No matching peer found

Asterisk Asterisk Support
1

I upgraded or better say I reinstalled asterisk from 1.4 to 1.8 last night.,

At this point is might be easier to get the ISO and start from scratch, but not sure .

None of the phones are registering.

I have checked the password on the phone and it matches the conf file.

here is part of the sip debug log, would it be easier to just start from scratch? I seem to remember the CD ISO install worked very well. Also having a problem with the Superfecta not being able to access the web php, permission denied. Does that have anything to do with this problem?

<------------>
<— SIP read from UDP:10.215.81.50:5060 —>
REGISTER sip:10.215.81.1:5060 SIP/2.0
Via: SIP/2.0/UDP 10.215.81.50;branch=z9hG4bK83879277347B4E48
From: “Mark Worsnop” sip:300@10.215.81.1;tag=9C233824-DC3DF231
To: sip:300@10.215.81.1
CSeq: 1 REGISTER
Call-ID: a63e4085-5fad01be-4f400b43@10.215.81.50
Contact: sip:300@10.215.81.50;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.0.1.0032
Max-Forwards: 70
Expires: 3600
Content-Length: 0

<------------->
[Jan 31 06:36:33] VERBOSE[8698] chan_sip.c: — (11 headers 0 lines) —
[Jan 31 06:36:33] VERBOSE[8698] chan_sip.c: Sending to 10.215.81.50:5060 (no NAT)
[Jan 31 06:36:33] VERBOSE[8698] chan_sip.c:
<— Transmitting (no NAT) to 10.215.81.50:5060 —>
SIP/2.0 100 Trying^M
Via: SIP/2.0/UDP 10.215.81.50;branch=z9hG4bK83879277347B4E48;received=10.215.81.50^M
From: “Mark Worsnop” sip:300@10.215.81.1;tag=9C233824-DC3DF231^M
To: sip:300@10.215.81.1^M
Call-ID: a63e4085-5fad01be-4f400b43@10.215.81.50^M
CSeq: 1 REGISTER^M
Server: Asterisk PBX 1.8.2.3^M
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH^M
Supported: replaces, timer^M
Content-Length: 0^M

<------------>
[Jan 31 06:36:33] VERBOSE[8698] chan_sip.c:
<— Transmitting (no NAT) to 10.215.81.50:5060 —>
SIP/2.0 401 Unauthorized^M
Via: SIP/2.0/UDP 10.215.81.50;branch=z9hG4bK83879277347B4E48;received=10.215.81.50^M
From: “Mark Worsnop” sip:300@10.215.81.1;tag=9C233824-DC3DF231^M
To: sip:300@10.215.81.1;tag=as4177dc05^M
Call-ID: a63e4085-5fad01be-4f400b43@10.215.81.50^M
CSeq: 1 REGISTER^M
Server: Asterisk PBX 1.8.2.3^M
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH^M
Supported: replaces, timer^M
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce=“47feecdf”^M
Content-Length: 0^M
^M

<------------>
[Jan 31 06:36:33] VERBOSE[8698] chan_sip.c: Scheduling destruction of SIP dialog ‘a63e4085-5fad01be-4f400b43@10.215.81.50’ in 32000 ms (Method: REGISTER)
[Jan 31 06:36:33] NOTICE[8698] chan_sip.c: Registration from ‘sip:300@10.215.81.1’ failed for ‘10.215.81.50:5060’ - No matching peer found
[Jan 31 06:36:33] VERBOSE[8698] chan_sip.c: Scheduling destruction of SIP dialog ‘a63e4085-5fad01be-4f400b43@10.215.81.50’ in 32000 ms (Method: REGISTER)
[Jan 31 06:36:33] VERBOSE[8698] chan_sip.c:
<— SIP read from UDP:10.215.81.50:5060 —>
REGISTER sip:10.215.81.1:5060 SIP/2.0
Via: SIP/2.0/UDP 10.215.81.50;branch=z9hG4bKca679afa7B6880CF
From: “Mark Worsnop” sip:300@10.215.81.1;tag=9C233824-DC3DF231
To: sip:300@10.215.81.1
CSeq: 2 REGISTER
Call-ID: a63e4085-5fad01be-4f400b43@10.215.81.50
Contact: sip:300@10.215.81.50;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.0.1.0032
Authorization: Digest username=“1234”, realm=“asterisk”, nonce=“47feecdf”, uri=“sip:10.215.81.1:5060”, response=“6b7fa51550a00d43b95e882531b999ec”, algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0

<------------->
[Jan 31 06:36:33] VERBOSE[8698] chan_sip.c: — (12 headers 0 lines) —
[Jan 31 06:36:33] VERBOSE[8698] chan_sip.c: Sending to 10.215.81.50:5060 (no NAT)
[Jan 31 06:36:33] VERBOSE[8698] chan_sip.c:
<— Transmitting (no NAT) to 10.215.81.50:5060 —>
SIP/2.0 100 Trying^M
Via: SIP/2.0/UDP 10.215.81.50;branch=z9hG4bKca679afa7B6880CF;received=10.215.81.50^M
From: “Mark Worsnop” sip:300@10.215.81.1;tag=9C233824-DC3DF231^M
To: sip:300@10.215.81.1^M
Call-ID: a63e4085-5fad01be-4f400b43@10.215.81.50^M
CSeq: 2 REGISTER^M
Server: Asterisk PBX 1.8.2.3^M
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH^M
Supported: replaces, timer^M
Content-Length: 0^M
^M

<------------>
[Jan 31 06:36:33] VERBOSE[8698] chan_sip.c:
<— Transmitting (no NAT) to 10.215.81.50:5060 —>
SIP/2.0 403 Forbidden (Bad auth)^M
Via: SIP/2.0/UDP 10.215.81.50;branch=z9hG4bKca679afa7B6880CF;received=10.215.81.50^M
From: “Mark Worsnop” sip:300@10.215.81.1;tag=9C233824-DC3DF231^M
To: sip:300@10.215.81.1;tag=as4177dc05^M
Call-ID: a63e4085-5fad01be-4f400b43@10.215.81.50^M
CSeq: 2 REGISTER^M
Server: Asterisk PBX 1.8.2.3^M
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH^M
Supported: replaces, timer^M
Content-Length: 0^M
^M

<------------>
[Jan 31 06:36:33] NOTICE[8698] chan_sip.c: Registration from ‘sip:300@10.215.81.1’ failed for ‘10.215.81.50:5060’ - No matching peer found
[Jan 31 06:36:33] VERBOSE[8698] chan_sip.c: Scheduling destruction of SIP dialog ‘a63e4085-5fad01be-4f400b43@10.215.81.50’ in 32000 ms (Method: REGISTER)
[Jan 31 06:36:34] VERBOSE[8698] chan_sip.c:
<— SIP read from UDP:10.215.81.50:5060 —>
SUBSCRIBE sip:224@10.215.81.1:5060 SIP/2.0
Via: SIP/2.0/UDP 10.215.81.50;branch=z9hG4bKf4c6ae9d6C56F7F6
From: “Mark Worsnop” sip:300@10.215.81.1;tag=333A54B2-1B015227
To: sip:224@10.215.81.1
CSeq: 1 SUBSCRIBE
Call-ID: a047bf1b-ea29d11c-20821c9@10.215.81.50
Contact: sip:300@10.215.81.50
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: message-summary
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.0.1.0032
Accept: application/simple-message-summary
Max-Forwards: 70
Expires: 3600
Content-Length: 0

<------------->
[Jan 31 06:36:34] VERBOSE[8698] chan_sip.c: — (14 headers 0 lines) —
[Jan 31 06:36:34] VERBOSE[8698] chan_sip.c:
<— Transmitting (no NAT) to 10.215.81.50:5060 —>
SIP/2.0 403 Forbidden (policy)^M
Via: SIP/2.0/UDP 10.215.81.50;branch=z9hG4bKf4c6ae9d6C56F7F6;received=10.215.81.50^M
From: “Mark Worsnop” sip:300@10.215.81.1;tag=333A54B2-1B015227^M
To: sip:224@10.215.81.1;tag=as37ee26a2^M
Call-ID: a047bf1b-ea29d11c-20821c9@10.215.81.50^M
CSeq: 1 SUBSCRIBE^M
Server: Asterisk PBX 1.8.2.3^M
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH^M
Supported: replaces, timer^M
Content-Length: 0^M

2

A slight possibility, What type of machine is this running on?

I would also recommend looking at the log itself. On occasion, non-visible characters can appear in the log, but not on the console.

3

Hi,

Did you find a solution to this? I have the same problem.

4

Without sip.conf, one cannot say.

5

On machines with an ARM cpu, this error can be cause by an issue reported in a different thread:

http://forums.digium.com/viewtopic.php?f=1&t=76909

The solution is also given in that thread.