Configuration of Asterisk and Pingtel Expressa phones

Has anyone got Pingtel Expressa phones to work with Asterisk? I’ve got several, but I can’t figure out how to configure them and Asterisk to work together. Here’s the exchange between extension 172 (IP address 192.168.46.172) with Asterisk server (192.168.46.178). It’s like it’s close, but no cigar :slight_smile:

172 -> AS
REGISTER sip:192.168.46.178 SIP/2.0
From: sip:[email protected]
To: sip:[email protected]
Call-Id: [email protected]
Cseq: 1 REGISTER
Contact: sip:[email protected]
Expires: 3600
Content-Length: 0
Accept-Language: en
Supported: sip-cc, sip-cc-01, timer
Www-Authenticate: Digest REALM=“asterisk”, NONCE=“1234567890”, OPAQUE="abcdefghij"
User-Agent: Pingtel/1.0.0 (VxWorks)
Via: SIP/2.0/UDP 192.168.46.172

AS -> 172
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.46.172;received=192.168.46.172
From: sip:[email protected]
To: sip:[email protected];tag=as7fea4e4c
Call-ID: [email protected]
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="367baa44"
Content-Length: 0

172 -> AS
REGISTER sip:192.168.46.178 SIP/2.0
From: sip:[email protected]
To: sip:[email protected]
Call-Id: [email protected]
Cseq: 2 REGISTER
Contact: sip:[email protected]
Expires: 3600
Content-Length: 0
Accept-Language: en
Supported: sip-cc, sip-cc-01, timer
Www-Authenticate: Digest REALM=“asterisk”, NONCE=“1234567890”, OPAQUE="abcdefghij"
User-Agent: Pingtel/1.0.0 (VxWorks)
Authorization: DIGEST USERNAME=“172”, REALM=“asterisk”, NONCE=“367baa44”, RESPONSE=“5effc6a3680cae80b3b22d39633a8e5d”, URI="sip:192.168.46.178"
Via: SIP/2.0/UDP 192.168.46.172

AS -> 172
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.46.172;received=192.168.46.172
From: sip:[email protected]
To: sip:[email protected];tag=as7fea4e4c
Call-ID: [email protected]
CSeq: 2 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="010db116"
Content-Length: 0

Here’s the relevant section of /etc/asterisk/sip_additional.conf:
[172]
deny=0.0.0.0/0.0.0.0
type=friend
secret=1234
qualify=yes
port=5060
pickupgroup=
permit=0.0.0.0/0.0.0.0
nat=never
[email protected]
host=192.168.46.178
dtmfmode=rfc2833
dial=SIP/172
context=from-internal
canreinvite=no
callgroup=
callerid=device <172>
accountcode=
call-limit=50

It’s not easy to show the configuration values of the Expressa phone here.

There must be perhaps one thing that’s misconfigured between them, but for the life of me I can’t see it, so any help would be welcome.

Its been years. I think you have to take qualify=no

Thanks for your response, but setting qualify=yes or qualify=no seems to make no difference: on attempted registration, Asterisk still says “Unauthorized” twice, then the Pingtel phone gives up.

Just what does “qualify” mean to Asterisk? I can’t find any references in the docs as to how one should choose what setting to use, or what it affects. And is “no” a valid setting? NAT, for example, is “yes” or “never” according to the docs. (It might help if the FreePBX GUI offered a pull-down with the acceptable values, rather than free text input.)

What is “host”? I’ve allocated fixed IP addresses (that’s how this part of our LAN is administered) to the Asterisk server and the phones. But does “host” refer to the IP address of the Asterisk server (that’s the guess I’ve taken so far), or of the phone?

Host is the host of the connected peer. It can either be static or dynamic.

The Digium sample configs distributed with the Asterisk source are the best references.

The voip-info page is also pretty good:

http://www.voip-info.org/wiki/view/Asterisk+config+sip.conf