Hi Douglas,
I've recently been through the process of setting opensips up as an
authenticating relay much the same as you describe. I strip the User-Agent
header but it would be a single line to add a replacement one in after this.
I also remove remove the g711 codecs due to bandwidth limitation
I am trying to get Snom's Contact List to work with RLS and XCAP. When I
enable the Snom phone it sends the following subscribe
SUBSCRIBE sip:9012732...@irock.com;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.70:3072;branch=z9hG4bK-o82o6mip4krp;rport
From: ;tag=uqpj4kshg7
To:
Call-ID: c60b313c
My Request URI is something like sip:+1206555.
However if I create a rule in dr_rules with prefix "+1206555", while
loading the rules to memory drmodule seems to be dropping those rows as it
contains non-numeric chars.
Currently I am doing something like the following (Strip +
Currently if in the opensips configuration, if I make a call to do_routing()
without any parameters it will automatically query dr_groups table to
determine the group. If it cannot find a group corresponding to the caller ,
it fails do_routing and the calls fails.
Is there a way to specify a de
Hello all,
A new module is available in trunk.
It provides an embedded http server that can be used by other modules.
So far, we have the mi_http module which takes advantage of the
embedded http server.
The embedded server is based on libmicrohttpd library.
More info here:
http://www.opensips.or
I am running in to the same issue as the below attached post. Any solutions
?
My dr tables have the following entries.
+--+--+--+---++---++
---+
| gwid | type | address | strip | pri_prefix | attrs | probe_mode |
descr
Hi Vlad,
Okay I got the functionality of"!" symbol & also got different return codes
while testing on a local server. But I am stuck with one thing that I am
getting this in the wireshark logs either the max_valuefor the
mf_process_maxfwd_headeris 1 or 10:
564d700c00f837-1--d87543-;rport..Max-
Hello!
Opensips 1.6.4-2 and rtpproxy
I want to use start_record() function for call recording.
There is one question, where in script should I use the function? After or
before I call rtpproxy_offer() (or rtpproxy_answer())?
Thank you for any help!
_
Hi,
On Jan 25, 2012, at 10:05 PM, Reda Aouad wrote:
> Hello,
>
> Is there a way to change Call Control's behavior when ending the session,
> replacing the BYE message with an INVITE to a media gateway?
> If not, is there a way in OpenSips to intercept the BYE message generated bye
> Call Contr
