SIP phone setup troubleshooting - SOLVED AND WORKING!

Status, setup, and issues:

I am running up to date ASL3 on a Pi4 w a lan side static ip and using a Polycom VVX201. I am using 1001 as the line in all the cfg files.
The line is registered on the phone which I finally solved by adding the 5060 port to the firewall in ASL3 via the cockpit. But, when dialing my node I get a fast busy signal. When I followed the troubleshooting steps I ran into a "no such command issue" mentioned below.

I am following the "official" ASL3 instructions here from the manual:

I started the CLI with "asterisk -rvvv" but the next command fails:

ASL48654*CLI> pjsip set logger on
No such command 'pjsip set logger on' (type 'core show help pjsip set' for other possible commands)

(Also just fyi, there is no "global" stanza in the modules.conf near the end of the file as mentioned in the instructions. I just put the provided stanza at the end of the file.)

Thanks for any assistance. (I tried to make this as clear as possible so this thread can help others.)

(thx to everyone who worked on ASL3 including the manual)

UPDATE- WORKING-

  • for some reason "pjsip set logger on" started working... not sure why.

  • I dialed my node from the phone and looked at the output in CLI- there was a bunch of info- but this line caught my eye:

"[2026-08-02 22:13:15.752] NOTICE[1773]: res_pjsip_session.c:3940 new_invite: 1001: Call (UDP:xxx.xxx.x.xxx:5060) to extension '48654' rejected because extension not found in context 'from-internal'"

  • I began comparing what the manual showed and what my cfg files showed and found the line in question that was in error in default ASL3:

The original stanza in pjsip.conf had:
"context=from-internal" and NOT "context=sip-phones"

It was also missing the callerID line. I adjusted as you see below and when tested it worked!

[1001]
type=endpoint
transport=transport-udp
context=sip-phones ;was: from-internal
disallow=all
allow=ulaw
allow=gsm
auth=1001
aors=1001
callerid="My CallerID"

Following your progress (someone just gave me a VVX-450).

I got it working- updated the original post. I will add info on configuring the Polycom VVX 201 as well.

Polycom cfg- only using Line 2 cfg page- the simple setup/line 1 pages are used for line 1 phone number.

Line 2 for ASL3:

  • display name shows on the phone screen
  • address is the extension used in the cfg files
  • user/pass are from the pjsip.conf file
  • ip is local LAN ip of the ASL3 node


Hope this helps someone!