Calling exit() during the initial request and right before creating the dialog
also increments the drop_requests statistic
De : Users De la part de Ben Newlin
Envoyé : jeudi 18 avril 2024 15:59
À : OpenSIPS users mailling list
Objet : Re: [OpenSIPS-Users] question on core statistics.
Are you c
hi Stefan
your question is far from stupid
have you tried manipulating the SIP headers to influence the negotiation
process between the provider and your PBX ?
you can do that by adjusting the Supported/Require headers in SIP messages
within OpenSIPS to indicate that only re-INVITE is supported
Hello Adrien
Try :
if (!rl_check("test1", 1, "RED"))
{
rl_dec_count("test1")
send_reply(487, "Request Terminated: CPS limit exceeded");
exit;
}
* If you are using ratelimit to limit only successful traffic, you need to
explicitely decrease the counter for the declined calls using the
Hello Michael,
Are you sure about that ?
As per the French implementation, it's essential to ensure that when an
operator processes an Identity Shaken call, they do not modify the "User"
section of the SIP From, To, and PAI headers.
Regarding portability, it's important to note that any additi
Hello
I have run into a problem with the STIR/SHAKEN verification process.
In the French implementation of StirShaken, the CRL of the operator
certificates is signed with a certificate that is different from the one used
to sign providers certificates.
and in such case, OpenSSL does not allow in
Hello Mickael
if the goal is only to keep your key out of opensips plain sight, the simplest
way is using exec.so module, load the private key from a local file or an
external data source with an external command, and then store it securely in
the local cache for future calls stirshaken signing
Hello,
I noticed a segmentation fault is caused by sst module :
Client B2BUA (UAC) invite has : Session-Expires: 1000 Min-SE: 90
Opensips SST min_se configured at 1800s, dialog has sst_flag.
This causes Opensips to crash :
opensips log :
Mar 3 13:06:10 test-boxopensips /usr/sbin/opensips[6088]:
Hello, question related to sst module.
Suppose a use case on which the UAC (A) offers during the invite
Session-Expires : 4000, MIN-SE :90
UAS (B) configured sessions-expires=5000s but responds : "Session-Expires :
4000, refresher=uas" to match UAC's request.
That means at roughly 2000 seconds th
Hello,
I am using the new rabbitmq module to send some informations as AMQP messages
to a rabbitmq server. Mostly fraud detection + CDR.
I wanted to know if rabbitmq_publish is considered a blocking function? for
example in the case of fraud detection warning, my use case is to publish the
mes
er, making it
far easier to send/receive multiple headers.
On Tue, Dec 27, 2022 at 8:55 AM Wadii ELMAJDI | Evenmedia
mailto:wa...@evenmedia.fr>> wrote:
>
> Hello,
>
>
> I am using Asterisk as a media server behind OpenSIPS.
> I need to send some additional data to opensi
ng it in the BYE or an
in-dialog request, then you'll have to do something out of band like HTTPS,
Redis, etc, which are not bad options. Have you looked into
"X-Asterisk-HangupCauseCode" to see if that can help you out?
On Tue, Dec 27, 2022 at 8:55 AM Wadii ELMAJDI |
Hello,
I am using Asterisk as a media server behind OpenSIPS.
I need to send some additional data to opensips during the call or at the
latest during the hangup. For instance a custom hang up reason.
Asterisk does not allow neither to send a sequential request withing an ongoing
dialog, nor add
hello , i do have a question related to rtpproxy module documentation.
The doc describes that rewriting sdp body should happen during either INVITE ,
200 OK or ACK.
In the case of SDP presence on invite <=> 200 , one should rtpproxy_offer
during the invite and rtpproxy_answer during the 200 OK.
I ended up using dialplan module with a new partition dedicated to CLID.
De : Users De la part de Wadii ELMAJDI |
Evenmedia
Envoyé : lundi 19 décembre 2022 15:44
À : OpenSIPS users mailling list
Objet : [OpenSIPS-Users] REPLICATION / SQL_CACHER / Local CacheDB Collection
Hello ,
I am trying
Hello ,
I am trying to cache an entire MySQL table containing CLIDs/Clients Mapping.
My use case is to check pre every call , if the $fU used belongs to the client
(via its ip source).
I am using SQL CACHER (ondemand = 0) mode module, my opensips box is clustered.
the cached_url for SQL_CACHER is
Hello,
I am using statistics module to store CPS data on demand (via an external
variable shv_set), keys are current seconds ,example :
opensips-cli -x mi get_statistics cps:
{
"cps:2:20221202143557": 1,
"cps:2:20221202143613": 1,
"cps:2:20221202143616": 1
}
The data is stored on th
Hello,
After running a system upgrade opensips-cli package was upgraded.
Since then, no opensips mi commands is working , error :
Nov 30 13:54:07 1 /usr/sbin/opensips[2484]: ERROR:mi_fifo:get_reply_filename:
forbidden filename: opensips_fifo_reply_1669816447.3951223:{"jsonrpc": "2.0",
"id": "82
Hello,
I was able to install ratecacher module and make it work.
But editing the clients hash size parameter generates an error.
"ERROR:core:set_mod_param_regex: parameter not found in
module "
however vendors_hash_size is correctly identified as a rate_cacher parameter
Everything works when rem
nvoyé : vendredi 18 novembre 2022 11:27
À : OpenSIPS users mailling list ; Wadii ELMAJDI |
Evenmedia
Objet : Re: [OpenSIPS-Users] relay invites UAC
Hi Wadii,
You can route the call to any SIP URI you want :)
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
https://www.opensips-solution
Hello Everyone,
I am using opensips as sip proxy, receiving invites from my clients (mostly
b2bua), that i should relay to some gateways.
I am using Dynamic route module for such purpose.
Some of those invites (depending on called DID or gateways probing) should be
relayed to a sip trunk service
Hello ,
I would like to know what would be the best method for such use case :
My opensip box will play a sip proxy ROLE, handling inbound calls fom pstn to
my clients (B2BUA PBXs).
My clients will have to register as some of them dont have a static ip.
Client (Asterisk) (A) < Registration =
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