Problem with incoming call

i’m working by this scheme:
PSTN-> dlink DVG-6004s -> asterisk 11.3.0 -> Cisco Unified IP Phone 7965G

Outgoing call works greate, but i have problem with incoming call, may be someone will help.

dlink recives incoming call from PSTN and sends notification to asterisk. Asterisk recives it and sends call on extension, but only on a few seconds, only two bees. Then call drops and after a few seconds it comes again on less then a second (on cisco ip phone) and caller hears normal beebs.

here is log with enabled debug of dlink ip address:
*asterisk - 192.168.1.80
*dlink - 192.168.1.73

[quote]<------------->
— (8 headers 0 lines) —

<— SIP read from UDP:192.168.1.73:5060 —>
INVITE sip:100@192.168.1.80:5060;user=phone SIP/2.0
Allow:INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,REFER,SUBSCRIBE,NOTIFY,UPDATE
Via:SIP/2.0/UDP 192.168.1.73:5060;branch=z9hG4bK5b4530c763a15de3
From: “Anonymous” sip:1001@192.168.1.80;user=phone;tag=3222c599-686617
To: sip:100@192.168.1.80:5060;user=phone
Call-ID:D1B9-125A-46686617963977A1F695-023@SipHost
CSeq:26 INVITE
Contact:sip:1001@192.168.1.73:5060
Expires:90
Max-Forwards:70
Authorization:Digest username=“1001”,realm=“asterisk”,nonce=“5914869b”,uri=“sip:100@192.168.1.80:5060;user=phone”,response=“7ae4685e0d4a7495e4dabea8edca28f6”,algorithm=MD5
Supported:replaces
User-Agent:dlink 12-37-61928258-0.9.5.1.735
Content-Type:application/sdp
Content-Length:257

v=0
o=1001 1793812580 1793812580 IN IP4 192.168.1.73
s=Session SDP
c=IN IP4 192.168.1.73
t=0 0
m=audio 9002 RTP/AVP 2 4 18 0 8
a=rtpmap:2 G726-32/8000/1
a=rtpmap:4 G723/8000/1
a=rtpmap:18 G729/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
<------------->
— (15 headers 11 lines) —
Sending to 192.168.1.73:5060 (NAT)
Using INVITE request as basis request - D1B9-125A-46686617963977A1F695-023@SipHost
Found peer ‘1001’ for ‘1001’ from 192.168.1.73:5060
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Found RTP audio format 2
Found RTP audio format 4
Found RTP audio format 18
Found RTP audio format 0
Found RTP audio format 8
Found audio description format G726-32 for ID 2
Found audio description format G723 for ID 4
Found audio description format G729 for ID 18
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Capabilities: us - (ulaw|alaw), peer - audio=(g723|ulaw|alaw|g726|g729)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 192.168.1.73:9002
Looking for 100 in from-internal (domain 192.168.1.80)
list_route: hop: sip:1001@192.168.1.73:5060

<— Transmitting (NAT) to 192.168.1.73:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.73:5060;branch=z9hG4bK5b4530c763a15de3;received=192.168.1.73;rport=5060
From: “Anonymous” sip:1001@192.168.1.80;user=phone;tag=3222c599-686617
To: sip:100@192.168.1.80:5060;user=phone
Call-ID: D1B9-125A-46686617963977A1F695-023@SipHost
CSeq: 26 INVITE
Server: FPBX-2.11.0(11.3.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:100@192.168.1.80:5060
Content-Length: 0

<------------>
– Executing [100@from-internal:1] Set(“SIP/1001-000002ec”, “__RINGTIMER=15”) in new stack
– Executing [100@from-internal:2] Macro(“SIP/1001-000002ec”, “exten-vm,novm,100,0,0,0”) in new stack
– Executing [s@macro-exten-vm:1] Macro(“SIP/1001-000002ec”, “user-callerid,”) in new stack
– Executing [s@macro-user-callerid:1] Set(“SIP/1001-000002ec”, “TOUCH_MONITOR=1372261323.802”) in new stack
– Executing [s@macro-user-callerid:2] Set(“SIP/1001-000002ec”, “AMPUSER=1001”) in new stack
– Executing [s@macro-user-callerid:3] GotoIf(“SIP/1001-000002ec”, “0?report”) in new stack
– Executing [s@macro-user-callerid:4] ExecIf(“SIP/1001-000002ec”, “1?Set(REALCALLERIDNUM=1001)”) in new stack
– Executing [s@macro-user-callerid:5] Set(“SIP/1001-000002ec”, “AMPUSER=1001”) in new stack
– Executing [s@macro-user-callerid:6] Set(“SIP/1001-000002ec”, “AMPUSERCIDNAME=1001”) in new stack
– Executing [s@macro-user-callerid:7] GotoIf(“SIP/1001-000002ec”, “0?report”) in new stack
– Executing [s@macro-user-callerid:8] Set(“SIP/1001-000002ec”, “AMPUSERCID=1001”) in new stack
– Executing [s@macro-user-callerid:9] Set(“SIP/1001-000002ec”, “__DIAL_OPTIONS=Ttr”) in new stack
– Executing [s@macro-user-callerid:10] Set(“SIP/1001-000002ec”, “CALLERID(all)=“1001” <1001>”) in new stack
– Executing [s@macro-user-callerid:11] GotoIf(“SIP/1001-000002ec”, “0?limit”) in new stack
– Executing [s@macro-user-callerid:12] ExecIf(“SIP/1001-000002ec”, “0?Set(GROUP(concurrency_limit)=1001)”) in new stack
– Executing [s@macro-user-callerid:13] GosubIf(“SIP/1001-000002ec”, “7?sub-ccss,s,1(macro-exten-vm,100)”) in new stack
– Executing [s@sub-ccss:1] ExecIf(“SIP/1001-000002ec”, “0?Return()”) in new stack
– Executing [s@sub-ccss:2] Set(“SIP/1001-000002ec”, “CCSS_SETUP=TRUE”) in new stack
– Executing [s@sub-ccss:3] GosubIf(“SIP/1001-000002ec”, “0?monitor_config,1(macro-exten-vm,100):monitor_default,1(macro-exten-vm,100)”) in new stack
– Executing [monitor_default@sub-ccss:1] GotoIf(“SIP/1001-000002ec”, “1?is_exten”) in new stack
– Goto (sub-ccss,monitor_default,4)
– Executing [monitor_default@sub-ccss:4] Set(“SIP/1001-000002ec”, “CALLCOMPLETION(cc_monitor_policy)=generic”) in new stack
– Executing [monitor_default@sub-ccss:5] Set(“SIP/1001-000002ec”, “CALLCOMPLETION(cc_max_monitors)=5”) in new stack
– Executing [monitor_default@sub-ccss:6] Return(“SIP/1001-000002ec”, “TRUE”) in new stack
– Executing [s@sub-ccss:4] GosubIf(“SIP/1001-000002ec”, “7?agent_config,1():agent_default,1()”) in new stack
– Executing [agent_config@sub-ccss:1] Set(“SIP/1001-000002ec”, “CALLCOMPLETION(cc_agent_policy)=generic”) in new stack
– Executing [agent_config@sub-ccss:2] Set(“SIP/1001-000002ec”, “CALLCOMPLETION(cc_offer_timer)=30”) in new stack
– Executing [agent_config@sub-ccss:3] Set(“SIP/1001-000002ec”, “CALLCOMPLETION(ccbs_available_timer)=”) in new stack
– Executing [agent_config@sub-ccss:4] Set(“SIP/1001-000002ec”, “CALLCOMPLETION(ccnr_available_timer)=”) in new stack
– Executing [agent_config@sub-ccss:5] Set(“SIP/1001-000002ec”, “CALLCOMPLETION(cc_callback_macro)=ccss-default”) in new stack
[Jun 26 19:42:03] WARNING[16347][C-00000189]: ccss.c:1000 ast_set_cc_callback_macro: Usage of cc_callback_macro is deprecated. Please use cc_callback_sub instead.
– Executing [agent_config@sub-ccss:6] ExecIf(“SIP/1001-000002ec”, “1?Set(CALLCOMPLETION(cc_recall_timer)=)”) in new stack
– Executing [agent_config@sub-ccss:7] ExecIf(“SIP/1001-000002ec”, “1?Set(CALLCOMPLETION(cc_max_agents)=)”) in new stack
– Executing [agent_config@sub-ccss:8] ExecIf(“SIP/1001-000002ec”, “0?Set(CALLCOMPLETION(cc_agent_dialstring)=Local/1001_100@from-ccss-)”) in new stack
– Executing [agent_config@sub-ccss:9] Set(“SIP/1001-000002ec”, “CALLCOMPLETION(cc_callback_macro)=ccss-default”) in new stack
[Jun 26 19:42:03] WARNING[16347][C-00000189]: ccss.c:1000 ast_set_cc_callback_macro: Usage of cc_callback_macro is deprecated. Please use cc_callback_sub instead.
– Executing [agent_config@sub-ccss:10] Return(“SIP/1001-000002ec”, “”) in new stack
– Executing [s@sub-ccss:5] Set(“SIP/1001-000002ec”, “DB(AMPUSER/1001/ccss/last_number)=100”) in new stack
– Executing [s@sub-ccss:6] Return(“SIP/1001-000002ec”, “”) in new stack
– Executing [s@macro-user-callerid:14] ExecIf(“SIP/1001-000002ec”, “0?Set(CHANNEL(language)=)”) in new stack
– Executing [s@macro-user-callerid:15] GotoIf(“SIP/1001-000002ec”, “0?continue”) in new stack
– Executing [s@macro-user-callerid:16] Set(“SIP/1001-000002ec”, “__TTL=64”) in new stack
– Executing [s@macro-user-callerid:17] GotoIf(“SIP/1001-000002ec”, “1?continue”) in new stack
– Goto (macro-user-callerid,s,28)
– Executing [s@macro-user-callerid:28] Set(“SIP/1001-000002ec”, “CALLERID(number)=1001”) in new stack
– Executing [s@macro-user-callerid:29] Set(“SIP/1001-000002ec”, “CALLERID(name)=1001”) in new stack
– Executing [s@macro-user-callerid:30] Set(“SIP/1001-000002ec”, “CDR(cnum)=1001”) in new stack
– Executing [s@macro-user-callerid:31] Set(“SIP/1001-000002ec”, “CDR(cnam)=1001”) in new stack
– Executing [s@macro-user-callerid:32] Set(“SIP/1001-000002ec”, “CHANNEL(language)=ru”) in new stack
– Executing [s@macro-exten-vm:2] Set(“SIP/1001-000002ec”, “RingGroupMethod=none”) in new stack
– Executing [s@macro-exten-vm:3] Set(“SIP/1001-000002ec”, “__EXTTOCALL=100”) in new stack
– Executing [s@macro-exten-vm:4] Set(“SIP/1001-000002ec”, “__PICKUPMARK=100”) in new stack
– Executing [s@macro-exten-vm:5] Set(“SIP/1001-000002ec”, “RT=15”) in new stack
– Executing [s@macro-exten-vm:6] Gosub(“SIP/1001-000002ec”, “sub-record-check,s,1(exten,100,)”) in new stack
– Executing [s@sub-record-check:1] Set(“SIP/1001-000002ec”, “REC_POLICY_MODE_SAVE=”) in new stack
– Executing [s@sub-record-check:2] GotoIf(“SIP/1001-000002ec”, “1?check”) in new stack
– Goto (sub-record-check,s,7)
– Executing [s@sub-record-check:7] Set(“SIP/1001-000002ec”, “__MON_FMT=wav”) in new stack
– Executing [s@sub-record-check:8] GotoIf(“SIP/1001-000002ec”, “1?next”) in new stack
– Goto (sub-record-check,s,11)
– Executing [s@sub-record-check:11] ExecIf(“SIP/1001-000002ec”, “0?Return()”) in new stack
– Executing [s@sub-record-check:12] ExecIf(“SIP/1001-000002ec”, “0?Set(__REC_POLICY_MODE=)”) in new stack
– Executing [s@sub-record-check:13] GotoIf(“SIP/1001-000002ec”, “0?exten,1”) in new stack
– Executing [s@sub-record-check:14] Set(“SIP/1001-000002ec”, “__REC_STATUS=INITIALIZED”) in new stack
– Executing [s@sub-record-check:15] Set(“SIP/1001-000002ec”, “NOW=1372261323”) in new stack
– Executing [s@sub-record-check:16] Set(“SIP/1001-000002ec”, “__DAY=26”) in new stack
– Executing [s@sub-record-check:17] Set(“SIP/1001-000002ec”, “__MONTH=06”) in new stack
– Executing [s@sub-record-check:18] Set(“SIP/1001-000002ec”, “__YEAR=2013”) in new stack
– Executing [s@sub-record-check:19] Set(“SIP/1001-000002ec”, “__TIMESTR=20130626-194203”) in new stack
– Executing [s@sub-record-check:20] Set(“SIP/1001-000002ec”, “__FROMEXTEN=1001”) in new stack
– Executing [s@sub-record-check:21] Set(“SIP/1001-000002ec”, “__CALLFILENAME=exten-100-1001-20130626-194203-1372261323.802”) in new stack
– Executing [s@sub-record-check:22] Goto(“SIP/1001-000002ec”, “exten,1”) in new stack
– Goto (sub-record-check,exten,1)
– Executing [exten@sub-record-check:1] GotoIf(“SIP/1001-000002ec”, “0?callee”) in new stack
– Executing [exten@sub-record-check:2] Set(“SIP/1001-000002ec”, “__REC_POLICY_MODE=dontcare”) in new stack
– Executing [exten@sub-record-check:3] GotoIf(“SIP/1001-000002ec”, “1?caller”) in new stack
– Goto (sub-record-check,exten,10)
– Executing [exten@sub-record-check:10] Set(“SIP/1001-000002ec”, “__REC_POLICY_MODE=dontcare”) in new stack
– Executing [exten@sub-record-check:11] GosubIf(“SIP/1001-000002ec”, “0?record,1(exten,100,1001)”) in new stack
– Executing [exten@sub-record-check:12] Return(“SIP/1001-000002ec”, “”) in new stack
– Executing [s@macro-exten-vm:7] Macro(“SIP/1001-000002ec”, “dial-one,15,Ttr,100”) in new stack
– Executing [s@macro-dial-one:1] Set(“SIP/1001-000002ec”, “DEXTEN=100”) in new stack
– Executing [s@macro-dial-one:2] Set(“SIP/1001-000002ec”, “DIALSTATUS_CW=”) in new stack
– Executing [s@macro-dial-one:3] GosubIf(“SIP/1001-000002ec”, “0?screen,1()”) in new stack
– Executing [s@macro-dial-one:4] GosubIf(“SIP/1001-000002ec”, “0?cf,1()”) in new stack
– Executing [s@macro-dial-one:5] GotoIf(“SIP/1001-000002ec”, “1?skip1”) in new stack
– Goto (macro-dial-one,s,8)
– Executing [s@macro-dial-one:8] GotoIf(“SIP/1001-000002ec”, “0?nodial”) in new stack
– Executing [s@macro-dial-one:9] GotoIf(“SIP/1001-000002ec”, “0?continue”) in new stack
– Executing [s@macro-dial-one:10] Set(“SIP/1001-000002ec”, “EXTHASCW=ENABLED”) in new stack
– Executing [s@macro-dial-one:11] GotoIf(“SIP/1001-000002ec”, “0?next1:cwinusebusy”) in new stack
– Goto (macro-dial-one,s,23)
– Executing [s@macro-dial-one:23] GotoIf(“SIP/1001-000002ec”, “1?next3:continue”) in new stack
– Goto (macro-dial-one,s,24)
– Executing [s@macro-dial-one:24] ExecIf(“SIP/1001-000002ec”, “0?Set(DIALSTATUS_CW=BUSY)”) in new stack
– Executing [s@macro-dial-one:25] GotoIf(“SIP/1001-000002ec”, “0?nodial”) in new stack
– Executing [s@macro-dial-one:26] GosubIf(“SIP/1001-000002ec”, “1?dstring,1():dlocal,1()”) in new stack
– Executing [dstring@macro-dial-one:1] Set(“SIP/1001-000002ec”, “DSTRING=”) in new stack
– Executing [dstring@macro-dial-one:2] Set(“SIP/1001-000002ec”, “DEVICES=100”) in new stack
– Executing [dstring@macro-dial-one:3] ExecIf(“SIP/1001-000002ec”, “0?Return()”) in new stack
– Executing [dstring@macro-dial-one:4] ExecIf(“SIP/1001-000002ec”, “0?Set(DEVICES=00)”) in new stack
– Executing [dstring@macro-dial-one:5] Set(“SIP/1001-000002ec”, “LOOPCNT=1”) in new stack
– Executing [dstring@macro-dial-one:6] Set(“SIP/1001-000002ec”, “ITER=1”) in new stack
– Executing [dstring@macro-dial-one:7] Set(“SIP/1001-000002ec”, “THISDIAL=SIP/100”) in new stack
– Executing [dstring@macro-dial-one:8] GosubIf(“SIP/1001-000002ec”, “1?zap2dahdi,1()”) in new stack
– Executing [zap2dahdi@macro-dial-one:1] ExecIf(“SIP/1001-000002ec”, “0?Return()”) in new stack
– Executing [zap2dahdi@macro-dial-one:2] Set(“SIP/1001-000002ec”, “NEWDIAL=”) in new stack
– Executing [zap2dahdi@macro-dial-one:3] Set(“SIP/1001-000002ec”, “LOOPCNT2=1”) in new stack
– Executing [zap2dahdi@macro-dial-one:4] Set(“SIP/1001-000002ec”, “ITER2=1”) in new stack
– Executing [zap2dahdi@macro-dial-one:5] Set(“SIP/1001-000002ec”, “THISPART2=SIP/100”) in new stack
– Executing [zap2dahdi@macro-dial-one:6] ExecIf(“SIP/1001-000002ec”, “0?Set(THISPART2=DAHDI/100)”) in new stack
– Executing [zap2dahdi@macro-dial-one:7] Set(“SIP/1001-000002ec”, “NEWDIAL=SIP/100&”) in new stack
– Executing [zap2dahdi@macro-dial-one:8] Set(“SIP/1001-000002ec”, “ITER2=2”) in new stack
– Executing [zap2dahdi@macro-dial-one:9] GotoIf(“SIP/1001-000002ec”, “0?begin2”) in new stack
– Executing [zap2dahdi@macro-dial-one:10] Set(“SIP/1001-000002ec”, “THISDIAL=SIP/100”) in new stack
– Executing [zap2dahdi@macro-dial-one:11] Return(“SIP/1001-000002ec”, “”) in new stack
– Executing [dstring@macro-dial-one:9] Set(“SIP/1001-000002ec”, “DSTRING=SIP/100&”) in new stack
– Executing [dstring@macro-dial-one:10] Set(“SIP/1001-000002ec”, “ITER=2”) in new stack
– Executing [dstring@macro-dial-one:11] GotoIf(“SIP/1001-000002ec”, “0?begin”) in new stack
– Executing [dstring@macro-dial-one:12] Set(“SIP/1001-000002ec”, “DSTRING=SIP/100”) in new stack
– Executing [dstring@macro-dial-one:13] Return(“SIP/1001-000002ec”, “”) in new stack
– Executing [s@macro-dial-one:27] GotoIf(“SIP/1001-000002ec”, “0?nodial”) in new stack
– Executing [s@macro-dial-one:28] GotoIf(“SIP/1001-000002ec”, “0?skiptrace”) in new stack
– Executing [s@macro-dial-one:29] GosubIf(“SIP/1001-000002ec”, “1?ctset,1():ctclear,1()”) in new stack
– Executing [ctset@macro-dial-one:1] Set(“SIP/1001-000002ec”, “DB(CALLTRACE/100)=1001”) in new stack
– Executing [ctset@macro-dial-one:2] Return(“SIP/1001-000002ec”, “”) in new stack
– Executing [s@macro-dial-one:30] Set(“SIP/1001-000002ec”, “D_OPTIONS=Ttr”) in new stack
– Executing [s@macro-dial-one:31] ExecIf(“SIP/1001-000002ec”, “0?SIPAddHeader(Alert-Info: )”) in new stack
– Executing [s@macro-dial-one:32] ExecIf(“SIP/1001-000002ec”, “0?SIPAddHeader()”) in new stack
– Executing [s@macro-dial-one:33] ExecIf(“SIP/1001-000002ec”, “0?Set(CHANNEL(musicclass)=)”) in new stack
– Executing [s@macro-dial-one:34] GosubIf(“SIP/1001-000002ec”, “0?qwait,1()”) in new stack
– Executing [s@macro-dial-one:35] Set(“SIP/1001-000002ec”, “__CWIGNORE=”) in new stack
– Executing [s@macro-dial-one:36] Set(“SIP/1001-000002ec”, “__KEEPCID=TRUE”) in new stack
– Executing [s@macro-dial-one:37] GotoIf(“SIP/1001-000002ec”, “0?usegoto,1”) in new stack
– Executing [s@macro-dial-one:38] GotoIf(“SIP/1001-000002ec”, “0?godial”) in new stack
– Executing [s@macro-dial-one:39] Set(“SIP/1001-000002ec”, “CONNECTEDLINE(name,i)=Anastasia”) in new stack
– Executing [s@macro-dial-one:40] Set(“SIP/1001-000002ec”, “CONNECTEDLINE(num)=100”) in new stack
– Executing [s@macro-dial-one:41] Set(“SIP/1001-000002ec”, “D_OPTIONS=TtrI”) in new stack
– Executing [s@macro-dial-one:42] Dial(“SIP/1001-000002ec”, “SIP/100,15,TtrI”) in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
– Called SIP/100

<— Transmitting (NAT) to 192.168.1.73:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.73:5060;branch=z9hG4bK5b4530c763a15de3;received=192.168.1.73;rport=5060
From: “Anonymous” sip:1001@192.168.1.80;user=phone;tag=3222c599-686617
To: sip:100@192.168.1.80:5060;user=phone;tag=as522edba3
Call-ID: D1B9-125A-46686617963977A1F695-023@SipHost
CSeq: 26 INVITE
Server: FPBX-2.11.0(11.3.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:100@192.168.1.80:5060
Content-Length: 0

<------------>
– Connected line update to SIP/1001-000002ec prevented.
– SIP/100-000002ed is ringing

<— Transmitting (NAT) to 192.168.1.73:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.73:5060;branch=z9hG4bK5b4530c763a15de3;received=192.168.1.73;rport=5060
From: “Anonymous” sip:1001@192.168.1.80;user=phone;tag=3222c599-686617
To: sip:100@192.168.1.80:5060;user=phone;tag=as522edba3
Call-ID: D1B9-125A-46686617963977A1F695-023@SipHost
CSeq: 26 INVITE
Server: FPBX-2.11.0(11.3.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:100@192.168.1.80:5060
Content-Length: 0

<------------>

<— SIP read from UDP:192.168.1.73:5060 —>
CANCEL sip:100@192.168.1.80:5060;user=phone SIP/2.0
Via:SIP/2.0/UDP 192.168.1.73:5060;branch=z9hG4bK5b4530c763a15de3
From: “Anonymous” sip:1001@192.168.1.80;user=phone;tag=3222c599-686617
To: sip:100@192.168.1.80:5060;user=phone
Call-ID:D1B9-125A-46686617963977A1F695-023@SipHost
CSeq:26 CANCEL
Max-Forwards:70
User-Agent:dlink 12-37-61928258-0.9.5.1.735
Content-Length:0

<------------->
— (9 headers 0 lines) —
Sending to 192.168.1.73:5060 (NAT)

<— Reliably Transmitting (NAT) to 192.168.1.73:5060 —>
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.1.73:5060;branch=z9hG4bK5b4530c763a15de3;received=192.168.1.73;rport=5060
From: “Anonymous” sip:1001@192.168.1.80;user=phone;tag=3222c599-686617
To: sip:100@192.168.1.80:5060;user=phone;tag=as522edba3
Call-ID: D1B9-125A-46686617963977A1F695-023@SipHost
CSeq: 26 INVITE
Server: FPBX-2.11.0(11.3.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------>

<— Transmitting (NAT) to 192.168.1.73:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.73:5060;branch=z9hG4bK5b4530c763a15de3;received=192.168.1.73;rport=5060
From: “Anonymous” sip:1001@192.168.1.80;user=phone;tag=3222c599-686617
To: sip:100@192.168.1.80:5060;user=phone;tag=as522edba3
Call-ID: D1B9-125A-46686617963977A1F695-023@SipHost
CSeq: 26 CANCEL
Server: FPBX-2.11.0(11.3.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------>
== Spawn extension (macro-dial-one, s, 42) exited non-zero on ‘SIP/1001-000002ec’ in macro ‘dial-one’
== Spawn extension (macro-exten-vm, s, 7) exited non-zero on ‘SIP/1001-000002ec’ in macro ‘exten-vm’
== Spawn extension (from-internal, 100, 2) exited non-zero on ‘SIP/1001-000002ec’
– Executing [h@from-internal:1] Hangup(“SIP/1001-000002ec”, “”) in new stack
== Spawn extension (from-internal, h, 1) exited non-zero on ‘SIP/1001-000002ec’

<— SIP read from UDP:192.168.1.73:5060 —>
ACK sip:100@192.168.1.80:5060;user=phone SIP/2.0
Via:SIP/2.0/UDP 192.168.1.73:5060;branch=z9hG4bK5b4530c763a15de3
From: “Anonymous” sip:1001@192.168.1.80;user=phone;tag=3222c599-686617
To: sip:100@192.168.1.80:5060;user=phone;tag=as522edba3
Call-ID:D1B9-125A-46686617963977A1F695-023@SipHost
CSeq:26 ACK
Max-Forwards:70
Content-Length:0

<------------->
— (8 headers 0 lines) —
Really destroying SIP dialog ‘D1B9-125A-46686617963977A1F695-023@SipHost’ Method: ACK
Really destroying SIP dialog ‘D1B9-125A-46686614E550F7BF2335-020@SipHost’ Method: REGISTER
Really destroying SIP dialog ‘D1B9-125A-4668661440AFDE309F3B-021@SipHost’ Method: REGISTER
Really destroying SIP dialog ‘D1B9-125A-4668661589D32B6F9245-022@SipHost’ Method: REGISTER
[/quote]

Could You post Your sip.conf (especially for the two devices, the DLink and the Cisco).
From the debug it seems to be either a problem with the local network or - more likely - with COLP.

The trace shows a dialplan that is so complicated that you need to consult the people who wrote it (it did not come with Asterisk), however it shows an incoming call which has ringback applied using the DIal r option, regardless of the outgoing side state, and then cancelled by the caller. There is no trace for the outgoing side. I hope that is because you used sip set debug ip, but, in any case, you needed to trace both sides.