Hello, trying to connect an Avaya 9621G to Freepbx.
UPDATE : Successful ! See the end of this post.
What I found is : Connecting in UDP does not seem to work well. When I try to connect (set username and password), the phone displays “Acquiring service…” forever.
The Avaya phone speaks on port 1025, and Asterisk tries to connect to 5060, although I have port=1025 in the phone config. I see lot of 0.0.0.0 adresses in the log. Here is SIP debug log (192.168.1.93 is the address of the phone) :
<--- SIP read from UDP:192.168.1.93:1025 --->
REGISTER sip:192.168.1.182 SIP/2.0
CSeq: 21 REGISTER
Via: SIP/2.0/UDP 0.0.0.0;branch=z9hG4bK15_386ddfa81b77ea011s4ai4k4v2j5x1v5d3i243x4i1p10165g2w_R4188.8.131.52
User-Agent: Avaya one-X Deskphone 184.108.40.206 (39)
--- (12 headers 0 lines) ---
Sending to 192.168.1.93:5060 (no NAT)
<--- Transmitting (no NAT) to 192.168.1.93:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 0.0.0.0;branch=z9hG4bK15_386ddfa81b77ea011s4ai4k4v2j5x1v5d3i243x4i1p10165g2w_R4220.127.116.11;received=192.168.1.93
CSeq: 21 REGISTER
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7f016178"
So it is better to connect in TCP. I set it in the CRAFT procedure of the phone, in SIP menu and SIP Proxy Server configuration.
I added in Asterisk SIP settings :
I also set nat=no, as recommended in this post.
Now the situation is :
Communication seems OK in the log. The port is the same for sending and receiving, and I no longer see 0.0.0.0 addresses. On the phone, the message sent appears on the screen.
If I deliberately set a wrong password, I get : 403 Forbidden (Bad Auth)
If I set the right password, I get : 403 Forbidden.
And no reason is indicated for the error.
I am stuck there and don’t know what may be wrong.
UPDATE : I found the problem.
The log indicated that this extension was not set for TCP.
I added transport=tcp in the extension configuration (sip_additional.conf) and now the phone works in FreeBBX !
I am using Elastix 2.5
I will post later a more detailed report of all the steps necessary to have this beautiful phone work with FreePBX / Elastix