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 <miconda(a)gmail.com>
Sent: Monday, April 24, 2023 4:11 PM
To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org>rg>; Kiss Zoltán
<kiss.zoltan(a)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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