Aastra 670i not registering


#1

Good evening,

Ive been testing with softphones for a while and finally got in some Aastra 6730i phones to continue learning on. Im having some issues however with registering them. I keep getting error messages regarding no user. Im running Debian with Asterisk 1.8.7.

The Phone settings i have included in the phones Global Sip Settings are:
Authentication name: 700
password: 1234
Proxy server and Registration server are set to IP of my server.

Ive also configured the same settings under Line 1 via the web based gui logged in as admin.

I feel like Im missing a registration setting regarding the user but im not sure what it is? Ive got the asterisk guide and a few other resources but I have the extension named [700] in Sip and thought that should be the user name, with the secret password?

Any help would be greatly appreciated as i continue to learn this amazing system

Extensions.conf File:
[default]
exten => 1001,1,Answer
exten => 1001,2,Playback(hello-world)
exten => 1001,3,Hangup

exten => 700,1,Dial(SIP/700)

My Sip.conf File:
[general]
bindport = 5060
bindaddr = 0.0.0.0

[700]
type=friend
secret=1234
host=dynamic
callerid=“Office”

Error Message and Sip show Peers results:

[Oct 27 18:28:22] NOTICE[2917]: chan_sip.c:24331 handle_request_register: Registration from ‘sip:No%20User@192.168.23.187:5060’ failed for ‘192.168.23.103:5060’ - No matching peer found
Scheduling destruction of SIP dialog ‘78dc39695ad49323’ in 32000 ms (Method: REGISTER)
Really destroying SIP dialog ‘78dc39695ad49323’ Method: REGISTER

<— SIP read from UDP:192.168.23.103:5060 —>
REGISTER sip:192.168.23.187:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.23.103:5060;branch=z9hG4bK9d0daefadf04be4cd.c60097147afb3ece0
Max-Forwards: 70
From: sip:No%20User@192.168.23.187:5060;tag=9ab32a4b13
To: sip:No%20User@192.168.23.187:5060
Call-ID: 78dc39695ad49323
CSeq: 15815 REGISTER
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
Allow-Events: talk, hold, conference, LocalModeStatus
Contact: “No User” sip:No%20User@192.168.23.103:5060;transport=udp;+sip.instance="urn:uuid:00000000-0000-1000-8000-00085D2BBE0A"
Supported: gruu, path
User-Agent: Aastra 6730i/2.6.0.1007
Content-Length: 0

<------------->
— (13 headers 0 lines) —
Sending to 192.168.23.103:5060 (no NAT)

<— Transmitting (no NAT) to 192.168.23.103:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.23.103:5060;branch=z9hG4bK9d0daefadf04be4cd.c60097147afb3ece0;received=192.168.23.103
From: sip:No%20User@192.168.23.187:5060;tag=9ab32a4b13
To: sip:No%20User@192.168.23.187:5060;tag=as2e70400a
Call-ID: 78dc39695ad49323
CSeq: 15815 REGISTER
Server: Asterisk PBX 1.8.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="3b1ca51e"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘78dc39695ad49323’ in 32000 ms (Method: REGISTER)
[Oct 27 18:58:22] NOTICE[2917]: chan_sip.c:24331 handle_request_register: Registration from ‘sip:No%20User@192.168.23.187:5060’ failed for ‘192.168.23.103:5060’ - No matching peer found
Scheduling destruction of SIP dialog ‘78dc39695ad49323’ in 32000 ms (Method: REGISTER)

<— SIP read from UDP:192.168.23.103:5060 —>
REGISTER sip:192.168.23.187:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.23.103:5060;branch=z9hG4bK14e64e04861412cf5.4b090717ba9592656
Max-Forwards: 70
From: sip:No%20User@192.168.23.187:5060;tag=9ab32a4b13
To: sip:No%20User@192.168.23.187:5060
Call-ID: 78dc39695ad49323
CSeq: 15816 REGISTER
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
Allow-Events: talk, hold, conference, LocalModeStatus
Authorization: Digest username=“No User”,realm=“asterisk”,nonce=“3b1ca51e”,uri=“sip:192.168.23.187:5060”,response=“3e3b57eea9b6c138b19bc6373518007b”,algorithm=MD5
Contact: “No User” sip:No%20User@192.168.23.103:5060;transport=udp;+sip.instance="urn:uuid:00000000-0000-1000-8000-00085D2BBE0A"
Supported: gruu, path
User-Agent: Aastra 6730i/2.6.0.1007
Content-Length: 0

<------------->
— (14 headers 0 lines) —
Sending to 192.168.23.103:5060 (no NAT)

<— Transmitting (no NAT) to 192.168.23.103:5060 —>
SIP/2.0 403 Forbidden (Bad auth)
Via: SIP/2.0/UDP 192.168.23.103:5060;branch=z9hG4bK14e64e04861412cf5.4b090717ba9592656;received=192.168.23.103
From: sip:No%20User@192.168.23.187:5060;tag=9ab32a4b13
To: sip:No%20User@192.168.23.187:5060;tag=as2e70400a
Call-ID: 78dc39695ad49323
CSeq: 15816 REGISTER
Server: Asterisk PBX 1.8.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

debian*CLI> sip show peers
Name/username Host Dyn Forcerport ACL Port Status
700 (Unspecified) D 0 Unmonitored
710 (Unspecified) D 0 Unmonitored
720 (Unspecified) D 0 Unmonitored
730 (Unspecified) D 0 Unmonitored
740 (Unspecified) D 0 Unmonitored
5 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 5 offline]


#2

Did you upgrade the telephone to the latest firmware version?

If you still have problems, please copy/paste the output of the “sip show users” Asterisk CLI command.


#3

Updating the firmware now but here is the output of the Show Users command:

Username Secret Accountcode Def.Context ACL ForcerPort
720 STRG@STT default No No
700 1234 default No No
730 STRG@STT default No No
710 STRG@STT default No No
740 STRG@STT default No No