Dear sir,
We install 3 openser 1.1.0 in public internet, and we find a BYE relay issue to caller SIP UA.
UA(a)-------Proxy1 --------Proxy2 ----Proxy 3 ---------UA(b)
We make call from UA(a) to UA (b), if UA(b) send Bye first, UA(a) can not get Bye.
The packet is from Proxy2 to Proxy1 as following
BYE sip:1.2.3.4:5060;nat=yes;ftag=ooqjvbi7ql;lr=on SIP/2.0 Record-Route: sip:5.6.7.8;lr=on;ftag=as1b41fbeb Record-Route: sip:5.6.7.8;lr=on;ftag=as1b41fbeb Via: SIP/2.0/UDP 5.6.7.8;branch=z9hG4bKc74c.bd89ce13.0 Via: SIP/2.0/UDP 5.6.7.8;branch=z9hG4bKc74c.ad89ce13.0 Via: SIP/2.0/UDP 100.200.100.200;branch=z9hG4bKc74c.321ef5e3.0 Via: SIP/2.0/UDP 100.200.100.199:5060;branch=z9hG4bK75e8d6f3;rport=5060 From: sip:0908900000@sip.edu.tw;user=phone;tag=as1b41fbeb To: "0909802020 kl" sip:0909802020@sip.edu.tw;tag=ooqjvbi7ql Call-ID: 3c275da915f9-iylvmp2wjuh0@snom360-00041323058B CSeq: 102 BYE User-Agent: Gentrice_IPPBX Max-Forwards: 67 Content-Length: 0
Then bye is looping in proxy1.
BYE sip:1.2.3.4:5060;nat=yes;ftag=ooqjvbi7ql;lr=on SIP/2.0 Record-Route: sip:5.6.7.8;lr=on;ftag=as1b41fbeb Record-Route: sip:5.6.7.8;lr=on;ftag=as1b41fbeb Via: SIP/2.0/UDP 1.2.3.4;branch=z9hG4bKc74c.1714a4e7.0 Via: SIP/2.0/UDP 5.6.7.8;branch=z9hG4bKc74c.bd89ce13.0 Via: SIP/2.0/UDP 5.6.7.8;branch=z9hG4bKc74c.ad89ce13.0 Via: SIP/2.0/UDP 100.200.100.200;branch=z9hG4bKc74c.321ef5e3.0 Via: SIP/2.0/UDP 100.200.100.199:5060;branch=z9hG4bK75e8d6f3;rport=5060 From: sip:0908900000@sip.edu.tw;user=phone;tag=as1b41fbeb To: "0909802020 kl" sip:0909802020@sip.edu.tw;tag=ooqjvbi7ql Call-ID: 3c275da915f9-iylvmp2wjuh0@snom360-00041323058B CSeq: 102 BYE User-Agent: Gentrice_IPPBX Max-Forwards: 66 Content-Length: 0
We used to use default script for proxy1, but the same... And we do record_route, loose_route relay as well.
Anyone know what happen?
best regards Thrli
Hi!
AS this is probably a loose-route problem you have to provide a log of the initial INVITE too.
regards klaus
kaiser wrote:
Dear sir,
We install 3 openser 1.1.0 in public internet, and we find a BYE relay issue to caller SIP UA.
UA(a)-------Proxy1 --------Proxy2 ----Proxy 3 ---------UA(b)
We make call from UA(a) to UA (b), if UA(b) send Bye first, UA(a) can not get Bye.
The packet is from Proxy2 to Proxy1 as following
BYE sip:1.2.3.4:5060;nat=yes;ftag=ooqjvbi7ql;lr=on SIP/2.0 Record-Route: sip:5.6.7.8;lr=on;ftag=as1b41fbeb Record-Route: sip:5.6.7.8;lr=on;ftag=as1b41fbeb Via: SIP/2.0/UDP 5.6.7.8;branch=z9hG4bKc74c.bd89ce13.0 Via: SIP/2.0/UDP 5.6.7.8;branch=z9hG4bKc74c.ad89ce13.0 Via: SIP/2.0/UDP 100.200.100.200;branch=z9hG4bKc74c.321ef5e3.0 Via: SIP/2.0/UDP 100.200.100.199:5060;branch=z9hG4bK75e8d6f3;rport=5060 From: sip:0908900000@sip.edu.tw;user=phone;tag=as1b41fbeb To: "0909802020 kl" sip:0909802020@sip.edu.tw;tag=ooqjvbi7ql Call-ID: 3c275da915f9-iylvmp2wjuh0@snom360-00041323058B CSeq: 102 BYE User-Agent: Gentrice_IPPBX Max-Forwards: 67 Content-Length: 0
Then bye is looping in proxy1.
BYE sip:1.2.3.4:5060;nat=yes;ftag=ooqjvbi7ql;lr=on SIP/2.0 Record-Route: sip:5.6.7.8;lr=on;ftag=as1b41fbeb Record-Route: sip:5.6.7.8;lr=on;ftag=as1b41fbeb Via: SIP/2.0/UDP 1.2.3.4;branch=z9hG4bKc74c.1714a4e7.0 Via: SIP/2.0/UDP 5.6.7.8;branch=z9hG4bKc74c.bd89ce13.0 Via: SIP/2.0/UDP 5.6.7.8;branch=z9hG4bKc74c.ad89ce13.0 Via: SIP/2.0/UDP 100.200.100.200;branch=z9hG4bKc74c.321ef5e3.0 Via: SIP/2.0/UDP 100.200.100.199:5060;branch=z9hG4bK75e8d6f3;rport=5060 From: sip:0908900000@sip.edu.tw;user=phone;tag=as1b41fbeb To: "0909802020 kl" sip:0909802020@sip.edu.tw;tag=ooqjvbi7ql Call-ID: 3c275da915f9-iylvmp2wjuh0@snom360-00041323058B CSeq: 102 BYE User-Agent: Gentrice_IPPBX Max-Forwards: 66 Content-Length: 0
We used to use default script for proxy1, but the same... And we do record_route, loose_route relay as well.
Anyone know what happen?
best regards Thrli
Users mailing list Users@openser.org http://openser.org/cgi-bin/mailman/listinfo/users
Hi,
Thanks for your help, this is the invite from initial proxy:
sip:0908900000@sip.ipox.org.tw:5060;user=phone SIP/2.0 Record-Route: sip:1.2.3.4:5060;nat=yes;ftag=ooqjvbi7ql;lr=on Via: SIP/2.0/UDP 1.2.3.4;branch=z9hG4bK2e5a.835198e5.0 Via: SIP/2.0/UDP 192.168.0.12:2051;received=59.120.208.208;branch=z9hG4bK- kblxgjx7vsj6;rport=2051 From: "0909802020 kl" sip:0709802020@sip.edu.tw;tag=ooqjvbi7ql To: sip:0908900000@sip.edu.tw;user=phone Call-ID: 3c275da915f9-iylvmp2wjuh0@snom360-00041323058B CSeq: 1 INVITE Max-Forwards: 69 Contact: sip:0909802020@59.120.208.208:2051;line=5bj76qg6;flow-id=1
do you mean we have to add username in record_route?
best regards kaiser
在 2007/5/22 上午 1:21 時，Klaus Darilion 寫到：
Hi!
AS this is probably a loose-route problem you have to provide a log of the initial INVITE too.
regards klaus
kaiser wrote:
Dear sir, We install 3 openser 1.1.0 in public internet, and we find a BYE relay issue to caller SIP UA. UA(a)-------Proxy1 --------Proxy2 ----Proxy 3 ---------UA(b) We make call from UA(a) to UA (b), if UA(b) send Bye first, UA(a) can not get Bye. The packet is from Proxy2 to Proxy1 as following BYE sip:1.2.3.4:5060;nat=yes;ftag=ooqjvbi7ql;lr=on SIP/2.0 Record-Route: sip:5.6.7.8;lr=on;ftag=as1b41fbeb Record-Route: sip:5.6.7.8;lr=on;ftag=as1b41fbeb Via: SIP/2.0/UDP 5.6.7.8;branch=z9hG4bKc74c.bd89ce13.0 Via: SIP/2.0/UDP 5.6.7.8;branch=z9hG4bKc74c.ad89ce13.0 Via: SIP/2.0/UDP 100.200.100.200;branch=z9hG4bKc74c.321ef5e3.0 Via: SIP/2.0/UDP 100.200.100.199:5060;branch=z9hG4bK75e8d6f3;rport=5060 From: sip:0908900000@sip.edu.tw;user=phone;tag=as1b41fbeb To: "0909802020 kl" sip:0909802020@sip.edu.tw;tag=ooqjvbi7ql Call-ID: 3c275da915f9-iylvmp2wjuh0@snom360-00041323058B CSeq: 102 BYE User-Agent: Gentrice_IPPBX Max-Forwards: 67 Content-Length: 0 Then bye is looping in proxy1. BYE sip:1.2.3.4:5060;nat=yes;ftag=ooqjvbi7ql;lr=on SIP/2.0 Record-Route: sip:5.6.7.8;lr=on;ftag=as1b41fbeb Record-Route: sip:5.6.7.8;lr=on;ftag=as1b41fbeb Via: SIP/2.0/UDP 1.2.3.4;branch=z9hG4bKc74c.1714a4e7.0 Via: SIP/2.0/UDP 5.6.7.8;branch=z9hG4bKc74c.bd89ce13.0 Via: SIP/2.0/UDP 5.6.7.8;branch=z9hG4bKc74c.ad89ce13.0 Via: SIP/2.0/UDP 100.200.100.200;branch=z9hG4bKc74c.321ef5e3.0 Via: SIP/2.0/UDP 100.200.100.199:5060;branch=z9hG4bK75e8d6f3;rport=5060 From: sip:0908900000@sip.edu.tw;user=phone;tag=as1b41fbeb To: "0909802020 kl" sip:0909802020@sip.edu.tw;tag=ooqjvbi7ql Call-ID: 3c275da915f9-iylvmp2wjuh0@snom360-00041323058B CSeq: 102 BYE User-Agent: Gentrice_IPPBX Max-Forwards: 66 Content-Length: 0 We used to use default script for proxy1, but the same... And we do record_route, loose_route relay as well. Anyone know what happen? best regards Thrli _______________________________________________ Users mailing list Users@openser.org http://openser.org/cgi-bin/mailman/listinfo/users
kaiser wrote:
Hi,
Thanks for your help, this is the invite from initial proxy:
sip:0908900000@sip.ipox.org.tw:5060;user=phone SIP/2.0 Record-Route: sip:1.2.3.4:5060;nat=yes;ftag=ooqjvbi7ql;lr=on Via: SIP/2.0/UDP 1.2.3.4;branch=z9hG4bK2e5a.835198e5.0 Via: SIP/2.0/UDP 192.168.0.12:2051;received=59.120.208.208;branch=z9hG4bK-kblxgjx7vsj6;rport=2051
From: "0909802020 kl" sip:0709802020@sip.edu.tw;tag=ooqjvbi7ql To: sip:0908900000@sip.edu.tw;user=phone Call-ID: 3c275da915f9-iylvmp2wjuh0@snom360-00041323058B CSeq: 1 INVITE Max-Forwards: 69 Contact: sip:0909802020@59.120.208.208:2051;line=5bj76qg6;flow-id=1
do you mean we have to add username in record_route?
no. But this is still to less log. Send the complete dialog: INVITE+100+180+200ok+ACK+BYE
regards klaus
Further, describe your setup: IP addresses of the proxies IP addresses of the clients
kaiser wrote:
Hi,
Thanks for your help, this is the invite from initial proxy:
sip:0908900000@sip.ipox.org.tw:5060;user=phone SIP/2.0 Record-Route: sip:1.2.3.4:5060;nat=yes;ftag=ooqjvbi7ql;lr=on Via: SIP/2.0/UDP 1.2.3.4;branch=z9hG4bK2e5a.835198e5.0 Via: SIP/2.0/UDP 192.168.0.12:2051;received=59.120.208.208;branch=z9hG4bK-kblxgjx7vsj6;rport=2051
From: "0909802020 kl" sip:0709802020@sip.edu.tw;tag=ooqjvbi7ql To: sip:0908900000@sip.edu.tw;user=phone Call-ID: 3c275da915f9-iylvmp2wjuh0@snom360-00041323058B CSeq: 1 INVITE Max-Forwards: 69 Contact: sip:0909802020@59.120.208.208:2051;line=5bj76qg6;flow-id=1
do you mean we have to add username in record_route?
best regards kaiser
在 2007/5/22 上午 1:21 時，Klaus Darilion 寫到：
Hi!
AS this is probably a loose-route problem you have to provide a log of the initial INVITE too.
regards klaus
kaiser wrote:
Dear sir, We install 3 openser 1.1.0 in public internet, and we find a BYE relay issue to caller SIP UA. UA(a)-------Proxy1 --------Proxy2 ----Proxy 3 ---------UA(b) We make call from UA(a) to UA (b), if UA(b) send Bye first, UA(a) can not get Bye. The packet is from Proxy2 to Proxy1 as following BYE sip:1.2.3.4:5060;nat=yes;ftag=ooqjvbi7ql;lr=on SIP/2.0 Record-Route: sip:5.6.7.8;lr=on;ftag=as1b41fbeb Record-Route: sip:5.6.7.8;lr=on;ftag=as1b41fbeb Via: SIP/2.0/UDP 5.6.7.8;branch=z9hG4bKc74c.bd89ce13.0 Via: SIP/2.0/UDP 5.6.7.8;branch=z9hG4bKc74c.ad89ce13.0 Via: SIP/2.0/UDP 100.200.100.200;branch=z9hG4bKc74c.321ef5e3.0 Via: SIP/2.0/UDP 100.200.100.199:5060;branch=z9hG4bK75e8d6f3;rport=5060 From: sip:0908900000@sip.edu.tw;user=phone;tag=as1b41fbeb To: "0909802020 kl" sip:0909802020@sip.edu.tw;tag=ooqjvbi7ql Call-ID: 3c275da915f9-iylvmp2wjuh0@snom360-00041323058B CSeq: 102 BYE User-Agent: Gentrice_IPPBX Max-Forwards: 67 Content-Length: 0 Then bye is looping in proxy1. BYE sip:1.2.3.4:5060;nat=yes;ftag=ooqjvbi7ql;lr=on SIP/2.0 Record-Route: sip:5.6.7.8;lr=on;ftag=as1b41fbeb Record-Route: sip:5.6.7.8;lr=on;ftag=as1b41fbeb Via: SIP/2.0/UDP 1.2.3.4;branch=z9hG4bKc74c.1714a4e7.0 Via: SIP/2.0/UDP 5.6.7.8;branch=z9hG4bKc74c.bd89ce13.0 Via: SIP/2.0/UDP 5.6.7.8;branch=z9hG4bKc74c.ad89ce13.0 Via: SIP/2.0/UDP 100.200.100.200;branch=z9hG4bKc74c.321ef5e3.0 Via: SIP/2.0/UDP 100.200.100.199:5060;branch=z9hG4bK75e8d6f3;rport=5060 From: sip:0908900000@sip.edu.tw;user=phone;tag=as1b41fbeb To: "0909802020 kl" sip:0909802020@sip.edu.tw;tag=ooqjvbi7ql Call-ID: 3c275da915f9-iylvmp2wjuh0@snom360-00041323058B CSeq: 102 BYE User-Agent: Gentrice_IPPBX Max-Forwards: 66 Content-Length: 0 We used to use default script for proxy1, but the same... And we do record_route, loose_route relay as well. Anyone know what happen? best regards Thrli _______________________________________________ Users mailing list Users@openser.org http://openser.org/cgi-bin/mailman/listinfo/users
Dear Sir,
This is the Packet in initial Proxy:
UA ----> Proxy1 ---> proxy2 ...... -----> called UA
Voice is ok, Invite, ACK, are fine, only BYE has trouble when called hang up first.
The call signal packets close to Proxy 1
thanks
-----Invite from UA
sip:0908900000@sip.edu;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.0.12:2051;branch=z9hG4bK-kblxgjx7vsj6;rport From: "0709802020 kl" sip:0709802020@sip.edu;tag=ooqjvbi7ql To: sip:0908900000@sip.edu;user=phone Call-ID: 3c275da915f9-iylvmp2wjuh0@snom360-00041323058B CSeq: 1 INVITE Max-Forwards: 70 Contact: sip:0709802020@192.168.0.12:2051;line=5bj76qg6;flow-id=1 P-Key-Flags: resolution="31x13", keys="4" User-Agent: snom360/6.5.1 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces, callerid Session-Expires: 3600;refresher=uas Min-SE: 90 Content-Type: application/sdp Content-Length: 424
v=0 o=root 1112263304 1112263304 IN IP4 192.168.0.12 s=call c=IN IP4 192.168.0.12 t=0 0 m=audio 50530 RTP/AVP 18 4 0 8 3 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline: 3GR6bicZ4axFL3M0iBo9rx84mGt9TNWJIfgyMprL a=rtpmap:18 g729/8000 a=rtpmap:4 g723/8000 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:3 gsm/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=encryption:optional a=sendrecv
SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 192.168.0.12:2051;branch=z9hG4bK- kblxgjx7vsj6;rport=2051;received=59.120.208.208 From: "0709802020 kl" sip:0709802020@sip.edu;tag=ooqjvbi7ql To: sip:0908900000@sip.edu;user=phone Call-ID: 3c275da915f9-iylvmp2wjuh0@snom360-00041323058B CSeq: 1 INVITE Content-Length: 0
------ Invite from 1st Proxy to 2nd Proxy
INVITE sip:0908900000@sip.org:5060;user=phone SIP/2.0 Record-Route: sip:1.2.3.4:5060;nat=yes;ftag=ooqjvbi7ql;lr=on Via: SIP/2.0/UDP 1.2.3.4;branch=z9hG4bK2e5a.835198e5.0 Via: SIP/2.0/UDP 192.168.0.12:2051;received=59.120.208.208;branch=z9hG4bK- kblxgjx7vsj6;rport=2051 From: "0709802020 kl" sip:0709802020@sip.edu;tag=ooqjvbi7ql To: sip:0908900000@sip.edu;user=phone Call-ID: 3c275da915f9-iylvmp2wjuh0@snom360-00041323058B CSeq: 1 INVITE Max-Forwards: 69 Contact: sip:0709802020@59.120.208.208:2051;line=5bj76qg6;flow-id=1 P-Key-Flags: resolution="31x13", keys="4" User-Agent: snom360/6.5.1 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces, callerid Session-Expires: 3600;refresher=uas Min-SE: 90 Content-Type: application/sdp Content-Length: 426
v=0 o=root 1112263304 1112263304 IN IP4 192.168.0.12 s=call c=IN IP4 1.2.3.4 t=0 0 m=audio 35054 RTP/AVP 18 4 0 8 3 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline: 3GR6bicZ4axFL3M0iBo9rx84mGt9TNWJIfgyMprL a=rtpmap:18 g729/8000 a=rtpmap:4 g723/8000 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:3 gsm/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=encryption:optional a=sendrecv
SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 1.2.3.4;branch=z9hG4bK2e5a.835198e5.0 Via: SIP/2.0/UDP 192.168.0.12:2051;received=59.120.208.208;branch=z9hG4bK- kblxgjx7vsj6;rport=2051 From: "0709802020 kl" sip:0709802020@sip.edu;tag=ooqjvbi7ql To: sip:0908900000@sip.edu;user=phone Call-ID: 3c275da915f9-iylvmp2wjuh0@snom360-00041323058B CSeq: 1 INVITE Content-Length: 0
------- Proxy receive Callee Answer packet
SIP/2.0 200 OK Via: SIP/2.0/UDP 1.2.3.4;branch=z9hG4bK2e5a.835198e5.0 Via: SIP/2.0/UDP 192.168.0.12:2051;received=59.120.208.208;branch=z9hG4bK- kblxgjx7vsj6;rport=2051 Record-Route: sip:16.3.0.100;lr=on;ftag=ooqjvbi7ql Record-Route: sip:5.6.7.8;lr=on;ftag=ooqjvbi7ql Record-Route: sip:1.2.3.4:5060;nat=yes;ftag=ooqjvbi7ql;lr=on From: "0709802020 kl" sip:0709802020@sip.edu;tag=ooqjvbi7ql To: sip:0908900000@sip.edu;user=phone;tag=as1b41fbeb Call-ID: 3c275da915f9-iylvmp2wjuh0@snom360-00041323058B CSeq: 1 INVITE User-Agent: Gentrice_IPPBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: sip:7777@163.30.0.199 Content-Type: application/sdp Content-Length: 287
v=0 o=root 23878 23878 IN IP4 163.30.0.199 s=session c=IN IP4 16.3.0.100 t=0 0 m=audio 57124 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv
----------OK relay to UA
SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.12:2051;received=59.120.208.208;branch=z9hG4bK- kblxgjx7vsj6;rport=2051 Record-Route: sip:16.3.0.100;lr=on;ftag=ooqjvbi7ql Record-Route: sip:5.6.7.8;lr=on;ftag=ooqjvbi7ql Record-Route: sip:1.2.3.4:5060;nat=yes;ftag=ooqjvbi7ql;lr=on From: "0709802020 kl" sip:0709802020@sip.edu;tag=ooqjvbi7ql To: sip:0908900000@sip.edu;user=phone;tag=as1b41fbeb Call-ID: 3c275da915f9-iylvmp2wjuh0@snom360-00041323058B CSeq: 1 INVITE User-Agent: Gentrice_IPPBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: sip:7777@163.30.0.199 Content-Type: application/sdp Content-Length: 289
v=0 o=root 23878 23878 IN IP4 163.30.0.199 s=session c=IN IP4 1.2.3.4 t=0 0 m=audio 35054 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv
------ACK received from UA
ACK sip:7777@163.30.0.199 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.12:2051;branch=z9hG4bK-us1wrcp6otau;rport Route: sip:1.2.3.4:5060;nat=yes;ftag=ooqjvbi7ql;lr=on Route: sip:5.6.7.8;lr=on;ftag=ooqjvbi7ql Route: sip:16.3.0.100;lr=on;ftag=ooqjvbi7ql From: "0709802020 kl" sip:0709802020@sip.edu;tag=ooqjvbi7ql To: sip:0908900000@sip.edu;user=phone;tag=as1b41fbeb Call-ID: 3c275da915f9-iylvmp2wjuh0@snom360-00041323058B CSeq: 1 ACK Max-Forwards: 70 Contact: sip:0709802020@192.168.0.12:2051;line=5bj76qg6;flow-id=1 Content-Length: 0
------ACK relay to next proxy
ACK sip:7777@163.30.0.199 SIP/2.0 Via: SIP/2.0/UDP 1.2.3.4;branch=z9hG4bK2e5a.835198e5.2 Via: SIP/2.0/UDP 192.168.0.12:2051;received=59.120.208.208;branch=z9hG4bK- us1wrcp6otau;rport=2051 Route: sip:5.6.7.8;lr=on;ftag=ooqjvbi7ql Route: sip:16.3.0.100;lr=on;ftag=ooqjvbi7ql From: "0709802020 kl" sip:0709802020@sip.edu;tag=ooqjvbi7ql To: sip:0908900000@sip.edu;user=phone;tag=as1b41fbeb Call-ID: 3c275da915f9-iylvmp2wjuh0@snom360-00041323058B CSeq: 1 ACK Max-Forwards: 69 Contact: sip:0709802020@192.168.0.12:2051;line=5bj76qg6;flow-id=1 Content-Length: 0
----Called party hangup
BYE sip:1.2.3.4:5060;nat=yes;ftag=ooqjvbi7ql;lr=on SIP/2.0 Record-Route: sip:5.6.7.8;lr=on;ftag=as1b41fbeb Record-Route: sip:5.6.7.8;lr=on;ftag=as1b41fbeb Via: SIP/2.0/UDP 5.6.7.8;branch=z9hG4bKc74c.bd89ce13.0 Via: SIP/2.0/UDP 5.6.7.8;branch=z9hG4bKc74c.ad89ce13.0 Via: SIP/2.0/UDP 16.3.0.100;branch=z9hG4bKc74c.321ef5e3.0 Via: SIP/2.0/UDP 163.30.0.199:5060;branch=z9hG4bK75e8d6f3;rport=5060 From: sip:0908900000@sip.edu;user=phone;tag=as1b41fbeb To: "0709802020 kl" sip:0709802020@sip.edu;tag=ooqjvbi7ql Call-ID: 3c275da915f9-iylvmp2wjuh0@snom360-00041323058B CSeq: 102 BYE User-Agent: Gentrice_IPPBX Max-Forwards: 67 Content-Length: 0
-----Looped Bye ...
BYE sip:1.2.3.4:5060;nat=yes;ftag=ooqjvbi7ql;lr=on SIP/2.0 Record-Route: sip:5.6.7.8;lr=on;ftag=as1b41fbeb Record-Route: sip:5.6.7.8;lr=on;ftag=as1b41fbeb Via: SIP/2.0/UDP 1.2.3.4;branch=z9hG4bKc74c.3714a4e7.0 Via: SIP/2.0/UDP 1.2.3.4;branch=z9hG4bKc74c.2714a4e7.0 Via: SIP/2.0/UDP 1.2.3.4;branch=z9hG4bKc74c.1714a4e7.0 Via: SIP/2.0/UDP 5.6.7.8;branch=z9hG4bKc74c.bd89ce13.0 Via: SIP/2.0/UDP 5.6.7.8;branch=z9hG4bKc74c.ad89ce13.0 Via: SIP/2.0/UDP 16.3.0.100;branch=z9hG4bKc74c.321ef5e3.0 Via: SIP/2.0/UDP 163.30.0.199:5060;branch=z9hG4bK75e8d6f3;rport=5060 From: sip:0908900000@sip.edu;user=phone;tag=as1b41fbeb To: "0709802020 kl" sip:0709802020@sip.edu;tag=ooqjvbi7ql Call-ID: 3c275da915f9-iylvmp2wjuh0@snom360-00041323058B CSeq: 102 BYE User-Agent: Gentrice_IPPBX Max-Forwards: 64 Content-Length: 0
在 2007/5/22 下午 8:29 時，Klaus Darilion 寫到：
Further, describe your setup: IP addresses of the proxies IP addresses of the clients
kaiser wrote:
Hi,
Thanks for your help, this is the invite from initial proxy:
sip:0908900000@sip.ipox.org.tw:5060;user=phone SIP/2.0 Record-Route: sip:1.2.3.4:5060;nat=yes;ftag=ooqjvbi7ql;lr=on Via: SIP/2.0/UDP 1.2.3.4;branch=z9hG4bK2e5a.835198e5.0 Via: SIP/2.0/UDP 192.168.0.12:2051;received=59.120.208.208;branch=z9hG4bK- kblxgjx7vsj6;rport=2051
From: "0909802020 kl" sip:0709802020@sip.edu.tw;tag=ooqjvbi7ql To: sip:0908900000@sip.edu.tw;user=phone Call-ID: 3c275da915f9-iylvmp2wjuh0@snom360-00041323058B CSeq: 1 INVITE Max-Forwards: 69 Contact: sip:0909802020@59.120.208.208:2051;line=5bj76qg6;flow-id=1
do you mean we have to add username in record_route?
best regards kaiser
在 2007/5/22 上午 1:21 時，Klaus Darilion 寫到：
Hi!
AS this is probably a loose-route problem you have to provide a log of the initial INVITE too.
regards klaus
kaiser wrote:
Dear sir, We install 3 openser 1.1.0 in public internet, and we find a BYE relay issue to caller SIP UA. UA(a)-------Proxy1 --------Proxy2 ----Proxy 3 ---------UA(b) We make call from UA(a) to UA (b), if UA(b) send Bye first, UA (a) can not get Bye. The packet is from Proxy2 to Proxy1 as following BYE sip:1.2.3.4:5060;nat=yes;ftag=ooqjvbi7ql;lr=on SIP/2.0 Record-Route: sip:5.6.7.8;lr=on;ftag=as1b41fbeb Record-Route: sip:5.6.7.8;lr=on;ftag=as1b41fbeb Via: SIP/2.0/UDP 5.6.7.8;branch=z9hG4bKc74c.bd89ce13.0 Via: SIP/2.0/UDP 5.6.7.8;branch=z9hG4bKc74c.ad89ce13.0 Via: SIP/2.0/UDP 100.200.100.200;branch=z9hG4bKc74c.321ef5e3.0 Via: SIP/2.0/UDP 100.200.100.199:5060;branch=z9hG4bK75e8d6f3;rport=5060 From: sip:0908900000@sip.edu.tw;user=phone;tag=as1b41fbeb To: "0909802020 kl" sip:0909802020@sip.edu.tw;tag=ooqjvbi7ql Call-ID: 3c275da915f9-iylvmp2wjuh0@snom360-00041323058B CSeq: 102 BYE User-Agent: Gentrice_IPPBX Max-Forwards: 67 Content-Length: 0 Then bye is looping in proxy1. BYE sip:1.2.3.4:5060;nat=yes;ftag=ooqjvbi7ql;lr=on SIP/2.0 Record-Route: sip:5.6.7.8;lr=on;ftag=as1b41fbeb Record-Route: sip:5.6.7.8;lr=on;ftag=as1b41fbeb Via: SIP/2.0/UDP 1.2.3.4;branch=z9hG4bKc74c.1714a4e7.0 Via: SIP/2.0/UDP 5.6.7.8;branch=z9hG4bKc74c.bd89ce13.0 Via: SIP/2.0/UDP 5.6.7.8;branch=z9hG4bKc74c.ad89ce13.0 Via: SIP/2.0/UDP 100.200.100.200;branch=z9hG4bKc74c.321ef5e3.0 Via: SIP/2.0/UDP 100.200.100.199:5060;branch=z9hG4bK75e8d6f3;rport=5060 From: sip:0908900000@sip.edu.tw;user=phone;tag=as1b41fbeb To: "0909802020 kl" sip:0909802020@sip.edu.tw;tag=ooqjvbi7ql Call-ID: 3c275da915f9-iylvmp2wjuh0@snom360-00041323058B CSeq: 102 BYE User-Agent: Gentrice_IPPBX Max-Forwards: 66 Content-Length: 0 We used to use default script for proxy1, but the same... And we do record_route, loose_route relay as well. Anyone know what happen? best regards Thrli _______________________________________________ Users mailing list Users@openser.org http://openser.org/cgi-bin/mailman/listinfo/users
kaiser wrote:
----Called party hangup
BYE sip:1.2.3.4:5060;nat=yes;ftag=ooqjvbi7ql;lr=on SIP/2.0 Record-Route: sip:5.6.7.8;lr=on;ftag=as1b41fbeb Record-Route: sip:5.6.7.8;lr=on;ftag=as1b41fbeb Via: SIP/2.0/UDP 5.6.7.8;branch=z9hG4bKc74c.bd89ce13.0 Via: SIP/2.0/UDP 5.6.7.8;branch=z9hG4bKc74c.ad89ce13.0 Via: SIP/2.0/UDP 16.3.0.100;branch=z9hG4bKc74c.321ef5e3.0 Via: SIP/2.0/UDP 163.30.0.199:5060;branch=z9hG4bK75e8d6f3;rport=5060 From: sip:0908900000@sip.edu;user=phone;tag=as1b41fbeb To: "0709802020 kl" sip:0709802020@sip.edu;tag=ooqjvbi7ql Call-ID: 3c275da915f9-iylvmp2wjuh0@snom360-00041323058B CSeq: 102 BYE User-Agent: Gentrice_IPPBX Max-Forwards: 67 Content-Length: 0
This looks strange. It looks like a BYE which was already looped one time. I suspect the Gentrice_IPPBX behaves strange.
For confidence, I need to see the BYE from caller to proxy3 and the BYE from proxy3 to proxy2.
regards klaus
Klaus,
I used to suspect the remote IP-PBX, after we test without this IP- PBX, we get the same result. I will collect the packet for remote site BYE later.
But I saw the Proxy2 remove the info from route header and put it in SIP URI, it makes the proxy has no where to go. And the BYE URI is sip:1.2.3.4:5060;nat=yes;ftag=ooqjvbi7ql;lr=on,
it is exactly the same with record_route which is generated by itself for ist Invite relay out to next hop.
regards Chilly
在 2007/5/22 下午 10:39 時，Klaus Darilion 寫到：
wrote:
----Called party hangup
BYE sip:1.2.3.4:5060;nat=yes;ftag=ooqjvbi7ql;lr=on SIP/2.0 Record-Route: sip:5.6.7.8;lr=on;ftag=as1b41fbeb Record-Route: sip:5.6.7.8;lr=on;ftag=as1b41fbeb Via: SIP/2.0/UDP 5.6.7.8;branch=z9hG4bKc74c.bd89ce13.0 Via: SIP/2.0/UDP 5.6.7.8;branch=z9hG4bKc74c.ad89ce13.0 Via: SIP/2.0/UDP 16.3.0.100;branch=z9hG4bKc74c.321ef5e3.0 Via: SIP/2.0/UDP 163.30.0.199:5060;branch=z9hG4bK75e8d6f3;rport=5060 From: sip:0908900000@sip.edu;user=phone;tag=as1b41fbeb To: "0709802020 kl" sip:0709802020@sip.edu;tag=ooqjvbi7ql Call-ID: 3c275da915f9-iylvmp2wjuh0@snom360-00041323058B CSeq: 102 BYE User-Agent: Gentrice_IPPBX Max-Forwards: 67 Content-Length: 0
This looks strange. It looks like a BYE which was already looped one time. I suspect the Gentrice_IPPBX behaves strange.
For confidence, I need to see the BYE from caller to proxy3 and the BYE from proxy3 to proxy2.
regards klaus
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Gentrice's kaiser -
kaiser
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Klaus Darilion