Hi everyone,
I have 1 Load Balancer and 2 SIP Servers, all of them are Kamailio. Does
anyone suggest how to add NAT for them?
Best Regard,
Nguyen Anh Tuan
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users
Greetings.
I try to add rls/xcap presence into resiprocate-based user agent.
May anyone tell me what module is responsible for parsing presence rules
documents?
I'm trying to setup "allow all" presence rules document but did not get
success...
Thank you!
Also a sip trunk account from any providers with strong credentials will
serve the purpose. Connecting to internet could be dynamic IP as provided
by many dsl providers. This will not give any security risks as you will
not open any inbound ports.
On Fri, Dec 7, 2012 at 8:00 PM, Jijo wrote:
> H
Hi,
Considering the security aspect of SIP Trunk and the cost of dedicated TDM
trunk, the cheapest option for you is to use your home PSTN line as TDM
trunk..
To do that buy a PCI FXO card for Asterisk and setup the routing in
asterisk either directly to your lan SIP phone(Xlite) or via
openser/k
Hi,
I'm using kamailio-3.3.2, it contains this changes, just checked. Yes db is
also workaround.
Thanks,
Pavel.
2012/12/7 Daniel-Constantin Mierla
> Hello,
>
> what version are you using? There was an enhancement to the ctl module to
> cope better with large data:
>
>
> http://git.sip-router.
Hello,
what version are you using? There was an enhancement to the ctl module
to cope better with large data:
http://git.sip-router.org/cgi-bin/gitweb.cgi/sip-router/?a=commit;h=3d195f2675569954a1f74128508db07cbc604ed9
An alternative will be to fetch the data from database and send via
mi/rp
Hi.
In our project we need to query current users, dialogs and ability to end
dialog using some rpc. Requirements are - do this remotely. I already tried
to use xmlrpc module. When we had 42 users online it returned 17kb broken
xml file. Then I tried to use binrpc protocol from CTL module. Usage e
While this is not Kamailio - it's related. If you want to use Kamailio in front
of Asterisk, you do need SIP Path header support in Asterisk to do it right.
Especially if you have one Kamailio handling Internet-facing communication and
one handling internal communication.
Please test!
/O
Vida
HI,
For this you need connect to a telecom operator in India or a sip trunk
provider. But I doubt that any telecom operator would do it. You can connect
to an international SIP trunk provider. There will be lot's of packages for
rates and payments. You can easily search the internet for possibl
As a word of advice.
You will have to pay someone for the mobile or PSTN calls. It is very
unsafe to open your system to the public without very good security,
especially very strong user passwords.
There are constant attacks by robots that attempt to break into systems
and get free (for them) ca
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