Hi,
In DB mode 2, DB only replication is not enough as the mem cache on
backup machine will never be updated with the registrations. So you end
up with a discrepancy (on backup) between what opensips has in memory
and what you have in DB (in terms of registered contacts).
You definitely need
Hi Satish,
The dialog module does TH at SIP level (and not at RTP level) - if you
want to do that, we need to use (in conjunction to dialog TH) a media
relay (as mediaproxy or rtpproxy) in order to hide also the RTP side.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
Hi Satish,
My question was if in logs you get any other ERR message (before the
ones you posted) from the b2b_entities module.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 27.08.2014 22:28, Satish Patel wrote:
Whenever I call those error
Hi Eugene,
For routing between 2 different networks you just need to do:
1) listen on both interfaces
2) enable mhomed
3) if you want to control when to cross between the networks, aside
setting the proper destination, use the force_send_socket() function to
set the outbound SIP socket to
Hi John,
Actually you should use the uac_replace_to() function from the uac module:
http://www.opensips.org/html/docs/modules/1.11.x/uac.html#id293710
This wil do a SIP-wise change over the TO header (a dialog persistent
change).
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
Hi Igor,
Right now no - the problem is that the nat pinging is not able to report
anything back ; flaging is not a problem here.
We plan to have such support for the next release.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 01.09.2014
Hi Satish,
SO you have the same call going twice through same FS box ? and when
hitting for the second time you get the 482 ? (once again, for the same
call)
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 01.09.2014 21:34, Satish Patel
2014-09-02 12:30 GMT+04:00 Bogdan-Andrei Iancu :
For routing between 2 different networks you just need to do:
1) listen on both interfaces
2) enable mhomed
3) ... force_send_socket() ...
It looks like not enought - see my first message with all details in
this thread. I need to configure
Hi all,
The second topic discussed during the last IRC meeting[1] was about
building a Fraud Detection module that prevents PBX or accounts hijacking.
Basically the module will allow you to define different dialing profiles
(the destination you are dialing, how often, how many parallel
Hi all,
For the next OpenSIPS release we discussed about building a
Quality-based routing module.
Using this new module we aim to gather statistics about gateways (PDD,
ASR, ACD, CCR, probably RTP stats) and use this information to reorder
the gateways. These statistics are dynamically
Hi all,
The first topic presented in the last public meeting[1] was related to
Message compression - the idea is to reduce the size of the SIP messages
to avoid UDP fragmentationor to save bandwidth and processing power.
This topic was split in two parts:
1) SIP-wise changes over the
Hi all,
Among the last discussion of the last IRC meeting[1] was related to
Asynchronous processing in OpenSIPS script - we want to add a new
mechanism that allows you to perform asynchronous operations (such as DB
, REST or exec operations) directly from the script. Using this feature
will
Hi Razvan,
My test server crashed and I couldnt extract the info from the core dump.
Wanted to do it on a new server; but realized that the patch doesnt exist any more athttp://pastebin.com/TuEZ26Qz
Can you please send me the patch again?
Thanks
-Gary
Sent:Tuesday, July 15, 2014 at
Hi Dmitry,
If for a single SIP reply you arm both failure and reply route, then the
reply route will be first executed and then the failure route.
For different replies (of the same transaction), the reply route may be
run in parallel, but failure route not !
Regards,
Bogdan-Andrei Iancu
Do you have solution or documents to do that?
On Tue, Sep 2, 2014 at 4:10 AM, Bogdan-Andrei Iancu bog...@opensips.org
wrote:
Hi Satish,
The dialog module does TH at SIP level (and not at RTP level) - if you
want to do that, we need to use (in conjunction to dialog TH) a media relay
(as
Yes, I have send you email in detail about my scenario, but Yes, you are
right, UA will register to opensips and dial 00123456789, opensips will
route that call to Freeswitch, Freeswitch has dialplan to send that call
back to opensips removing 00 prefix and then opensips will forward that
call to
I'm curious what the failure cases look like here..
What happens when your long running db query is excessively long? Is
there a timeout? What happens if you have 1000s of backed up processes.
Would love to see some sort of fifo command somehow quantify the
effectiveness of it being async. Not
Hello,
SIP body compression using gzip has already been implemented in a few places:
http://blog.krisk.org/2013/09/learning-stupid-nat-tricks-from-apple.html
https://jira.freeswitch.org/browse/FS-5814
Compliance with this standard in OpenSIPS would be a good thing :).
On Tue, Sep 2, 2014
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