Asterisk ACD and Polycom Phones

Asterisk Asterisk Support
1

I am using Asterisk 1.8 and Polycom 650 phones and trying to get the Agent Login/Logout function working.
Using firmware 3.3.4 using the Admin Guide I was able to get the ASignIn softkey on the phone but the functionality is not there. I can log in/out using the agent callback extension but I can not go unavailable using Menu, 1, 6.

I am getting a 489 from Asterisk. Has anyone gotten this to work?

Scheduling destruction of SIP dialog ‘b90e9270-2c3732d-8bb60afa@192.168.50.201’ in 32000 ms (Method: SUBSCRIBE)

<— SIP read from UDP:192.168.50.201:5060 —>
SUBSCRIBE sip:301@192.168.50.10:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.50.201;branch=z9hG4bKadd64ed7DB2CC3C4
From: “301” sip:301@192.168.50.10;tag=CA8C21C-6A421899
To: sip:301@192.168.50.10:5060
CSeq: 2 SUBSCRIBE
Call-ID: b90e9270-2c3732d-8bb60afa@192.168.50.201
Contact: sip:301@192.168.50.201
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: as-feature-event
User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.3.4.0085
Accept-Language: en
Accept: application/x-as-feature-event+xml
Authorization: Digest username=“301”, realm=“asterisk”, nonce=“1ad9f855”, uri=“sip:301@192.168.50.10:5060”, response=“57a6a25a12f1bf40f031cb38430dc575”, algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0

<------------->
— (16 headers 0 lines) —
Creating new subscription
Sending to 192.168.50.201:5060 (NAT)
Found peer ‘301’ for ‘301’ from 192.168.50.201:5060
Looking for 301 in default (domain 192.168.50.10)

<— Transmitting (NAT) to 192.168.50.201:5060 —>
SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP 192.168.50.201;branch=z9hG4bKadd64ed7DB2CC3C4;received=192.168.50.201;rport=5060
From: “301” sip:301@192.168.50.10;tag=CA8C21C-6A421899
To: sip:301@192.168.50.10:5060;tag=as015cf7a3
Call-ID: b90e9270-2c3732d-8bb60afa@192.168.50.201
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX 1.8.9.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

2

Why are you including a trace for a SUBSCRIBE, i.e. the phone asking to be informed of events, when you are asking about giving a command?

As far as I know Asterisk only supports simple presence, message waiting, and under the hood, transfer progress, with subscribe.

Any command to make the agent unavailable from an agent login would have to be done with DTMF, or by re-invites to hold. I’m not sure if the former is possible without help from AMI, and I’m not sure if hold is factored in to the queue scheduling.

3

Hello drreim,

the Polycom Admin Guide clearly states:

The SIP signaling used for this implementation is described in the Device Key Synchronization Requirements Document;
Release R14 sp2; Document version 1.6. Contact Polycom Product Management for more information.

Above means that this feature was implemented for a Polycom Partner and in your Case Asterisk is simply not aware of the requirements in signalling.

Please work via your Polycom Reseller.

Best Regards

Steffen Baier