Hello,
Could you please help me correctly to configure Kamailio with load balancing between two asterisk servers and topology hidding with topoh module. In attach sip trace from kamailio and my config.
Currently my problem is that after call UP on Asterisk side, Asterisk send 10 times Retransmitting to <-------------> Retransmitting #5 (no NAT) to 37.148.171.162:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 37.148.171.162;branch=z9hG4bKf534.157d7d1321019895cc6ba367dd3ae94c.0;received=37.148.171.162 Via: SIP/2.0/UDP 37.148.171.162;branch=z9hG4bKsr-mDYdVEts1x8-yJc91MhEVMtOVMhC6pni1uai1uSAQiytLvyKkMqHB5JsSg98PMYayEyHzESE1gdA6E9C From: "testaccount" sip:37125511039@213.21.197.50;tag=as7abbf1f1 To: sip:100@37.148.171.162:5060;tag=as143173ac Call-ID: !!:1MeMfMhpfGtgfo6MQu6HfMNH1v-515SC1omDQocT1Efc1MhEVMtOVMhC6pni1uai1uSA CSeq: 102 INVITE Server: Asterisk PBX GIT-master-c83a44c Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: sip:100@37.148.171.164:5060 Content-Type: application/sdp Require: timer Content-Length: 267
v=0 o=root 1502512559 1502512559 IN IP4 37.148.171.164 s=Asterisk PBX GIT-master-c83a44c c=IN IP4 37.148.171.164 t=0 0 m=audio 19658 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv
And no ACK answer from Kamailio. After 32 sec call drops.
I have call flow: Provider 213.21.197.50 -> Kamailio 37.148.171.162 -> Asterisk 37.148.171.163.
kamailio -v version: kamailio 5.2.0 (x86_64/linux) 535e13 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 ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144 MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 535e13 compiled on 01:27:51 Dec 16 2018 with gcc 4.8.5
On Asterisk side I have config
[general] language=en maxexpirey=3600 defaultexpirey=3600 bindport=5060 subscribecontext=lab allowsubscribe=yes limitonpeers=yes notifyringing=yes notifyhold=yes disallow=all alwaysauthreject=yes allowguest=no allow=alaw bindaddr=37.148.171.163 t38pt_udptl=yes,redundancy faxdetect=yes directmedia=no rtptimeout=60 rtpholdtimeout=300
[kamailio] type=friend host=37.148.171.162 port=5060 nat=no qualify=no canreinvite=yes insecure=invite context=from-pstn ;directmedia=yes
Many thanks !
BR, Alex
i have similar problem, but only with streams due i used NAT with rtpproxy.. some help are apreciate in details!
Lenz McKAY Gerardo (PICCORO) http://qgqlochekone.blogspot.com
El mar., 19 de feb. de 2019 a la(s) 03:22, Techinnovation ( techinnovation@inbox.lv) escribió:
Hello,
Could you please help me correctly to configure Kamailio with load balancing between two asterisk servers and topology hidding with topoh module. In attach sip trace from kamailio and my config.
Currently my problem is that after call UP on Asterisk side, Asterisk send 10 times Retransmitting to <-------------> Retransmitting #5 (no NAT) to 37.148.171.162:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 37.148.171.162;branch=z9hG4bKf534.157d7d1321019895cc6ba367dd3ae94c.0;received=37.148.171.162 Via: SIP/2.0/UDP 37.148.171.162;branch=z9hG4bKsr-mDYdVEts1x8-yJc91MhEVMtOVMhC6pni1uai1uSAQiytLvyKkMqHB5JsSg98PMYayEyHzESE1gdA6E9C From: "testaccount" sip:37125511039@213.21.197.50;tag=as7abbf1f1 To: sip:100@37.148.171.162:5060;tag=as143173ac Call-ID: !!:1MeMfMhpfGtgfo6MQu6HfMNH1v-515SC1omDQocT1Efc1MhEVMtOVMhC6pni1uai1uSA CSeq: 102 INVITE Server: Asterisk PBX GIT-master-c83a44c Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: sip:100@37.148.171.164:5060 Content-Type: application/sdp Require: timer Content-Length: 267
v=0 o=root 1502512559 1502512559 IN IP4 37.148.171.164 s=Asterisk PBX GIT-master-c83a44c c=IN IP4 37.148.171.164 t=0 0 m=audio 19658 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv
And no ACK answer from Kamailio. After 32 sec call drops.
I have call flow: Provider 213.21.197.50 -> Kamailio 37.148.171.162 -> Asterisk 37.148.171.163.
kamailio -v version: kamailio 5.2.0 (x86_64/linux) 535e13 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 ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144 MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 535e13 compiled on 01:27:51 Dec 16 2018 with gcc 4.8.5
On Asterisk side I have config
[general] language=en maxexpirey=3600 defaultexpirey=3600 bindport=5060 subscribecontext=lab allowsubscribe=yes limitonpeers=yes notifyringing=yes notifyhold=yes disallow=all alwaysauthreject=yes allowguest=no allow=alaw bindaddr=37.148.171.163 t38pt_udptl=yes,redundancy faxdetect=yes directmedia=no rtptimeout=60 rtpholdtimeout=300
[kamailio] type=friend host=37.148.171.162 port=5060 nat=no qualify=no canreinvite=yes insecure=invite context=from-pstn ;directmedia=yes
Many thanks !
BR, Alex
*-- Best regards, Techinnovation * mailto:techinnovation@inbox.lv techinnovation@inbox.lv _______________________________________________ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Hello,
I found that same problem also is with BYE
call flow: Provider 213.21.197.50 -> Kamailio 37.148.171.162 -> Asterisk 37.148.171.163
This BYE send iniciator
U 2019/02/20 18:44:02.341540 213.21.197.50:5060 -> 37.148.171.162:5060 #12299 BYE sip:37.148.171.162;line=sr-BgYAQMhA1hcE6pnO6u9s1oBOVMhg1Eai1uSA SIP/2.0. Via: SIP/2.0/UDP 213.21.197.50:5060;branch=z9hG4bK6b28c7be. Max-Forwards: 70. From: "Yealink VC Mobile" sip:37125511039@213.21.197.50;tag=as4be8f9bb. To: sip:100@37.148.171.162:5060;tag=as5e2133c0. Call-ID: 4eb0425445ea0d41269ef2b70e82a773@213.21.197.50:5060. CSeq: 103 BYE. User-Agent: Asterisk PBX 14.6.2. X-Asterisk-HangupCause: Normal Clearing. X-Asterisk-HangupCauseCode: 16. Content-Length: 0.
And this we see on kamailio side
U 2019/02/20 20:45:24.545345 213.21.197.50:5060 -> 37.148.171.162:5060 #136 BYE sip:37.148.171.162;line=sr-BgYAQMhA1hcE6pnO6u9s1oBOVMhg1Eai1uSA SIP/2.0. Via: SIP/2.0/UDP 213.21.197.50:5060;branch=z9hG4bK3c5c07d9. Max-Forwards: 70. From: "Yealink VC Mobile" sip:37125511039@213.21.197.50;tag=as627af367. To: sip:100@37.148.171.162:5060;tag=as2795e468. Call-ID: 5e90daae78528a5b1d7447fc6c9eae29@213.21.197.50:5060. CSeq: 103 BYE. User-Agent: Asterisk PBX 14.6.2. X-Asterisk-HangupCause: Normal Clearing. X-Asterisk-HangupCauseCode: 16. Content-Length: 0. . erver: Asterisk
U 2019/02/20 20:45:24.546130 37.148.171.162:5060 -> 213.21.197.50:5060 #137 SIP/2.0 404 Not here. Via: SIP/2.0/UDP 213.21.197.50:5060;branch=z9hG4bK3c5c07d9. From: "Yealink VC Mobile" sip:37125511039@213.21.197.50;tag=as627af367. To: sip:100@37.148.171.162:5060;tag=as2795e468. Call-ID: 5e90daae78528a5b1d7447fc6c9eae29@213.21.197.50:5060. CSeq: 103 BYE. Server: kamailio (5.2.1 (x86_64/linux)). Content-Length: 0.
BR, Alex
Tuesday, February 19, 2019, 5:09:48 PM, you wrote:
Could you please help me correctly to configure Kamailio with load balancing between two asterisk servers and topology hidding with topoh module. In attach sip trace from kamailio and my config.
Currently my problem is that after call UP on Asterisk side, Asterisk send 10 times Retransmitting to <-------------> Retransmitting #5 (no NAT) to 37.148.171.162:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 37.148.171.162;branch=z9hG4bKf534.157d7d1321019895cc6ba367dd3ae94c.0;received=37.148.171.162 Via: SIP/2.0/UDP 37.148.171.162;branch=z9hG4bKsr-mDYdVEts1x8-yJc91MhEVMtOVMhC6pni1uai1uSAQiytLvyKkMqHB5JsSg98PMYayEyHzESE1gdA6E9C From: "testaccount" sip:37125511039@213.21.197.50;tag=as7abbf1f1 To: sip:100@37.148.171.162:5060;tag=as143173ac Call-ID: !!:1MeMfMhpfGtgfo6MQu6HfMNH1v-515SC1omDQocT1Efc1MhEVMtOVMhC6pni1uai1uSA CSeq: 102 INVITE Server: Asterisk PBX GIT-master-c83a44c Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: sip:100@37.148.171.164:5060 Content-Type: application/sdp Require: timer Content-Length: 267
v=0 o=root 1502512559 1502512559 IN IP4 37.148.171.164 s=Asterisk PBX GIT-master-c83a44c c=IN IP4 37.148.171.164 t=0 0 m=audio 19658 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv
And no ACK answer from Kamailio. After 32 sec call drops.
I have call flow: Provider 213.21.197.50 -> Kamailio 37.148.171.162 -> Asterisk 37.148.171.163.
kamailio -v version: kamailio 5.2.0 (x86_64/linux) 535e13 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 ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144 MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 535e13 compiled on 01:27:51 Dec 16 2018 with gcc 4.8.5
On Asterisk side I have config
[general] language=en maxexpirey=3600 defaultexpirey=3600 bindport=5060 subscribecontext=lab allowsubscribe=yes limitonpeers=yes notifyringing=yes notifyhold=yes disallow=all alwaysauthreject=yes allowguest=no allow=alaw bindaddr=37.148.171.163 t38pt_udptl=yes,redundancy faxdetect=yes directmedia=no rtptimeout=60 rtpholdtimeout=300
[kamailio] type=friend host=37.148.171.162 port=5060 nat=no qualify=no canreinvite=yes insecure=invite context=from-pstn ;directmedia=yes
Many thanks !
BR, Alex
-
PICCORO McKAY Lenz -
Techinnovation