Hi,
sip.conf
[5001]
type=friend
username=5001
fromuser=5001
qualify=yes
host=xx.xxx.xxx.xx
[5002]
type=friend
username=5002
fromuser=5002
qualify=yes
host=xx.xxx.xxx.xx
[5003]
type=friend
username=5003
fromuser=5003
qualify=yes
host=xx.xxx.xxx.xx
[5004]
type=friend
username=5004
fromuser=5004
qualify=yes
host=xx.xxx.xxx.xx
[root@localhost asterisk]# asterisk -rx “sip show peers”| grep OK
5001/5001 xx.xxx.xxx.xx N 5060 OK (59 ms)
5002/5002 xx.xxx.xxx.xx N 5060 OK (59 ms)
5003/5003 xx.xxx.xxx.xx N 5060 OK (60 ms)
5004/5004 xx.xxx.xxx.xx N 5060 OK (77 ms)
then sip set debug on
this is the log file output. what do i need to change the fix this (URI not recognized) in Asterisk.
OPTIONS sip:64.245.205.20 SIP/2.0
ÿVia: SIP/2.0/UDP 172.16.215.20:5060;branch=z9hG4bK3bf87cd7;rport
ÿMax-Forwards: 70
ÿFrom: “asterisk” sip:5001@172.16.215.20;tag=as072b1694
ÿTo: sip:64.245.205.20
ÿContact: sip:5001@172.16.215.20:5060
ÿCall-ID: 45cb3499140d39ab61a93670773abd46@172.16.215.20:5060
ÿCSeq: 102 OPTIONS
ÿUser-Agent: Asterisk PBX 11.7.0
ÿDate: Wed, 13 Aug 2014 07:54:17 GMT
ÿAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
ÿSupported: replaces, timer
ÿContent-Length: 0
ÿ
ÿ
ÿ—
[2014-08-13 02:54:17] VERBOSE[2942] chan_sip.c: Reliably Transmitting (NAT) to 64.245.205.20:5060:
ÿOPTIONS sip:64.245.205.20 SIP/2.0
ÿVia: SIP/2.0/UDP 172.16.215.20:5060;branch=z9hG4bK0f5c05fa;rport
ÿMax-Forwards: 70
ÿFrom: “asterisk” sip:5002@172.16.215.20;tag=as0fa3c3f7
ÿTo: sip:64.245.205.20
ÿContact: sip:5002@172.16.215.20:5060
ÿCall-ID: 0732c6e52f4cdb772d871c7525a7394e@172.16.215.20:5060
ÿCSeq: 102 OPTIONS
ÿUser-Agent: Asterisk PBX 11.7.0
ÿDate: Wed, 13 Aug 2014 07:54:17 GMT
ÿAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
ÿSupported: replaces, timer
ÿContent-Length: 0
ÿ
ÿ
ÿ—
[2014-08-13 02:54:17] VERBOSE[2942] chan_sip.c: Reliably Transmitting (NAT) to 64.245.205.20:5060:
ÿOPTIONS sip:64.245.205.20 SIP/2.0
ÿVia: SIP/2.0/UDP 172.16.215.20:5060;branch=z9hG4bK7e1fd1c7;rport
ÿMax-Forwards: 70
ÿFrom: “asterisk” sip:5003@172.16.215.20;tag=as1fd14311
ÿTo: sip:64.245.205.20
ÿContact: sip:5003@172.16.215.20:5060
ÿCall-ID: 23f1151c6290e95644811aff63d632b1@172.16.215.20:5060
ÿCSeq: 102 OPTIONS
ÿUser-Agent: Asterisk PBX 11.7.0
ÿDate: Wed, 13 Aug 2014 07:54:17 GMT
ÿAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
ÿSupported: replaces, timer
ÿContent-Length: 0
ÿ
ÿ
ÿ—
[2014-08-13 02:54:17] VERBOSE[2942] chan_sip.c:
ÿ<— SIP read from UDP:64.245.205.20:5060 —>
ÿSIP/2.0 403 From: URI not recognized
ÿVia: SIP/2.0/UDP 172.16.215.20:5060;received=198.211.207.90;branch=z9hG4bK3bf87cd7;rport=43489
ÿFrom: “asterisk” sip:5001@172.16.215.20;tag=as072b1694
ÿTo: sip:64.245.205.20;tag=192.168.0.40+1+6d0408+cebfc105
ÿCall-ID: 45cb3499140d39ab61a93670773abd46@172.16.215.20:5060
ÿCSeq: 102 OPTIONS
ÿServer: DC-SIP/2.0
ÿOrganization: MetaSwitch
ÿContent-Length: 0
ÿ
ÿ<------------->
[2014-08-13 02:54:17] VERBOSE[2942] chan_sip.c: — (9 headers 0 lines) —
[2014-08-13 02:54:17] VERBOSE[2942] chan_sip.c: Really destroying SIP dialog ‘45cb3499140d39ab61a93670773abd46@172.16.215.20:5060’ Method: OPTIONS
[2014-08-13 02:54:17] VERBOSE[2942] chan_sip.c:
ÿ<— SIP read from UDP:64.245.205.20:5060 —>
ÿSIP/2.0 403 From: URI not recognized
ÿVia: SIP/2.0/UDP 172.16.215.20:5060;received=198.211.207.90;branch=z9hG4bK0f5c05fa;rport=43489
ÿFrom: “asterisk” sip:5002@172.16.215.20;tag=as0fa3c3f7
ÿTo: sip:64.245.205.20;tag=192.168.0.40+1+6d690a+f756da6f
ÿCall-ID: 0732c6e52f4cdb772d871c7525a7394e@172.16.215.20:5060
ÿCSeq: 102 OPTIONS
ÿServer: DC-SIP/2.0
ÿOrganization: MetaSwitch
ÿContent-Length: 0
ÿ
ÿ<------------->
[2014-08-13 02:54:17] VERBOSE[2942] chan_sip.c: — (9 headers 0 lines) —
[2014-08-13 02:54:17] VERBOSE[2942] chan_sip.c: Really destroying SIP dialog ‘0732c6e52f4cdb772d871c7525a7394e@172.16.215.20:5060’ Method: OPTIONS
[2014-08-13 02:54:17] VERBOSE[2942] chan_sip.c:
ÿ<— SIP read from UDP:64.245.205.20:5060 —>
ÿSIP/2.0 403 From: URI not recognized
ÿVia: SIP/2.0/UDP 172.16.215.20:5060;received=198.211.207.90;branch=z9hG4bK7e1fd1c7;rport=43489
ÿFrom: “asterisk” sip:5003@172.16.215.20;tag=as1fd14311
ÿTo: sip:64.245.205.20;tag=192.168.0.40+1+6dc705+55d445b9
ÿCall-ID: 23f1151c6290e95644811aff63d632b1@172.16.215.20:5060
ÿCSeq: 102 OPTIONS
ÿServer: DC-SIP/2.0
ÿOrganization: MetaSwitch
ÿContent-Length: 0
ÿ
ÿ<------------->
[2014-08-13 02:54:17] VERBOSE[2942] chan_sip.c: — (9 headers 0 lines) —
[2014-08-13 02:54:17] VERBOSE[2942] chan_sip.c: Really destroying SIP dialog ‘23f1151c6290e95644811aff63d632b1@172.16.215.20:5060’ Method: OPTIONS
[2014-08-13 02:54:17] VERBOSE[2942] chan_sip.c: Reliably Transmitting (NAT) to 64.245.205.20:5060:
ÿOPTIONS sip:64.245.205.20 SIP/2.0
ÿVia: SIP/2.0/UDP 172.16.215.20:5060;branch=z9hG4bK768cfd75;rport
ÿMax-Forwards: 70
ÿFrom: “asterisk” sip:5004@172.16.215.20;tag=as389b85d2
ÿTo: sip:64.245.205.20
ÿContact: sip:5004@172.16.215.20:5060
ÿCall-ID: 4307e87e28f9135439cf012a23c5a1bf@172.16.215.20:5060
ÿCSeq: 102 OPTIONS
ÿUser-Agent: Asterisk PBX 11.7.0
ÿDate: Wed, 13 Aug 2014 07:54:17 GMT
ÿAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
ÿSupported: replaces, timer
ÿContent-Length: 0
ÿ
ÿ
ÿ—
[2014-08-13 02:54:18] VERBOSE[2942] chan_sip.c:
ÿ<— SIP read from UDP:64.245.205.20:5060 —>
ÿSIP/2.0 403 From: URI not recognized
ÿVia: SIP/2.0/UDP 172.16.215.20:5060;received=198.211.207.90;branch=z9hG4bK768cfd75;rport=43489
ÿFrom: “asterisk” sip:5004@172.16.215.20;tag=as389b85d2
ÿTo: sip:64.245.205.20;tag=192.168.0.40+1+6d0a13+f635b4d1
ÿCall-ID: 4307e87e28f9135439cf012a23c5a1bf@172.16.215.20:5060
ÿCSeq: 102 OPTIONS
ÿServer: DC-SIP/2.0
ÿOrganization: MetaSwitch
ÿContent-Length: 0
ÿ
ÿ<------------->
[2014-08-13 02:54:18] VERBOSE[2942] chan_sip.c: — (9 headers 0 lines) —
[2014-08-13 02:54:18] VERBOSE[2942] chan_sip.c: Really destroying SIP dialog ‘4307e87e28f9135439cf012a23c5a1bf@172.16.215.20:5060’ Method: OPTIONS
[2014-08-13 02:54:18] VERBOSE[2942] chan_sip.c:
ÿ<— SIP read from UDP:64.245.205.20:5060 —>
ÿOPTIONS sip:metaswitch@172.16.215.20:5060;transport=udp SIP/2.0
ÿVia: SIP/2.0/UDP 64.245.205.20:5060;branch=z9hG4bK66sm1n20e87g4icnj311.1
ÿAllow-Events: message-summary, refer, dialog, line-seize, presence, call-info, as-feature-event
ÿMax-Forwards: 69
ÿCall-ID: 6C60E3A9@192.168.0.40
ÿFrom: sip:metaswitch@64.245.205.20:5060;tag=192.168.0.40+1+6dbe0f+8e2da025
ÿCSeq: 1031686864 OPTIONS
ÿOrganization: MetaSwitch
ÿSupported: resource-priority, 100rel
ÿContent-Length: 0
ÿContact: sip:metaswitch@64.245.205.20:5060;transport=udp
ÿTo: sip:metaswitch@172.16.215.20
ÿ
ÿ<------------->
[2014-08-13 02:54:18] VERBOSE[2942] chan_sip.c: — (12 headers 0 lines) —
[2014-08-13 02:54:18] VERBOSE[2942] chan_sip.c: Sending to 64.245.205.20:5060 (NAT)
[2014-08-13 02:54:18] VERBOSE[2942] chan_sip.c: Looking for metaswitch in default (domain 172.16.215.20)
[2014-08-13 02:54:18] VERBOSE[2942] chan_sip.c:
ÿ<— Transmitting (NAT) to 64.245.205.20:5060 —>
ÿSIP/2.0 404 Not Found
ÿVia: SIP/2.0/UDP 64.245.205.20:5060;branch=z9hG4bK66sm1n20e87g4icnj311.1;received=64.245.205.20;rport=5060
ÿFrom: sip:metaswitch@64.245.205.20:5060;tag=192.168.0.40+1+6dbe0f+8e2da025
ÿTo: sip:metaswitch@172.16.215.20;tag=as40e7d992
ÿCall-ID: 6C60E3A9@192.168.0.40
ÿCSeq: 1031686864 OPTIONS
ÿServer: Asterisk PBX 11.7.0
ÿAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
ÿSupported: replaces, timer
ÿAccept: application/sdp
ÿContent-Length: 0
ÿ
ÿ
ÿ<------------>
[2014-08-13 02:54:18] VERBOSE[2942] chan_sip.c: Scheduling destruction of SIP dialog ‘6C60E3A9@192.168.0.40’ in 32000 ms (Method: OPTIONS)
[2014-08-13 02:54:19] VERBOSE[2942] chan_sip.c: Really destroying SIP dialog ‘27217EFA@192.168.0.40’ Method: OPTIONS
[2014-08-13 02:54:41] VERBOSE[14033] asterisk.c: – Remote UNIX connection disconnected
[2014-08-13 02:54:48] VERBOSE[2942] chan_sip.c:
ÿ<— SIP read from UDP:64.245.205.20:5060 —>
ÿOPTIONS sip:metaswitch@172.16.215.20:5060;transport=udp SIP/2.0
ÿVia: SIP/2.0/UDP 64.245.205.20:5060;branch=z9hG4bK099m5310fo603isp75n0.1
ÿAllow-Events: message-summary, refer, dialog, line-seize, presence, call-info, as-feature-event
ÿMax-Forwards: 69
ÿCall-ID: A452F0F5@192.168.0.40
ÿFrom: sip:metaswitch@64.245.205.20:5060;tag=192.168.0.40+1+b141c+413a8d85
ÿCSeq: 265694914 OPTIONS
ÿOrganization: MetaSwitch
ÿSupported: resource-priority, 100rel
ÿContent-Length: 0
ÿContact: sip:metaswitch@64.245.205.20:5060;transport=udp
ÿTo: sip:metaswitch@172.16.215.20
ÿ
ÿ<------------->
[2014-08-13 02:54:48] VERBOSE[2942] chan_sip.c: — (12 headers 0 lines) —
[2014-08-13 02:54:48] VERBOSE[2942] chan_sip.c: Sending to 64.245.205.20:5060 (NAT)
[2014-08-13 02:54:48] VERBOSE[2942] chan_sip.c: Looking for metaswitch in default (domain 172.16.215.20)
[2014-08-13 02:54:48] VERBOSE[2942] chan_sip.c:
ÿ<— Transmitting (NAT) to 64.245.205.20:5060 —>
ÿSIP/2.0 404 Not Found
ÿVia: SIP/2.0/UDP 64.245.205.20:5060;branch=z9hG4bK099m5310fo603isp75n0.1;received=64.245.205.20;rport=5060
ÿFrom: sip:metaswitch@64.245.205.20:5060;tag=192.168.0.40+1+b141c+413a8d85
ÿTo: sip:metaswitch@172.16.215.20;tag=as2e4d3ca7
ÿCall-ID: A452F0F5@192.168.0.40
ÿCSeq: 265694914 OPTIONS
ÿServer: Asterisk PBX 11.7.0
ÿAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
ÿSupported: replaces, timer
ÿAccept: application/sdp
ÿContent-Length: 0
ÿ
ÿ
help me to get clear idea about this