2 Nov
2017
2 Nov
'17
3:35 p.m.
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
2 Nov
2 Nov
4:43 p.m.
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
5:16 p.m.
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(a)lists.kamailio.org> on behalf of Vasiliy Ganchev
<vancecezar(a)gmail.com>
Sent: Thursday, November 2, 2017 3:43 PM
To: sr-users(a)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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2 comments
2 participants
participants (2)
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hanoulia salameh -
Vasiliy Ganchev