[SR-Users] Kamalio call issue
Yogendra Gupta
yogendra at konstantinfosolutions.com
Thu Mar 19 13:50:07 CET 2015
Hello,
When I am calling with other SIP user then I did not see any INVITE . that
have issue with DNS.
If we call with different DNS that is working fine then we see INVITE option
like
U 2015/03/19 12:39:01.744616 117.215.244.16:63380 -> 23.253.110.48:5060
SIP/2.0 180 Ringing.
CSeq: 2 INVITE.
Call-ID: d83c4bc1e75e54df5ebd06b74f9089ef at 0:0:0:0:0:0:0:0.
From: "tester1" <sip:tester1 at 23.253.110.48>;tag=ef809ce0.
To: <sip:tester2 at 23.253.110.48>;tag=23a5eaea.
Via: SIP/2.0/UDP
23.253.110.48;branch=z9hG4bKa1a3.21d8ef51bac2678fc26eca5975ae7b00.0,SIP/2.0/
UDP
192.168.0.217:5060;rport=62554;received=115.252.208.170;branch=z9hG4bK-34393
8-6f1017bcd9e693a4959717c9eabdc26e.
Record-Route: <sip:23.253.110.48;lr=on;nat=yes>.
Contact: "tester2"
<sip:tester2 at 192.168.0.100:5060;transport=udp;registering_acc=23_253_110_48>
.
User-Agent: Jitsi2.6.5390Windows 7.
Content-Length: 0.
.
U 2015/03/19 12:39:01.744870 23.253.110.48:5060 -> 115.252.208.170:62554
SIP/2.0 180 Ringing.
CSeq: 2 INVITE.
Call-ID: d83c4bc1e75e54df5ebd06b74f9089ef at 0:0:0:0:0:0:0:0.
From: "tester1" <sip:tester1 at 23.253.110.48>;tag=ef809ce0.
To: <sip:tester2 at 23.253.110.48>;tag=23a5eaea.
Via: SIP/2.0/UDP
192.168.0.217:5060;rport=62554;received=115.252.208.170;branch=z9hG4bK-34393
8-6f1017bcd9e693a4959717c9eabdc26e.
Record-Route: <sip:23.253.110.48;lr=on;nat=yes>.
Contact: "tester2"
<sip:tester2 at 192.168.0.100:5060;transport=udp;registering_acc=23_253_110_48;
alias=117.215.244.16~63380~1>.
User-Agent: Jitsi2.6.5390Windows 7.
Content-Length: 0.
Can you tell me what can be issue of firewall dropping?
When I checked at server firewall:
sudo ufw status
Status: inactive
Let me know what can be other issue for it..
Thanks
From: Daniel-Constantin Mierla [mailto:miconda at gmail.com]
Sent: Thursday, March 19, 2015 5:50 PM
To: Yogendra Gupta; 'Kamailio (SER) - Users Mailing List'
Subject: Re: [SR-Users] Kamalio call issue
Hello,
OPTIONS is not the request for initiating the calls, that is INVITE. You
would need to know SIP a bit in order to be able to understand and configure
Kamailio.
If you don't see any INVITE on kamailo server via ngrep when you call, then
the issue is on client side or there is a firewall dropping it.
Cheers,
Daniel
On 19/03/15 11:39, Yogendra Gupta wrote:
Hello,
Thanks for nice support.
When we call to test2 user and run this command at server
ngrep -d any -qt -W byline "sip" port 5060
then we found following response at server:
--
Daniel-Constantin Mierla
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio World Conference, May 27-29, 2015
Berlin, Germany - http://www.kamailioworld.com
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