Directed pickup then place on 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=z9hG4bKca036a7023a07250 From: ;tag=e801b9d7fe932bec To: ;tag=as7b6ef03b Contact: Replaces: pickup-8ee7cc384d9b55fd@192.168.5.129 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="7faafe58", response="41af609ec16ce7c163b73aaa53d48c94" Call-ID: 7362f43a2bbb5079@192.168.5.129 CSeq: 31596 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 5072 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 <-------------> --- (18 headers 16 lines) --- <--- Reliably Transmitting (no NAT) to 192.168.5.129:5066 ---> SIP/2.0 400 Bad request Via: SIP/2.0/UDP 192.168.5.129:5066;branch=z9hG4bKca036a7023a07250;received=192.168.5.129 From: ;tag=e801b9d7fe932bec To: ;tag=as7b6ef03b Call-ID: 7362f43a2bbb5079@192.168.5.129 CSeq: 31596 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 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 <------------> <--- 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=z9hG4bKca036a7023a07250 From: ;tag=e801b9d7fe932bec To: ;tag=as7b6ef03b Contact: Supported: path X-Grandstream-PBX: true Authorization: Digest username="GXP0013", realm="asterisk", algorithm=MD5, uri="sip:8612@192.168.5.41:5060", nonce="7faafe58", response="41af609ec16ce7c163b73aaa53d48c94" Call-ID: 7362f43a2bbb5079@192.168.5.129 CSeq: 31596 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) --- Retransmitting #1 (no NAT) to 192.168.5.129:5066: SIP/2.0 400 Bad request Via: SIP/2.0/UDP 192.168.5.129:5066;branch=z9hG4bKca036a7023a07250;received=192.168.5.129 From: ;tag=e801b9d7fe932bec To: ;tag=as7b6ef03b Call-ID: 7362f43a2bbb5079@192.168.5.129 CSeq: 31596 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 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- Retransmitting #2 (no NAT) to 192.168.5.129:5066: SIP/2.0 400 Bad request Via: SIP/2.0/UDP 192.168.5.129:5066;branch=z9hG4bKca036a7023a07250;received=192.168.5.129 From: ;tag=e801b9d7fe932bec To: ;tag=as7b6ef03b Call-ID: 7362f43a2bbb5079@192.168.5.129 CSeq: 31596 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 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- Retransmitting #3 (no NAT) to 192.168.5.129:5066: SIP/2.0 400 Bad request Via: SIP/2.0/UDP 192.168.5.129:5066;branch=z9hG4bKca036a7023a07250;received=192.168.5.129 From: ;tag=e801b9d7fe932bec To: ;tag=as7b6ef03b Call-ID: 7362f43a2bbb5079@192.168.5.129 CSeq: 31596 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 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- Retransmitting #4 (no NAT) to 192.168.5.129:5066: SIP/2.0 400 Bad request Via: SIP/2.0/UDP 192.168.5.129:5066;branch=z9hG4bKca036a7023a07250;received=192.168.5.129 From: ;tag=e801b9d7fe932bec To: ;tag=as7b6ef03b Call-ID: 7362f43a2bbb5079@192.168.5.129 CSeq: 31596 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 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- Retransmitting #5 (no NAT) to 192.168.5.129:5066: SIP/2.0 400 Bad request Via: SIP/2.0/UDP 192.168.5.129:5066;branch=z9hG4bKca036a7023a07250;received=192.168.5.129 From: ;tag=e801b9d7fe932bec To: ;tag=as7b6ef03b Call-ID: 7362f43a2bbb5079@192.168.5.129 CSeq: 31596 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 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- Retransmitting #6 (no NAT) to 192.168.5.129:5066: SIP/2.0 400 Bad request Via: SIP/2.0/UDP 192.168.5.129:5066;branch=z9hG4bKca036a7023a07250;received=192.168.5.129 From: ;tag=e801b9d7fe932bec To: ;tag=as7b6ef03b Call-ID: 7362f43a2bbb5079@192.168.5.129 CSeq: 31596 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 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- Retransmitting #7 (no NAT) to 192.168.5.129:5066: SIP/2.0 400 Bad request Via: SIP/2.0/UDP 192.168.5.129:5066;branch=z9hG4bKca036a7023a07250;received=192.168.5.129 From: ;tag=e801b9d7fe932bec To: ;tag=as7b6ef03b Call-ID: 7362f43a2bbb5079@192.168.5.129 CSeq: 31596 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 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- -- Channel 0/1, span 1 got hangup request, cause 16 -- Executing [h@incoming:1] Verbose("DAHDI/i1/5604866-6", "0,NAME= NUM='5604866' SUBADDR='' DNID='8612' DNIDSUBADDR='' RDNIS=''") in new stack NAME= NUM='5604866' SUBADDR='' DNID='8612' DNIDSUBADDR='' RDNIS='' Scheduling destruction of SIP dialog '7362f43a2bbb5079@192.168.5.129' in 32000 ms (Method: INVITE) set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.5.129:5066 Reliably Transmitting (no NAT) to 192.168.5.129:5066: BYE sip:GXP0013@192.168.5.129:5066;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.5.41:5060;branch=z9hG4bK60d9e41f Max-Forwards: 70 From: ;tag=as7b6ef03b To: ;tag=e801b9d7fe932bec Call-ID: 7362f43a2bbb5079@192.168.5.129 CSeq: 102 BYE User-Agent: Asterisk PBX SVN-trunk-r281257 Proxy-Authorization: Digest username="GXP0013", realm="asterisk", algorithm=MD5, uri="192.168.5.41", nonce="", response="ddf3a5d044c6920c2c12f0e8ee2d09ec" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- == Spawn extension (incoming, 8612, 1) exited non-zero on 'DAHDI/i1/5604866-6' -- Hungup 'DAHDI/i1/5604866-6' == Extension Changed 8626[trusted] new state Idle for Notify User GXP0001 <--- SIP read from UDP:192.168.5.129:5066 ---> SIP/2.0 481 No Such Call Via: SIP/2.0/UDP 192.168.5.41:5060;branch=z9hG4bK60d9e41f From: ;tag=as7b6ef03b To: ;tag=e801b9d7fe932bec Call-ID: 7362f43a2bbb5079@192.168.5.129 CSeq: 102 BYE User-Agent: Grandstream GXP2010 1.2.3.5 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Retransmitting #8 (no NAT) to 192.168.5.129:5066: SIP/2.0 400 Bad request Via: SIP/2.0/UDP 192.168.5.129:5066;branch=z9hG4bKca036a7023a07250;received=192.168.5.129 From: ;tag=e801b9d7fe932bec To: ;tag=as7b6ef03b Call-ID: 7362f43a2bbb5079@192.168.5.129 CSeq: 31596 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 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- test*CLI>