<html><head><title>Re: [SR-Users] Kamailio Asterisk Retransmitting problem</title>
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<span style=" font-family:'Courier New'; font-size: 9pt;">Hello,<br>
<br>
I found that same problem also is with BYE<br>
<br>
call flow: Provider 213.21.197.50 -> Kamailio 37.148.171.162 -> Asterisk 37.148.171.163<br>
<br>
This BYE send iniciator<br>
<br>
U 2019/02/20 18:44:02.341540 213.21.197.50:5060 -> 37.148.171.162:5060 #12299<br>
BYE sip:37.148.171.162;line=sr-BgYAQMhA1hcE6pnO6u9s1oBOVMhg1Eai1uSA SIP/2.0.<br>
Via: SIP/2.0/UDP 213.21.197.50:5060;branch=z9hG4bK6b28c7be.<br>
Max-Forwards: 70.<br>
From: "Yealink VC Mobile" <sip:37125511039@213.21.197.50>;tag=as4be8f9bb.<br>
To: <sip:100@37.148.171.162:5060>;tag=as5e2133c0.<br>
Call-ID: 4eb0425445ea0d41269ef2b70e82a773@213.21.197.50:5060.<br>
CSeq: 103 BYE.<br>
User-Agent: Asterisk PBX 14.6.2.<br>
X-Asterisk-HangupCause: Normal Clearing.<br>
X-Asterisk-HangupCauseCode: 16.<br>
Content-Length: 0.<br>
<br>
<br>
And this we see on kamailio side<br>
<br>
U 2019/02/20 20:45:24.545345 213.21.197.50:5060 -> 37.148.171.162:5060 #136<br>
BYE sip:37.148.171.162;line=sr-BgYAQMhA1hcE6pnO6u9s1oBOVMhg1Eai1uSA SIP/2.0.<br>
Via: SIP/2.0/UDP 213.21.197.50:5060;branch=z9hG4bK3c5c07d9.<br>
Max-Forwards: 70.<br>
From: "Yealink VC Mobile" <sip:37125511039@213.21.197.50>;tag=as627af367.<br>
To: <sip:100@37.148.171.162:5060>;tag=as2795e468.<br>
Call-ID: 5e90daae78528a5b1d7447fc6c9eae29@213.21.197.50:5060.<br>
CSeq: 103 BYE.<br>
User-Agent: Asterisk PBX 14.6.2.<br>
X-Asterisk-HangupCause: Normal Clearing.<br>
X-Asterisk-HangupCauseCode: 16.<br>
Content-Length: 0.<br>
.<br>
erver: Asterisk<br>
<br>
U 2019/02/20 20:45:24.546130 37.148.171.162:5060 -> 213.21.197.50:5060 #137<br>
SIP/2.0 404 Not here.<br>
Via: SIP/2.0/UDP 213.21.197.50:5060;branch=z9hG4bK3c5c07d9.<br>
From: "Yealink VC Mobile" <sip:37125511039@213.21.197.50>;tag=as627af367.<br>
To: <sip:100@37.148.171.162:5060>;tag=as2795e468.<br>
Call-ID: 5e90daae78528a5b1d7447fc6c9eae29@213.21.197.50:5060.<br>
CSeq: 103 BYE.<br>
Server: kamailio (5.2.1 (x86_64/linux)).<br>
Content-Length: 0.<br>
<br>
BR,<br>
Alex<br>
<br>
Tuesday, February 19, 2019, 5:09:48 PM, you wrote:<br>
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<span style=" font-family:'courier new'; font-size: 9pt;">Could you please help me correctly to configure Kamailio with load balancing between two asterisk servers and topology hidding with topoh module.<br>
In attach sip trace from kamailio and my config.<br>
<br>
Currently my problem is that after call UP on Asterisk side, Asterisk send 10 times Retransmitting to <br>
<-------------><br>
Retransmitting #5 (no NAT) to </span><a style=" font-family:'courier new'; font-size: 9pt;" href="http://37.148.171.162:5060">37.148.171.162:5060</a><span style=" font-family:'courier new'; font-size: 9pt;">:<br>
SIP/2.0 200 OK<br>
Via: SIP/2.0/UDP 37.148.171.162;branch=z9hG4bKf534.157d7d1321019895cc6ba367dd3ae94c.0;received=37.148.171.162<br>
Via: SIP/2.0/UDP 37.148.171.162;branch=z9hG4bKsr-mDYdVEts1x8-yJc91MhEVMtOVMhC6pni1uai1uSAQiytLvyKkMqHB5JsSg98PMYayEyHzESE1gdA6E9C<br>
From: "testaccount" <</span><a style=" font-family:'courier new'; font-size: 9pt;" href="mailto:sip%3A37125511039@213.21.197.50">sip:37125511039@213.21.197.50</a><span style=" font-family:'courier new'; font-size: 9pt;">>;tag=as7abbf1f1<br>
To: <</span><a style=" font-family:'courier new'; font-size: 9pt;" href="http://sip:100@37.148.171.162:5060">sip:100@37.148.171.162:5060</a><span style=" font-family:'courier new'; font-size: 9pt;">>;tag=as143173ac<br>
Call-ID: !!:1MeMfMhpfGtgfo6MQu6HfMNH1v-515SC1omDQocT1Efc1MhEVMtOVMhC6pni1uai1uSA<br>
CSeq: 102 INVITE<br>
Server: Asterisk PBX GIT-master-c83a44c<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE<br>
Supported: replaces, timer<br>
Session-Expires: 1800;refresher=uas<br>
Contact: <</span><a style=" font-family:'courier new'; font-size: 9pt;" href="http://sip:100@37.148.171.164:5060">sip:100@37.148.171.164:5060</a><span style=" font-family:'courier new'; font-size: 9pt;">><br>
Content-Type: application/sdp<br>
Require: timer<br>
Content-Length: 267<br>
<br>
v=0<br>
o=root 1502512559 1502512559 IN IP4 37.148.171.164<br>
s=Asterisk PBX GIT-master-c83a44c<br>
c=IN IP4 37.148.171.164<br>
t=0 0<br>
m=audio 19658 RTP/AVP 8 101<br>
a=rtpmap:8 PCMA/8000<br>
a=rtpmap:101 telephone-event/8000<br>
a=fmtp:101 0-16<br>
a=ptime:20<br>
a=maxptime:150<br>
a=sendrecv<br>
<br>
And no ACK answer from Kamailio. After 32 sec call drops.<br>
<br>
I have call flow: Provider 213.21.197.50 -> Kamailio 37.148.171.162 -> Asterisk 37.148.171.163.<br>
<br>
kamailio -v<br>
version: kamailio 5.2.0 (x86_64/linux) 535e13<br>
flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES<br>
ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144 MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB<br>
poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.<br>
id: 535e13<br>
compiled on 01:27:51 Dec 16 2018 with gcc 4.8.5<br>
<br>
On Asterisk side I have config<br>
<br>
[general]<br>
language=en<br>
maxexpirey=3600<br>
defaultexpirey=3600<br>
bindport=5060<br>
subscribecontext=lab<br>
allowsubscribe=yes<br>
limitonpeers=yes<br>
notifyringing=yes<br>
notifyhold=yes<br>
disallow=all<br>
alwaysauthreject=yes<br>
allowguest=no<br>
allow=alaw<br>
bindaddr=37.148.171.163<br>
t38pt_udptl=yes,redundancy<br>
faxdetect=yes<br>
directmedia=no<br>
rtptimeout=60<br>
rtpholdtimeout=300<br>
<br>
[kamailio]<br>
type=friend<br>
host=37.148.171.162<br>
port=5060<br>
nat=no<br>
qualify=no<br>
canreinvite=yes<br>
insecure=invite<br>
context=from-pstn<br>
;directmedia=yes<br>
<br>
Many thanks !<br>
<br>
BR,<br>
Alex<br>
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<span style=" font-size: 8pt; color: #c0c0c0;"><i>-- <br>
Best regards,<br>
Techinnovation </i></span></span><a style=" font-family:'courier new'; font-size: 9pt;" href="mailto:techinnovation@inbox.lv">mailto:techinnovation@inbox.lv</a><br>
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<span style=" font-family:'arial'; color: #c0c0c0;"><i>-- <br>
Best regards,<br>
Techinnovation </i></span><a style=" font-family:'arial';" href="mailto:techinnovation@inbox.lv">mailto:techinnovation@inbox.lv</a></body></html>