Hi Thomas,
Please look at t_new_request:
https://opensips.org/docs/modules/3.2.x/tm.html#func_t_new_request
> 23 авг. 2021 г., в 22:59, Thomas Grande via Users
> написал(а):
>
> Greetings,
>
> I'm considering migrating from Kamailio to OpenSIPS due to some modules that
> are missing on
Hi, yes there is an issue.
Please follow here: https://github.com/OpenSIPS/opensips/issues/2433
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Alexey Vasilyev
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But config is for 2.4
A lot of things has changed.
You should go through the whole config and follow first this:
https://www.opensips.org/Documentation/Migration-2-4-0-to-3-0-0
and then this:
https://www.opensips.org/Documentation/Migration-3-0-0-to-3-1-0
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Alexey Vasilyev
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Hi Mark,
try this:
if (nat_uac_test("123")) {
if (is_method("REGISTER")) {
fix_nated_register();
} else {
fix_nated_contact();
}
}
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Alexey Vasilyev
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Hi Pasan,
This was explained here:
http://opensips-open-sip-server.1449251.n2.nabble.com/TLS-handshake-failure-td7619394.html
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Alexey Vasilyev
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/
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Alexey Vasilyev
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t;transport=tls".
Check this please.
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Alexey Vasilyev
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("sbc.ip-sentinel.com:5091;transport=tls");
of course without: add_rr_param(";r2=on");
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Alexey Vasilyev
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answer that
if(is_method("ACK") && has_body("application/sdp")){
xlog("[RTPPROXY] onreply_route[handle_nat] ACK rtpproxy_answer\n");
rtpproxy_answer("co", "137.117,136.143");
}
and change comma to point.
May be this can help?
Hi James,
Just to be sure, that this is not a typo:
check your SDP
o= 137.117,136.143
c=IN IP4 137.117,136.143
This will definitely fail.
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Alexey Vasilyev
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difference was RTPEngine to transcode
SRTP-RTP.
And within test lab I've tested not only calls, but transfers worked fine
too.
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Alexey Vasilyev
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"dr_enable_probing 0" after
starting opensips.
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Alexey Vasilyev
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.
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Alexey Vasilyev
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Please forward requests to Asterisk with simple record_route().
Only to MS Teams you should use preset with domain name.
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Alexey Vasilyev
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Hi Jeff,
Transactions are not replicated.
Here
https://blog.opensips.org/2018/03/21/full-anycast-support-in-opensips-2-4/
Razvan explains why. Section "Distributed transactions handling".
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Alexey Vasilyev
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Hi Callum,
I have almost the same problem. But with 2.4.6.
Just today I opened issue: https://github.com/OpenSIPS/opensips/issues/1976
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Alexey Vasilyev
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Hi Mark,
I'd use uaCSTA, if your phones support it. For example, snom or yealinks
definitely can be controlled over SIP.
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Alexey Vasilyev
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com/OpenSIPS/opensips/issues/1769>
In the case of UDP, may be it make sense to use simple mirroring mode with
nathelper pings.
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Alexey Vasilyev
alexei.vasil...@gmail.com <mailto:alexei.vasil...@gmail.com>
> 27 Sep 2019, в 22:58, Igor Olhovskiy [via OpenSIPS (Open SIP Server)]
Hi Bogdan,
Yes, this is exactly what I need. I'll change it and will see, how it works
with next switching.
Thanks.
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Alexey Vasilyev
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define DEFAULT_TCP_LISTEN_BACKLOG 10
If we have thousands TLS/TCP connections at the same time, and for example
32 processes, so we'll have only 10*32=320 socket backlog limit for
OpenSIPS? Is this right?
Should it be increased?
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Alexey Vasilyev
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Thanks, Giovanni and Bogdan!
If somebody has questions, I think we can discuss here.
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Alexey Vasilyev
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uZYcpOhf1g/h+oG
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
m=audio 60812 RTP/AVP 9 8 101
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
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Alexey Vasilye
, there is parameter for domain:
modparam("tls_mgm", "ca_client_send", "[dom]1")
For this domain OpenSIPS adds DNs to Certificate request.
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Alexey Vasilyev
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, that they can choose to send or not its'
certificate?
If you like to test it in your environment?
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Alexey Vasilyev
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encrypted or unencrypted.
You should choose, what you want. Use only one profile as parameter.
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Alexey Vasilyev
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eaders. But it seems, that function topology_hiding()
encodes only RR headers from original UAC request, not those ones that were
added by OpenSIPS.
So the question - how to pass this in-dialog data?
Should we add functions add_th_param/get_th_param for this purpose?
Thanks.
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Alexey Vasilyev
Hi Bogdan,
Sorry that I mentioned He-Who-Must-Not-Be-Named. Just to simplify search later:
https://github.com/OpenSIPS/opensips/issues/1651
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Alexey Vasilyev
alexei.vasil...@gmail.com
> 28 Mar 2019, в 16:45, Bogdan-Andrei Iancu написал(а):
>
> Hi Alexey,
>
> oh, if i
e case - we have only 1 destination target, but we should
> use
> > several TLS connections to this target with different TLS certificates.
> So
> > first connection will be successful, but SIP message for second domain
> which
> > should use another certificate will try to reuse this
exists. But for TLS it is not
enough. We additionally should check, what certificate uses this connection (or
what domain it is related).
If on server side we have SNI, why not to have more control for client TLS side?
-
Alexey Vasilyev
alexei.vasil...@gmail.com
> 25 Mar 2019, в 18
> > but it simplifies total work.
> >> Let's imagine 3 nodes A,B,C. A is Active. A and B lost interconnection.
> C
> > is down. Then C is up and has 2 hosts for syncing. But A already has 200
> > phones re-registered for some r
. But again, if Active node restarts and still Active
- we will have a problem.
-
Alexey Vasilyev
> 31 Dec 2018, в 18:04, John Quick написал(а):
>
> Hi Alexei,
>
> Many thanks for your reply to my query about syncing the seed node for
> usrloc registrations.
> I just
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