PJSIP with AMI header

Asterisk Asterisk Endpoints
1

Hello I am trying to use PJSIP asterisk 22.5.0 with AMI call files and set a header to auto-answer

Action: Originate
Async: yes
Channel: PJSIP/402
Context: dialout
Exten: mycall
Priority: 1
Timeout: 40000
Variable: PJSIP_HEADER(add,Alert-Info)=“Ring Answer”
ActionID: 100000

The Polycom phone (which auto-answered under chan_sip) - does not auto answer.

What am I not doing correctly for the header ?

Thanks

Jerry

2

What actually goes out in the SIP INVITE?

3

INVITE sip:402@192.168.1.22 SIP/2.0^M
Via: SIP/2.0/UDP 192.168.1.8:5060;rport;branch=z9hG4bKPj45734a77-517f-4f7b-8de2-711980f3818b^M
From: “Jerry Geis 101” sip:XXXXXXX@192.168.1.8;tag=2a971096-9851-4e71-a8fc-9b499f3390c9^M
To: sip:402@192.168.1.22^M
Contact: sip:asterisk@192.168.1.8:5060^M
Call-ID: 37d77278-5faf-4347-aeef-b69f1231ac71^M
CSeq: 5098 INVITE^M
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER^M
Supported: 100rel, timer, replaces, norefersub, histinfo^M
Session-Expires: 1800^M
Min-SE: 90^M
Alert-Info: Ring Answer^M
Max-Forwards: 70^M
User-Agent: Asterisk PBX^M
Content-Type: application/sdp^M
Content-Length: 282^M

Seems like its there - but the polycom phone does not autoanswer like it used to.

jerry

4

Actually I tried a different phone - it works on Polycom 335 - but not on Polycom 331

Weird.

Jerry

5

Alas, I can only confirm the resulting INVITE. Polycom is outside my knowledge. I’ve moved your topic over to Endpoints, someone else may know the difference.

6

Is this a new phone? Have you enabled it on the phone? It’s a security risk, so it will not be enabled out of the box.

7

Actually OLD phone - used it for years

Thanks!

Jerry

8

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