URI not recognized

Asterisk Asterisk Support
1

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

2

Add extension metaswitch to the default domain.

Normally this is not a problem as the fact that the peer gets a response confirms there is connectivity and checking that is the only purpose of the OPTIONS request.

3

Hi David,

Thanks for your reply.

The same instance PBX was working in another premise, Since changed the location we are facing this problem, all network setting modified and disabled firewall rules also.

One more doubts is, why it’s showing like below debug messages.

[quote] ÿ<— SIP read from UDP:64.245.205.20:5060 —>
ÿOPTIONS [color=#0000FF]sip:metaswitch[/color]@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: [color=#0000FF]sip:metaswitch[/color]@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: [color=#0000FF]sip:metaswitch[/color]@64.245.205.20:5060;tag=192.168.0.40+1+6dbe0f+8e2da025
ÿTo: [color=#0000FF]sip:metaswitch[/color]@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 [color=#0000FF]sip:metaswitch[/color]@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: [color=#0000BF]sip:metaswitch[/color]@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: [color=#0000FF]sip:metaswitch[/color]@64.245.205.20:5060;tag=192.168.0.40+1+b141c+413a8d85
ÿTo: [color=#0000FF]sip:metaswitch[/color]@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
[/quote]

4

Because the peer is sending an OPTIONS request to metaswitch@… but you don’t have an extension with that name. OPTIONS is normally sent to test the connection, and, as I already said, you normally treat not found as being successful, as it still confirms that there is two way communication.

5

Thanks david.