On 21 Jan 2011, at 18:32, Jeff Pyle wrote:
> Dan,
>
> Brett and I communicated privately. It turns out the majority of both our
> headaches with this issue is from the same, rather large carrier. Does
> anything come to mind on how we might handle it?
>
> I'd love an option I could include
Well, your problem seems pretty obvious. You seem to have a race condition,
where the original media stream continues to flow past the 2nd 183, so even
though mediaproxy resets the ports and expects to re-learn them, it will still
lock to the old one as the stream continues to flow past the rese
Thanks for that clarification!
So just to be sure I'm clear on this. $Ts rounds down ( truncates ) the
current second. So using $avp(s:start_time) = $Ts.$Tsm; would give something
like "12343253.543233" and always be accurate?
Further (standard ACC [without dialog]) you could just put $Tsm in the
Anca,
When using t_replicate, is opensips then expecting a response from the
server it is replicated to? I'm guessing not and also therefore will not
send retries when/if no response is received. Then on the recording server
you can just use tcpdump to save the messages.
Further you would need to
On 01/24/2011 07:19 PM, Toyima Dias wrote:
Thanks Anca,
I can send the call as many destinations i want, right? as many
t_replicate, many destinations? but now i have a doubt, is it right
what i want to do? i mean: sending the INVITE to both the final
destination and the recorder server to rec
Thanks Anca,
I can send the call as many destinations i want, right? as many t_replicate,
many destinations? but now i have a doubt, is it right what i want to do? i
mean: sending the INVITE to both the final destination and the recorder
server to record conversations between peers? should i treat
Hi to all,
I am new here and I have a problem with accounting on RADIUS.
OpenSIPS sends start and stop accounting messages properly, but in case
of update sends messages with Acct-Status-Type = 0, which are not
recognized by the RADIUS server.
I tried to add a definition to Acct-Status-Type = 0 in
Hi,
thanks for your reply.
I am using the script reported in the Opensips book ("Building telephony
systems with Opensips") at page 81, that is :
if (has_totag() ) {
# sequential request withing a dialog should
# take the path determined by record-routing
if (loose_route() ) {
if (
Hi Toyima,
You can use t_replicate function:
http://www.opensips.org/html/docs/modules/devel/tm.html#treplicate.
Regards,
--
Anca Vamanu
www.voice-system.ro
On 01/24/2011 02:35 PM, Toyima Dias wrote:
Hello,
I have a requeriment which indicates that all initiation calls
(INVITES), should
Hi all,
Just a short reminder on OpenSIPS Social Event that will take place in
Miami on 3rd of February :
"This is an *do what ever you like* event where developers, community
people, users, fans, etc will get together, talk (even about project :)
), have fun, etc. It will be an opportunity
also try to increase this in config.h, then recompile:
#define PKG_MEM_POOL_SIZE 1024*1024
set it to 2*1024*1024 for testing
2011/1/24 Denis Putyato
> Hello!
>
>
>
> I have such error messages in syslog while using 1.6.4
>
>
>
> Jan 24 09:40:04 opensips /usr/local/opensips1.6.4/sbin/opensips[10
Hello,
I have a requeriment which indicates that all initiation calls (INVITES),
should reach not only the destination, also another server (a SIP recorder
server), something like the following:
A>PROXY--->B
-
-
-
SIP Recorder SE
Hi Bogdan,
I seecould you open a bug report on the tracker (
http://www.opensips.org/Development/Tracker) and I will take care of it
asap.
Regards,
Bogdan
rad bogdan wrote:
Hi Bogdan,
I added
local_route {
xlog("LOCAL_ROUTE\n");
setflag(22);
sip
Hello!
I have such error messages in syslog while using 1.6.4
Jan 24 09:40:04 opensips /usr/local/opensips1.6.4/sbin/opensips[10176]:
WARNING:core:fm_malloc: Not enough free memory, will atempt defragmenation
Jan 24 09:40:04 opensips /usr/local/opensips1.6.4/sbin/opensips[10176]:
ERROR:c
Update:
i think to use opensips as outbound proxy too,
configuring fs with nat and setting fs_path parameter for bleg.
Regards,
Alessandro
-Messaggio originale-
Da: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] Per conto di Alessandro Illiano
Inviato: lunedì 2
Hello,
The function fix_route_dialog() needs loose_route() to be called, in
order to set the current dialog pointer that fix_route_dialog() will
attempt to match and fix accordingly. It isn't mandatory to call
validate_dialog() first, but it is best practice, because a call to
validate_dialog
Hi,
Ok I'm going to try, a simple question:
Incoming call works fine with opensips+rtpproxy in bridged mode.
But what if a second call is generated directly from freeswitch and bridged
to the frist one?
The second call flow bypass opensips+rtpproxy but goes directly to the
destination (at the mome
Hi Bogdan,
I added
local_route {
xlog("LOCAL_ROUTE\n");
setflag(22);
sip_trace();
if (is_method("BYE") ) {
xlog("BYE\n");
}
}
to the script but I'm running into a crash (see bellow the log excerpt) caused
b
Hi,
I encountered some problems while trying to use fix_route_dialog()
function.
Basically what I'd like to do is to use this function to correct Route
headers of the BYE messages that
* have wrong or no Route header at all
* have valid to/from & call-id information ( 'valid' in res
Hi Kyle,
I think also the only solution is in textops module - subs_body function:
http://www.opensips.org/html/docs/modules/1.6.x/textops.html#id292763
Regards,
--
Anca Vamanu
www.voice-system.ro
On 01/22/2011 01:17 AM, Kyle Haefner wrote:
Hi All,
I need to change the number in the Messag
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