SOLVED How do I enable paging/intercom for Polycom vvx 310, 600

KUMARULLAL

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I have incrediblepbbx with Asterisk 11 and freepbx 2.11 running at digital ocean.
I modified the mysql dabase in paging table from
Polycom ALERTINFO Alert-Info: info=Auto Answer
To
Polycom ALERTINFO Alert-Info: info=Ring Answer

The phone setup throught the web browser has auto answer option, when enabled, it simple answers even if you dial the extensions. SO I had to disable it.
What am I missing?
Thanks and advance.
 

KUMARULLAL

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After a several hours of trial and error, I could finally resolve the issue. Polycom has very poor documentation for asterisk.
Here is how I did it.
On Freepbx I did the following:
I modified the mysql dabase in paging table from
Polycom ALERTINFO Alert-Info: info=Auto Answer
To
Polycom ALERTINFO Alert-Info: info=Ring Answer
Next
in [macro-extension] in extention_additional.conf
I changed from

exten => s,n,ExecIf($["${phone:0:7}" = "Polycom"]?Set(ALERTINFO=Alert-Info: info=Auto Answer))
To
exten => s,n,ExecIf($["${phone:0:7}" = "Polycom"]?Set(ALERTINFO=Alert-Info: info=Ring Answer))

Then just ran a reload from cli.

On Polycom vvx series phones, though the web browser of the extension, I selected utilities ---> Imprt/Export ---> export (All device setting) and saved it to my desktop.
Opend the file in notepad, and added 3 lines:
In the reg directive, I added this line
reg.1.auth.password="xxxxxxxxxxxxxxx" (The password for that extension)
This is because there is comment, that the auth password is not exported. So it is important to include this line or else the phone will not be registered.
In the server directive, after port=5060 I added these 2 lines:
voIpProt.SIP.alertInfo.1.class="4"
voIpProt.SIP.alertInfo.1.value="Ring Answer"
Save the file with the same file name.
Then through the web browser, I selected the Import option (Select file). Wen to my desktop folder where the file was saved and clicked on "Import"
After the Import was successful, I clicked on --->utilities-->"restart phone"
That is all.
I did that for all the extensions.
After the phones restarted, paging and Intercom started working flawlessly.
Cheers!!!!!
P.S. Do Not eb=nable auto answer option in the phone through the web interface. Leave everything as default.
 
Last edited:

QueBall

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I am trying to figure out how to make this work myself but cannot crack it.

I have Polycom Soundpoint IP 501 phones and an AAstra 6757 phone.
All were setup using the OSS endpoint manager to generate the phone config files. From what I can see looking at those config files they already have the XML lines setup correctly so I don't think I need to do anything there.
in sip_318.cfg that all phones pull when booting:
<alertInfo voIpProt.SIP.alertInfo.1.value="Ring Answer" voIpProt.SIP.alertInfo.1.class="4"/>
and
<RING_ANSWER se.rt.4.name="Ring Answer" se.rt.4.type="ring-answer" se.rt.4.timeout="2000" se.rt.4.ringer="2" se.rt.4.callWait="6" se.rt.4.mod="1"/>

I changed the mysql table and the extensions_additional.conf to change "Auto Answer" to say "Ring Answer" and rebooted.
The Aastra intercom feature works great, no problems, the Polycom does not.

I turned on sip debugging on one of the Polycom units and captured this in the logs:
Code:
0825094444|sip  |0|01|<<<Packet Received
0825094444|sip  |0|01|    INVITE sip:[email protected];intercom=true SIP/2.0
0825094444|sip  |0|01|    Via: SIP/2.0/UDP 192.168.0.87:5060;branch=z9hG4bK5230c1b9;rport
0825094444|sip  |0|01|    Max-Forwards: 70
0825094444|sip  |0|01|    From: "Me" <sip:[email protected]>;tag=as430de771
0825094444|sip  |0|01|    To: <sip:[email protected];intercom=true>
0825094444|sip  |0|01|    Contact: <sip:[email protected]:5060>
0825094444|sip  |0|01|    Call-ID: [email protected]:5060
0825094444|sip  |0|01|    CSeq: 102 INVITE
0825094444|sip  |0|01|    User-Agent: FPBX-12.0.74(13.7.2)
0825094444|sip  |0|01|    Date: Thu, 25 Aug 2016 15:44:46 GMT
0825094444|sip  |0|01|    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
0825094444|sip  |0|01|    Supported: replaces, timer
0825094444|sip  |0|01|    Call-Info: <uri>;answer-after=0
0825094444|sip  |0|01|    Alert-Info: Ring Answer
0825094444|sip  |0|01|    Content-Type: application/sdp
0825094444|sip  |0|01|    Content-Length: 298
0825094444|sip  |0|01|
0825094444|sip  |0|01|    v=0
0825094444|sip  |0|01|    o=root 1459553215 1459553215 IN IP4 192.168.0.87
0825094444|sip  |0|01|    s=Asterisk PBX 13.7.2
So it is sending the new Alert-Info string as Ring Answer. The phone just does not pickup.
I see in the To: header is has intercom=true. I also see the Alert-Info: Ring Answer header is present.

Must have missed something. Anyone else have insights on what I can do?
 

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