Hello,
Thanks for your replay.
I made a test with the following scheme:
Browser (SIP.JS) -> OpenSips (Rtp engine) ---> ITSP
Zoiper --> OpenSips(rtpproxy) ---> ITSP
All works fine.
If I want to make the following connection, what RTP must I use:
Zoiper <---> OpenSip()
Dragomir,
Do you intend on having interoperability between standard(AVPF/AVP) sip
devices and WEBRTC? If yes I think rtpengine in the only media relay that
supports translation. Also consider using a library that supports sip
headers. JSSIP or SIPJS
Thanks,
Tito
On Thu, Apr 13, 2017 at 3:00 PM,
Hello,
For WebRTC I must to use rtpengine.
In this case I need to stop rtpproxy?
Best regards,
Dragomir
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
In this case, the opensips docs would also have helped you:
http://www.opensips.org/html/docs/modules/2.2.x/mmgeoip.html#idp55120
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
OpenSIPS Summit May 2017 Amsterdam
Thanks bogdan, ill see how I can work around it keeping launch() in mind.
-Qasim
On Apr 13, 2017 4:28 PM, "Bogdan-Andrei Iancu" wrote:
> The async(), as based on transactions, works only for requests, not for
> replies. This is limitation that will be changed in the next
Yes, this file for IPv4.
For IPv6 /usr/share/GeoIP/GeoLiteCityv6.dat
On 13/04/17 16:37, Bogdan-Andrei Iancu wrote:
Good !!
Is the correct file /usr/share/GeoIP/GeoLiteCity.dat ?
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
OpenSIPS Summit
Hi, John!
The do_routing() function is not adding a branch so it must be added by
someone/something else.
What branch is first sent? The one to the gateway, or the one? Can you
provide a trace for such a call? If privacy is a concern, send it privately.
Best regards,
Răzvan Crainea
OpenSIPS
Of course it is:
https://github.com/OpenSIPS/opensips/blob/master/modules/python/doc/python_admin.xml
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
OpenSIPS Summit May 2017 Amsterdam
http://www.opensips.org/events/Summit-2017Amsterdam.html
Hi Andrej,
Is there any parallel forking ? And, are you sure that the 500 reply has
a Disconnected header ?
Best regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
OpenSIPS Summit May 2017 Amsterdam
Hello,
While testing with v2.2.3 it looks like the behaviour of the do_routing()
function in the DROUTING module has changed.
From packet captures, I can see the call is parallel forking to the original
R-URI *and* to the first choice gateway destination from DRouting.
OpenSIPS sends two INVITE's
Oh, is your documentation in Git?
Robert Mundkowsky
From: Bogdan-Andrei Iancu-2 [via OpenSIPS (Open SIP Server)]
[mailto:ml-node+s1449251n7606955...@n2.nabble.com]
Sent: Thursday, April 13, 2017 5:26 AM
To: Mundkowsky, Robert
Subject: Re: $du not expanded/evaluated?
In
Good !!
Is the correct file /usr/share/GeoIP/GeoLiteCity.dat ?
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
OpenSIPS Summit May 2017 Amsterdam
http://www.opensips.org/events/Summit-2017Amsterdam.html
On 04/13/2017 04:28 PM, Serge S.
I managed to resolve this by reading module docs from Kamailio ;)
Problem was in datafile: GeoIP.dat is GeoLite-Country while module
expects GeoLite-City so after loading correct file resolving working as
expected.
Sorry for noise.
On 13/04/17 15:28, Serge S. Yuriev wrote:
Yes, I tried all
Yes, I tried all types and no param at all. Result the same
On 13/04/17 14:32, Bogdan-Andrei Iancu wrote:
Ok, so the IP to be looked up is properly determined. Have you tried to
change the cache_type to other values ?
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
Ok, so the IP to be looked up is properly determined. Have you tried to
change the cache_type to other values ?
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
OpenSIPS Summit May 2017 Amsterdam
The async(), as based on transactions, works only for requests, not for
replies. This is limitation that will be changed in the next releases
(but not in 2.3).
What you could do in 2.3 is to use launch() statement to run your async
stuff.
launch( blocking_function(...) );
The launch will
Hi,
As I wrote before
DBG:mmgeoip:mmg_lookup_cmd: '62.112.8.216'--> 'Unknown'.
Called as follows
if(mmg_lookup("cc:lon:lat","$si","$avp(lat_lon)")) {
..
}
On 13/04/17 12:28, Bogdan-Andrei Iancu wrote:
Hi,
So, if you run with log_level=4 in script, what are the log messages
from the mmgeoip
Hi Bogdan,
Yes i have to do some accounting calls based on reply recieved. So i guess
that means that async is not available in reply route? Is it still planned
for future release or maybe in 2.3?
Regards,
Qasim
On Thu, Apr 13, 2017 at 3:15 PM, Bogdan-Andrei Iancu
wrote:
Hello. OpenSIPs 2.2.2
Continue http://www.opensips.org/pipermail/users/2009-September/008294.html
"failure_route[1] {
if (t_check_status("500")) {
if ( $(hdr(Disconnected)) != NULL ) {
t_reply("480", "Temporarily Unavailable");
Hi Qasim,
This looks very very similar to an old report:
http://lists.opensips.org/pipermail/users/2017-January/036231.html
And this was fixed in 2.2:
https://github.com/OpenSIPS/opensips/commit/4023a5d797960d245b52eb73dc9fc26b8cdf2914
I guess you try to do some async stuff in the reply route,
Hi Bogdan,
PFA required logs. Also fund below some more info on the setup i am using.
*OS:*
Distributor ID: SUSE LINUX
Description:openSUSE Leap 42.1 (x86_64)
Release:42.1
*Opensips:*
version: opensips 2.2.3 (x86_64/linux)
flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP,
Hi Qasim,
Thank you for your report. Could you please run a "bt full" in gdb a
post the output ?
Best regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
OpenSIPS Summit May 2017 Amsterdam
In terms of both code and documentation, anyone is free and more than
welcome to contribute with improvements or fixes. This is one of the
golden rules of Open Sources Softwares ;)
In regards to the python threads, I do not have the python knowledge to
formulate a comment.
Regards,
Pat,
The Call Center is built in such a way that requires a SIP entity to do
the playback. So, for the announcements and on-hold you must use a SIP
media server to provide the RTP.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
OpenSIPS
Hi,
I have upgraded my script from 1.11 to 2.2.3 which works fine until i put
async function on a single rest_get query. When the async line is executed
i get following errors:
2017-04-13T14:03:45.320300+05:00 sip01 kernel: [31710929.448312]
> opensips[11475]: segfault at 10 ip 00427260
Hi, Khaled!
What patch did you apply?
Can you start a trace and see if the command that is sent from rtpproxy
has the same IP as the one provisioned in the rtpproxy_sock parameter?
RTPProxy does this check and prints that error when the IP is different.
Perhaps rtpproxy is using a private IP
26 matches
Mail list logo