Hi Team,
Greetings!
I was trying to use HTTP to do the subscriber management (CRUD operations) & for that I am using JSONRPC-S module (JSONRPC over HTTP). But like other modules e.g. dispatcher, location I am unable to find any RPC commands for doing subscriber management (checked in auth_db) module. So is there a way to achieve subscriber management using JSONRPC over HTTP.
Few HTTP examples which I am using for dispatcher module are given below. Similar commands I am looking for subscriber management as well.
curl http://localhost:8081/RPC -H "Content-Type: application/json" -d " {\"jsonrpc\":\"2.0\",\"method\":\"dispatcher.add\", \"params\":[2, \"sip:127.0.0.1:5061\"], \"id\": \"1\"}"
curl http://localhost:8081/RPC -H "Content-Type: application/json" -d " {\"jsonrpc\":\"2.0\",\"method\":\"dispatcher.remove\”, \"params\":[2, \"sip:127.0.0.1:5061\"], \"id\": \"1\"}"
Surabhi Garg
Mobile - 91 783 888 7523
Hi,
I'm having the following scenario
UA <---> Prox1 <----> Proxy2 <-----> B2BUA <------> UA
Proxy2 and B2BUA are connected on LAN and Proxy1 is adding his own Contact
Alias and Proxy 2 is adding his own Contact Alias
I'm adding Alias via using the following function set_contact_alias("1") as
it will remove other contact alias but is causing the problem in the case
of BYE coming from B2BUA.
Is there is a way to find if the contact alias is already set?
How I can replace the local IP alias with domain name when request is
coming from B2BUA and alias is added by Proxy2.
Here are logs to show problem
in 200 sent by B2BUA and proxy2 add alias 10.44.44.2~6000~1 Now in ACK it
is causing the problem
SIP/2.0 200 OK
Via: SIP/2.0/UDP sip.prox1.example.com:5060
;rport=5060;received=3.18.1.21;branch=z9hG4bK2d66.a89039bedfe3d1113132f04c94b8a757.0
Via: SIP/2.0/UDP 192.168.1.168:50279
;received=109.8.2.1;rport=45607;branch=z9hG4bKPj84875d08eafc48bfa6c29de5fe245cea
Record-Route: <sip:sip.proxy2.example;lr;did=e84.87e2;rtp=bridge>
Record-Route: <sip:sip.prox1.example.com;lr;did=e84.7452;rtp=bridge>
From: "8001" <sip:8001@sip.example.com>;tag=0b9915abae654e21845fd4d7f577ede9
To: <sip:2501@sip.example.com>;tag=as1f4f6b1c
Call-ID: a0095b9868924b0681225a8884a06c22
CSeq: 3687 INVITE
Server: Voip
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:2501@10.44.44.2:6000;alias=10.44.44.2~6000~1>
Content-Type: application/sdp
Require: timer
Content-Length: 281
Ack for 200
ACK sip:2501@10.44.44.2:6000 SIP/2.0
Via: SIP/2.0/UDP sip.proxy2.example.com:5060
;branch=z9hG4bK2d66.3c10a3b45e5c8caacbc5853d5e6418b2.0
Via: SIP/2.0/UDP sip.proxy1.example.com:5060;rport=5060;received= 3.18.1.21
;branch=z9hG4bK2d66.788312006d5f7aedc058435cd203fed3.0
Via: SIP/2.0/UDP 192.168.1.168:50279;received= 109.8.2.1
;rport=45607;branch=z9hG4bKPjf003c3bfe4ba423e9a0dedd647511dd3
Max-Forwards: 68
From: "8001" <sip:8001@sip.example.com>;tag=0b9915abae654e21845fd4d7f577ede9
To: <sip:2501@sip.example.com>;tag=as1f4f6b1c
Call-ID: a0095b9868924b0681225a8884a06c22
CSeq: 3687 ACK
Content-Length: 0
Thanks in advance
I am unable to base64url decode the json payload in identity header generated by secsipid.
(Using python for test)
decoded_payload = url64.decode(‘payload’)
UnicodeDecodeError: 'utf-8' codec can't decode byte 0xc2 in position 27: invalid continuation byte
Header decodes fine this way but not payload.
Is this an issue with the payload encoding?
Kamailio 5.5
Daniel W. Graham, CTO
CMSInter.net LLC
DIRECT (989) 400-4230
INTERNET | TELEPHONE | MANAGED IT
Hi,
[Note I posted this yesterday from the wrong email address.]
I am struggling to see why my SDP is not being set correctly on the INVITE to my supplier with Proxy-Authentication when I use uac_auth().
The initial INVITE to my supplier has correct SDP set by rtpengine_manage. Then supplier replies with a 407. My failure route correctly handles the Auth, and also calls NATMANAGE again... but this time the SDP is unchanged and the private IP and original media information from the original device is relayed to my supplier.
kamailio.cfg isbased on the default config. Running kamailio 5.4.6 and using example from uac module for uac_auth.
My failure route calls NATMANAGE. Is there anything special about uac_auth? Do I need some extra magic to apply the message body changes after I have run rtpengine_manage().
I can see that the NATMANAGEr test for nat_uac_test("8") is true, and rtpengine_manage() s being called. But the outgoing SDP is not changed.
Thanks for any hints!
Paul
====================
Extract of kamailio.cfg
failure_route[TRUNKAUTH] {
if (t_is_canceled()) {
exit;
}
route(NATMANAGE);
xlog("L_INFO","In failure route, just finshed NATMANAGE and now body is $mb");
if(t_check_status("401|407")) {
# $avp(auser) = "test";
# $avp(apass) = "test";
# $avp(apass) = "36d0a02793542b4961e8348347236dbf";
if (uac_auth()) {
t_relay();
}
exit;
}
}
Hi,
I am struggling to see why my SDP is not being set correctly on the
INVITE to my supplier with Proxy-Authentication when I use uac_auth().
The initial INVITE to my supplier has correct SDP set by
rtpengine_manage. Then supplier replies with a 407. My failure route
correctly handles the Auth, and also calls NATMANAGE again... but this
time the SDP is unchanged and the private IP and original media
information from the original device is relayed to my supplier.
kamailio.cfg isbased on the default config. Running kamailio 5.4.6 and
using example from uac module for uac_auth.
My failure route calls NATMANAGE. Is there anything special about
uac_auth? Do I need some extra magic to apply the message body changes
after I have run rtpengine_manage().
====================
Extract of kamailio.cfg
failure_route[TRUNKAUTH] {
if (t_is_canceled()) {
exit;
}
route(NATMANAGE);
xlog("L_INFO","In failure route, just finshed NATMANAGE and now body is $mb");
if(t_check_status("401|407")) {
# $avp(auser) = "test";
# $avp(apass) = "test";
# $avp(apass) = "36d0a02793542b4961e8348347236dbf";
if (uac_auth()) {
t_relay();
}
exit;
}
}
Hello,
I am trying to use Kamailio in our application which would be deployed in certain environment in addition to a cloud infrastructure. As I understand it, parts of the Kamailio code are covered under the BSD license - are you able to share what those are? I am trying to determine if Kamailio can be used in a customer environment or should it be strictly restricted to cloud environments due to the GPL v2 licenses
Thanks,Steve
