Hi,
I want to handle the responses for forked Requests in kamailio.
Please suggest me that where i can handle this in configuration.
Here is my requirement.
If Clinet A is registered with kamailio server on pc 1 and pc 2. If Client
B calls A, Server is sending INVITE to A for both PC 1 and PC 2.
As per current implementation Server closing the INVITE session if any
one(pc1/pc2) responds with 486.
But i want to handle the 486 response, and wait till other one(pc1/pc2)
also responds instead of closing INVITE session.
Regards,
Raman
Klaus,
Thanks for the tip. I have been using (uri==myself) up to know but I was
trying to move towards a more multidomain configuration, was kinda
hoping to get rid of hardcoded stuff in particular servers.
Daniel,
Can the fact that port in ruri makes host part not longer matching as
own domain be considered a bug or it is the accepted working mechanism?
The "register_myself" parameter made me think that it should match host
part at least (port is not important for me), but in my case port in
ruri makes the whole host not matching. Hope I have clearly expressed my
thoughts ;).
Ta,
DanB
Guys,
I was wondering if I am doing something wrong or domain module not
considering properly local ips within is_uri_host_local().
Although I have domain loaded with register_myself on 1, the request
going to my IP is simply not matching local domain (listening on
127.0.0.1 port 5070).
Bellow the trace of such request:
"""
#
U 2013/05/30 12:35:56.436485 10.10.10.21:5060 -> 127.0.0.1:5070
OPTIONS sip:127.0.0.1:5070 SIP/2.0.
Via: SIP/2.0/UDP
10.10.10.21;branch=z9hG4bK4866.b4f45983000000000000000000000000.0.
To: <sip:127.0.0.1:5070>.
From: <sip:ep@iec.itsyscom.com>;tag=ae9b2706b606c3acb0ebe4f1c8f81cee-f20d.
CSeq: 10 OPTIONS.
Call-ID: 467de807489c4482-3002(a)10.10.10.21.
Max-Forwards: 70.
Content-Length: 0.
User-Agent: iClass4-EP 4.0.0.
.
#
U 2013/05/30 12:35:56.440410 127.0.0.1:5070 -> 10.10.10.21:5060
SIP/2.0 484 Address Incomplete.
Via: SIP/2.0/UDP
10.10.10.21;branch=z9hG4bK4866.b4f45983000000000000000000000000.0.
To: <sip:127.0.0.1:5070>;tag=46a6e639fa023622ac1ba4fea686e961.d61e.
From: <sip:ep@iec.itsyscom.com>;tag=ae9b2706b606c3acb0ebe4f1c8f81cee-f20d.
CSeq: 10 OPTIONS.
Call-ID: 467de807489c4482-3002(a)10.10.10.21.
Server: iClass4-AP 4.0.0.
Content-Length: 0.
"""
My script looks something like bellow, so Address Incomplete should
never be reached:
"""
#!define LISTEN_IP 127.0.0.1
#!define LISTEN_PORT 5070
...
listen=LISTEN_IP
port=LISTEN_PORT
auto_aliases=yes
...
# ----- domain params -----
modparam("domain", "db_url", DBURL)
modparam("domain", "register_myself", 1)
...
if (is_method("OPTIONS") && is_uri_host_local()) {
options_reply();
exit;
}
...
if ($rU==$null) {
# request with no Username in RURI
sl_send_reply("484","Address Incomplete");
exit;
}
"""
I am running on git master with the test patch Daniel did few days back
for me.
Thanks in advance for any tip!
DanB
Dear All,
I am trying to integrate kamailio 4.0.1 server with Asterisk 11.4.0 in
Ubuntu 12.04 LTS version. I tried it by using the following link :
http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb.
But after all, when i tried to start Kmailio server with
/etc/init.d/kamailio start command , i am getting the following errors:
asterisk-kamailio@asterisk-kamailio:/$ sudo /etc/init.d/kamailio start
[sudo] password for asterisk-kamailio:
/etc/init.d/kamailio: 1: /etc/init.d/kamailio: sed: not found
/etc/init.d/kamailio: 118: /etc/init.d/kamailio: arithmetic expression:
expecting primary: ""
what could be the wrong, and how can i solve it?
Please share your valuable suggestions on this.
Any help will greatly appreciate.
*Thank you and Best regards,*
*R Ravindra **Gowda*
*Engineer - Systems and Products*.
*Thrikasa Technologies*
*| Office: +91 40 23260434
| Mob: +918885284050
*
*| Fax:+91 40 23221045*
*| email: ravindra(a)thrikasa.in*
*| Web: www.thrikasa.in*
'If you don't give up,you cannot fail'.
The information contained in this communication is proprietary to Thrikasa
Technologies., and/or third parties, may contain classified or privileged
information, and is intended only for the use of the intended addressee
thereof. If you are not the intended addressee, please be aware that any
use, disclosure, distribution and/or copying of this communication is
strictly prohibited.If you receive this communication in error, please
notify the sender immediately and delete it from your computer.
Hi,
When I want to unregister, I have 2 ways
1. Append ;expires=0 in Contact field
2. Add another Expires : 0 header
with CSeq, Call-ID not the same with when i register
Does Kamailio check for the saem Cseq, Call-ID to allow unregistration ?
--
Khoa Pham
HCMC University of Science
www.fantageek.com
Dear All,
I am trying to integrate kamailio 4.0.1 server with Asterisk 11.4.0 in
Ubuntu 12.04 LTS version. I tried it by using the following link :
http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb.
But i am unable to accomplish it, i just ended with some mysql errors.
I am very new to this aspect. So can anyone suggest me any other standard
reference procedure or any tutorials to do this.
PS: my host Architecture: i686
Any help will greatly appreciate.
*Thank you and Best regards,*
*R Ravindra **Gowda*
*Thrikasa Technologies*
*| Office: +91 40 23260434
| Mob: +918885284050
*
*| Fax:+91 40 23221045*
*| email: ravindra(a)thrikasa.in*
*| Web: www.thrikasa.in*
'If you don't give up,you cannot fail'.
The information contained in this communication is proprietary to Thrikasa
Technologies., and/or third parties, may contain classified or privileged
information, and is intended only for the use of the intended addressee
thereof. If you are not the intended addressee, please be aware that any
use, disclosure, distribution and/or copying of this communication is
strictly prohibited.If you receive this communication in error, please
notify the sender immediately and delete it from your computer.
Hi,
Authentication for users is working fine for me, I add a user to the db and the phone can register. I am looking to connect a SIP trunk but there doesn't seem to be any authentication? What I want is to allow users to connect from any IP but only allow SIP trunks to connect from certain IPs and authenticate.
Thanks,
Keith
