Hello,
are you using topoh module?
A pcap with all sip messages part of the dialog, from the initial INVITE
to the last BYE transaction would be important to have in order to
investigate how the headers are changed.
Cheers,
Daniel
On 28/12/2016 13:11, Nihar Ranjan Deb wrote:
>
> Hi All,
>
>
Anyone any help on this.
Regards,
Nihar
From: Nihar Ranjan Deb [mailto:nihar.ran...@microtalkgroup.com]
Sent: Wednesday, December 28, 2016 5:42 PM
To: 'sr-users@lists.sip-router.org'
Subject: BYE issue kamailio + rtpproxy "481 call leg transaction doesn't
exists".
Hi All,
I
Hi All,
I am a new user of kamailio. We have setup a kamilio server with RTP proxy.
Most of the features are working fine except BYE.
Say User A and B are registered in kamailio server. A (caller) connected to
B(called), call established.
1. If Caller (A) disconnects the call, eve
@lists.sip-router.org] *On
> Behalf Of *Nelson Migliaro
> *Sent:* 17 October 2016 18:23
> *To:* Kamailio (SER) - Users Mailing List
> *Subject:* [SR-Users] BYE issue
>
>
>
> Hello everybody,
>
>
>
> I am having issues with one SIP vendor.
>
>
>
> I have a
Is it not just the case that the vendor does not support loose routing?
-Original Message-
From: sr-users [mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of Alex
Balashov
Sent: 18 October 2016 08:40
To: Kamailio (SER) - Users Mailing List
Subject: Re: [SR-Users] BYE issue
Handling NAT, perhaps, but not correctly. The RURI doesn't change. But not does
it have to determine where the request is actually sent on the network and
transport layers.
-- Alex
--
Principal, Evariste Systems LLC (www.evaristesys.com)
Sent from my Google Nexus.
__
need to rewrite the
headers yourself.
From: sr-users [mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of
Nelson Migliaro
Sent: 17 October 2016 18:23
To: Kamailio (SER) - Users Mailing List
Subject: [SR-Users] BYE issue
Hello everybody,
I am having issues with one SIP vendor.
I have a
Nelson,
You are very correct. The request URI of the BYE (and all other
in-dialog requests, such as reinvites) should equal the Contact URI of
the party to which it is being sent, and this should not change even if
the in-dialog request is being sent through Kamailio because of
Record-Route.
Hello everybody,
I am having issues with one SIP vendor.
I have a Kamailio in bridge mode (private IP / Public IP) and some Asterisk
and Media Gateways.
Calls get established and I have two way audio but when the remote party
hangs up the call, the BYE arrives to the Kamailio and does not move
f
Dear experts ,
I am using kamailio with rtp proxy module. I have 2 questions /issues .
1. When caller or callee ends the call the other end call is not
disocnnecting .
UA is pjsip based and behind NAT router. Present call flow is
pjsipUA (LAN_ip)->Router (Publicip)>Kamailio_with_
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