Hello
Sorry if I wasn’t clear.
We would like to implement a dialog stateless SIP proxy. Once the correct SIP
routing is found, we would like to allow the SIP-based elements to perform
midcall signaling directly with one another.
Best regards,
De : Users De la part de Bogdan-Andrei Iancu
Thank you Bogdan!
It's worth noting that, if using {s.escape.user}, it won't detect a SQL
injection, however, it may detect other potentially problematic
characters, so one then has to apply both checks individually, e.g.
if ( $fU != $(fU{s.escape.common}) || $tU != $(tU{s.escape.common}) )
Thanks, I'll check it out.
On Tue, 5 Dec 2023 at 16:29, Bogdan-Andrei Iancu
wrote:
> Then take a look at the call_center module:
> https://opensips.org/html/docs/modules/3.4.x/call_center.html
>
> Regards
>
> Bogdan-Andrei Iancu
>
> OpenSIPS Founder and Developer
>
Hi,
Your post is a bit confusing when comes to state (and statefull) - are
you talking about transaction statefull or dialog statefull ?
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
https://www.opensips-solutions.com
https://www.siphub.com
On 05.12.2023 13:01,
Andrew,
Please capture (or xlog the Contact hdr) the 200 OK reply you receive
from the callee. Most probably there is something weird with it, making
the ACK to be badly routed.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
https://www.opensips-solutions.com
I usually make a mix f.e. on OPTIONS, subscribe, notify I send 200 OK with
sl_send_reply.
INVITE handling I always do stateful.
Normally, the dependencies of a module are described in the module
documentation.
I hope this helps.
BR, Johan.
Op di 5 dec 2023 om 12:04 schreef :
> Hi opensips
Hi opensips experts,
We have always worked Opensips in a statefull mode and the first words in the
routing module description are "OpenSIPS is basically only a transaction
statefull proxy".
I've also seen that there was a stateless module "The SL module allows OpenSIPS
to act as a stateless
Then take a look at the call_center module:
https://opensips.org/html/docs/modules/3.4.x/call_center.html
Regards
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
https://www.opensips-solutions.com
https://www.siphub.com
On 05.12.2023 12:49, Prathibha B wrote:
Yes, you are right.
On
Hi,
Could you please detail a bit your case? what kind of advertising are
you using? per socket, global? ALso, an off-list pcap (covering both in
and out traffic for opensips) will help.
In regards the t_relay, as per docs, just do t_relay("no-auto-477")
Regards,
Bogdan-Andrei Iancu
Yes, you are right.
On Tue, 5 Dec 2023 at 15:04, Bogdan-Andrei Iancu
wrote:
> What do you mean by "free call taker in the group" ? something like in
> call centers?
>
> Bogdan-Andrei Iancu
>
> OpenSIPS Founder and Developer
> https://www.opensips-solutions.com
> https://www.siphub.com
>
>
Hi Razvan
The error I see is
No ack received for an extended period of time
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Hi Bogdan,
I have the same issue, i have just upgraded the opensips to 3.4.2 version.
It did not fix the issue but the other issue iam getting is t_relay is not
accepting any parameter.
The documentation is not clearly saying how to put *no-auto-477* - (old
*0x02* flag). in my old 3.3.2 version i
What do you mean by "free call taker in the group" ? something like in
call centers?
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
https://www.opensips-solutions.com
https://www.siphub.com
On 04.12.2023 14:30, Prathibha B wrote:
How to identify a free call taker in the group?
On
Hi Gregory,
As it is said, there is no single way to skin the cat :). Your approach
is a valid one, by using the escaping transformation. Maybe you should
check the s.escape.user [1].
Such checks make sense when using avp_db_query(), so raw queries. The
internal queries (like auth, etc) are
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