Hello, When we can change our DNS IP then it works with following :
U 2015/03/20 10:06:06.504009 23.253.110.48:5060 -> 202.157.76.21:64051 ACK sip:tester2@192.168.0.100:5060;transport=udp;registering_acc=23_253_110_48 SIP/2.0. Call-ID: 143610160aa023568302b9c999b79f45@0:0:0:0:0:0:0:0. CSeq: 2 ACK. Via: SIP/2.0/UDP 23.253.110.48;branch=z9hG4bK7.601b877c92120368c17b3c0da0802af6.0. Via: SIP/2.0/UDP 192.168.0.217:5060;rport=63789;received=202.157.76.21;branch=z9hG4bK-353035- 5aa153400d9ce60dfc573738a4b55232. From: "tester1" <sip:tester1@23.253.110.48>;tag=3fdb1d0f. To: "tester2" <sip:tester2@23.253.110.48>;tag=31532119. Max-Forwards: 69. Contact: "tester1" <sip:tester1@192.168.0.217:5060;transport=udp;registering_acc=23_253_110_48; alias=202.157.76.21~63789~1>. User-Agent: Jitsi2.6.5390Windows 7. Content-Length: 0. But when we used another DNS IP (internet) and call then it showing only initialize.. If it has firewall issue then it will not work at all DNS IPs. I have attached before my config file for kamalio. Can you tell me what can be issue that it works at some DNS IP and not at all? Thanks From: Daniel-Constantin Mierla [mailto:mico...@gmail.com] Sent: Friday, March 20, 2015 2:14 AM To: Yogendra Gupta; 'Kamailio (SER) - Users Mailing List' Subject: Re: [SR-Users] Kamalio call issue These are replies to INVITE requests, but if you see them, the INVITE passed through the server as well. If you are not aware of a firewall, then perhaps you don't have one unless is a default installation with it enabled or one on the network. I suggest you do sip tracing on the client machine to see if the invite requests leave to the proper IP. Ultimately can be also a problem caused by a NAT router with ALG, if the client is behind such device. Cheers, Daniel On 19/03/15 13:50, Yogendra Gupta wrote: 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@0:0:0:0:0:0:0:0. From: "tester1" <sip:tester1@23.253.110.48> <sip:tester1@23.253.110.48>;tag=ef809ce0. To: <sip:tester2@23.253.110.48> <sip:tester2@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> <sip:23.253.110.48;lr=on;nat=yes>. Contact: "tester2" <sip:tester2@192.168.0.100:5060;transport=udp;registering_acc=23_253_110_48> <sip:tester2@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@0:0:0:0:0:0:0:0. From: "tester1" <sip:tester1@23.253.110.48> <sip:tester1@23.253.110.48>;tag=ef809ce0. To: <sip:tester2@23.253.110.48> <sip:tester2@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> <sip:23.253.110.48;lr=on;nat=yes>. Contact: "tester2" <sip:tester2@192.168.0.100:5060;transport=udp;registering_acc=23_253_110_48; alias=117.215.244.16~63380~1> <sip:tester2@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:mico...@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://twitter.com/#%21/miconda> - http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015 Berlin, Germany - http://www.kamailioworld.com -- 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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