Hello,
I have a problem with kamailio (v4.4), when i try to call from a sip phone through kamailio, the call is interrupted and hang up automatically but after 4s it continues to retry (I can see in the log file Session Progress) and goes through the asterisk (certified/13.13-cert4) and establishes a normal call (It rings at the client side) while it's still disconnected from the sip phone side But when i try again it works fine sometimes. Am I missing something in my configuration file? What can I show in order to help me?
Regards
hi!
try to: - look in tcpdump of your SIP traffic (use wireshark) - add logging to your kamailio.cfg
both points to understand the difference between 2 calls (good and failed).
The description that you prepared is not informative enough - to understand what is wrong (important point: kamailio is sipproxy, and does not operate with such things as "call")
try to dig more, prepare more info on the point of view of SIP and share here.
cheers
-- Sent from: http://sip-router.1086192.n5.nabble.com/Users-f3.html
Hello,
Below you can find attached the 2 files of tcpdump
and for the logging I only see an error when it fails in this section where xlog("L_ERR","666666666666666") :
route[4] {
if (is_method("BYE|CANCEL")) { unforce_rtp_proxy(); } else if (is_method("INVITE")){ xlog("L_INFO","4444444444444444"); if (nat_uac_test("8")) { rtpproxy_offer(); #force_rtp_proxy(); } else { rtpproxy_offer(); #force_rtp_proxy(); } xlog("L_ERR","666666666666666"); t_on_failure("FAIL_ONE"); t_on_reply("REPLY_ONE");
};
if (isflagset(5)) search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes'); }
Regards
________________________________ From: sr-users sr-users-bounces@lists.kamailio.org on behalf of Vasiliy Ganchev vancecezar@gmail.com Sent: Thursday, November 2, 2017 3:43 PM To: sr-users@lists.sip-router.org Subject: Re: [SR-Users] Kamailio not calling each time
hi!
try to: - look in tcpdump of your SIP traffic (use wireshark) - add logging to your kamailio.cfg
both points to understand the difference between 2 calls (good and failed).
The description that you prepared is not informative enough - to understand what is wrong (important point: kamailio is sipproxy, and does not operate with such things as "call")
try to dig more, prepare more info on the point of view of SIP and share here.
cheers
-- Sent from: http://sip-router.1086192.n5.nabble.com/Users-f3.html
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