Almost there with Avaya 9641G

Hello I have an issue.

I’m trying to register an Avaya 9641 (SIP firmware S96x1_SALBR6_2_2r17_V4r70.tar) to FreePBX 17 / Asterisk 21 using PJSIP.

Current status:

  • Phone provisions successfully from my local server.

  • It fetches 96x1Supgrade.txt and 46xxsettings.txt.

  • PBX IP is 192.168.50.175.

  • Extension 1000 exists and loads correctly in PJSIP again.

  • pjsip show endpoint 1000 shows the endpoint is present.

  • Identify match is 192.168.50.146/32.

The problem:

  • The phone sends REGISTER.

  • FreePBX responds 401 Unauthorized.

  • The phone sends another REGISTER, but never includes an Authorization header.

  • This repeats forever, so no contact is created and the phone stays on “Acquiring Service…”.

Important details:

  • The phone is reaching the correct PBX IP.

  • Provisioning is no longer the issue.

  • Endpoint loading is no longer the issue.

  • The remaining failure is that the Avaya does not answer the digest challenge.

Example SIP behavior:

  • REGISTER sip:192.168.50.175

  • 401 Unauthorized

  • REGISTER sip:192.168.50.175

  • 401 Unauthorized

  • repeated, with no Authorization header ever sent

Things already tried:

  • identify_by=ip

  • matching by 192.168.50.146/32

  • custom provisioning values

  • attempts to relax/remove auth in PJSIP

  • restoring endpoint when bad overrides broke loading

Question:
Has anyone successfully registered an Avaya 9641/9611 to FreePBX PJSIP when the phone refuses to respond to digest auth? Is the right path to:

  1. switch this extension to chan_sip,

  2. use a fully manual IP-authenticated PJSIP endpoint,
    or

  3. set a specific Avaya provisioning/auth parameter so the phone actually sends Authorization after 401?

Howdy! Welcome to the forums.

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Please paste them at pastebin.com and post the link here.

root@freepbx:~# nano /etc/asterisk/pjsip.endpoint_custom_post.confroot@freepbx - Pastebin.com Hi, please let me know if this is the correct info.

@avayax

In the extension, set Rewrite Contact to No.
Submit, Apply Config, test.

https://pastebin.com/QJZqwPrq

Sorry, my mistake.
Set Force rport to No.

Bizarre, line 24 of the paste (Via header) shows that the 401 reply should go to port 5060, but line 32 shows it being sent to port 1025.

Please post a screenshot of the Advanced tab for extension 1000. Also, at the Asterisk command prompt, type
pjsip show endpoint 1000
and post the output.

OK, so the extension setting for Force rport is correct (No), but line 72 of the paste shows it’s wrong (true). For extension 1000, try doing a Submit and Apply Config again.
Recheck the output of pjsip show endpoint 1000. If force rport now show false, retest registration. If it still shows true, try rebooting the server and check again. If it’s false now, retest registration. If it’s still true, there is a bug in the module, but we can probably get it working with a custom config file.

I rebooted because it was still true. This is the result after rebooting and pjsip show endpoint 1000

I don’t know the actual cause of this conflict, but strongly believe it’s related to the unusual settings (especially identify by IP) that you have applied to extension 1000.

I suggest that you proceed by temporarily shutting down the Avaya, deleting extension 1000, Submit and Apply Config, reboot server. Check that endpoint 1000 is gone.

Now create a new extension 1000. Initially, set up the minimum for the Avaya to register: Set Secret to match the SIP password in the Avaya and set Force rport and Rewrite Contact to No. Submit and Apply Config, boot up the Avaya and see whether it registers. If it does, it should be able to make and receive calls. If so, then make any settings for the extension that you need (Display Name, Outbound CID, etc.) and retest.

SUCCESS !!! Thank you soooo very much! I appreciate your support infinitely! I have been wrestling with this issue for 2 weeks and because of your help you made it possible.