Directed pickup using **8612 then hold

test*CLI> hold
No such command 'hold' (type 'core show help hold' for other possible commands)

<--- SIP read from UDP:192.168.5.129:5066 --->
INVITE sip:**8612@192.168.5.41:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.5.129:5066;branch=z9hG4bK5a7b1510f417a613
From: <sip:GXP0013@192.168.5.41>;tag=ae6af6d8bcf74147
To: <sip:**8612@192.168.5.41>;tag=as3ce7e9fa
Contact: <sip:GXP0013@192.168.5.129:5066;transport=udp>
Supported: replaces, timer, path
X-Grandstream-PBX: true
P-Early-Media: Supported
Session-Expires: 1800;refresher=uas
Authorization: Digest username="GXP0013", realm="asterisk", algorithm=MD5, uri="sip:**8612@192.168.5.41:5060", nonce="77ec65f4", response="3bf5cc60ae31b654055e0636a4868c63"
Call-ID: 5afa39d4015496f9@192.168.5.129
CSeq: 34510 INVITE
User-Agent: Grandstream GXP2010 1.2.3.5
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Content-Type: application/sdp
Content-Length: 323

v=0
o=GXP0013 8000 8002 IN IP4 192.168.5.129
s=SIP Call
c=IN IP4 0.0.0.0
t=0 0
m=audio 5096 RTP/AVP 8 4 18 2 97 9 3
a=sendonly
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=rtpmap:9 G722/8000
a=rtpmap:3 GSM/8000
a=ptime:20

<------------->
--- (17 headers 16 lines) ---
Sending to 192.168.5.129:5066 (no NAT)
Found RTP audio format 8
Found RTP audio format 4
Found RTP audio format 18
Found RTP audio format 2
Found RTP audio format 97
Found RTP audio format 9
Found RTP audio format 3
Found audio description format PCMA for ID 8
Found audio description format G723 for ID 4
Found audio description format G729 for ID 18
Found audio description format G726-32 for ID 2
Found audio description format iLBC for ID 97
Found audio description format G722 for ID 9
Found audio description format GSM for ID 3
Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x1d0b (g723|gsm|alaw|g726|g729|ilbc|g722)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xa (gsm|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 0.0.0.0:5096

<--- Transmitting (no NAT) to 192.168.5.129:5066 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.5.129:5066;branch=z9hG4bK5a7b1510f417a613;received=192.168.5.129
From: <sip:GXP0013@192.168.5.41>;tag=ae6af6d8bcf74147
To: <sip:**8612@192.168.5.41>;tag=as3ce7e9fa
Call-ID: 5afa39d4015496f9@192.168.5.129
CSeq: 34510 INVITE
Server: Asterisk PBX SVN-trunk-r281257
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Require: timer
Session-Expires: 1800;refresher=uas
Contact: <sip:**8612@192.168.5.41:5060>
Content-Length: 0


<------------>
Audio is at 5060
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP

<--- Reliably Transmitting (no NAT) to 192.168.5.129:5066 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.5.129:5066;branch=z9hG4bK5a7b1510f417a613;received=192.168.5.129
From: <sip:GXP0013@192.168.5.41>;tag=ae6af6d8bcf74147
To: <sip:**8612@192.168.5.41>;tag=as3ce7e9fa
Call-ID: 5afa39d4015496f9@192.168.5.129
CSeq: 34510 INVITE
Server: Asterisk PBX SVN-trunk-r281257
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Require: timer
Session-Expires: 1800;refresher=uas
Contact: <sip:**8612@192.168.5.41:5060>
Content-Type: application/sdp
Content-Length: 211

v=0
o=root 211929038 211929039 IN IP4 192.168.5.41
s=Asterisk PBX SVN-trunk-r281257
c=IN IP4 192.168.5.41
t=0 0
m=audio 13190 RTP/AVP 8 3
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=ptime:20
a=recvonly

<------------>
    -- Started music on hold, class 'default', on DAHDI/i1/5604866-8

<--- SIP read from UDP:192.168.5.129:5066 --->
ACK sip:**8612@192.168.5.41:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.5.129:5066;branch=z9hG4bK0cbafb383b00a281
From: <sip:GXP0013@192.168.5.41>;tag=ae6af6d8bcf74147
To: <sip:**8612@192.168.5.41>;tag=as3ce7e9fa
Contact: <sip:GXP0013@192.168.5.129:5066;transport=udp>
Supported: path
X-Grandstream-PBX: true
Authorization: Digest username="GXP0013", realm="asterisk", algorithm=MD5, uri="sip:**8612@192.168.5.41:5060", nonce="77ec65f4", response="3bf5cc60ae31b654055e0636a4868c63"
Call-ID: 5afa39d4015496f9@192.168.5.129
CSeq: 34510 ACK
User-Agent: Grandstream GXP2010 1.2.3.5
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Content-Length: 0


<------------->
--- (14 headers 0 lines) ---
test*CLI>