rendered paste bodyContact: <sip:1153@10.54.2.131>
Call-ID: 2e97ea0367795ce9712d92d478829693@10.54.2.131
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
loca*CLI>
<-- SIP read from 10.54.2.123:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.54.2.131:5060;branch=z9hG4bK36c0c8bf;rport
From: <sip:1141@10.54.2.131>;tag=as3732ef65
To: "Chris" <sip:1153@10.54.2.131>;tag=f1436e8d465ab9b7
Call-ID: 22ca4ebc8683b387@10.54.2.123
CSeq: 102 INVITE
User-Agent: Grandstream GXP2000 1.1.0.14
Contact: <sip:1153@10.54.2.123:5060>
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Content-Type: application/sdp
Supported: replaces, timer
Content-Length: 153
v=0
o=1153 8001 8002 IN IP4 10.54.2.123
s=SIP Call
c=IN IP4 10.54.2.123
t=0 0
m=audio 5006 RTP/AVP 0
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ptime:20
--- (12 headers 9 lines)---
Found RTP audio format 0
Peer audio RTP is at port 10.54.2.123:5006
Found description format PCMU
Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
list_route: hop: <sip:1153@10.54.2.123:5060>
set_destination: Parsing <sip:1153@10.54.2.123:5060> for address/port to send to
set_destination: set destination to 10.54.2.123, port 5060
Transmitting (no NAT) to 10.54.2.123:5060:
ACK sip:1153@10.54.2.123:5060 SIP/2.0
Via: SIP/2.0/UDP 10.54.2.131:5060;branch=z9hG4bK086e4687;rport
From: <sip:1141@10.54.2.131>;tag=as3732ef65
To: "Chris" <sip:1153@10.54.2.131>;tag=f1436e8d465ab9b7
Contact: <sip:1141@10.54.2.131>
Call-ID: 22ca4ebc8683b387@10.54.2.123
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
loca*CLI>
<-- SIP read from 10.54.2.123:5060:
REFER sip:7809694975@10.54.2.131 SIP/2.0
Via: SIP/2.0/UDP 10.54.2.123:5060;branch=z9hG4bK2b4ed58646a7b279
From: <sip:1153@10.54.2.123:5060>;tag=e4508aa8e6ea244c
To: "Admin Asst Back" <sip:7809694975@10.54.2.131>;tag=as5ad38f51
Contact: <sip:1153@10.54.2.123:5060>
Supported: replaces
Refer-To: <sip:1141@10.54.2.131?Replaces=22ca4ebc8683b387%4010.54.2.123%3Bfrom-tag%3Df1436e8d465ab9b7%3Bto-tag%3Das3732ef65>
Referred-By: <sip:1153@10.54.2.131>
Call-ID: 5c90b01760f1a8b61819d5d37934d79e@10.54.2.131
CSeq: 27454 REFER
User-Agent: Grandstream GXP2000 1.1.0.14
Proxy-Require: 10.54.2.131
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Content-Length: 0
--- (15 headers 0 lines)---
Transfer to 1141 in internal
Transfer from 1153 in internal
-- Stopped music on hold on Zap/1-1
Transmitting (no NAT) to 10.54.2.123:5060:
SIP/2.0 202 Accepted
Via: SIP/2.0/UDP 10.54.2.123:5060;branch=z9hG4bK2b4ed58646a7b279;received=10.54.2.123
From: <sip:1153@10.54.2.123:5060>;tag=e4508aa8e6ea244c
To: "Admin Asst Back" <sip:7809694975@10.54.2.131>;tag=as5ad38f51
Call-ID: 5c90b01760f1a8b61819d5d37934d79e@10.54.2.131
CSeq: 27454 REFER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:7809694975@10.54.2.131>
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing
---
set_destination: Parsing <sip:1153@10.54.2.123:5060> for address/port to send to
set_destination: set destination to 10.54.2.123, port 5060
Reliably Transmitting (no NAT) to 10.54.2.123:5060:
NOTIFY sip:1153@10.54.2.123:5060 SIP/2.0
Via: SIP/2.0/UDP 10.54.2.131:5060;branch=z9hG4bK5ca5174a;rport
From: "Admin Asst Back" <sip:7809694975@10.54.2.131>;tag=as5ad38f51
To: <sip:1153@10.54.2.123:5060>;tag=e4508aa8e6ea244c
Contact: <sip:7809694975@10.54.2.131>
Call-ID: 5c90b01760f1a8b61819d5d37934d79e@10.54.2.131
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: refer;id=27454
Subscription-state: terminated;reason=noresource
Content-Type: message/sipfrag;version=2.0
Content-Length: 14
SIP/2.0 200 OK
---
set_destination: Parsing <sip:1153@10.54.2.123:5060> for address/port to send to
set_destination: set destination to 10.54.2.123, port 5060
Reliably Transmitting (no NAT) to 10.54.2.123:5060:
BYE sip:1153@10.54.2.123:5060 SIP/2.0
Via: SIP/2.0/UDP 10.54.2.131:5060;branch=z9hG4bK01f522d2;rport
From: "Admin Asst Back" <sip:7809694975@10.54.2.131>;tag=as5ad38f51
To: <sip:1153@10.54.2.123:5060>;tag=e4508aa8e6ea244c
Contact: <sip:7809694975@10.54.2.131>
Call-ID: 5c90b01760f1a8b61819d5d37934d79e@10.54.2.131
CSeq: 104 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
X-Asterisk-HangupCause: Normal Clearing
Content-Length: 0
---
set_destination: Parsing <sip:1153@10.54.2.123:5060> for address/port to send to
set_destination: set destination to 10.54.2.123, port 5060
Reliably Transmitting (no NAT) to 10.54.2.123:5060:
BYE sip:1153@10.54.2.123:5060 SIP/2.0
Via: SIP/2.0/UDP 10.54.2.131:5060;branch=z9hG4bK1dbf02e8;rport
From: <sip:1141@10.54.2.131>;tag=as3732ef65
To: "Chris" <sip:1153@10.54.2.131>;tag=f1436e8d465ab9b7
Contact: <sip:1141@10.54.2.131>
Call-ID: 22ca4ebc8683b387@10.54.2.123
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
== Spawn extension (incoming, 650137, 2) exited non-zero on 'SIP/1153-b481bcb0<ZOMBIE>'
loca*CLI>
<-- SIP read from 10.54.2.123:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.54.2.131:5060;branch=z9hG4bK5ca5174a;rport
From: "Admin Asst Back" <sip:7809694975@10.54.2.131>;tag=as5ad38f51
To: <sip:1153@10.54.2.123:5060>;tag=e4508aa8e6ea244c
Call-ID: 5c90b01760f1a8b61819d5d37934d79e@10.54.2.131
CSeq: 103 NOTIFY
User-Agent: Grandstream GXP2000 1.1.0.14
Contact: <sip:1153@10.54.2.123:5060>
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Supported: replaces, timer
Content-Length: 0
--- (11 headers 0 lines)---
loca*CLI>
<-- SIP read from 10.54.2.123:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.54.2.131:5060;branch=z9hG4bK01f522d2;rport
From: "Admin Asst Back" <sip:7809694975@10.54.2.131>;tag=as5ad38f51
To: <sip:1153@10.54.2.123:5060>;tag=e4508aa8e6ea244c
Call-ID: 5c90b01760f1a8b61819d5d37934d79e@10.54.2.131
CSeq: 104 BYE
User-Agent: Grandstream GXP2000 1.1.0.14
Contact: <sip:1153@10.54.2.123:5060>
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Supported: replaces, timer
Content-Length: 0
--- (11 headers 0 lines)---
Destroying call '5c90b01760f1a8b61819d5d37934d79e@10.54.2.131'
loca*CLI>
<-- SIP read from 10.54.2.123:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.54.2.131:5060;branch=z9hG4bK1dbf02e8;rport
From: <sip:1141@10.54.2.131>;tag=as3732ef65
To: "Chris" <sip:1153@10.54.2.131>;tag=f1436e8d465ab9b7
Call-ID: 22ca4ebc8683b387@10.54.2.123
CSeq: 103 BYE
User-Agent: Grandstream GXP2000 1.1.0.14
Contact: <sip:1153@10.54.2.123:5060>
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Supported: replaces, timer
Content-Length: 0
--- (11 headers 0 lines)---
Destroying call '22ca4ebc8683b387@10.54.2.123'
loca*CLI>
<-- SIP read from 10.54.2.123:5060:
--- (0 headers 0 lines) Nat keepalive ---
loca*CLI>
<-- SIP read from 10.54.2.105:5060:
--- (0 headers 0 lines) Nat keepalive ---
-- Channel 0/1, span 1 got hangup request
set_destination: Parsing <sip:1141@10.54.2.105:5060> for address/port to send to
set_destination: set destination to 10.54.2.105, port 5060
Reliably Transmitting (no NAT) to 10.54.2.105:5060:
BYE sip:1141@10.54.2.105:5060 SIP/2.0
Via: SIP/2.0/UDP 10.54.2.131:5060;branch=z9hG4bK468b297e;rport
From: "Back Office" <sip:1153@10.54.2.131>;tag=as1ea073c4
To: <sip:1141@10.54.2.105:5060>;tag=33c82fbcea293afa
Contact: <sip:1153@10.54.2.131>
Call-ID: 2e97ea0367795ce9712d92d478829693@10.54.2.131
CSeq: 104 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
== Spawn extension (macro-extensions, s, 1) exited non-zero on 'Zap/1-1' in macro