Hi,
Am Dienstag, dem 25.10.2022 um 22:28 -0400 schrieb Saint Michael:
> Dear Friends
> local_route
>{
>if (is_method("200"))
> {
>}
> }
>
> I need to intercept the code at the moment when a call connects, so I
> may notify that information to the database. What spec
Dear Friends
local_route
{
if (is_method("200"))
{
}
}
I need to intercept the code at the moment when a call connects, so I
may notify that information to the database. What specific code can I
use inside a Local Route so the 200 OK corresponds with a real
connection
in 3.3 too:
https://opensips.org/docs/modules/3.3.x/tm.html#param_disable_6xx_block
---
I'm SoCIaL, MayBe
El 25/10/2022 a las 4:44 p. m., Joseph Jackson escribió:
Its listed here -
https://opensips.org/html/docs/modules/3.1.x/tm.html#param_disable_6xx_block
So yes it should work.
-
Its listed here -
https://opensips.org/html/docs/modules/3.1.x/tm.html#param_disable_6xx_block
So yes it should work.
From: Users on behalf of Saint Michael
Sent: Tuesday, October 25, 2022 9:52 AM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users
There is no max length in the RFC. I've seen them up to 256.
From: Users on behalf of Saint Michael
Sent: Tuesday, October 25, 2022 12:41 PM
To: OpenSIPS users mailling list
Subject: [OpenSIPS-Users] How long can the callID be in SIP
I just caught a callID
I just caught a callID that is 160 characters long.
What is the max length that it should be? I am using 128.
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Hi,
Not sure if this is the intended behaviour or I’m just misunderstanding these
parameters…
Assume a simple config with a few sockets
socket=tcp:10.44.30.12:7700
socket=tcp:10.44.30.13:7730
And setting the outbound socket in route[relay] with
force_send_socket(“tcp:10.44.30.12:7700”);
Th
This is a Linphone problem. A simple UA has no way of knowing that all the
other UAs are unable to take the call. The Linphone developers should be
encouraged to change the default response to a rejected call. Perhaps a
universal reject could be an option but not the default.
On Tuesday, Octobe
Will that work on version 3.1X
On Tue, Oct 25, 2022, 10:36 AM Social Boh wrote:
> maybe:
>
>
>-
>
>*modparam("tm", "disable_6xx_block", 1)*
>
> to TM module
>
> ---
> I'm SoCIaL, MayBe
>
> El 25/10/2022 a las 9:22 a. m., Richard Robson escribió:
>
> I am currently testing version 3.3 an
good catch.
On 25/10/2022 16:33, Social Boh wrote:
maybe:
*
*modparam("tm", "disable_6xx_block", 1)*
to TM module
---
I'm SoCIaL, MayBe
El 25/10/2022 a las 9:22 a. m., Richard Robson escribió:
I am currently testing version 3.3 and I have a scenario where we
will receive a 603 to an i
maybe:
*
*modparam("tm", "disable_6xx_block", 1)*
to TM module
---
I'm SoCIaL, MayBe
El 25/10/2022 a las 9:22 a. m., Richard Robson escribió:
I am currently testing version 3.3 and I have a scenario where we will
receive a 603 to an invite. this needs to be failed over to the next
desti
I am currently testing version 3.3 and I have a scenario where we will
receive a 603 to an invite. this needs to be failed over to the next
destination. we are using dispatcher for this.
While testing I am finding that opensips is treating the 603 a a relpy
and not as a failure and its being h
Hi,
this is actually fully SIP/RFC compliant. By definition, any 6xx answer
provides the "ultimate, global answer", to put it in simple words: "I
cannot reply to the request and I know no one else can". On the contrary as
per the definition/RFC, a 5xx answer indicates, that a single endpoint is
n
Dear friends
is there a command I can send to Opensips 3.1, so all calls are shut
down and BYEs are sent to both the caller and the callee? Then of
course, opensips closes down.
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Hi Takeshi,
Thanks for the note here, I will check the PR.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
https://www.opensips-solutions.com
OpenSIPS Bootcamp 5-16 Dec 2022, online
https://www.opensips.org/training/OpenSIPS_eBootcamp_2022/
On 10/22/22 12:53 PM, mayamatakeshi
Hi Bogdan-Andrei,thanks for your reply. I had set up a opensips service on my
Ubuntu 18.04.6 , and opensips version: opensips 3.2.6 (x86_64/linux).
When I test that case, I use two desktop linphone login same sip accout
01010101 (callee A/B). and I use my Android linphone(caller C) invite A/B,
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