Hello,
I have a working implementation of the B2BUA modules for OpenSIPS. One problem
I am having is that the B2BUA module doesn't seem to respect the From: header
that I updated before B2B_INIT was called (at the end of the script). Before I
call B2B_INIT, my script looks up the correct
In addition, the uac_replace_from(display,uri); function does not seem to work
either. I tested this function by putting:
uac_replace_from(TEST, sip:12...@1.2.3.4);
immediately above the call to B2B_INIT_REQUEST and the outbound From: header
remains unchanged (incorrect).
Brett Woollum
Hi David,
The only thing that won't work as you like is the append_hf - it won't
work as you like since the b2bua module won't see that header. You need
to do that for the new request generated by the b2bua and put that code
in the local_route.
Regards,
--
Anca Vamanu
www.voice-system.ro
Hi Brett,
The b2bua generates a new request with the info from the received one.
The problem is with the changes upon a request not being visible to the
following functions called from the script.
So you have to change the header on the generated one, in local route.
Regards,
Anca
On
Why don't you explain the problem you want to solve?
Maybe that to modify contact in this way is not the right solution.
s
Il 05/10/2010 18:34, David Santiago ha scritto:
Hi all,
I need to modify the host part of a contact header. I'm trying something like:
if (
Hi All,
I've got a problem I'm not 100% sure how to resolve.
Ok the scenario is, client A is on a public interface, Client B is on a
private IP address and has cflag 6 set in the location table.
Client A calls Client B via Opensips.
Now during the initial INVITE, opensips locates Client B,
Hi Anca,
Does this mean that I would need to make the changes after I call
B2B_INIT_REQUEST instead of before? If that's the case it sounds like I might
want to call B2B_INIT_REQUEST very early in the script...
Thanks
Brett Woollum
br...@woollum.com
- Original Message -
From:
Hi Doug,
I had similar problem. My solution is to use record-route variable:
add_rr_param(mr=1) on initial INVITE at same time as first call to
use_media_relay(). The later within loose_route block, I use
check_route_param(mr=1) for re-INVITES and then re-invoke media relay if
found. In that
Hi Kennard,
Would you be so kind as to post what you did in regards to the reply
handler? Also do you set the bflag if check_route_param succeeds and
then let your reply handler check for a bflag?
In regards to the private IP address, thats handled easily with
Hi Doug,
Here are the relevant sections (with a lot of other stuff deleted).
Unfortunately, due the embedded biz logic I cannot post whole script.
if (loose_route()) {
if(is_method(INVITE)) {
t_check_trans();
if (check_route_param(mr=1)) {
Hi Anca,
I figured out the solution that you were referring to. I added the local_route
section and changed the header in there, and it works.
The only problem now is that the AVP variable I set in route doesn't seem to be
persistent into the new request inside local_route. This is a problem
I have dialogs getting stuck because my origination gateway (OpenSIPS proxy)
fails to proxy 200 OKs to some INVITEs. In the logs, I see:
/usr/local/sbin/opensips[30336]: ERROR:uac:decode_uri: invalid base64
stringAAUECAEIAwUHCQsip:12.24.48.96
/usr/local/sbin/opensips[30336]:
Hi ,
I followed the instruction in build telephony system with opensips 1.6
to install opensips-cp. However, an error comes up when access the link
http://localhost/cp.
Failed to issue query, error message: MDB2 Edrror: no such table.
I did import the two tables for cdr and smonitor,
To add a little more detail, after doing some from uri normalization, this
is eventually passed on to an OpenSIPS instance running b2bua for top
hiding. I'm assuming this error indicates that the header param that is
used to rebuild the original from uri is different than expected, but I
don't
Hi,
I want to solve the nat traversal problem, and I read the great book
Building Telephony System with Opensips 1.6.0. Now I encounter some
doubts.
From the book,I know the STUN solution is considered a near-end NAT
traversal solution. And I think a near-end NAT traversal solution is
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