Background:
I have a dozen or so Polycom soundpoint IP phones that are registered with an Asterisk server (Asterisk 1.6.2). I am trying to implement a way to simulate a legacy key system. Directed Call Pickup with the use of presence and hints will hopefully allow me to do this. When a call comes in for phone 1, I should be able to pickup the blinking line on phone 2. Also, When phone 2 puts that call on hold, I should be able to press the button next to the flashing red light on phone 1 and continue the conversation from that phone.
Working So Far:
I have presence working on the Polycoms and the hints work great. When a call comes in for phone 1, the corresponding attendant on phone 2 blinks green and the “pick up” soft key displays on the screen.
The Problem:
When I hit the “pick up” soft key during an incoming call, it doesn’t do anything and the following error displays in the Asterisk CLI:
[Mar 20 11:26:29] NOTICE[30413]: chan_sip.c:20582 handle_request_invite: Nothing to pick up for 894d3315-bab2041c-6d274007@192.168.1.107
[Mar 20 11:26:32] NOTICE[30413]: chan_sip.c:20582 handle_request_invite: Nothing to pick up for 12332203-80097552-78133265@192.168.1.107
[Mar 20 11:26:33] NOTICE[30413]: chan_sip.c:20582 handle_request_invite: Nothing to pick up for 6b5cfe81-a7282dd8-a2801ed3@192.168.1.107sip.conf:
[general]
callcounter=yes
Autocreatepeer=no
bindport=5060
bindaddr=0.0.0.0
context=sip ;default
maxexpiry=3600
subscribecontext=sip
allowsubscribe=yes
notifyringing=yes
notifyhold=yes
limitonpeer=yes
checkmwi=10
vmexten=voicemail
videosupport=yes
rtcachefriends=yes
disallow=all
allow=g722
allow=gsm
allow=ulaw
allow=alaw
allow=g729
allow=g723.1
allow=h263
allow=h263p
allow=h264
rtptimeout=60
dtmfmode=auto
canreinvite=no
auth=md5
qualify=yes
[phone1]
type=friend
username=phone1
secret=password
host=dynamic
dtmfmode=rfc2833
mailbox=0001
context=sip
subscribecontext=sip
canreinvite=no
nat=yes
realm=xxx.xxx.xxx.xxx
callerid="phone 1" <555-555-5555>
[phone2]
type=friend
username=phone2
secret=password
host=dynamic
dtmfmode=rfc2833
mailbox=0002
context=sip
subscribecontext=sip
canreinvite=no
nat=yes
realm=xxx.xxx.xxx.xxx
callerid="phone 2" <555-555-5555>extensions.conf:
[general]
static = yes
writeprotect = yes
autofallthrough = yes
clearglobalvars = no
priorityjumping = no
DYNAMIC_FEATURES = automon
[sip]
include => parkedcalls
exten => _*8.,1,SET(GLOBAL(PICKUPMARK)=${EXTEN:2})
exten => _*8.,n,Pickup(${EXTEN:2}@PICKUPMARK)
exten => 0001,hint,SIP/phone1
exten => 0001,1,Dial(SIP/phone1,20,r)
exten => 0001,2,VoiceMail(0000)
exten => 0002,hint,SIP/phone2
exten => 0002,1,Dial(SIP/phone2,20,r)
exten => 0002,2,VoiceMail(0000)Excerpt from -registration.cfg
<attendant
attendant.reg="1"
attendant.resourceList.1.address="sip:0001@192.168.1.6"
attendant.resourceList.1.label="Phone 1"
attendant.resourceList.2.address="sip:0002@192.168.1.6"
attendant.resourceList.1.label="Phone 2"
/>Many thanks to anyone with advice or ideas!
-Matt