Re: [SR-Users] Kamalio call issue
Hello, Thanks for nice support. When kamalio is not working at some DNS with audio call then how we can know what is issue at client side? Can you give us any online communication like Skype or hangout ID? So we can check it is server or client side issue? Thanks From: Daniel-Constantin Mierla [mailto:mico...@gmail.com] Sent: Friday, March 20, 2015 5:24 PM To: Yogendra Gupta; 'Kamailio (SER) - Users Mailing List' Subject: Re: [SR-Users] Kamalio call issue Hello, again, if the SIP messages don't get to kamailio box, then the problem is on client side, not on server side. If the request gets to kamailio and cannot do dns requests or they don't resolve, you will see a sip reply from kamailio. There is not an issue that can be revealed by the config of kamailio. On the client side, try to use tools like host, dig and see what happens with dns requests. Also, use there ngrep to see if there are sip packets sent out of that box. Cheers, Daniel On 20/03/15 11:20, Yogendra Gupta wrote: 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_48SI P/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 sip:tester1@23.253.110.48;tag=3fdb1d0f. To: tester2 sip:tester2@23.253.110.48 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 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 -- 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 ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Kamalio call issue
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
Re: [SR-Users] Kamalio call issue
Hello, again, if the SIP messages don't get to kamailio box, then the problem is on client side, not on server side. If the request gets to kamailio and cannot do dns requests or they don't resolve, you will see a sip reply from kamailio. There is not an issue that can be revealed by the config of kamailio. On the client side, try to use tools like host, dig and see what happens with dns requests. Also, use there ngrep to see if there are sip packets sent out of that box. Cheers, Daniel On 20/03/15 11:20, Yogendra Gupta wrote: 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_48SIP/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 -- 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 ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Kamalio call issue
Hello, look at the sip traffic on server to see if it is forwarded or not, and if yes, where. You can use ngrep, like: ngrep -d any -qt -W byline sip port 5060 Also, check your syslog file to see if any error message is printed there. It is not possible to guess what a fix would be if the real issue is not known. Cheers, Daniel On 19/03/15 09:59, Yogendra Gupta wrote: Hello, We have setup kamalio at server and it is working at some DNS with call/chat. At some IP chat is working but when we call then it is not working. At some DNS(Net IP) it is working . What can be problem at server? Following is login for SIP test user: tester1@23.253.110.48 123456 Host IP : 23.253.110.48 I have attached mu kamalio config file. Let me know what we need to change at config file or server Thanks ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- 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 ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
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 ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Kamalio call issue
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;tag=ef809ce0. To: 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. Contact: tester2 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;tag=ef809ce0. To: 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. Contact: tester2 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://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015 Berlin, Germany - http://www.kamailioworld.com ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Kamalio call issue
:5060;transport=udp;registering_acc=23_253_110_48 . User-Agent: Jitsi2.6.5390Windows 7. Content-Type: application/sdp. Content-Length: 430. . v=0. o=tester2-jitsi.org 0 0 IN IP4 192.168.0.100. s=-. c=IN IP4 192.168.0.100. t=0 0. m=audio 0 RTP/AVP 96 97 98 9 100 102 0 8 103 3 104 4 101. m=video 5061 RTP/AVP 105 99. a=inactive. a=rtpmap:105 H264/9. a=fmtp:105 profile-level-id=4DE01f;packetization-mode=1. a=imageattr:105 send * recv [x=[0-1600],y=[0-900]]. a=rtpmap:99 H264/9. a=fmtp:99 profile-level-id=4DE01f. a=imageattr:99 send * recv [x=[0-1600],y=[0-900]]. U 2015/03/19 13:00:02.345786 23.253.110.48:5060 - 115.252.208.170:62554 SIP/2.0 200 OK. CSeq: 2 INVITE. Call-ID: 09b36cf7988131e179e345af90922a4c@0:0:0:0:0:0:0:0. From: tester1 sip:tester1@23.253.110.48;tag=aca2e78b. To: sip:tester2@23.253.110.48;tag=53a85e1d. Via: SIP/2.0/UDP 192.168.0.217:5060;rport=62554;received=115.252.208.170;branch=z9hG4bK-34393 8-bbb85207996f538b1ac664a6a065d1fc. Record-Route: 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. User-Agent: Jitsi2.6.5390Windows 7. Content-Type: application/sdp. Content-Length: 449. . v=0. o=tester2-jitsi.org 0 0 IN IP4 23.253.110.48. s=-. c=IN IP4 23.253.110.48. t=0 0. m=audio 0 RTP/AVP 96 97 98 9 100 102 0 8 103 3 104 4 101. m=video 24322 RTP/AVP 105 99. a=inactive. a=rtpmap:105 H264/9. a=fmtp:105 profile-level-id=4DE01f;packetization-mode=1. a=imageattr:105 send * recv [x=[0-1600],y=[0-900]]. a=rtpmap:99 H264/9. a=fmtp:99 profile-level-id=4DE01f. a=imageattr:99 send * recv [x=[0-1600],y=[0-900]]. a=nortpproxy:yes. U 2015/03/19 13:00:02.632817 115.252.208.170:62554 - 23.253.110.48:5060 ACK sip:tester2@192.168.0.100:5060;transport=udp;registering_acc=23_253_110_48;a lias=117.215.244.16~63380~1 SIP/2.0. Call-ID: 09b36cf7988131e179e345af90922a4c@0:0:0:0:0:0:0:0. CSeq: 2 ACK. Via: SIP/2.0/UDP 192.168.0.217:5060;branch=z9hG4bK-343938-155090537c14abf0495a2fba45f5683a. From: tester1 sip:tester1@23.253.110.48;tag=aca2e78b. To: tester2 sip:tester2@23.253.110.48;tag=53a85e1d. Max-Forwards: 70. Route: sip:23.253.110.48;lr=on;nat=yes. Contact: tester1 sip:tester1@192.168.0.217:5060;transport=udp;registering_acc=23_253_110_48 . User-Agent: Jitsi2.6.5390Windows 7. Content-Length: 0. . U 2015/03/19 13:00:02.633118 23.253.110.48:5060 - 117.215.244.16:63380 ACK sip:tester2@192.168.0.100:5060;transport=udp;registering_acc=23_253_110_48 SIP/2.0. Call-ID: 09b36cf7988131e179e345af90922a4c@0:0:0:0:0:0:0:0. CSeq: 2 ACK. Via: SIP/2.0/UDP 23.253.110.48;branch=z9hG4bKf5a.1f70aef5a57bc3412e94ec02b865d6b1.0. Via: SIP/2.0/UDP 192.168.0.217:5060;rport=62554;received=115.252.208.170;branch=z9hG4bK-34393 8-155090537c14abf0495a2fba45f5683a. From: tester1 sip:tester1@23.253.110.48;tag=aca2e78b. To: tester2 sip:tester2@23.253.110.48;tag=53a85e1d. Max-Forwards: 69. Contact: tester1 sip:tester1@192.168.0.217:5060;transport=udp;registering_acc=23_253_110_48; alias=115.252.208.170~62554~1. User-Agent: Jitsi2.6.5390Windows 7. Content-Length: 0. 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://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015 Berlin, Germany - http://www.kamailioworld.com ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
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;tag=ef809ce0. To: 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-343938-6f1017bcd9e693a4959717c9eabdc26e. Record-Route: 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. 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;tag=ef809ce0. To: 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-343938-6f1017bcd9e693a4959717c9eabdc26e. Record-Route: 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. 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 ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
