This is what i did but its not working, getting error SIP/2.0 403
Forbidden, it is thinking number i am dialing is local and checking in
local DB . by the way SIP provider Trunk is already registered using UAC
module. I am using Multi-domain setup.
# do lookup with method filtering
if (!
Hi,
The source for this string to int conversion error was found, it was just a
minor glitch in an if statement! Man, I feel stupid...
Anyways, the problem about calls not going through still persists. This I
located to the rtpengine_offer() call in a branch route. The sdp is not
changed and this
Hello,
I'm having trouble with this scenario (Kamailio and Asterisk are working on
the same server, Asterisk listens on 4060 instead of 5060): the UAC sends an
ACK request with the following R-URI: sip:955*95%23@
:4060.
When I'm doing a capture on loopback interface, I just see an ACK request
On Wednesday 20 August 2014 14:54:42 Satish Patel wrote:
> I am new in Kamailio so could you please give me code example how to use
> t_relay() to forward traffic to Provide, I know how to use rewritehost()
> but i never use t_relay() function
Well, my guess is your routing ends with t_relay().
B
My example don`t help you. you must read about t_relay there.
t_relay() is not central thing. I must stop your attension at SIP invite
that goes to provider. t_relay simple to use- just customise your INVITE
and call t_relay() from your route.
http://kamailio.org/docs/modules/devel/modules/tm.htm
I am new in Kamailio so could you please give me code example how to use
t_relay() to forward traffic to Provide, I know how to use rewritehost()
but i never use t_relay() function
On Wed, Aug 20, 2014 at 8:22 AM, Yuriy Gorlichenko
wrote:
> You can use t_relay() too. One thing that you need - t
You can use t_relay() too. One thing that you need - to have right packet,
that will be relays to Provider. I have multiple providers and manually
change packets that will send to provider.
2014-08-20 15:57 GMT+04:00 Satish Patel :
> Great! I registered remote Trunk using UAC module. so now i ca
Great! I registered remote Trunk using UAC module. so now i can just use
following function to forward my call right?
rewritehost()
On Wed, Aug 20, 2014 at 12:33 AM, Yuriy Gorlichenko
wrote:
> Use UAC module for this
> 20.08.2014 7:40 пользователь "Satish Patel"
> написал:
>
>> We have setup
Djemel,
Even though your file /usr/premises/etc./kamailio/kamailio.cfg might look ok,
even if it is perfect...
It is at the wrong location.
The init script expects the config file to be at
/usr/local/etc/kamaiilio/kamailio.cfg
So you either copy your config file to that location or edit the
Sweet!!!
Thanks Carsten!
-Original Message-
From: sr-users-boun...@lists.sip-router.org
[mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of Carsten Bock
Sent: Wednesday, August 20, 2014 11:44 AM
To: Kamailio (SER) - Users Mailing List
Subject: Re: [SR-Users] Generate SIP MESSAGE
I don know how to do one thing:
As I understand I need to set values for variables
modparam("uac","auth_realm_avp","$avp(i:10)")
modparam("uac","auth_username_avp","$avp(i:11)")
modparam("uac","auth_password_avp","$avp(i:12)")
Am I right?
If I get query results with password and realm through s
Hi Grant,
you could try uac_req_send():
http://kamailio.org/docs/modules/devel/modules/uac.html#uac.f.uac_req_send()
Kind regards,
Carsten
2014-08-20 11:40 GMT+02:00 Grant Bagdasarian :
> Hello,
>
>
>
> Is it possible for Kamailio to generate a SIP MESSAGE message when receiving
> a HTTP request
Hello,
Is it possible for Kamailio to generate a SIP MESSAGE message when receiving a
HTTP request?
Take the content of the HTTP message and form it into a SIP MESSAGE and route
it to a certain destination?
I can't find a module that does this, or I'm just blind.. :)
Regards,
Grant
Hello,
see avpops or sqlops for how to load values from database and access
them in config file via variables.
Cheers,
Daniel
On 19/08/14 22:53, Yuriy Gorlichenko wrote:
Hello. I suceesfully authenticate some accounts of providers from UAc
using DB.
When I try to Call to any provider it re
Use rtpengine for this. You may use rtpproxy-ng module to manipulate
options of rtpengine.
20.08.2014 11:07 пользователь написал:
> Hi,
>
> We are using Kamailio as a WebRTC proxy. We have converted the signaling
> successfully.
>
> Now, for media, is it possible to convert srtp to rtp using rtpp
Hi,
We are using Kamailio as a WebRTC proxy. We have converted the signaling
successfully.
Now, for media, is it possible to convert srtp to rtp using rtpproxy_ng or
mediaproxy? If yes can you provide me with some details?
Thanks in advanced!
Sent from my “contract free” BlackBerry® smartphon
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