You do not need to manipulate core variables. You have to add a header
to pass the source ip to asterisk.
esample append_hf("X-src-ip: $si\r\n")
Il 10/10/2013 02.05, bluerain ha scritto:
Are you sure? Can you tell my which function call in opensips? I know how
to manipulate the core variabl
Hello!
I'm testing OpenSIPS 1.10.0 version now and I've strange behavior of
topology_hiding() in dialog module.
topology_hiding() function cut off all VIA headers and FreeSWITCH drop this
messages when receive it.
U 2013/10/09 20:58:07.836413 192.168.56.101:5060 -> 192.168.56.102:5080
INVITE
Are you sure? Can you tell my which function call in opensips? I know how
to manipulate the core variable, but $si is read only. And I think if you
define a "peering" resource in asterisk, it will try to match it by the
source IP at the network layer and not within the INVITE. Please tell me
wh
opensips can add an header with the real IP
and asterisk can use that header to know the real IP
Il 09/10/2013 17.02, bluerain ha scritto:
I've try to search on internet but not much info. I currently have Asterisk
server setup to have sip trunk with customers on a "peer" type. This way,
no re
I've try to search on internet but not much info. I currently have Asterisk
server setup to have sip trunk with customers on a "peer" type. This way,
no registration need and that asterisk server will identify the inbound call
base on "IP address" matching. But now I would like to put OPENSIPS i
Hello!
Please help me resolving the nasty issue,
OpenSIPS have been setup as a proxy in front of 3xFreeSWITCH servers
passing managing all of the requests to them for further handling,
including
REGISTRATION requests, using DISPATHER
the problem is:
After a certain while (may vary from minutes to
Hi all,
To integrate OpenSIPS with Asterisk, both should be in same server? or
If we can install OpenSIPS ans Asterisk in two different servers, how can I
connect those two? Could any one please tell me, where is the configuration
for that?
Thanks in advance
On Wed, Oct 9, 2013 at 10:57 AM, Si
HI,
I have setup Oversip for Websockets, Opensips as a registrar, RTPProxy as
a proxy server.
Before introducing Websockets through Oversip, The Video and Audio calls
where connecting without any issues within the network as well as outside
network. Where i had to introduce Nathelper and RTPPro
Codec issue.
Thanks & Regards,
Aamir Chougule
Cell: 08097989101
Skype-ID: aamir_ryu
--- Sent from my BlackBerry ---
-Original Message-
From: "Rajesh Babu"
Sender: users-boun...@lists.opensips.org
Date: Wed, 9 Oct 2013 13:20:50
To: 'OpenSIPS users mailling list'
Reply-To: OpenSIPS use
Opensips registration through the network, then connect the asterisk, while
talking on the phone, sometimes can play out, but sometimes an error
again.Call failed:Service Unavailable.
What reason is this excuse me.___
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