[Asterisk-Users] Polycom 500 "Group Call Pickup Feature" and *

[Chris Coulthurst]

If you activate (via sip.cfg) the feature Group Call Pickup, its no
surprise that asterisk doesn't know what to do with this feature
request.  But it is being sent as a SIP SUBSCRIBE request, and I'm
wondering if, as asterisk stands, there is a way to take advantage of
this feature to emulate the "*8#" normal behavior.


If anyone has any input, there is also a call parking function that I
think is SIP SUBSCRIBE-based.


Here is the 'sip debug' snippet from when I pressed the New Call ->
Pickup -> Group softkeys:


Sip read: 
SUBSCRIBE sip:groupcallpickup at 192.168.0.9:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.234;branch=z9hG4bKa58a6cc24AEA0129
From: "Chris Office" <sip:201 at 192.168.0.9>;tag=569A308-31C12E4D
To: <sip:groupcallpickup at 192.168.0.9>
CSeq: 1 SUBSCRIBE
Call-ID: d4b32c74-68b2cfb6-70db113 at 192.168.0.234
Contact: <sip:201 at 192.168.0.234>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE,
NOTIFY, PRACK, UPDATE, REFER
Event: dialog
User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.5.2.0054
Accept: application/dialog-info+xml
Max-Forwards: 70
Expires: 0
Content-Length: 0


14 headers, 0 lines
Using latest SUBSCRIBE request as basis request
Sending to 192.168.0.234 : 5060 (non-NAT)
Found peer '201'
Transmitting (no NAT):
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.0.234;branch=z9hG4bKa58a6cc24AEA0129
From: "Chris Office" <sip:201 at 192.168.0.9>;tag=569A308-31C12E4D
To: <sip:groupcallpickup at 192.168.0.9>;tag=as1b873db6
Call-ID: d4b32c74-68b2cfb6-70db113 at 192.168.0.234
CSeq: 1 SUBSCRIBE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:groupcallpickup at 192.168.0.9>
Proxy-Authenticate: Digest realm="asterisk", nonce="5041eff0"
Content-Length: 0


 to 192.168.0.234:5060
Scheduling destruction of call 'd4b32c74-68b2cfb6-70db113 at 192.168.0.234'
in 15000 ms
morse*CLI> 

Sip read: 
SUBSCRIBE sip:groupcallpickup at 192.168.0.9:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.234;branch=z9hG4bK802f53579213D6EA
From: "Chris Office" <sip:201 at 192.168.0.9>;tag=569A308-31C12E4D
To: <sip:groupcallpickup at 192.168.0.9>
CSeq: 2 SUBSCRIBE
Call-ID: d4b32c74-68b2cfb6-70db113 at 192.168.0.234
Contact: <sip:201 at 192.168.0.234>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE,
NOTIFY, PRACK, UPDATE, REFER
Event: dialog
User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.5.2.0054
Accept: application/dialog-info+xml
Proxy-Authorization: Digest username="201", realm="asterisk",
nonce="5041eff0", uri="sip:groupcallpickup at 192.168.0.9:5060",
response="b48b989d85958a6ce18c9431058ce6f3", algorithm=MD5
Max-Forwards: 70
Expires: 0
Content-Length: 0


15 headers, 0 lines
Using latest SUBSCRIBE request as basis request
Sending to 192.168.0.234 : 5060 (non-NAT)
Found peer '201'
Looking for groupcallpickup in default
Transmitting (no NAT):
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.0.234;branch=z9hG4bK802f53579213D6EA
From: "Chris Office" <sip:201 at 192.168.0.9>;tag=569A308-31C12E4D
To: <sip:groupcallpickup at 192.168.0.9>;tag=as1b873db6
Call-ID: d4b32c74-68b2cfb6-70db113 at 192.168.0.234
CSeq: 2 SUBSCRIBE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:groupcallpickup at 192.168.0.9>
Content-Length: 0


 to 192.168.0.234:5060
Destroying call 'd4b32c74-68b2cfb6-70db113 at 192.168.0.234'
morse*CLI> sip no debug
SIP Debugging Disabled

Chris Coulthurst
chris at shuksan.com