HI Anca
i am trying to use the b2b_request + b2b_reply
route{
...
if(is_method(INVITE) !(src_ip == 192.168.1.249 src_port ==5060))
{
if (! t_newtran()){
sl_reply_error();
exit;
};
Hi Bogdan
got it :)
1 more question about the flag
modparam(nathelper, sipping_bflag, 7)
modparam(usrloc, nat_bflag, 7)
the modem ADSL will close the port after 3 mins(some minutes), so Opensips
should send OPTION message(sip ping) to modem to keep port that should open for
UA
the
Hi Ha`,
The excerpt from your script shows that you don't have a good
understanding of the opensips scripting logic. First, the *route* block
will only be called for SIP Requests.
Calling this is not right:
/ if(status==200)
route(b2b_reply);
/The replies
Hi Ha,
the two flags are different and may have different values - one is used
as NAT marker, the other one is used as SIP-based pinging marker.
so, you can use different flags and both of them will be saved in cflag
mask.
Regards,
Bogdan
ha do wrote:
Hi Bogdan
got it :)
1 more question
Hi!
I'm a newbie with OpenSIPS administration and configuration and I searched
on the mail archives regarding limiting the channels but only found the site
regarding Concurrent calls limitation. I've been trying to grasp the whole
idea about AVPops and dialog module but unfortunately I'm having a
Hi,
I'm using permission module's function check_source_address(), the problem
I'm facing is that I can not add not than more 8 IPs in address table, but I
want to permit more than 100 IPs. I only want to use these IPs on group 0
what I am using. When I enter more than 8 IPs in address table and
Hi list,
Im trying to find a way to extract and the re-write the number in the
INVITE itself, Mera and a few other switch manufacturers are all of a
sudden are trying to route on INVITE and not the To field, when the
INVITE has a prefix in it they fail to complete the call then. I can't
seem to
On the contrary, rewriting the Request URI (which is what I assume you
mean by rewrite the number in the INVITE itself), including the user
part of the Request URI (the part before the @), is one of the most
basic functions the proxy *can* perform.
See the pseudovariables $ru (and for the user
Hi Johnson,
The idea is to use dialog profiles to keep trace of the ongoing calls
(per resource). As a first step take a look at the following tutorial:
http://www.opensips.org/Resources/DocsTutConcurrentCalls
Regards,
Bogdan
Johnson Pajayat wrote:
Hi!
I'm a newbie with OpenSIPS
Hi,
for bridging IPv4 and IPv6 you can use opensips (nathelper) for doing
the signalling part and rtpproxy for doing the RTP part.
You need to configure opensips to listen both on an IPv4 and IPv6
interface (and do the routing between). Also RTPproxy does bridging
between 2 interface (ipv4
Hi All
I'm trying to get drouting working, the issue is I can not work out
which tables and field its complaining about when it give the error.
'drouting:dr_load_routing_info: route 1 does not exist'
In 'dr_rules' I have a routeid of 1 for a number of rows, and if I
change this to a 2 or 5 then
Hello Bogdan,
I appreciate a lot your response regarding my inquiry. I've been reading
that tutorial as well as the AVPops and dialog modules documentation for
about a month now. I tried to adapt that route block for inbound calls and
here's a portion of what I have on our OpenSIPS 1.5 config
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