I have a Siemens/Unify phone that doesn’t work with paging and intercom out of the box. The wiki for Siemens OpenStage phones says this.
When automatic call answering is allowed on the phone, automatic call answering can be requested from within the incoming call via the SIP Alert-Info header. In the Asterisk extensions.conf file, the SIPAddHeader function must be programmed like this: SIPAddHeader(“Alert-Info:hxxp://example.com;info=alert-autoanswer”)
How do I implement this inside FreePBX?
I’ve been playing around with the paging_autoanswer table and I can’t get freepbx to detect my phone’s user agent.
I’ve tried both these
| OpenStage | ALERTINFO | info=alert-autoanswer |
| OpenStage_60_V3 | ALERTINFO | info=alert-autoanswer
phones UA is: OpenStage_60_V3 R3.32.0 SIP 141208
only works with CHan-SIP. If you are using PJSip, you will need to use the PJ-SIP specific dialplan function that does that.
I’m going to guess that you will need to add an extension specific dialplan in your extensions_custom.conf file to handle the specifics of this phone.
I’m trying to remember the exact bit of the screen that does this, but IIRC, there is a setting on the Extensions page that allows you to specify custom alert information. The tricky bit is that it’s design to work specifically with Sangoma phones. You can populate the field, but I don’t think you can use it like a pull-down. Might be remembering something else, though, so if you don’t find the entry right away, move on to writing a custom context for your extensions_custom.conf dialplan.
I’m using Chan sip on FreePBX 13.0.167.
Is there a way to check and see if the phone is even working properly?
Yeah.
The first should be to use the asterisk cli.Log into the console as ‘root’ and type ‘asterisk -vr’ and watch the fun. This is the server’s perspective, but it’s usually enough to get you close enough to the problem to identify the root cause.
If you don’t see what you’re looking for there, you can set up a SIP trace. This, however, is beyond my skill set for a quick answer (I’d actually have to look it up; I don’t do it very often). Using a SIP trace, you can see what’s happening between the phone and the server.
It works!
Problem 1: For some reason these unify openstage phones put the setting for enabling auto answer in the user settings area.
Settings>User>Configuration>Incoming Calls>CTI Calls>AutoAnswer = YES
Problem 2: The MySQL query I was running to add the alertinfo string didn’t have an escaped backslash.
I used this:
mysql
use asterisk
insert into paging_autoanswer VALUES('OpenStage','ALERTINFO','<http://example.com>\\;info=alert-autoanswer');`
exit
fwconsole reload