I would like to forward all incoming calls that are not answered from kamailio,
to my asterisk voicemail server. - I have not setup any Realtime Integration.
I am using the "Redirected Dialed Number Information Service" header and
calling "554@192.168.1.134" which is my voicemail server.
It’s exposed as $rU, as many attributes of a message are:
https://www.kamailio.org/wiki/cookbooks/5.3.x/pseudovariables
—
Sent from mobile, with due apologies for brevity and errors.
>> On Nov 7, 2019, at 9:51 AM, Laura wrote:
> Dears..
>
> how is possibile to read that data 9052562462354464
Or $rU :)
On Thu, 7 Nov 2019 at 14:58, Sergiu Pojoga wrote:
> That's $rU
>
> For more:
> https://www.kamailio.org/wiki/cookbooks/5.3.x/pseudovariables
>
> On Thu, Nov 7, 2019 at 9:51 AM Laura wrote:
>
>> Dears..
>>
>> how is possibile to read that data 9052562462354464 inside the first
>>
That’s the URI, off the top of my head, you can get it fro the
pseudo-variable $oU I believe
On Thu, 7 Nov 2019 at 14:51, Laura wrote:
> Dears..
>
> how is possibile to read that data 9052562462354464 inside the first
> INVITE message
>
>
> === Example ===
>
> U 2019/11/07 14:46:59.320936
That's $rU
For more:
https://www.kamailio.org/wiki/cookbooks/5.3.x/pseudovariables
On Thu, Nov 7, 2019 at 9:51 AM Laura wrote:
> Dears..
>
> how is possibile to read that data 9052562462354464 inside the first
> INVITE message
>
>
> === Example ===
>
> U 2019/11/07 14:46:59.320936
Dears..
how is possibile to read that data 9052562462354464 inside the first
INVITE message
=== Example ===
U 2019/11/07 14:46:59.320936 192.168.1.1:5060 -> 192.168.1.242:5060
INVITE sip:9052562462354464@192.168.1.242 SIP/2.0.
Record-Route:
.
Max-Forwards: 66.
Via: SIP/2.0/UDP
Definitely it does not work getting the keys from redis. Also, until not
long ago, trying to failover from REDIS in tls calls was crashing rtpengine.
I believe it can work on DTLS if we start with a "pristine" reinvite, doing
ICE and all things again, like it was a first invite. Something like
Hello,
the plan right now is to release v5.3.1 during the next Kamailio
Developers Meeting, if we succeed to coordinate on site, therefore
expect it to happen next week during Nov 14-15. Besides just doing this
specific release, to goal is to try to build a more automatic process
for releases,
Hi Giovanni,
i have an SRTP and WebRTC DTLS setup with pacemaker/corosync and failover
works for SRTP (with REINVITES).
I use rtpengine with redis backend. On DTLS side, i dont got it working
with REINVITES.
AFAIK the session keys are not stored like SRTP in SIP Signaling.
So i thought, that
Hi Igor, hi kamailions,
i create an kamailio logfile with sipflow via $mb. Its corresponding to the
kamailio.cfg in this thread.
Callflow is B2BUA 172.20.170.1 --> 172.20.120.59 -->
172.20.120.101/212.XX.XX.XXX --> MS-Teams.
172.20.170.1 B2BUA, unencrypted media
172.20.120.59 INTERNAL
( but yes, it works on DTLS, I had not really read you were talking about
DTLS. You must reinvite reusing the original SDP peers sent to you)
On Thu, Nov 7, 2019 at 1:54 PM Giovanni Maruzzelli
wrote:
> I believe the problem is that there is no more tcp connection.
>
> Eg, if you generate a
I believe the problem is that there is no more tcp connection.
Eg, if you generate a reinvite over udp, it works (with due care, you can
have the keys renegotiated as per beginning)
But... you have no more tcp (tls is tcp) connection to send the reinvite to
So, it works on udp, but udp is no
Can you give an example of SIP trace where it's not working as expected?
Cause reading through config will not help much without setting up a lab.
On Nov 7 2019, at 11:27 am, Karsten Horsmann wrote:
> Hello Mailinglist,
>
> i try to figure out, how to solve RE-INVITES and SRTP:
>
> i have an
You can reuse the grp database table (from the group module) but do the
query you need using sql_query() (from sqlops) module. So you don't have
to use the functions of the group module if they do not fit your needs,
anyhow behind is_user_in() is an sql query.
If you authenticate the user with
Hello Mailinglist,
i try to figure out, how to solve RE-INVITES and SRTP:
i have an kamailio route names route[MEDIAPROXY] that i used to controll
rtpengine in an private and public ip setup. This works fine for me now.
It use the permission module and the adress table to figure out
Hello Igor,
to make it clearer – if you want to use the regular expression matching, have a
look to the module documentation. There is e.g. a regular expression column
that you need to fill, not the one that you quoted below in your select.
If you want to use the quoted table, use the
Thanks!
Looks ok. (Despite the fact it's looking like write-only statement : )
Idea was, that I want to avoid creating additional tables or schema of ACL.
Means if I can use already built-in mechanism/tables/schema - than why to
implement own. Code reuse, all this.
On Nov 7 2019, at 10:56 am,
On Wed, Nov 06, 2019 at 07:15:42PM +0100, Igor Olhovskiy wrote:
>
> Hm... Maybe there is other module to achieve such functions?
> Best if it would be with cache :)
> But if no - regex also fine.
But if it works it works! I don't think there is a specific module to do
this but it is easy to
Hi,
AFAIK the keys of an DTLS session are not restorable so after failover will
come with an stale DTLS call.
Only SRTP can recovered with RE-INVITES if you use some kind session
storage.
Am Di., 30. Okt. 2018 um 12:07 Uhr schrieb Жан Базаров :
> I need to send re-invite after pacemaker fails
Hello,
On 06.11.19 20:46, Karsten Horsmann wrote:
> Hi,
>
> the sips Uri schemata is not used for tls with dispatcher.
jumping in to clarify a bit about sips protocol schema. It doesn't imply
TLS as one may think HTTPS does it for HTTP. The sips is mandating that
the traffic goes over secure
Hello Henning,
Thank you!
Regards,
Igor.
De : Henning Westerholt
Envoyé : vendredi 1 novembre 2019 18:32
À : Kamailio (SER) - Users Mailing List ;
igor.potjevle...@gmail.com; mico...@gmail.com
Objet : Re: [SR-Users] lookup(aliases) issues with 5.2
Hello Igor,
functional
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