Hi,

We have to test every scenario, but the latest issue was we have one way rtp 
and the call is dropped after 6 seconds cc.
In the test the calle was the GS phone which is registered via Kamailio, and 
the called party was an another phone witch was registered directly tot he 
backend Asterisk.
After switching GrandStream phone to sip scheme, then everything is working 
fine again.

Zoltan

From: Daniel-Constantin Mierla <mico...@gmail.com>
Sent: Monday, April 24, 2023 4:11 PM
To: Kamailio (SER) - Users Mailing List <sr-users@lists.kamailio.org>; Kiss 
Zoltán <kiss.zol...@adertis.hu>
Subject: Re: [SR-Users] sips to sip with TLS proxy

Hello,

just to clarify: you cannot initiate calls from the phone or you can't sent 
calls to the phone?

Cheers,
Daniel

On 24.04.23 15:58, Kiss Zoltán wrote:
Hi all,

We have a working Kamailio setup, lets call it a transparent proxy for Asterisk 
boxes. Its based on domain and dispatcher modules and everything is working as 
expected with the test clients (more or less microsip, softphone for ios, etc). 
We are tried to register with a Grandstream deskphone today, and we see that 
the phone sending sips:xxx in the Reg Contact field for example. Because the 
sips schema, the register is working, but we cannot initiate calls from this 
phone. If we are turning SIP scheme to sip from sips in the phone, then 
everything is working as expected.
I think we can transform those requests from sips to sip with Kamailio, but 
currently we dont know where can we start.
Has anybody a suggestion about this issue? I know that we can transform ruri, 
contact, etc with textops, nathelper and a lot of other modules, but what is 
the best for this sips->sip translation?

Thanks for your help.

With kind regards,
Zoltan



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