To make this more interesting.. I actually don't get the $json variable in
a failure route either. There definitely seems to be something up with this
variable type's scope.
For now, I save the json into a dlg_val, and restore as needed. However,
it'd be nice to let it persist. Is there a way to d
Not sure if anyone else experienced this but "server_header" and
Polycom don't play nice when it comes to registering. Setting
"server_signature = no" allows SIP messaging to resume normally.
PS I can test further if anyone likes.
N.
On 1/9/13, Nick Khamis wrote:
> Duane,
>
> Thank you so much
Duane,
Thank you so much for your response! I temporarily commented out
"server_header="example.test.com", and we have two way communication
happening now. Not quite sure why sine the "server_header" is not
"opensips.example.com", and is actually a FQDN (i.e, known to public
DNSs).
Will look into
Sorry. After looking at this more I see that this is a siptrace from the
phone itself so the 200 OK is obviously making it to the phone. So yeah
your "opensips.example.com." might be whats messing it up.
Look at this example of a 200 OK
http://www.rfc-ref.org/RFC-TEXTS/3261/chapter24.html
What
Hahah, trying so hard to make everyone (in this case internal/external
UAs) Happy! The external (i.e., outside of the network) phones are all
working nicely. It's just the phones within the network that are
complaining. Looking at the NAT box, I see that they have 5060 port
forwarded to the OpenSIP
So you say you "closed gaping holes". Are you saying that the local
polycom phone are behind a firewall? If they are behind the firewall then
you need to figure out why the 200 OK is not making it to the polycom
phones. It could be because Polycom doesn't support rport (just a shot in
the dark f
Hello Everyone,
After scrambling to close up some gaping holes and getting RTP
relaying some media,
all of which is playing so nicely together. I have been accused of
breaking the "REGISTER"
functionality on the local Polycom phones. The remote ones all seem to
be ok but for some reason, I have th
Hi,
The same happens sometimes with INVITE requests, if it takes to long the UA
might start retransmitting.
I solve this by sending a provisional reply right upon receiving the request
if (is_method("INVITE"))
{
sl_send_reply("100", "Trying -- your call is important to us");
}
And then us
Hi Bogdan,
Understood. Thanks again!
Mariana.
On Wed, Jan 9, 2013 at 5:09 PM, Bogdan-Andrei Iancu wrote:
> **
> Hi Mariana,
>
> no, you do not need to create the INVITE trasaction in advance - do it as
> usually : at the end, on t_relay(). Because if the INVITE transaction does
> not exist (no
Hi Mariana,
no, you do not need to create the INVITE trasaction in advance - do it
as usually : at the end, on t_relay(). Because if the INVITE transaction
does not exist (not yet relayed), the t_check_trans() for CANCEL will
fail and CANCEL will be dropped (not relayed). Only one of the follo
Hi Chen-Che,
The rtpproxy cannot help you with such an algorithm. It knows which rtpp
are alive, but has no clue on the load.
What you could try is to use rtpproxy in combination with dialog. Put
each rtpp in a different rtpp set and make each set to correspond to a
dialog profile - use dial
yes, using that MI function make sense only if you have an already
running server that needs to take over the calls (like an active-active
setup or cluster setup).
Also be careful as moving the dialog from one server to another should
be done with preserving the IP of the server, otherwise in-
Hey All,
I'm trying to add $json variables to db_extra. I'm 100% sure the variables
are set as I'm using them all over my script. However, when acc wants to
write, I get errors like this:
Jan 9 18:10:52 development /usr/local/sbin/opensips[10469]:
ERROR:json:pv_get_json: Variable named:cdata1 no
Some one here know how i can download the .dsc files of opensips packages
for debian?
I need to rebuild the package, but if i can download the dsc will be more
easy.
Sorry my english,
Tnks...
___
Users mailing list
Users@lists.opensips.org
http://lists.
Happy New Year to you too.
Maybe it is doing default STUN to the SIP server :)
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 01/09/2013 05:55 PM, Филиппов Юрий wrote:
Thank you Bogdan,
Will do it definitely.
Strange thing, the client I'm
Hi Bogdan-Andrei,
Thanks for your reply. I am sorry that I did not state my problem clearly.
In my scenario, one SIP server communicates with 10 RTP proxies. When
receiving
an INVITE request, the SIP server will look for one RTP proxy satisfying the
following criteria
1) the RTP proxy is aliv
Hi Bogdan,
So I understand that the only way to recognize the transaction of a CANCEL
message received before relaying the INVITE is to create the transaction
using t_newtran() just after receiving the INVITE, but this would affect
some of our features.
In this case, we´ll have to work on the tim
Hi,
If the second server is started on-demand (e.g. keepalived) the dialogs
are loaded into memory from the DB (where the other opensips stored the
dialogs , by db_mode realtime..).
So there would be no need to use dlg_db_sync in this simple failover
scenario, right?
Best regards
Max M.
Just convert the ocn deck to npanxx and continue to roll on as usual
-Brett
On Tue, Jan 8, 2013 at 3:22 PM, Jamuel Starkey wrote:
> Hi,
>
> A few carriers that we using here domestically (in the USA) are now
> offering their termination rate decks by OCN (operating company name
> identifier) as
See the MI command dlg_db_sync
http://www.opensips.org/html/docs/modules/1.8.x/dialog.html#id295605
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 01/09/2013 02:59 PM, samuel wrote:
My point was exactly what you state in the last sentence
Hi Mariana,
According to RFC3261, the CANCEL processing (as cancelling the call) is
not up to a proxy, but up to the end devices. So a proxy has to simply
relay all received info (INVITE and CANCEL) - it cannot terminate the
call by itself.
So you have to relay both INVITE and CANCEL and let
Hi Matt,
The problem is your "group" column is not integer, but char (as type).
Permissions module expects an integer value there.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 01/04/2013 04:19 PM, l...@mattn.com wrote:
Hi Guys,
First po
Hi Julien,
Currently you cannot push a MI command from script (even this was
discussed several times).
So, what you try to do is, from script, in the context of one dialog, to
terminate another dialog ?
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutio
My point was exactly what you state in the last sentence about failover
scenario. Is there some way to use current dialog module to handle these
scenarios?
Thanks a lot,
Samuel
El 09/01/2013 12:50, "Bogdan-Andrei Iancu" va
escriure:
> **
> Hi Samuel,
>
> For dialog module, the primary storage (f
Hi Larry,
Most of the scripts do already have support for Messaging - there is
nothing special to add there. Just give it a try and see if it works or not.
For more on scripting, see the webinars from the website:
http://www.opensips.org/Resources/Webinars#toc9
Regards,
Bogdan-Andrei Ian
Hi Peter,
You cannot do that directly, but you can use the exec module to run an
extern script to do the rpc call
http://www.opensips.org/html/docs/modules/1.8.x/exec.html
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 01/07/2013 12:59 A
Hi Chen-Che,
opensipsctl fifo nh_show_rtpp
http://www.opensips.org/html/docs/modules/1.8.x/rtpproxy.html#id293424
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 01/08/2013 05:33 PM, microx wrote:
Hi all,
Can anyone give me a suggestion that
Hi Nick,
The linking between the dialplan and drouting must be done by you, at
script level, to reflect whatever you try to achieve - this is so
because of flexibility reasons.
Now imagine you use DR to determine what kind of number the user has
dialed - like national , international, etc (s
Hi Jamuel,
Currently there is no way to do http queries directly from OpenSIPS
script, except maybe playing with db_http module - a module that allows
you to do SQL like queries via HTTP
(http://www.opensips.org/html/docs/modules/1.8.x/db_http.html).
In short / medium term we are planing to
Hi Mariana,
If CANCEL hits opensips while the the INVITE is still under processing
(in a different process), the INVITE transaction will not exist yet (it
is created by t_relay), so the t_check_trans() will return false ->
script will exit without doing anything for the CANCEL - more or less
Hi Samuel,
For dialog module, the primary storage (for dialog info) is all the time
the mem cache. DB is only a secondary storage and data is flushed from
mem to DB. At runtime, opensips never reads from DB, but only from mem
(the primary storage). (db mode REALTIME means the DB storage is upd
Hi Flavio,
I've checked but, as far as I know, this feature only allows to distribute
the profiles, not the data of the dialog themselves.
Doing some tests, I have 2 instances of opensips sharing the same
database. I've also set the timeout to a low value and make several calls.
Stopping an opens
Hi Samuel,
I suggest you investigate the new feature called distributed dialog
profiles http://lists.opensips.org/pipermail/users/2012-February/020657.html
Flavio E. Goncalves
www.sippulse.com
2013/1/7 samuel
> Hi folks,
>
> I'm started reading about dialog module and how to use it in a dis
Hi Mariana,
There is the t_flush_flags to push changes in the flags after t_newtran().
For CANCEL, if you call t_check_trans, you can relay the CANCEL
automagically based on the transaction. It is like a shortcut. It is in the
default script.
if (is_method("CANCEL")) {
if (t_check_trans
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