Hi,
Installed asterisk and sjphones successfully but in the asterisk CLI i get the following error message.
Please resolve the same if any one has the solution.
CLI> – Got SIP response 405 “Method Not Allowed” back from 10.1.1.177
– Got SIP response 405 “Method Not Allowed” back from 10.1.1.160
This is probably related to IM/presence/… which is supported by the client but not by the server. You should check the complete debug output to see what happens before that error message.
Thank very much for your valuable feedback.
here is my debugging output …pl have a look and help me as im new to this asterisk domain.
I also did get what you actually you meant by IM/presence pl make it clear 
Pl help me im struggling to solve this issue from last two days and finally thought i will get help from the forum.
when i enter: " sip set debug " in the CLI i get the following
<— SIP read from 10.1.1.177:5060 —>
OPTIONS sip:10.1.1.161 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.177;rport;branch=z9hG4bK0a0101b10000000b46540f440000506e00002885
Content-Length: 0
Call-ID: 7B8FEC9F-EC73-4CC1-8EAE-39069108A6B7@10.1.1.177
CSeq: 508 OPTIONS
From: sip:santest@10.1.1.161;tag=866297963492
Max-Forwards: 70
To: sip:10.1.1.161
<------------->
— (8 headers 0 lines) —
Looking for s in default (domain 10.1.1.161)
<— Transmitting (no NAT) to 10.1.1.177:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.1.177;branch=z9hG4bK0a0101b10000000b46540f440000506e00002885;received=10.1.1.177;rport=5060
From: sip:santest@10.1.1.161;tag=866297963492
To: sip:10.1.1.161;tag=as5f1a5ce9
Call-ID: 7B8FEC9F-EC73-4CC1-8EAE-39069108A6B7@10.1.1.177
CSeq: 508 OPTIONS
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:10.1.1.161
Accept: application/sdp
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog ‘7B8FEC9F-EC73-4CC1-8EAE-39069108A6B7@10.1.1.177’ in 32000 ms (Method: OPTIONS)
<— SIP read from 10.1.1.177:5060 —>
REGISTER sip:10.1.1.161 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.177;rport;branch=z9hG4bK0a0101b10000000b46540f48000040a30000288f
Content-Length: 0
Contact: sip:santest@10.1.1.177:5060
Call-ID: 9ABFFA86-AD71-4C93-81CA-F8F9137F9CD2@10.1.1.177
CSeq: 178 REGISTER
From: sip:santest@10.1.1.161;tag=8663364025308
Max-Forwards: 70
To: sip:santest@10.1.1.161
User-Agent: SJphone/1.60.289a (SJ Labs)
<------------->
— (10 headers 0 lines) —
Using latest REGISTER request as basis request
Sending to 10.1.1.177 : 5060 (NAT)
<— Transmitting (no NAT) to 10.1.1.177:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.1.1.177;branch=z9hG4bK0a0101b10000000b46540f48000040a30000288f;received=10.1.1.177;rport=5060
From: sip:santest@10.1.1.161;tag=8663364025308
To: sip:santest@10.1.1.161
Call-ID: 9ABFFA86-AD71-4C93-81CA-F8F9137F9CD2@10.1.1.177
CSeq: 178 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:santest@10.1.1.161
Content-Length: 0
<------------>
<— Transmitting (no NAT) to 10.1.1.177:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.1.1.177;branch=z9hG4bK0a0101b10000000b46540f48000040a30000288f;received=10.1.1.177;rport=5060
From: sip:santest@10.1.1.161;tag=8663364025308
To: sip:santest@10.1.1.161;tag=as08f7d7da
Call-ID: 9ABFFA86-AD71-4C93-81CA-F8F9137F9CD2@10.1.1.177
CSeq: 178 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="051f080a"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog ‘9ABFFA86-AD71-4C93-81CA-F8F9137F9CD2@10.1.1.177’ in 32000 ms (Method: REGISTER)
<— SIP read from 10.1.1.177:5060 —>
REGISTER sip:10.1.1.161 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.177;rport;branch=z9hG4bK0a0101b10000000b46540f48000071be00002892
Content-Length: 0
Contact: sip:santest@10.1.1.177:5060
Call-ID: 9ABFFA86-AD71-4C93-81CA-F8F9137F9CD2@10.1.1.177
CSeq: 179 REGISTER
From: sip:santest@10.1.1.161;tag=8663364014286
Max-Forwards: 70
To: sip:santest@10.1.1.161
User-Agent: SJphone/1.60.289a (SJ Labs)
Authorization: Digest username=“santest”,realm=“asterisk”,nonce=“051f080a”,uri=“sip:10.1.1.161”,response=“25d2a81a0c2dedd73fd060c8523f7aaa”,algorithm=“MD5”
<------------->
— (11 headers 0 lines) —
Using latest REGISTER request as basis request
Sending to 10.1.1.177 : 5060 (NAT)
<— Transmitting (no NAT) to 10.1.1.177:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.1.1.177;branch=z9hG4bK0a0101b10000000b46540f48000071be00002892;received=10.1.1.177;rport=5060
From: sip:santest@10.1.1.161;tag=8663364014286
To: sip:santest@10.1.1.161
Call-ID: 9ABFFA86-AD71-4C93-81CA-F8F9137F9CD2@10.1.1.177
CSeq: 179 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:santest@10.1.1.161
Content-Length: 0
<------------>
<— Transmitting (no NAT) to 10.1.1.177:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.1.177;branch=z9hG4bK0a0101b10000000b46540f48000071be00002892;received=10.1.1.177;rport=5060
From: sip:santest@10.1.1.161;tag=8663364014286
To: sip:santest@10.1.1.161;tag=as08f7d7da
Call-ID: 9ABFFA86-AD71-4C93-81CA-F8F9137F9CD2@10.1.1.177
CSeq: 179 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Expires: 120
Contact: sip:santest@10.1.1.177:5060;expires=120
Date: Wed, 23 May 2007 15:18:49 GMT
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog ‘9ABFFA86-AD71-4C93-81CA-F8F9137F9CD2@10.1.1.177’ in 32000 ms (Method: REGISTER)
Reliably Transmitting (no NAT) to 10.1.1.177:5060:
OPTIONS sip:santest@10.1.1.177:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.161:5060;branch=z9hG4bK5a1721cc;rport
From: “asterisk” sip:asterisk@10.1.1.161;tag=as54b32b09
To: sip:santest@10.1.1.177:5060
Contact: sip:asterisk@10.1.1.161
Call-ID: 0431f61f378f335b5bf64b194d208966@10.1.1.161
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 23 May 2007 15:18:49 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
<— SIP read from 10.1.1.177:5060 —>
SIP/2.0 405 Method Not Allowed
Via: SIP/2.0/UDP 10.1.1.161:5060;rport=5060;received=10.1.1.161;branch=z9hG4bK5a1721cc
Content-Length: 0
Allow: INVITE, ACK, CANCEL, BYE, REFER, NOTIFY
Call-ID: 0431f61f378f335b5bf64b194d208966@10.1.1.161
CSeq: 102 OPTIONS
From: "asterisk"sip:asterisk@10.1.1.161;tag=as54b32b09
Server: SJphone/1.60.289a (SJ Labs)
To: "santesting"sip:santest@10.1.1.177:5060
<------------->
— (9 headers 0 lines) —
Really destroying SIP dialog ‘0431f61f378f335b5bf64b194d208966@10.1.1.161’ Method: OPTIONS
[May 23 20:48:49] NOTICE[18816]: chan_sip.c:7177 sip_reg_timeout: – Registration for ‘santest@10.1.1.177’ timed out, trying again (Attempt #49)
REGISTER 12 headers, 0 lines
Reliably Transmitting (no NAT) to 10.1.1.177:5060:
REGISTER sip:10.1.1.177 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.161:5060;branch=z9hG4bK76a78b7b;rport
From: sip:santest@10.1.1.177;tag=as50d4e797
To: sip:santest@10.1.1.177
Call-ID: 0888c2f359384eb729bdd4012cc85855@127.0.0.1
CSeq: 151 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Expires: 120
Contact: sip:789@10.1.1.161
Event: registration
Content-Length: 0
[May 23 20:48:49] NOTICE[18816]: chan_sip.c:7177 sip_reg_timeout: – Registration for ‘lakshmi@10.1.1.160’ timed out, trying again (Attempt #49)
REGISTER 12 headers, 0 lines
Reliably Transmitting (no NAT) to 10.1.1.160:5060:
REGISTER sip:10.1.1.160 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.161:5060;branch=z9hG4bK2b23fe32;rport
From: sip:lakshmi@10.1.1.160;tag=as6ac47106
To: sip:lakshmi@10.1.1.160
Call-ID: 15abc62979fdc5776d9417456abbf5cf@127.0.0.1
CSeq: 151 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Expires: 120
Contact: sip:456@10.1.1.161
Event: registration
Content-Length: 0
Method OPTIONS is not supported by your client as it not shown in Allow: line. Try current version of SJphone.
Hi , im curious about one thing that,
i was able to configure the same phone with the same version with asterisk pbx installed on windows machine, and was working fine.
Now im running asterisk on fedora6 and the phone on windows xp and getting this message.
The other thing is though it says 405 error iam able to make a call and receive the call from win xp to linux and viceversa ,but no audio is heard on either side only when i dial a number i get the dial tone , once the call is established i dnt have any audio, pl guide me on the same to resolve this issue i will be very much thank full to u…
thanks again for responding to my mails