reply) is read from the network by a child
(any of them) and it will be processed (to the end) only by that child.
A child processes only one message at a time..
Regards,
Bogdan
rajib deka wrote:
> Hi All,
>
> Can anybody please explain me the use of op
Thanks.
Base svn revision 6624, I check files in db_berkeley/opensips/ again. I
found the files which have no METADATA_KEY value as below:
dialplan
dr_gateways
dr_groups
dr_rules
globalblacklist
nh_sockets
userblacklist
maybe you need fix them also.
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> I have the following problem
>
> With install OpenSIPS 1.6 and Control Panel 4.0.
>
> When try the access the Control Panel didn't accept the user o password
> (Invalid Username / Password)
>
> With the commands opensips-c, monit-t said that the configuration files
> are fine.
>
> I can regi
Hi,
Does pua/rls modules properly support pidf-diff+xml?
When subscribing to a rls service (that contains an embedded resource list)
I am able to get presence state notifications for buddies that publish their
state using pidf full (pidf+xml).
PUBLISH request is as below:
PUBLISH sip:5...@raftma
Hi Dimitri,
reloading the data should not delete the dialog profiles, unless you
change the name of the resources (in the destination definition).
Is this your case?
Regards,
Bogdan
Dmitri G. wrote:
> Hi,
>
> Is there a way to keep the number of the dialogs after running the
> lb_reload comman
Sorry I missed this email,
I'll think about using db_virtual, but back to global vars issue, for a
test I defined just a global counter variable, and created new function
inside siptrace module, so every time a siptrace function is called I
also call this function and do counter ++. since count
Is it possible to download the movie files themselves? Not everyone can
or is allowed to use Flash.
Thank you,
noah
Bogdan-Andrei Iancu wrote:
> The recording of this webinar is now available under :
>http://www.opensips.org/html/docs/video/webinar005/
>
> Regards,
> Bogdan
>
>
>
> B
I have the following problem
With install OpenSIPS 1.6 and Control Panel 4.0.
When try the access the Control Panel didn't accept the user o password
(Invalid Username / Password)
With the commands opensips-c, monit-t said that the configuration files
are fine.
I can register phone.
The
On Mar 10, 2010, at 7:49 PM, Thomas Gelf wrote:
> Adrian Georgescu schrieb:
>>> * How does it figure out it's reflective candidate if I didn't
>>> configure a STUN server?
>>
>> The stun server must be in DNS of the caller domain or you can add
>> them manually in Account -> Advanced -> Nat Trav
Hi Bogdan,
I'm running OpenSIPS 1.6.1
version: opensips 1.6.1-notls (i386/linux)
flags: STATS: Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, SHM_MEM,
SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
MAX_URI_SIZE 1024
Bogdan,
I am using the fix_nated_contact() because if I don't, it tries to register
with the private IP of the device on my network... still not the public IP
of it.
Perhaps I'm attacking this the wrong way, maybe you can clarify for me what
I actually want to do. The end I'm trying to reach is t
Tyler,
The script logic (for REGISTER) is a bit broken - you authenticate the
REGISTER, you save it locally, send a reply to UAC and then relay the
REGISTER to asterisk; which will send you back a reply, so you will end
up with 2 replies back to UAC :)
Regarding the contact part, I see you are
Hi,
I guess what you are describing is the same as this bug report:
https://sourceforge.net/tracker/?func=detail&aid=2941492&group_id=232389&atid=1086410
Am I right ?
Regards,
Bogdan
liuf wrote:
> I'm sorry. My real question is about where to set dialog pointer to the new
> transaction of
The recording of this webinar is now available under :
http://www.opensips.org/html/docs/video/webinar005/
Regards,
Bogdan
Bogdan-Andrei Iancu wrote:
> Next webinar is scheduled for 25th of February 2010.
>
> The topic is "Explaining the default script" - A detailed going though
> and e
On Mar 10, 2010, at 2:49 PM, Thomas Gelf wrote:
> As I work on Linux only
If you get the latest SIP SIMPLE command line clients (they work on
Debian at least) they all have the ICE feature and display the outcome
of ICE negotiation when RTP stream is active.
http://sipsimpleclient.com/wiki/
2010/3/10 Pascal Maugeri :
> Hi Anca
> We just tested the patch you sent us but we don't see any difference.
Could you paste a SIP flow for the SUBSCRIBE and NOTIFY?
--
Iñaki Baz Castillo
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> * How does it figure out it's reflective candidate if I didn't
> configure
> a STUN server?
The stun server must be in DNS of the caller domain or you can add
them manually in Account -> Advanced -> Nat Traversal section
> * Configuring a STUN server AND enabling ICE seems to cause
> pr
Bogdan, or anyone,
How can I relay the public IP of my UAD off to Asterisk instead of the IP of
the OpenSIPS server?
Everything I've tried has not changed anything. I am using
fix_nated_contact(), as before using it, it was trying to register the
private IP of the UAD.
-Tyler
-Original Mes
Hi Pascal,
Please try this revised patch. It is also against svn, take a new copy
of the tm module and apply it.
Thanks and regards,
--
Anca Vamanu
www.voice-system.ro
Pascal Maugeri wrote:
Hi Anca
We just tested the patch you sent us but we don't see any difference.
Just in case, we re
Hi Anca
We just tested the patch you sent us but we don't see any difference.
Just in case, we recompiled the whole server (first we updated to rev 6700)
and applied the patch you kindly sent us:
svn update
...
patch -p0 < tm_proto_fix.patch
make all
make install
service opensips restart
The N
Thank you Bogdan. I am clear now.
Regards
Rajib
On Wed, Mar 10, 2010 at 5:20 PM, Bogdan-Andrei Iancu wrote:
> Hi Rajib,
>
> I meant "to the end" for that SIP message - a SIP message is a request
> or a reply.
>
> For example, an INVITE request may be processed (entirely) by proc P1,
> while its
Hi,
I've configured OpenSIPS + FreeRadius Accounting and Authentication setup,
which was implemented success full. Using Authentication via Freeradius can
anybody tell me how can I populate data on radius database tables? Mean
what sort of values do I required for its tables so I can authenticat
Hi Mike,
RT_Txx are the retransmission timers (as defined per RFC3261). and you
can change them via:
http://www.opensips.org/html/docs/modules/1.6.x/tm.html#id228598
It seams that the delay comes from RADIUS level, when doing RADIUS acc:
Mar 10 00:09:25 OpenSIPS /sbin/opensips[13343]
Hi Dimitri,
Ok, so you do lb_reload just because you need to add some more
destinations in the balancer. You do not change the properties of the
already existing destination - this was what I was interested in.
Let me run some checks to see what about. BTW, what opensips version are
you using
Hi Rajib,
I meant "to the end" for that SIP message - a SIP message is a request
or a reply.
For example, an INVITE request may be processed (entirely) by proc P1,
while its 200 OK reply may be handled by proc P2.
Regards,
Bogdan
rajib deka wrote:
> Thank you Bogdan. When you say 'to the end
Hi,
you are right - I made a fix to solve this issue. SVN is update on
trunk, 1.6 and 1.5 branches.
This fix will be part of the new 1.6.2 release from tomorrow.
Thanks and regards,
Bogdan
liuf wrote:
> I know this source directory, I have read all file in this directory before.
>
> Can I put
Hi Iñaki,
Iñaki Baz Castillo wrote:
> El Martes, 9 de Febrero de 2010, Bogdan-Andrei Iancu escribió:
>
>> Iñaki Baz Castillo wrote:
>>
>>> El Martes, 9 de Febrero de 2010, Bogdan-Andrei Iancu escribió:
>>>
Hi Iñaki,
uh...you are totally right :D...
propagating a
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