SIPSTATION inbound calls result in "You have reached a number that has been disconnected

Fresh install of CentOS 5.3 Asterisk 1.6 FreePBX 2.6. All modules loaded and updated. Station is a Cisco 7960 running SIP.
Purchased 2 SIP trunks and 2 phone numbers thru the SIPSTATION module. Allowed the module to configure the trunks, inbound and outbound routes.
Outbound calls work fine. Station to Station works fine. Trunks are registered in Asterisk. SIPSTATION Firewall Test Passed.
I have tried allow anonymous sip calls yes and no. I’ve tried putting the host outside the DMZ on the router. I’ve tried adding incoming route context:
Context: freepbx-in
type=user
context=from-pstn
host=trunk1.freepbx.com
Inbound calls ring the station but dead air. Calling phone (my cell) hears dead air then two rings then the telco recording “you have reached a number that has been disconnected or is no longer in service.” Station will continue to ring even after I hear the message and hang up on the cell phone. After I pick up on the station and hang up it rings again a few seconds later.
Here is the CLI session for an incoming call:
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using UDPTL TOS bits 184
== Using UDPTL CoS mark 5
– Executing [[email protected]:1] Set(“SIP/fpbx-1-f36er788-b7118310”, “__FROM_DID=NXXNXXXXXX”) in new stack
– Executing [[email protected]:2] Gosub(“SIP/fpbx-1-f36er788-b7118310”, “app-blacklist-check,s,1”) in new stack
– Executing [[email protected]:1] GotoIf(“SIP/fpbx-1-f36er788-b7118310”, “0?blacklisted”) in new stack
– Executing [[email protected]:2] Set(“SIP/fpbx-1-f36er788-b7118310”, “CALLED_BLACKLIST=1”) in new stack
– Executing [[email protected]:3] Return(“SIP/fpbx-1-f36er788-b7118310”, “”) in new stack
– Executing [[email protected]:3] ExecIf(“SIP/fpbx-1-f36er788-b7118310”, “0 ?Set(CALLERID(name)=NXXNXXXXXX)”) in new stack
– Executing [[email protected]:4] Set(“SIP/fpbx-1-f36er788-b7118310”, “FAX_RX=”) in new stack
– Executing [[email protected]:5] Set(“SIP/fpbx-1-f36er788-b7118310”, “__CALLINGPRES_SV=allowed_not_screened”) in new stack
– Executing [[email protected]:6] Set(“SIP/fpbx-1-f36er788-b7118310”, “CALLERPRES()=allowed_not_screened”) in new stack
– Executing [[email protected]:7] Goto(“SIP/fpbx-1-f36er788-b7118310”, “from-did-direct,69999,1”) in new stack
– Goto (from-did-direct,69999,1)
– Executing [[email protected]:1] Macro(“SIP/fpbx-1-f36er788-b7118310”, “exten-vm,novm,69999”) in new stack
– Executing [[email protected]:1] Macro(“SIP/fpbx-1-f36er788-b7118310”, “user-callerid”) in new stack
– Executing [[email protected]:1] Set(“SIP/fpbx-1-f36er788-b7118310”, “AMPUSER=NXXNXXXXXX”) in new stack
– Executing [[email protected]:2] GotoIf(“SIP/fpbx-1-f36er788-b7118310”, “0?report”) in new stack
– Executing [[email protected]:3] ExecIf(“SIP/fpbx-1-f36er788-b7118310”, “1?Set(REALCALLERIDNUM=NXXNXXXXXX)”) in new stack
– Executing [[email protected]:4] Set(“SIP/fpbx-1-f36er788-b7118310”, “AMPUSER=”) in new stack
– Executing [[email protected]:5] Set(“SIP/fpbx-1-f36er788-b7118310”, “AMPUSERCIDNAME=”) in new stack
– Executing [[email protected]:6] GotoIf(“SIP/fpbx-1-f36er788-b7118310”, “1?report”) in new stack
– Goto (macro-user-callerid,s,10)
– Executing [[email protected]:10] GotoIf(“SIP/fpbx-1-f36er788-b7118310”, “0?continue”) in new stack
– Executing [[email protected]:11] Set(“SIP/fpbx-1-f36er788-b7118310”, “__TTL=64”) in new stack
– Executing [[email protected]:12] GotoIf(“SIP/fpbx-1-f36er788-b7118310”, “1?continue”) in new stack
– Goto (macro-user-callerid,s,19)
– Executing [[email protected]:19] NoOp(“SIP/fpbx-1-f36er788-b7118310”, “Using CallerID “+1NXXNXXXXXX” “) in new stack
– Executing [[email protected]:2] Set(“SIP/fpbx-1-f36er788-b7118310”, “RingGroupMethod=none”) in new stack
– Executing [[email protected]:3] Set(“SIP/fpbx-1-f36er788-b7118310”, “VMBOX=novm”) in new stack
– Executing [[email protected]:4] Set(“SIP/fpbx-1-f36er788-b7118310”, “EXTTOCALL=69999”) in new stack
– Executing [[email protected]:5] Set(“SIP/fpbx-1-f36er788-b7118310”, “CFUEXT=”) in new stack
– Executing [[email protected]:6] Set(“SIP/fpbx-1-f36er788-b7118310”, “CFBEXT=”) in new stack
– Executing [[email protected]:7] Set(“SIP/fpbx-1-f36er788-b7118310”, “RT=”””) in new stack
– Executing [[email protected]:8] Macro(“SIP/fpbx-1-f36er788-b7118310”, “record-enable,69999,IN”) in new stack
– Executing [[email protected]:1] GotoIf(“SIP/fpbx-1-f36er788-b7118310”, “1?check”) in new stack
– Goto (macro-record-enable,s,4)
– Executing [[email protected]:4] AGI(“SIP/fpbx-1-f36er788-b7118310”, “recordingcheck,20091120-123422,1258742062.42”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck,20091120-123422,1258742062.42: Inbound recording not enabled
– AGI Script recordingcheck completed, returning 0
– Executing [[email protected]:5] MacroExit(“SIP/fpbx-1-f36er788-b7118310”, “”) in new stack
– Executing [[email protected]:9] Macro(“SIP/fpbx-1-f36er788-b7118310”, “dial,”",tr,69999") in new stack
– Executing [[email protected]:1] GotoIf(“SIP/fpbx-1-f36er788-b7118310”, “1?dial”) in new stack
– Goto (macro-dial,s,3)
– Executing [[email protected]:3] AGI(“SIP/fpbx-1-f36er788-b7118310”, “dialparties.agi”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
dialparties.agi: Starting New Dialparties.agi
dialparties.agi: Caller ID name is ‘+1NXXNXXXXXX’ number is ‘NXXNXXXXXX’

dialparties.agi: USE_CONFIRMATION: 'FALSE’
dialparties.agi: RINGGROUP_INDEX: ''
dialparties.agi: Methodology of ring is ‘none’
– dialparties.agi: Added extension 69999 to extension map
dialparties.agi: Extension 69999 has call screening off
– dialparties.agi: Extension 69999 cf is disabled
– dialparties.agi: Extension 69999 do not disturb is disabled
dialparties.agi: extnum 69999 has: cw: 1; hascfb: 0 [] hascfu: 0 []
dialparties.agi: EXTENSION_STATE: 0 (NOT_INUSE)
– dialparties.agi: dbset CALLTRACE/69999 to NXXNXXXXXX
– dialparties.agi: Filtered ARG3: 69999
– AGI Script dialparties.agi completed, returning 0
– Executing [[email protected]:7] Dial(“SIP/fpbx-1-f36er788-b7118310”, “SIP/69999,”",tr") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using UDPTL TOS bits 184
== Using UDPTL CoS mark 5
– Called 69999
asterisk2CLI> Use of uninitialized value in hash element at /var/www/html/panel/op_server.pl line 3367.
Use of uninitialized value in concatenation (.) or string at /var/www/html/panel/op_server.pl line 3372.
Use of uninitialized value in pattern match (m//) at /var/www/html/panel/op_server.pl line 3374.
– SIP/69999-08dac6d0 is ringing
– SIP/69999-08dac6d0 answered SIP/fpbx-1-f36er788-b7118310
– Executing [[email protected]:1] Macro(“SIP/fpbx-1-f36er788-b7118310”, “hangupcall”) in new stack
– Executing [[email protected]:1] GotoIf(“SIP/fpbx-1-f36er788-b7118310”, “1?skiprg”) in new stack
– Goto (macro-hangupcall,s,4)
– Executing [[email protected]:4] GotoIf(“SIP/fpbx-1-f36er788-b7118310”, “1?skipblkvm”) in new stack
– Goto (macro-hangupcall,s,7)
– Executing [[email protected]:7] GotoIf(“SIP/fpbx-1-f36er788-b7118310”, “1?theend”) in new stack
– Goto (macro-hangupcall,s,9)
– Executing [[email protected]:9] Hangup(“SIP/fpbx-1-f36er788-b7118310”, “”) in new stack
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on ‘SIP/fpbx-1-f36er788-b7118310’ in macro ‘hangupcall’
== Spawn extension (macro-dial, h, 1) exited non-zero on ‘SIP/fpbx-1-f36er788-b7118310’
== Spawn extension (macro-dial, s, 7) exited non-zero on ‘SIP/fpbx-1-f36er788-b7118310’ in macro ‘dial’
== Spawn extension (macro-exten-vm, s, 9) exited non-zero on ‘SIP/fpbx-1-f36er788-b7118310’ in macro ‘exten-vm’
== Spawn extension (from-did-direct, 69999, 1) exited non-zero on 'SIP/fpbx-1-f36er788-b7118310’
asterisk2
CLI> Use of uninitialized value in concatenation (.) or string at /var/www/html/panel/op_server.pl line 3360.
Use of uninitialized value in hash element at /var/www/html/panel/op_server.pl line 3367.
Use of uninitialized value in concatenation (.) or string at /var/www/html/panel/op_server.pl line 3372.
Use of uninitialized value in pattern match (m//) at /var/www/html/panel/op_server.pl line 3374.
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using UDPTL TOS bits 184
== Using UDPTL CoS mark 5
– Executing [[email protected]:1] Set(“SIP/fpbx-1-f36er788-b7111b88”, “__FROM_DID=NXXNXXXXXX”) in new stack
– Executing [[email protected]:2] Gosub(“SIP/fpbx-1-f36er788-b7111b88”, “app-blacklist-check,s,1”) in new stack
– Executing [[email protected]:1] GotoIf(“SIP/fpbx-1-f36er788-b7111b88”, “0?blacklisted”) in new stack
– Executing [[email protected]:2] Set(“SIP/fpbx-1-f36er788-b7111b88”, “CALLED_BLACKLIST=1”) in new stack
– Executing [[email protected]:3] Return(“SIP/fpbx-1-f36er788-b7111b88”, “”) in new stack
– Executing [[email protected]:3] ExecIf(“SIP/fpbx-1-f36er788-b7111b88”, “0 ?Set(CALLERID(name)=NXXNXXXXXX)”) in new stack
– Executing [[email protected]:4] Set(“SIP/fpbx-1-f36er788-b7111b88”, “FAX_RX=”) in new stack
– Executing [[email protected]:5] Set(“SIP/fpbx-1-f36er788-b7111b88”, “__CALLINGPRES_SV=allowed_not_screened”) in new stack
– Executing [[email protected]:6] Set(“SIP/fpbx-1-f36er788-b7111b88”, “CALLERPRES()=allowed_not_screened”) in new stack
– Executing [[email protected]:7] Goto(“SIP/fpbx-1-f36er788-b7111b88”, “from-did-direct,69999,1”) in new stack
– Goto (from-did-direct,69999,1)
– Executing [[email protected]:1] Macro(“SIP/fpbx-1-f36er788-b7111b88”, “exten-vm,novm,69999”) in new stack
– Executing [[email protected]:1] Macro(“SIP/fpbx-1-f36er788-b7111b88”, “user-callerid”) in new stack
– Executing [[email protected]:1] Set(“SIP/fpbx-1-f36er788-b7111b88”, “AMPUSER=NXXNXXXXXX”) in new stack
– Executing [[email protected]:2] GotoIf(“SIP/fpbx-1-f36er788-b7111b88”, “0?report”) in new stack
– Executing [[email protected]:3] ExecIf(“SIP/fpbx-1-f36er788-b7111b88”, “1?Set(REALCALLERIDNUM=NXXNXXXXXX)”) in new stack
– Executing [[email protected]:4] Set(“SIP/fpbx-1-f36er788-b7111b88”, “AMPUSER=”) in new stack
– Executing [[email protected]:5] Set(“SIP/fpbx-1-f36er788-b7111b88”, “AMPUSERCIDNAME=”) in new stack
– Executing [[email protected]:6] GotoIf(“SIP/fpbx-1-f36er788-b7111b88”, “1?report”) in new stack
– Goto (macro-user-callerid,s,10)
– Executing [[email protected]:10] GotoIf(“SIP/fpbx-1-f36er788-b7111b88”, “0?continue”) in new stack
– Executing [[email protected]:11] Set(“SIP/fpbx-1-f36er788-b7111b88”, “__TTL=64”) in new stack
– Executing [[email protected]:12] GotoIf(“SIP/fpbx-1-f36er788-b7111b88”, “1?continue”) in new stack
– Goto (macro-user-callerid,s,19)
– Executing [[email protected]:19] NoOp(“SIP/fpbx-1-f36er788-b7111b88”, “Using CallerID “+1NXXNXXXXXX” “) in new stack
– Executing [[email protected]:2] Set(“SIP/fpbx-1-f36er788-b7111b88”, “RingGroupMethod=none”) in new stack
– Executing [[email protected]:3] Set(“SIP/fpbx-1-f36er788-b7111b88”, “VMBOX=novm”) in new stack
– Executing [[email protected]:4] Set(“SIP/fpbx-1-f36er788-b7111b88”, “EXTTOCALL=69999”) in new stack
– Executing [[email protected]:5] Set(“SIP/fpbx-1-f36er788-b7111b88”, “CFUEXT=”) in new stack
– Executing [[email protected]:6] Set(“SIP/fpbx-1-f36er788-b7111b88”, “CFBEXT=”) in new stack
– Executing [[email protected]:7] Set(“SIP/fpbx-1-f36er788-b7111b88”, “RT=”””) in new stack
– Executing [[email protected]:8] Macro(“SIP/fpbx-1-f36er788-b7111b88”, “record-enable,69999,IN”) in new stack
– Executing [[email protected]:1] GotoIf(“SIP/fpbx-1-f36er788-b7111b88”, “1?check”) in new stack
– Goto (macro-record-enable,s,4)
– Executing [[email protected]:4] AGI(“SIP/fpbx-1-f36er788-b7111b88”, “recordingcheck,20091120-123454,1258742094.44”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck,20091120-123454,1258742094.44: Inbound recording not enabled
– AGI Script recordingcheck completed, returning 0
– Executing [[email protected]:5] MacroExit(“SIP/fpbx-1-f36er788-b7111b88”, “”) in new stack
– Executing [[email protected]:9] Macro(“SIP/fpbx-1-f36er788-b7111b88”, “dial,”",tr,69999") in new stack
– Executing [[email protected]:1] GotoIf(“SIP/fpbx-1-f36er788-b7111b88”, “1?dial”) in new stack
– Goto (macro-dial,s,3)
– Executing [[email protected]:3] AGI(“SIP/fpbx-1-f36er788-b7111b88”, “dialparties.agi”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
dialparties.agi: Starting New Dialparties.agi
dialparties.agi: Caller ID name is ‘+1NXXNXXXXXX’ number is 'NXXNXXXXXX’
dialparties.agi: USE_CONFIRMATION: 'FALSE’
dialparties.agi: RINGGROUP_INDEX: ''
dialparties.agi: Methodology of ring is ‘none’
– dialparties.agi: Added extension 69999 to extension map
dialparties.agi: Extension 69999 has call screening off
– dialparties.agi: Extension 69999 cf is disabled
– dialparties.agi: Extension 69999 do not disturb is disabled
dialparties.agi: extnum 69999 has: cw: 1; hascfb: 0 [] hascfu: 0 []
dialparties.agi: EXTENSION_STATE: 0 (NOT_INUSE)
– dialparties.agi: dbset CALLTRACE/69999 to NXXNXXXXXX
– dialparties.agi: Filtered ARG3: 69999
– AGI Script dialparties.agi completed, returning 0
– Executing [[email protected]:7] Dial(“SIP/fpbx-1-f36er788-b7111b88”, “SIP/69999,”",tr") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using UDPTL TOS bits 184
== Using UDPTL CoS mark 5
– Called 69999
asterisk2CLI> Use of uninitialized value in hash element at /var/www/html/panel/op_server.pl line 3367.
Use of uninitialized value in concatenation (.) or string at /var/www/html/panel/op_server.pl line 3372.
Use of uninitialized value in pattern match (m//) at /var/www/html/panel/op_server.pl line 3374.
– SIP/69999-08dac6d0 is ringing
– SIP/69999-08dac6d0 answered SIP/fpbx-1-f36er788-b7111b88
– Executing [[email protected]:1] Macro(“SIP/fpbx-1-f36er788-b7111b88”, “hangupcall”) in new stack
– Executing [[email protected]:1] GotoIf(“SIP/fpbx-1-f36er788-b7111b88”, “1?skiprg”) in new stack
– Goto (macro-hangupcall,s,4)
– Executing [[email protected]:4] GotoIf(“SIP/fpbx-1-f36er788-b7111b88”, “1?skipblkvm”) in new stack
– Goto (macro-hangupcall,s,7)
– Executing [[email protected]:7] GotoIf(“SIP/fpbx-1-f36er788-b7111b88”, “1?theend”) in new stack
– Goto (macro-hangupcall,s,9)
– Executing [[email protected]:9] Hangup(“SIP/fpbx-1-f36er788-b7111b88”, “”) in new stack
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on ‘SIP/fpbx-1-f36er788-b7111b88’ in macro ‘hangupcall’
== Spawn extension (macro-dial, h, 1) exited non-zero on ‘SIP/fpbx-1-f36er788-b7111b88’
== Spawn extension (macro-dial, s, 7) exited non-zero on ‘SIP/fpbx-1-f36er788-b7111b88’ in macro ‘dial’
== Spawn extension (macro-exten-vm, s, 9) exited non-zero on ‘SIP/fpbx-1-f36er788-b7111b88’ in macro ‘exten-vm’
== Spawn extension (from-did-direct, 69999, 1) exited non-zero on 'SIP/fpbx-1-f36er788-b7111b88’
asterisk2
CLI> Use of uninitialized value in concatenation (.) or string at /var/www/html/panel/op_server.pl line 3360.
Use of uninitialized value in hash element at /var/www/html/panel/op_server.pl line 3367.
Use of uninitialized value in concatenation (.) or string at /var/www/html/panel/op_server.pl line 3372.
Use of uninitialized value in pattern match (m//) at /var/www/html/panel/op_server.pl line 3374.
Any help/suggestions would be greatly appreciated.

Well your lucky. I have no logging in CLI for inbound calls. I have signed up for 3 DID and none of them are working.

I have a broadvoice account and everything is working well there. Except broadvoice doesn’t allow outbound CID.

If anyone else has similar problems please log in the forums and let us know.

Outbound freePBX is working but no inbound. I am wondering if I have to set up something in freePBX account web page but I haven’t found anything there, not even a way to change your credit card information, or to cancel service.

Are either of you using the SIPSTATION module?

Yes, I used the SIPSTATION module to configure the trunks.

What is 69999?

Yup I used Sipstation.

I also tried creating my own inbound route. neither worked.

btw what is 69999?

69999 is the internal extension number.

Note:

The message I get is
The number you have reached is not in service. Please check the number and dial it again.

Uggh!!!

It seesm from you call trace that asterisk is trying to dial 69999 externally. Please check your call flow/dialplan to see why.

Here is another trace with all numbers left intact. This one includes a sip debugging.

<------------->
— (16 headers 11 lines) —
Really destroying SIP dialog ‘[email protected]’ Method: OPTIONS
asterisk1*CLI>
<— SIP read from UDP://216.82.225.24:5060 —>
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 216.82.225.24;rport;branch=z9hG4bKUQ3vZH8ajN0Ne
Max-Forwards: 51
From: “+14798998070” sip:[email protected];tag=76DZc5Fg1cZ6H
To: sip:[email protected]
Call-ID: c544580e-52fe-122d-6fac-0015c5eaaddb
CSeq: 123381674 INVITE
Contact: sip:[email protected]:5060
User-Agent: FreePBX Trunking
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
Supported: timer, precondition, path, replaces
Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 271
Remote-Party-ID: “+14798998070” sip:[email protected];party=calling;screen=yes;privacy=off

v=0
o=Sonus_UAC 28939 21363 IN IP4 192.168.27.72
s=SIP Media Capabilities
c=IN IP4 67.231.0.102
t=0 0
m=audio 5940 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=maxptime:20

<------------->
— (16 headers 12 lines) —
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using UDPTL TOS bits 184
== Using UDPTL CoS mark 5
Sending to 216.82.225.24 : 5060 (NAT)
Using INVITE request as basis request - c544580e-52fe-122d-6fac-0015c5eaaddb
No user ‘4798998070’ in SIP users list
Found peer ‘fpbx-1-e03ad727’ for ‘4798998070’ from 216.82.225.24:5060
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 67.231.0.102:5940
Found audio description format PCMU for ID 0
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - 0x104 (ulaw|g729), peer - audio=0x104 (ulaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x104 (ulaw|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 67.231.0.102:5940
Looking for 4799669999 in from-pstn (domain 98.172.119.87)
list_route: hop: sip:[email protected]:5060

<— Transmitting (NAT) to 216.82.225.24:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 216.82.225.24;branch=z9hG4bKUQ3vZH8ajN0Ne;received=216.82.225.24;rport=5060
From: “+14798998070” sip:[email protected];tag=76DZc5Fg1cZ6H
To: sip:[email protected]
Call-ID: c544580e-52fe-122d-6fac-0015c5eaaddb
CSeq: 123381674 INVITE
User-Agent: Asterisk PBX 1.6.0.17
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: -1;refresher=uas
Contact: sip:[email protected]
Content-Length: 0

<------------>
– Executing [[email protected]:1] Set(“SIP/fpbx-1-e03ad727-097c89e8”, “__FROM_DID=4799669999”) in new stack
– Executing [[email protected]:2] Gosub(“SIP/fpbx-1-e03ad727-097c89e8”, “app-blacklist-check,s,1”) in new stack
– Executing [[email protected]:1] GotoIf(“SIP/fpbx-1-e03ad727-097c89e8”, “0?blacklisted”) in new stack
– Executing [[email protected]:2] Set(“SIP/fpbx-1-e03ad727-097c89e8”, “CALLED_BLACKLIST=1”) in new stack
– Executing [[email protected]:3] Return(“SIP/fpbx-1-e03ad727-097c89e8”, “”) in new stack
– Executing [[email protected]:3] ExecIf(“SIP/fpbx-1-e03ad727-097c89e8”, “0 ?Set(CALLERID(name)=4798998070)”) in new stack
– Executing [[email protected]:4] Set(“SIP/fpbx-1-e03ad727-097c89e8”, “FAX_RX=”) in new stack
– Executing [[email protected]:5] Set(“SIP/fpbx-1-e03ad727-097c89e8”, “__CALLINGPRES_SV=allowed_not_screened”) in new stack
– Executing [[email protected]:6] Set(“SIP/fpbx-1-e03ad727-097c89e8”, “CALLERPRES()=allowed_not_screened”) in new stack
– Executing [[email protected]:7] Goto(“SIP/fpbx-1-e03ad727-097c89e8”, “from-did-direct,69999,1”) in new stack
– Goto (from-did-direct,69999,1)
– Executing [[email protected]:1] Macro(“SIP/fpbx-1-e03ad727-097c89e8”, “exten-vm,novm,69999”) in new stack
– Executing [[email protected]:1] Macro(“SIP/fpbx-1-e03ad727-097c89e8”, “user-callerid”) in new stack
– Executing [[email protected]:1] Set(“SIP/fpbx-1-e03ad727-097c89e8”, “AMPUSER=4798998070”) in new stack
– Executing [[email protected]:2] GotoIf(“SIP/fpbx-1-e03ad727-097c89e8”, “0?report”) in new stack
– Executing [[email protected]:3] ExecIf(“SIP/fpbx-1-e03ad727-097c89e8”, “1?Set(REALCALLERIDNUM=4798998070)”) in new stack
– Executing [[email protected]:4] Set(“SIP/fpbx-1-e03ad727-097c89e8”, “AMPUSER=”) in new stack
– Executing [[email protected]:5] Set(“SIP/fpbx-1-e03ad727-097c89e8”, “AMPUSERCIDNAME=”) in new stack
– Executing [[email protected]:6] GotoIf(“SIP/fpbx-1-e03ad727-097c89e8”, “1?report”) in new stack
– Goto (macro-user-callerid,s,10)
– Executing [[email protected]:10] GotoIf(“SIP/fpbx-1-e03ad727-097c89e8”, “0?continue”) in new stack
– Executing [[email protected]:11] Set(“SIP/fpbx-1-e03ad727-097c89e8”, “__TTL=64”) in new stack
– Executing [[email protected]:12] GotoIf(“SIP/fpbx-1-e03ad727-097c89e8”, “1?continue”) in new stack
– Goto (macro-user-callerid,s,19)
– Executing [[email protected]:19] NoOp(“SIP/fpbx-1-e03ad727-097c89e8”, “Using CallerID “+14798998070” <4798998070>”) in new stack
– Executing [[email protected]:2] Set(“SIP/fpbx-1-e03ad727-097c89e8”, “RingGroupMethod=none”) in new stack
– Executing [[email protected]:3] Set(“SIP/fpbx-1-e03ad727-097c89e8”, “VMBOX=novm”) in new stack
– Executing [[email protected]:4] Set(“SIP/fpbx-1-e03ad727-097c89e8”, “EXTTOCALL=69999”) in new stack
– Executing [[email protected]:5] Set(“SIP/fpbx-1-e03ad727-097c89e8”, “CFUEXT=”) in new stack
– Executing [[email protected]:6] Set(“SIP/fpbx-1-e03ad727-097c89e8”, “CFBEXT=”) in new stack
– Executing [[email protected]:7] Set(“SIP/fpbx-1-e03ad727-097c89e8”, “RT=”"") in new stack
– Executing [[email protected]:8] Macro(“SIP/fpbx-1-e03ad727-097c89e8”, “record-enable,69999,IN”) in new stack
– Executing [[email protected]:1] GotoIf(“SIP/fpbx-1-e03ad727-097c89e8”, “1?check”) in new stack
– Goto (macro-record-enable,s,4)
– Executing [[email protected]:4] AGI(“SIP/fpbx-1-e03ad727-097c89e8”, “recordingcheck,20091123-121341,1259000021.17”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck,20091123-121341,1259000021.17: Inbound recording not enabled
– <SIP/fpbx-1-e03ad727-097c89e8>AGI Script recordingcheck completed, returning 0
– Executing [[email protected]:5] MacroExit(“SIP/fpbx-1-e03ad727-097c89e8”, “”) in new stack
– Executing [[email protected]:9] Macro(“SIP/fpbx-1-e03ad727-097c89e8”, “dial,”",tr,69999") in new stack
– Executing [[email protected]:1] GotoIf(“SIP/fpbx-1-e03ad727-097c89e8”, “1?dial”) in new stack
– Goto (macro-dial,s,3)
– Executing [[email protected]:3] AGI(“SIP/fpbx-1-e03ad727-097c89e8”, “dialparties.agi”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
dialparties.agi: Starting New Dialparties.agi
dialparties.agi: Caller ID name is ‘+14798998070’ number is ‘4798998070’
> dialparties.agi: USE_CONFIRMATION: ‘FALSE’
> dialparties.agi: RINGGROUP_INDEX: ''
dialparties.agi: Methodology of ring is ‘none’
– dialparties.agi: Added extension 69999 to extension map
> dialparties.agi: Extension 69999 has call screening off
– dialparties.agi: Extension 69999 cf is disabled
– dialparties.agi: Extension 69999 do not disturb is disabled
> dialparties.agi: extnum 69999 has: cw: 1; hascfb: 0 [] hascfu: 0 []
dialparties.agi: EXTENSION_STATE: 0 (NOT_INUSE)
– dialparties.agi: dbset CALLTRACE/69999 to 4798998070
– dialparties.agi: Filtered ARG3: 69999
– <SIP/fpbx-1-e03ad727-097c89e8>AGI Script dialparties.agi completed, returning 0
– Executing [[email protected]:7] Dial(“SIP/fpbx-1-e03ad727-097c89e8”, “SIP/69999,”",tr") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using UDPTL TOS bits 184
== Using UDPTL CoS mark 5
Audio is at 192.168.0.4 port 17796
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.0.253:5060:
INVITE sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.0.4:5060;branch=z9hG4bK152e08f5
Max-Forwards: 70
From: “+14798998070” sip:[email protected];tag=as52ae5b74
To: sip:[email protected]:5060;transport=udp
Contact: sip:[email protected]
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.0.17
Date: Mon, 23 Nov 2009 18:13:41 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 262

v=0
o=root 1228763685 1228763685 IN IP4 192.168.0.4
s=Asterisk PBX 1.6.0.17
c=IN IP4 192.168.0.4
t=0 0
m=audio 17796 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv


-- Called 69999

asterisk1*CLI>
<— Transmitting (NAT) to 216.82.225.24:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 216.82.225.24;branch=z9hG4bKUQ3vZH8ajN0Ne;received=216.82.225.24;rport=5060
From: “+14798998070” sip:[email protected];tag=76DZc5Fg1cZ6H
To: sip:[email protected];tag=as30aa6d49
Call-ID: c544580e-52fe-122d-6fac-0015c5eaaddb
CSeq: 123381674 INVITE
User-Agent: Asterisk PBX 1.6.0.17
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: -1;refresher=uas
Contact: sip:[email protected]
Content-Length: 0

<------------>
asterisk1*CLI>
<— SIP read from UDP://192.168.0.253:50975 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.4:5060;branch=z9hG4bK152e08f5
From: “+14798998070” sip:[email protected];tag=as52ae5b74
To: sip:[email protected]:5060;transport=udp
Call-ID: [email protected]
Date: Mon, 23 Nov 2009 18:13:41 GMT
CSeq: 102 INVITE
Server: Cisco-CP7960G/8.0
Contact: sip:6[email protected]:5060;transport=udp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE
Content-Length: 0

<------------->
— (11 headers 0 lines) —
asterisk1*CLI>
<— SIP read from UDP://216.82.225.24:5060 —>
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 216.82.225.24;rport;branch=z9hG4bKUQ3vZH8ajN0Ne
Max-Forwards: 51
From: “+14798998070” sip:[email protected];tag=76DZc5Fg1cZ6H
To: sip:[email protected]
Call-ID: c544580e-52fe-122d-6fac-0015c5eaaddb
CSeq: 123381674 INVITE
Contact: sip:[email protected]:5060
User-Agent: FreePBX Trunking
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
Supported: timer, precondition, path, replaces
Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 271
Remote-Party-ID: “+14798998070” sip:[email protected];party=calling;screen=yes;privacy=off

v=0
o=Sonus_UAC 28939 21363 IN IP4 192.168.27.72
s=SIP Media Capabilities
c=IN IP4 67.231.0.102
t=0 0
m=audio 5940 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=maxptime:20

<------------->
— (16 headers 12 lines) —
Ignoring this INVITE request

<— Transmitting (NAT) to 216.82.225.24:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 216.82.225.24;branch=z9hG4bKUQ3vZH8ajN0Ne;received=216.82.225.24;rport=5060
From: “+14798998070” sip:[email protected];tag=76DZc5Fg1cZ6H
To: sip:[email protected]
Call-ID: c544580e-52fe-122d-6fac-0015c5eaaddb
CSeq: 123381674 INVITE
User-Agent: Asterisk PBX 1.6.0.17
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: -1;refresher=uas
Contact: sip:[email protected]
Content-Length: 0

<------------>
asterisk1*CLI>
<— SIP read from UDP://192.168.0.253:50975 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.4:5060;branch=z9hG4bK152e08f5
From: “+14798998070” sip:[email protected];tag=as52ae5b74
To: sip:[email protected]:5060;transport=udp;tag=0014a97146da00b603a8c04b-081b33b9
Call-ID: [email protected]
Date: Mon, 23 Nov 2009 18:13:41 GMT
CSeq: 102 INVITE
Server: Cisco-CP7960G/8.0
Contact: sip:[email protected]:5060;transport=udp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE
Remote-Party-ID: “9669999” sip:[email protected];party=called;id-type=subscriber;privacy=off;screen=yes
Content-Length: 0

<------------->
— (12 headers 0 lines) —
– SIP/69999-097cf6f0 is ringing
asterisk1*CLI>
<— Transmitting (NAT) to 216.82.225.24:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 216.82.225.24;branch=z9hG4bKUQ3vZH8ajN0Ne;received=216.82.225.24;rport=5060
From: “+14798998070” sip:[email protected];tag=76DZc5Fg1cZ6H
To: sip:[email protected];tag=as30aa6d49
Call-ID: c544580e-52fe-122d-6fac-0015c5eaaddb
CSeq: 123381674 INVITE
User-Agent: Asterisk PBX 1.6.0.17
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: -1;refresher=uas
Contact: sip:[email protected]
Content-Length: 0

<------------>
asterisk1*CLI>
<— SIP read from UDP://192.168.0.253:50975 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.4:5060;branch=z9hG4bK152e08f5
From: “+14798998070” sip:[email protected];tag=as52ae5b74
To: sip:[email protected]:5060;transport=udp;tag=0014a97146da00b603a8c04b-081b33b9
Call-ID: [email protected]
Date: Mon, 23 Nov 2009 18:13:42 GMT
CSeq: 102 INVITE
Server: Cisco-CP7960G/8.0
Contact: sip:[email protected]:5060;transport=udp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE
Remote-Party-ID: “9669999” sip:[email protected];party=called;id-type=subscriber;privacy=off;screen=yes
Supported: replaces,join,norefersub
Content-Length: 207
Content-Type: application/sdp
Content-Disposition: session;handling=optional

v=0
o=Cisco-SIPUA 26710 0 IN IP4 192.168.0.253
s=SIP Call
t=0 0
m=audio 23700 RTP/AVP 0 101
c=IN IP4 192.168.0.253
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv

<------------->
— (15 headers 10 lines) —
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 192.168.0.253:23700
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x104 (ulaw|g729), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.0.253:23700
list_route: hop: sip:[email protected]:5060;transport=udp
set_destination: Parsing sip:[email protected]:5060;transport=udp for address/port to send to
set_destination: set destination to 192.168.0.253, port 5060
Transmitting (no NAT) to 192.168.0.253:5060:
ACK sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.0.4:5060;branch=z9hG4bK5614444f
Max-Forwards: 70
From: “+14798998070” sip:[email protected];tag=as52ae5b74
To: sip:[email protected]:5060;transport=udp;tag=0014a97146da00b603a8c04b-081b33b9
Contact: sip:[email protected]
Call-ID: [email protected]
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.0.17
Content-Length: 0


-- SIP/69999-097cf6f0 answered SIP/fpbx-1-e03ad727-097c89e8

Audio is at 98.172.119.87 port 14546
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
asterisk1*CLI>
<— Reliably Transmitting (NAT) to 216.82.225.24:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 216.82.225.24;branch=z9hG4bKUQ3vZH8ajN0Ne;received=216.82.225.24;rport=5060
From: “+14798998070” sip:[email protected];tag=76DZc5Fg1cZ6H
To: sip:[email protected];tag=as30aa6d49
Call-ID: c544580e-52fe-122d-6fac-0015c5eaaddb
CSeq: 123381674 INVITE
User-Agent: Asterisk PBX 1.6.0.17
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: -1;refresher=uas
Contact: sip:[email protected]
Content-Type: application/sdp
Content-Length: 311

v=0
o=root 338654095 338654095 IN IP4 98.172.119.87
s=Asterisk PBX 1.6.0.17
c=IN IP4 98.172.119.87
t=0 0
m=audio 14546 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
asterisk1*CLI>
<— SIP read from UDP://216.82.225.24:5060 —>
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 216.82.225.24;rport;branch=z9hG4bKUQ3vZH8ajN0Ne
Max-Forwards: 51
From: “+14798998070” sip:[email protected];tag=76DZc5Fg1cZ6H
To: sip:[email protected]
Call-ID: c544580e-52fe-122d-6fac-0015c5eaaddb
CSeq: 123381674 INVITE
Contact: sip:[email protected]:5060
User-Agent: FreePBX Trunking
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
Supported: timer, precondition, path, replaces
Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 271
Remote-Party-ID: “+14798998070” sip:[email protected];party=calling;screen=yes;privacy=off

v=0
o=Sonus_UAC 28939 21363 IN IP4 192.168.27.72
s=SIP Media Capabilities
c=IN IP4 67.231.0.102
t=0 0
m=audio 5940 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=maxptime:20

<------------->
— (16 headers 12 lines) —
Ignoring this INVITE request

<— Transmitting (NAT) to 216.82.225.24:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 216.82.225.24;branch=z9hG4bKUQ3vZH8ajN0Ne;received=216.82.225.24;rport=5060
From: “+14798998070” sip:[email protected];tag=76DZc5Fg1cZ6H
To: sip:[email protected]
Call-ID: c544580e-52fe-122d-6fac-0015c5eaaddb
CSeq: 123381674 INVITE
User-Agent: Asterisk PBX 1.6.0.17
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: -1;refresher=uas
Contact: sip:[email protected]
Content-Length: 0

<------------>
Audio is at 98.172.119.87 port 14546
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<— Transmitting (NAT) to 216.82.225.24:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 216.82.225.24;branch=z9hG4bKUQ3vZH8ajN0Ne;received=216.82.225.24;rport=5060
From: “+14798998070” sip:[email protected];tag=76DZc5Fg1cZ6H
To: sip:[email protected];tag=as30aa6d49
Call-ID: c544580e-52fe-122d-6fac-0015c5eaaddb
CSeq: 123381674 INVITE
User-Agent: Asterisk PBX 1.6.0.17
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: -1;refresher=uas
Contact: sip:[email protected]
Content-Type: application/sdp
Content-Length: 311

v=0
o=root 338654095 338654096 IN IP4 98.172.119.87
s=Asterisk PBX 1.6.0.17
c=IN IP4 98.172.119.87
t=0 0
m=audio 14546 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
Retransmitting #1 (NAT) to 216.82.225.24:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 216.82.225.24;branch=z9hG4bKUQ3vZH8ajN0Ne;received=216.82.225.24;rport=5060
From: “+14798998070” sip:[email protected];tag=76DZc5Fg1cZ6H
To: sip:[email protected];tag=as30aa6d49
Call-ID: c544580e-52fe-122d-6fac-0015c5eaaddb
CSeq: 123381674 INVITE
User-Agent: Asterisk PBX 1.6.0.17
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: -1;refresher=uas
Contact: sip:[email protected]
Content-Type: application/sdp
Content-Length: 311

v=0
o=root 338654095 338654095 IN IP4 98.172.119.87
s=Asterisk PBX 1.6.0.17
c=IN IP4 98.172.119.87
t=0 0
m=audio 14546 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv


Retransmitting #2 (NAT) to 216.82.225.24:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 216.82.225.24;branch=z9hG4bKUQ3vZH8ajN0Ne;received=216.82.225.24;rport=5060
From: “+14798998070” sip:[email protected];tag=76DZc5Fg1cZ6H
To: sip:[email protected];tag=as30aa6d49
Call-ID: c544580e-52fe-122d-6fac-0015c5eaaddb
CSeq: 123381674 INVITE
User-Agent: Asterisk PBX 1.6.0.17
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: -1;refresher=uas
Contact: sip:[email protected]
Content-Type: application/sdp
Content-Length: 311

v=0
o=root 338654095 338654095 IN IP4 98.172.119.87
s=Asterisk PBX 1.6.0.17
c=IN IP4 98.172.119.87
t=0 0
m=audio 14546 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv


Retransmitting #3 (NAT) to 216.82.225.24:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 216.82.225.24;branch=z9hG4bKUQ3vZH8ajN0Ne;received=216.82.225.24;rport=5060
From: “+14798998070” sip:[email protected];tag=76DZc5Fg1cZ6H
To: sip:[email protected];tag=as30aa6d49
Call-ID: c544580e-52fe-122d-6fac-0015c5eaaddb
CSeq: 123381674 INVITE
User-Agent: Asterisk PBX 1.6.0.17
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: -1;refresher=uas
Contact: sip:[email protected]
Content-Type: application/sdp
Content-Length: 311

v=0
o=root 338654095 338654095 IN IP4 98.172.119.87
s=Asterisk PBX 1.6.0.17
c=IN IP4 98.172.119.87
t=0 0
m=audio 14546 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv


Retransmitting #4 (NAT) to 216.82.225.24:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 216.82.225.24;branch=z9hG4bKUQ3vZH8ajN0Ne;received=216.82.225.24;rport=5060
From: “+14798998070” sip:[email protected];tag=76DZc5Fg1cZ6H
To: sip:[email protected];tag=as30aa6d49
Call-ID: c544580e-52fe-122d-6fac-0015c5eaaddb
CSeq: 123381674 INVITE
User-Agent: Asterisk PBX 1.6.0.17
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: -1;refresher=uas
Contact: sip:[email protected]
Content-Type: application/sdp
Content-Length: 311

v=0
o=root 338654095 338654095 IN IP4 98.172.119.87
s=Asterisk PBX 1.6.0.17
c=IN IP4 98.172.119.87
t=0 0
m=audio 14546 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv


asterisk1*CLI>
<— SIP read from UDP://216.82.225.24:5060 —>
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 216.82.225.24;rport;branch=z9hG4bKUQ3vZH8ajN0Ne
Max-Forwards: 51
From: “+14798998070” sip:[email protected];tag=76DZc5Fg1cZ6H
To: sip:[email protected]
Call-ID: c544580e-52fe-122d-6fac-0015c5eaaddb
CSeq: 123381674 INVITE
Contact: sip:[email protected]:5060
User-Agent: FreePBX Trunking
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
Supported: timer, precondition, path, replaces
Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 271
Remote-Party-ID: “+14798998070” sip:[email protected];party=calling;screen=yes;privacy=off

v=0
o=Sonus_UAC 28939 21363 IN IP4 192.168.27.72
s=SIP Media Capabilities
c=IN IP4 67.231.0.102
t=0 0
m=audio 5940 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=maxptime:20

<------------->
— (16 headers 12 lines) —
Ignoring this INVITE request

<— Transmitting (NAT) to 216.82.225.24:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 216.82.225.24;branch=z9hG4bKUQ3vZH8ajN0Ne;received=216.82.225.24;rport=5060
From: “+14798998070” sip:[email protected];tag=76DZc5Fg1cZ6H
To: sip:[email protected]
Call-ID: c544580e-52fe-122d-6fac-0015c5eaaddb
CSeq: 123381674 INVITE
User-Agent: Asterisk PBX 1.6.0.17
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: -1;refresher=uas
Contact: sip:[email protected]
Content-Length: 0

<------------>
Audio is at 98.172.119.87 port 14546
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<— Transmitting (NAT) to 216.82.225.24:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 216.82.225.24;branch=z9hG4bKUQ3vZH8ajN0Ne;received=216.82.225.24;rport=5060
From: “+14798998070” sip:[email protected];tag=76DZc5Fg1cZ6H
To: sip:[email protected];tag=as30aa6d49
Call-ID: c544580e-52fe-122d-6fac-0015c5eaaddb
CSeq: 123381674 INVITE
User-Agent: Asterisk PBX 1.6.0.17
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: -1;refresher=uas
Contact: sip:[email protected]
Content-Type: application/sdp
Content-Length: 311

v=0
o=root 338654095 338654097 IN IP4 98.172.119.87
s=Asterisk PBX 1.6.0.17
c=IN IP4 98.172.119.87
t=0 0
m=audio 14546 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
Retransmitting #5 (NAT) to 216.82.225.24:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 216.82.225.24;branch=z9hG4bKUQ3vZH8ajN0Ne;received=216.82.225.24;rport=5060
From: “+14798998070” sip:[email protected];tag=76DZc5Fg1cZ6H
To: sip:[email protected];tag=as30aa6d49
Call-ID: c544580e-52fe-122d-6fac-0015c5eaaddb
CSeq: 123381674 INVITE
User-Agent: Asterisk PBX 1.6.0.17
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: -1;refresher=uas
Contact: sip:[email protected]
Content-Type: application/sdp
Content-Length: 311

v=0
o=root 338654095 338654095 IN IP4 98.172.119.87
s=Asterisk PBX 1.6.0.17
c=IN IP4 98.172.119.87
t=0 0
m=audio 14546 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv


asterisk1*CLI>
<— SIP read from UDP://216.82.225.24:5060 —>
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 216.82.225.24;rport;branch=z9hG4bKUQ3vZH8ajN0Ne
Max-Forwards: 51
From: “+14798998070” sip:[email protected];tag=76DZc5Fg1cZ6H
To: sip:[email protected]
Call-ID: c544580e-52fe-122d-6fac-0015c5eaaddb
CSeq: 123381674 INVITE
Contact: sip:[email protected]:5060
User-Agent: FreePBX Trunking
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
Supported: timer, precondition, path, replaces
Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 271
Remote-Party-ID: “+14798998070” sip:[email protected];party=calling;screen=yes;privacy=off

v=0
o=Sonus_UAC 28939 21363 IN IP4 192.168.27.72
s=SIP Media Capabilities
c=IN IP4 67.231.0.102
t=0 0
m=audio 5940 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=maxptime:20

<------------->
— (16 headers 12 lines) —
Ignoring this INVITE request

<— Transmitting (NAT) to 216.82.225.24:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 216.82.225.24;branch=z9hG4bKUQ3vZH8ajN0Ne;received=216.82.225.24;rport=5060
From: “+14798998070” sip:[email protected];tag=76DZc5Fg1cZ6H
To: sip:[email protected]
Call-ID: c544580e-52fe-122d-6fac-0015c5eaaddb
CSeq: 123381674 INVITE
User-Agent: Asterisk PBX 1.6.0.17
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: -1;refresher=uas
Contact: sip:[email protected]
Content-Length: 0

<------------>
Audio is at 98.172.119.87 port 14546
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<— Transmitting (NAT) to 216.82.225.24:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 216.82.225.24;branch=z9hG4bKUQ3vZH8ajN0Ne;received=216.82.225.24;rport=5060
From: “+14798998070” sip:[email protected];tag=76DZc5Fg1cZ6H
To: sip:[email protected];tag=as30aa6d49
Call-ID: c544580e-52fe-122d-6fac-0015c5eaaddb
CSeq: 123381674 INVITE
User-Agent: Asterisk PBX 1.6.0.17
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: -1;refresher=uas
Contact: sip:[email protected]
Content-Type: application/sdp
Content-Length: 311

v=0
o=root 338654095 338654098 IN IP4 98.172.119.87
s=Asterisk PBX 1.6.0.17
c=IN IP4 98.172.119.87
t=0 0
m=audio 14546 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
Retransmitting #6 (NAT) to 216.82.225.24:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 216.82.225.24;branch=z9hG4bKUQ3vZH8ajN0Ne;received=216.82.225.24;rport=5060
From: “+14798998070” sip:[email protected];tag=76DZc5Fg1cZ6H
To: sip:[email protected];tag=as30aa6d49
Call-ID: c544580e-52fe-122d-6fac-0015c5eaaddb
CSeq: 123381674 INVITE
User-Agent: Asterisk PBX 1.6.0.17
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: -1;refresher=uas
Contact: sip:[email protected]
Content-Type: application/sdp
Content-Length: 311

v=0
o=root 338654095 338654095 IN IP4 98.172.119.87
s=Asterisk PBX 1.6.0.17
c=IN IP4 98.172.119.87
t=0 0
m=audio 14546 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv


-- Executing [[email protected]:1] Macro("SIP/fpbx-1-e03ad727-097c89e8", "hangupcall") in new stack
-- Executing [[email protected]:1] GotoIf("SIP/fpbx-1-e03ad727-097c89e8", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,4)
-- Executing [[email protected]:4] GotoIf("SIP/fpbx-1-e03ad727-097c89e8", "1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,7)
-- Executing [[email protected]:7] GotoIf("SIP/fpbx-1-e03ad727-097c89e8", "1?theend") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [[email protected]:9] Hangup("SIP/fpbx-1-e03ad727-097c89e8", "") in new stack

== Spawn extension (macro-hangupcall, s, 9) exited non-zero on ‘SIP/fpbx-1-e03ad727-097c89e8’ in macro ‘hangupcall’
== Spawn extension (macro-dial, h, 1) exited non-zero on 'SIP/fpbx-1-e03ad727-097c89e8’
Scheduling destruction of SIP dialog ‘[email protected]’ in 10624 ms (Method: INVITE)
set_destination: Parsing sip:[email protected]:5060;transport=udp for address/port to send to
set_destination: set destination to 192.168.0.253, port 5060
Reliably Transmitting (no NAT) to 192.168.0.253:5060:
BYE sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.0.4:5060;branch=z9hG4bK3e2f85a6
Max-Forwards: 70
From: “+14798998070” sip:[email protected];tag=as52ae5b74
To: sip:[email protected]:5060;transport=udp;tag=0014a97146da00b603a8c04b-081b33b9
Call-ID: [email protected]
CSeq: 103 BYE
User-Agent: Asterisk PBX 1.6.0.17
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


== Spawn extension (macro-dial, s, 7) exited non-zero on ‘SIP/fpbx-1-e03ad727-097c89e8’ in macro ‘dial’
== Spawn extension (macro-exten-vm, s, 9) exited non-zero on ‘SIP/fpbx-1-e03ad727-097c89e8’ in macro ‘exten-vm’
== Spawn extension (from-did-direct, 69999, 1) exited non-zero on 'SIP/fpbx-1-e03ad727-097c89e8’
asterisk1*CLI>
<— SIP read from UDP://192.168.0.253:50975 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.4:5060;branch=z9hG4bK3e2f85a6
From: “+14798998070” sip:[email protected];tag=as52ae5b74
To: sip:[email protected]:5060;transport=udp;tag=0014a97146da00b603a8c04b-081b33b9
Call-ID: [email protected]
Date: Mon, 23 Nov 2009 18:13:52 GMT
CSeq: 103 BYE
Server: Cisco-CP7960G/8.0
Content-Length: 0

<------------->
— (9 headers 0 lines) —
Really destroying SIP dialog ‘[email protected]’ Method: INVITE
asterisk1*CLI>
<— SIP read from UDP://216.82.225.24:5060 —>
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 216.82.225.24;rport;branch=z9hG4bKUQ3vZH8ajN0Ne
Max-Forwards: 51
From: “+14798998070” sip:[email protected];tag=76DZc5Fg1cZ6H
To: sip:[email protected]
Call-ID: c544580e-52fe-122d-6fac-0015c5eaaddb
CSeq: 123381674 INVITE
Contact: sip:[email protected]:5060
User-Agent: FreePBX Trunking
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
Supported: timer, precondition, path, replaces
Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 271
Remote-Party-ID: “+14798998070” sip:[email protected];party=calling;screen=yes;privacy=off

v=0
o=Sonus_UAC 28939 21363 IN IP4 192.168.27.72
s=SIP Media Capabilities
c=IN IP4 67.231.0.102
t=0 0
m=audio 5940 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=maxptime:20

<------------->
— (16 headers 12 lines) —
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using UDPTL TOS bits 184
== Using UDPTL CoS mark 5
Sending to 216.82.225.24 : 5060 (NAT)
Using INVITE request as basis request - c544580e-52fe-122d-6fac-0015c5eaaddb
No user ‘4798998070’ in SIP users list
Found peer ‘fpbx-1-e03ad727’ for ‘4798998070’ from 216.82.225.24:5060
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 67.231.0.102:5940
Found audio description format PCMU for ID 0
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - 0x104 (ulaw|g729), peer - audio=0x104 (ulaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x104 (ulaw|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 67.231.0.102:5940
Looking for 4799669999 in from-pstn (domain 98.172.119.87)
list_route: hop: sip:[email protected]:5060

<— Transmitting (NAT) to 216.82.225.24:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 216.82.225.24;branch=z9hG4bKUQ3vZH8ajN0Ne;received=216.82.225.24;rport=5060
From: “+14798998070” sip:[email protected];tag=76DZc5Fg1cZ6H
To: sip:[email protected]
Call-ID: c544580e-52fe-122d-6fac-0015c5eaaddb
CSeq: 123381674 INVITE
User-Agent: Asterisk PBX 1.6.0.17
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
equire: timer>
Session-Expires: -1;refresher=uas
Contact: sip:[email protected]
Content-Length: 0

<------------>
– Executing [[email protected]:1] Set(“SIP/fpbx-1-e03ad727-09820028”, “__FROM_DID=4799669999”) in new stack
– Executing [[email protected]:2] Gosub(“SIP/fpbx-1-e03ad727-09820028”, “app-blacklist-check,s,1”) in new stack
– Executing [[email protected]:1] GotoIf(“SIP/fpbx-1-e03ad727-09820028”, “0?blacklisted”) in new stack
– Executing [[email protected]:2] Set(“SIP/fpbx-1-e03ad727-09820028”, “CALLED_BLACKLIST=1”) in new stack
– Executing [[email protected]:3] Return(“SIP/fpbx-1-e03ad727-09820028”, “”) in new stack
– Executing [[email protected]:3] ExecIf(“SIP/fpbx-1-e03ad727-09820028”, “0 ?Set(CALLERID(name)=4798998070)”) in new stack
– Executing [[email protected]:4] Set(“SIP/fpbx-1-e03ad727-09820028”, “FAX_RX=”) in new stack
– Executing [[email protected]:5] Set(“SIP/fpbx-1-e03ad727-09820028”, “__CALLINGPRES_SV=allowed_not_screened”) in new stack
– Executing [[email protected]:6] Set(“SIP/fpbx-1-e03ad727-09820028”, “CALLERPRES()=allowed_not_screened”) in new stack
– Executing [[email protected]:7] Goto(“SIP/fpbx-1-e03ad727-09820028”, “from-did-direct,69999,1”) in new stack
– Goto (from-did-direct,69999,1)
– Executing [[email protected]:1] Macro(“SIP/fpbx-1-e03ad727-09820028”, “exten-vm,novm,69999”) in new stack
– Executing [[email protected]:1] Macro(“SIP/fpbx-1-e03ad727-09820028”, “user-callerid”) in new stack
– Executing [[email protected]:1] Set(“SIP/fpbx-1-e03ad727-09820028”, “AMPUSER=4798998070”) in new stack
– Executing [[email protected]:2] GotoIf(“SIP/fpbx-1-e03ad727-09820028”, “0?report”) in new stack
– Executing [[email protected]:3] ExecIf(“SIP/fpbx-1-e03ad727-09820028”, “1?Set(REALCALLERIDNUM=4798998070)”) in new stack
– Executing [[email protected]:4] Set(“SIP/fpbx-1-e03ad727-09820028”, “AMPUSER=”) in new stack
– Executing [[email protected]:5] Set(“SIP/fpbx-1-e03ad727-09820028”, “AMPUSERCIDNAME=”) in new stack
– Executing [[email protected]:6] GotoIf(“SIP/fpbx-1-e03ad727-09820028”, “1?report”) in new stack
– Goto (macro-user-callerid,s,10)
– Executing [[email protected]:10] GotoIf(“SIP/fpbx-1-e03ad727-09820028”, “0?continue”) in new stack
– Executing [[email protected]:11] Set(“SIP/fpbx-1-e03ad727-09820028”, “__TTL=64”) in new stack
– Executing [[email protected]:12] GotoIf(“SIP/fpbx-1-e03ad727-09820028”, “1?continue”) in new stack
– Goto (macro-user-callerid,s,19)
– Executing [[email protected]:19] NoOp(“SIP/fpbx-1-e03ad727-09820028”, “Using CallerID “+14798998070” <4798998070>”) in new stack
– Executing [[email protected]:2] Set(“SIP/fpbx-1-e03ad727-09820028”, “RingGroupMethod=none”) in new stack
– Executing [[email protected]:3] Set(“SIP/fpbx-1-e03ad727-09820028”, “VMBOX=novm”) in new stack
– Executing [[email protected]:4] Set(“SIP/fpbx-1-e03ad727-09820028”, “EXTTOCALL=69999”) in new stack
– Executing [[email protected]:5] Set(“SIP/fpbx-1-e03ad727-09820028”, “CFUEXT=”) in new stack
– Executing [[email protected]:6] Set(“SIP/fpbx-1-e03ad727-09820028”, “CFBEXT=”) in new stack
– Executing [[email protected]:7] Set(“SIP/fpbx-1-e03ad727-09820028”, “RT=”"") in new stack
– Executing [[email protected]:8] Macro(“SIP/fpbx-1-e03ad727-09820028”, “record-enable,69999,IN”) in new stack
– Executing [[email protected]:1] GotoIf(“SIP/fpbx-1-e03ad727-09820028”, “1?check”) in new stack
– Goto (macro-record-enable,s,4)
– Executing [[email protected]:4] AGI(“SIP/fpbx-1-e03ad727-09820028”, “recordingcheck,20091123-121356,1259000036.19”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck,20091123-121356,1259000036.19: Inbound recording not enabled
– <SIP/fpbx-1-e03ad727-09820028>AGI Script recordingcheck completed, returning 0
– Executing [[email protected]:5] MacroExit(“SIP/fpbx-1-e03ad727-09820028”, “”) in new stack
– Executing [[email protected]:9] Macro(“SIP/fpbx-1-e03ad727-09820028”, “dial,”",tr,69999") in new stack
– Executing [[email protected]:1] GotoIf(“SIP/fpbx-1-e03ad727-09820028”, “1?dial”) in new stack
– Goto (macro-dial,s,3)
– Executing [[email protected]:3] AGI(“SIP/fpbx-1-e03ad727-09820028”, “dialparties.agi”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
dialparties.agi: Starting New Dialparties.agi
dialparties.agi: Caller ID name is ‘+14798998070’ number is ‘4798998070’
> dialparties.agi: USE_CONFIRMATION: ‘FALSE’
> dialparties.agi: RINGGROUP_INDEX: ''
dialparties.agi: Methodology of ring is ‘none’
– dialparties.agi: Added extension 69999 to extension map
> dialparties.agi: Extension 69999 has call screening off
– dialparties.agi: Extension 69999 cf is disabled
– dialparties.agi: Extension 69999 do not disturb is disabled
> dialparties.agi: extnum 69999 has: cw: 1; hascfb: 0 [] hascfu: 0 []
dialparties.agi: EXTENSION_STATE: 0 (NOT_INUSE)
– dialparties.agi: dbset CALLTRACE/69999 to 4798998070
– dialparties.agi: Filtered ARG3: 69999
– <SIP/fpbx-1-e03ad727-09820028>AGI Script dialparties.agi completed, returning 0
– Executing [[email protected]:7] Dial(“SIP/fpbx-1-e03ad727-09820028”, “SIP/69999,”",tr") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using UDPTL TOS bits 184
== Using UDPTL CoS mark 5
Audio is at 192.168.0.4 port 18784
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.0.253:5060:
INVITE sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.0.4:5060;branch=z9hG4bK60a18d72
Max-Forwards: 70
From: “+14798998070” sip:[email protected];tag=as43e496ad
To: sip:[email protected]:5060;transport=udp
Contact: sip:[email protected]
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.0.17
Date: Mon, 23 Nov 2009 18:13:57 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 260

v=0
o=root 277980194 277980194 IN IP4 192.168.0.4
s=Asterisk PBX 1.6.0.17
c=IN IP4 192.168.0.4
t=0 0
m=audio 18784 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv


-- Called 69999

<— Transmitting (NAT) to 216.82.225.24:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 216.82.225.24;branch=z9hG4bKUQ3vZH8ajN0Ne;received=216.82.225.24;rport=5060
From: “+14798998070” sip:[email protected];tag=76DZc5Fg1cZ6H
To: sip:[email protected];tag=as75d29534
Call-ID: c544580e-52fe-122d-6fac-0015c5eaaddb
CSeq: 123381674 INVITE
User-Agent: Asterisk PBX 1.6.0.17
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: -1;refresher=uas
Contact: sip:[email protected]
Content-Length: 0

<------------>
asterisk1*CLI>
<— SIP read from UDP://192.168.0.253:50975 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.4:5060;branch=z9hG4bK60a18d72
From: “+14798998070” sip:[email protected];tag=as43e496ad
To: sip:[email protected]:5060;transport=udp
Call-ID: [email protected]
Date: Mon, 23 Nov 2009 18:13:56 GMT
CSeq: 102 INVITE
Server: Cisco-CP7960G/8.0
Contact: sip:[email protected]:5060;transport=udp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE
Content-Length: 0

<------------->
— (11 headers 0 lines) —
asterisk1*CLI>
<— SIP read from UDP://192.168.0.253:50975 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.4:5060;branch=z9hG4bK60a18d72
From: “+14798998070” sip:[email protected];tag=as43e496ad
To: sip:[email protected]:5060;transport=udp;tag=0014a97146da00b7193ddf46-7992fe64
Call-ID: [email protected]
Date: Mon, 23 Nov 2009 18:13:56 GMT
CSeq: 102 INVITE
Server: Cisco-CP7960G/8.0
Contact: sip:[email protected]:5060;transport=udp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE
Remote-Party-ID: “9669999” sip:[email protected];party=called;id-type=subscriber;privacy=off;screen=yes
Content-Length: 0

<------------->
— (12 headers 0 lines) —
– SIP/69999-b7cfa010 is ringing

<— Transmitting (NAT) to 216.82.225.24:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 216.82.225.24;branch=z9hG4bKUQ3vZH8ajN0Ne;received=216.82.225.24;rport=5060
From: “+14798998070” sip:[email protected];tag=76DZc5Fg1cZ6H
To: sip:[email protected];tag=as75d29534
Call-ID: c544580e-52fe-122d-6fac-0015c5eaaddb
CSeq: 123381674 INVITE
User-Agent: Asterisk PBX 1.6.0.17
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: -1;refresher=uas
Contact: sip:[email protected]
Content-Length: 0

<------------>
Reliably Transmitting (no NAT) to 192.168.0.253:5060:
OPTIONS sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.0.4:5060;branch=z9hG4bK771d5183
Max-Forwards: 70
From: “Unknown” sip:[email protected];tag=as45278ced
To: sip:[email protected]:5060;transport=udp
Contact: sip:[email protected]
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.0.17
Date: Mon, 23 Nov 2009 18:14:04 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


asterisk1*CLI>
<— SIP read from UDP://192.168.0.253:51094 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.4:5060;branch=z9hG4bK771d5183
From: “Unknown” sip:[email protected];tag=as45278ced
To: sip:[email protected]:5060;transport=udp;tag=0014a97146da00b81c67d062-60499352
Call-ID: [email protected]
Date: Mon, 23 Nov 2009 18:14:03 GMT
CSeq: 102 OPTIONS
Server: Cisco-CP7960G/8.0
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE
Accept: application/sdp,multipart/mixed,multipart/alternative
Accept-Encoding: identity
Accept-Language: en
Supported: replaces,join,norefersub
Content-Length: 237
Content-Type: application/sdp
Content-Disposition: session;handling=optional

v=0
o=Cisco-SIPUA 5385 0 IN IP4 192.168.0.253
s=SIP Call
t=0 0
m=audio 0 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

<------------->
— (16 headers 11 lines) —
Really destroying SIP dialog ‘[email protected]’ Method: OPTIONS
Reliably Transmitting (no NAT) to 192.168.0.253:5060:
OPTIONS sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.0.4:5060;branch=z9hG4bK7e442278
Max-Forwards: 70
From: “Unknown” sip:[email protected];tag=as3b834e4a
To: sip:[email protected]:5060;transport=udp
Contact: sip:[email protected]
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.0.17
Date: Mon, 23 Nov 2009 18:14:04 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


asterisk1*CLI>
<— SIP read from UDP://192.168.0.253:51095 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.4:5060;branch=z9hG4bK7e442278
From: “Unknown” sip:[email protected];tag=as3b834e4a
To: sip:[email protected]:5060;transport=udp;tag=0014a97146da00b9575acab0-2d752a68
Call-ID: [email protected]
Date: Mon, 23 Nov 2009 18:14:03 GMT
CSeq: 102 OPTIONS
Server: Cisco-CP7960G/8.0
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE
Accept: application/sdp,multipart/mixed,multipart/alternative
Accept-Encoding: identity
Accept-Language: en
Supported: replaces,join,norefersub
Content-Length: 237
Content-Type: application/sdp
Content-Disposition: session;handling=optional

v=0
o=Cisco-SIPUA 7377 0 IN IP4 192.168.0.253
s=SIP Call
t=0 0
m=audio 0 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

<------------->
— (16 headers 11 lines) —
Really destroying SIP dialog ‘[email protected]’ Method: OPTIONS
asterisk1*CLI>
<— SIP read from UDP://192.168.0.253:50975 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.4:5060;branch=z9hG4bK60a18d72
From: “+14798998070” sip:[email protected];tag=as43e496ad
To: sip:[email protected]:5060;transport=udp;tag=0014a97146da00b7193ddf46-7992fe64
Call-ID: [email protected]
Date: Mon, 23 Nov 2009 18:14:06 GMT
CSeq: 102 INVITE
Server: Cisco-CP7960G/8.0
Contact: sip:[email protected]:5060;transport=udp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE
Remote-Party-ID: “9669999” sip:[email protected];party=called;id-type=subscriber;privacy=off;screen=yes
Supported: replaces,join,norefersub
Content-Length: 206
Content-Type: application/sdp
Content-Disposition: session;handling=optional

v=0
o=Cisco-SIPUA 7873 0 IN IP4 192.168.0.253
s=SIP Call
t=0 0
m=audio 23820 RTP/AVP 0 101
c=IN IP4 192.168.0.253
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv

<------------->
— (15 headers 10 lines) —
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 192.168.0.253:23820
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x104 (ulaw|g729), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.0.253:23820
list_route: hop: sip:[email protected]:5060;transport=udp
set_destination: Parsing sip:[email protected]:5060;transport=udp for address/port to send to
set_destination: set destination to 192.168.0.253, port 5060
Transmitting (no NAT) to 192.168.0.253:5060:
ACK sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.0.4:5060;branch=z9hG4bK6b0d8d5c
Max-Forwards: 70
From: “+14798998070” sip:[email protected];tag=as43e496ad
To: sip:[email protected]:5060;transport=udp;tag=0014a97146da00b7193ddf46-7992fe64
Contact: sip:[email protected]
Call-ID: [email protected]
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.0.17
Content-Length: 0


-- SIP/69999-b7cfa010 answered SIP/fpbx-1-e03ad727-09820028

Audio is at 98.172.119.87 port 13290
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<— Reliably Transmitting (NAT) to 216.82.225.24:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 216.82.225.24;branch=z9hG4bKUQ3vZH8ajN0Ne;received=216.82.225.24;rport=5060
From: “+14798998070” sip:[email protected];tag=76DZc5Fg1cZ6H
To: sip:[email protected];tag=as75d29534
all-ID: c544580e-52fe-122d-6fac-0015c5eaaddb
CSeq: 123381674 INVITE
User-Agent: Asterisk PBX 1.6.0.17
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: -1;refresher=uas
Contact: sip:[email protected]
Content-Type: application/sdp
Content-Length: 313

v=0
o=root 1445253653 1445253653 IN IP4 98.172.119.87
s=Asterisk PBX 1.6.0.17
c=IN IP4 98.172.119.87
t=0 0
m=audio 13290 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
Retransmitting #1 (NAT) to 216.82.225.24:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 216.82.225.24;branch=z9hG4bKUQ3vZH8ajN0Ne;received=216.82.225.24;rport=5060
From: “+14798998070” sip:[email protected];tag=76DZc5Fg1cZ6H
To: sip:[email protected];tag=as75d29534
Call-ID: c544580e-52fe-122d-6fac-0015c5eaaddb
CSeq: 123381674 INVITE
User-Agent: Asterisk PBX 1.6.0.17
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: -1;refresher=uas
Contact: sip:[email protected]
Content-Type: application/sdp
Content-Length: 313

v=0
o=root 1445253653 1445253653 IN IP4 98.172.119.87
s=Asterisk PBX 1.6.0.17
c=IN IP4 98.172.119.87
t=0 0
m=audio 13290 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv


Retransmitting #2 (NAT) to 216.82.225.24:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 216.82.225.24;branch=z9hG4bKUQ3vZH8ajN0Ne;received=216.82.225.24;rport=5060
From: “+14798998070” sip:[email protected];tag=76DZc5Fg1cZ6H
To: sip:[email protected];tag=as75d29534
Call-ID: c544580e-52fe-122d-6fac-0015c5eaaddb
CSeq: 123381674 INVITE
User-Agent: Asterisk PBX 1.6.0.17
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: -1;refresher=uas
Contact: sip:[email protected]
Content-Type: application/sdp
Content-Length: 313

v=0
o=root 1445253653 1445253653 IN IP4 98.172.119.87
s=Asterisk PBX 1.6.0.17
c=IN IP4 98.172.119.87
t=0 0
m=audio 13290 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv


asterisk1*CLI>
<— SIP read from UDP://192.168.0.253:50975 —>
BYE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.0.253:5060;branch=z9hG4bK5e25a1a3
From: sip:[email protected]:5060;transport=udp;tag=0014a97146da00b7193ddf46-7992fe64
To: “+14798998070” sip:[email protected];tag=as43e496ad
Call-ID: [email protected]
Max-Forwards: 70
Date: Mon, 23 Nov 2009 18:14:08 GMT
CSeq: 101 BYE
User-Agent: Cisco-CP7960G/8.0
Content-Length: 0

<------------->
— (10 headers 0 lines) —
Sending to 192.168.0.253 : 5060 (no NAT)

<— Transmitting (no NAT) to 192.168.0.253:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.253:5060;branch=z9hG4bK5e25a1a3;received=192.168.0.253
From: sip:[email protected]:5060;transport=udp;tag=0014a97146da00b7193ddf46-7992fe64
To: “+14798998070” sip:[email protected];tag=as43e496ad
Call-ID: [email protected]
CSeq: 101 BYE
User-Agent: Asterisk PBX 1.6.0.17
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

<------------>
– Executing [[email protected]:1] Macro(“SIP/fpbx-1-e03ad727-09820028”, “hangupcall”) in new stack
– Executing [[email protected]:1] GotoIf(“SIP/fpbx-1-e03ad727-09820028”, “1?skiprg”) in new stack
– Goto (macro-hangupcall,s,4)
– Executing [[email protected]:4] GotoIf(“SIP/fpbx-1-e03ad727-09820028”, “1?skipblkvm”) in new stack
– Goto (macro-hangupcall,s,7)
– Executing [[email protected]:7] GotoIf(“SIP/fpbx-1-e03ad727-09820028”, “1?theend”) in new stack
– Goto (macro-hangupcall,s,9)
– Executing [[email protected]:9] Hangup(“SIP/fpbx-1-e03ad727-09820028”, “”) in new stack
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on ‘SIP/fpbx-1-e03ad727-09820028’ in macro ‘hangupcall’
== Spawn extension (macro-dial, h, 1) exited non-zero on ‘SIP/fpbx-1-e03ad727-09820028’
== Spawn extension (macro-dial, s, 7) exited non-zero on ‘SIP/fpbx-1-e03ad727-09820028’ in macro ‘dial’
== Spawn extension (macro-exten-vm, s, 9) exited non-zero on ‘SIP/fpbx-1-e03ad727-09820028’ in macro ‘exten-vm’
== Spawn extension (from-did-direct, 69999, 1) exited non-zero on 'SIP/fpbx-1-e03ad727-09820028’
Scheduling destruction of SIP dialog ‘c544580e-52fe-122d-6fac-0015c5eaaddb’ in 6400 ms (Method: INVITE)
Really destroying SIP dialog ‘[email protected]’ Method: BYE
Retransmitting #3 (NAT) to 216.82.225.24:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 216.82.225.24;branch=z9hG4bKUQ3vZH8ajN0Ne;received=216.82.225.24;rport=5060
From: “+14798998070” sip:[email protected];tag=76DZc5Fg1cZ6H
To: sip:[email protected];tag=as75d29534
Call-ID: c544580e-52fe-122d-6fac-0015c5eaaddb
CSeq: 123381674 INVITE
User-Agent: Asterisk PBX 1.6.0.17
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: -1;refresher=uas
Contact: sip:[email protected]
Content-Type: application/sdp
Content-Length: 313

v=0
o=root 1445253653 1445253653 IN IP4 98.172.119.87
s=Asterisk PBX 1.6.0.17
c=IN IP4 98.172.119.87
t=0 0
m=audio 13290 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv


Retransmitting #4 (NAT) to 216.82.225.24:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 216.82.225.24;branch=z9hG4bKUQ3vZH8ajN0Ne;received=216.82.225.24;rport=5060
From: “+14798998070” sip:[email protected];tag=76DZc5Fg1cZ6H
To: sip:[email protected];tag=as75d29534
Call-ID: c544580e-52fe-122d-6fac-0015c5eaaddb
CSeq: 123381674 INVITE
User-Agent: Asterisk PBX 1.6.0.17
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: -1;refresher=uas
Contact: sip:[email protected]
Content-Type: application/sdp
Content-Length: 313

v=0
o=root 1445253653 1445253653 IN IP4 98.172.119.87
s=Asterisk PBX 1.6.0.17
c=IN IP4 98.172.119.87
t=0 0
m=audio 13290 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv


Retransmitting #5 (NAT) to 216.82.225.24:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 216.82.225.24;branch=z9hG4bKUQ3vZH8ajN0Ne;received=216.82.225.24;rport=5060
From: “+14798998070” sip:[email protected];tag=76DZc5Fg1cZ6H
To: sip:[email protected];tag=as75d29534
Call-ID: c544580e-52fe-122d-6fac-0015c5eaaddb
CSeq: 123381674 INVITE
User-Agent: Asterisk PBX 1.6.0.17
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: -1;refresher=uas
Contact: sip:[email protected]
Content-Type: application/sdp
Content-Length: 313

v=0
o=root 1445253653 1445253653 IN IP4 98.172.119.87
s=Asterisk PBX 1.6.0.17
c=IN IP4 98.172.119.87
t=0 0
m=audio 13290 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv


REGISTER 13 headers, 0 lines
Reliably Transmitting (NAT) to 216.82.231.10:5060:
REGISTER sip:trunk2.freepbx.com SIP/2.0
Via: SIP/2.0/UDP 98.172.119.87:5060;branch=z9hG4bK203b53c5;rport
Max-Forwards: 70
From: sip:[email protected];tag=as789ef87a
To: sip:[email protected]
Call-ID: [email protected]
CSeq: 108 REGISTER
User-Agent: Asterisk PBX 1.6.0.17
Authorization: Digest username=“e03ad727”, realm=“trunk2.freepbx.com”, algorithm=MD5, uri=“sip:trunk2.freepbx.com”, nonce=“fdf02a03-69b4-4b24-ba1d-0224c1175980”, response=“cfd38493224719ed52ed544444615ef2”, qop=auth, cnonce=“183f6075”, nc=00000006
Expires: 120
Contact: sip:[email protected]
Event: registration
Content-Length: 0


REGISTER 13 headers, 0 lines
Reliably Transmitting (NAT) to 216.82.225.24:5060:
REGISTER sip:trunk1.freepbx.com SIP/2.0
Via: SIP/2.0/UDP 98.172.119.87:5060;branch=z9hG4bK4160b5d0;rport
Max-Forwards: 70
From: sip:[email protected];tag=as3e90bd6b
To: sip:[email protected]
Call-ID: [email protected]
CSeq: 108 REGISTER
User-Agent: Asterisk PBX 1.6.0.17
Authorization: Digest username=“e03ad727”, realm=“trunk1.freepbx.com”, algorithm=MD5, uri=“sip:trunk1.freepbx.com”, nonce=“c6a02b28-d85a-11de-8068-1fcf8b0b91b7”, response=“487ba8148b61fd2b489192f958b945db”, qop=auth, cnonce=“5785ab89”, nc=00000006
Expires: 120
Contact: sip:[email protected]
Event: registration
Content-Length: 0


<— SIP read from UDP://216.82.231.10:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 98.172.119.87:5060;branch=z9hG4bK203b53c5;rport=5060
From: sip:[email protected];tag=as789ef87a
To: sip:[email protected];tag=jS03y4Uy72NZm
Call-ID: [email protected]
CSeq: 108 REGISTER
Contact: sip:[email protected];expires=120
Date: Mon, 23 Nov 2009 18:14:11 GMT
User-Agent: FreePBX Trunking
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
Supported: timer, precondition, path, replaces
Content-Length: 0

<------------->
— (12 headers 0 lines) —
Scheduling destruction of SIP dialog ‘[email protected]’ in 32000 ms (Method: REGISTER)
asterisk1*CLI> sip se
<— SIP read from UDP://216.82.231.10:5060 —>
NOTIFY sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 216.82.231.10;rport;branch=z9hG4bK10Fjve7pDX2Qa
Max-Forwards: 70
From: sip:[email protected];tag=K2Sv0Zc24Bcjg
To: sip:[email protected]
Call-ID: d749ee5f-52fe-122d-84a4-001372fb8c08
CSeq: 123381689 NOTIFY
Contact: sip:[email protected]:5060
User-Agent: FreePBX Trunking
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
Supported: timer, precondition, path, replaces
Event: message-summary
Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Subscription-State: terminated;reason=timeout
Content-Type: application/simple-message-summary
Content-Length: 74

Messages-Waiting: no
Message-Account: sip:[email protected]

<------------->
— (16 headers 3 lines) —

<— Transmitting (NAT) to 216.82.231.10:5060 —>
SIP/2.0 489 Bad event
Via: SIP/2.0/UDP 216.82.231.10;branch=z9hG4bK10Fjve7pDX2Qa;received=216.82.231.10;rport=5060
From: sip:[email protected];tag=K2Sv0Zc24Bcjg
To: sip:[email protected];tag=as61151a44
Call-ID: d749ee5f-52fe-122d-84a4-001372fb8c08
CSeq: 123381689 NOTIFY
User-Agent: Asterisk PBX 1.6.0.17
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

<------------>
asterisk1*CLI> sip se
<— SIP read from UDP://216.82.225.24:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 98.172.119.87:5060;branch=z9hG4bK4160b5d0;rport=5060
From: sip:[email protected];tag=as3e90bd6b
To: sip:[email protected];tag=m4yBaK34Q97Be
Call-ID: [email protected]
CSeq: 108 REGISTER
Contact: sip:[email protected];expires=120
Date: Mon, 23 Nov 2009 18:14:09 GMT
User-Agent: FreePBX Trunking
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
Supported: timer, precondition, path, replaces
Content-Length: 0

<------------->
— (12 headers 0 lines) —
Scheduling destruction of SIP dialog ‘[email protected]’ in 32000 ms (Method: REGISTER)
asterisk1*CLI> sip set
<— SIP read from UDP://216.82.225.24:5060 —>
NOTIFY sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 216.82.225.24;rport;branch=z9hG4bKtgvpFtD9U6UFr
Max-Forwards: 70
From: sip:[email protected];tag=QZapF4NFF4a4g
To: sip:[email protected]
Call-ID: d75dd3cf-52fe-122d-6fac-0015c5eaaddb
CSeq: 123381689 NOTIFY
Contact: sip:[email protected]:5060
User-Agent: FreePBX Trunking
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
Supported: timer, precondition, path, replaces
Event: message-summary
Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Subscription-State: terminated;reason=timeout
Content-Type: application/simple-message-summary
Content-Length: 74

Messages-Waiting: no
Message-Account: sip:[email protected]

<------------->
— (16 headers 3 lines) —

<— Transmitting (NAT) to 216.82.225.24:5060 —>
SIP/2.0 489 Bad event
Via: SIP/2.0/UDP 216.82.225.24;branch=z9hG4bKtgvpFtD9U6UFr;received=216.82.225.24;rport=5060
From: sip:[email protected];tag=QZapF4NFF4a4g
To: sip:[email protected];tag=as23d96056
Call-ID: d75dd3cf-52fe-122d-6fac-0015c5eaaddb
CSeq: 123381689 NOTIFY
User-Agent: Asterisk PBX 1.6.0.17
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

<------------>
Really destroying SIP dialog ‘[email protected]’ Method: REGISTER
asterisk1*CLI> sip set debug
<— SIP read from UDP://216.82.225.24:5060 —>
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 216.82.225.24;rport;branch=z9hG4bKUQ3vZH8ajN0Ne
Max-Forwards: 51
From: “+14798998070” sip:[email protected];tag=76DZc5Fg1cZ6H
To: sip:[email protected]
Call-ID: c544580e-52fe-122d-6fac-0015c5eaaddb
CSeq: 123381674 INVITE
Contact: sip:[email protected]:5060
User-Agent: FreePBX Trunking
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
Supported: timer, precondition, path, replaces
Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 271
Remote-Party-ID: “+14798998070” sip:[email protected];party=calling;screen=yes;privacy=off

v=0
o=Sonus_UAC 28939 21363 IN IP4 192.168.27.72
s=SIP Media Capabilities
c=IN IP4 67.231.0.102
t=0 0
m=audio 5940 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=maxptime:20

<------------->
— (16 headers 12 lines) —
Ignoring this INVITE request

<— Transmitting (NAT) to 216.82.225.24:5060 —>
SIP/2.0 503 Unavailable
Via: SIP/2.0/UDP 216.82.225.24;branch=z9hG4bKUQ3vZH8ajN0Ne;received=216.82.225.24;rport=5060
From: “+14798998070” sip:[email protected];tag=76DZc5Fg1cZ6H
To: sip:[email protected];tag=as75d29534
Call-ID: c544580e-52fe-122d-6fac-0015c5eaaddb
CSeq: 123381674 INVITE
User-Agent: Asterisk PBX 1.6.0.17
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: -1;refresher=uas
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘c544580e-52fe-122d-6fac-0015c5eaaddb’ in 6400 ms (Method: INVITE)
asterisk1CLI> sip set debug off
SIP Debugging Disabled
asterisk1
CLI>

Hmm, I wonder if this isn’t a sipstation issue…

I just tried configuring one of the trunks manually without SIPSTATION and the inbound call behaves exactly the same as before.

Hi,
I just wrote you back on your personal email via our support system, but just in case others are having the same issue…

I have to do a SIP trace with you live to confirm this but I think you have a router or firewall that’s doing some packet inspection and rewrites. Probably SIP ALG is enabled somewhere in your router, or some similar feature.

I say this because:

  • You are receiving our initial invites, which means at least port 5060 is open
  • We are not receiving your Trying/OK responses from the looks of it
  • Your registration and challenges are successful, which means that sometimes the two-way traffic must be flowing OK

That is the only theory I can come up with at this point.

I have requested that you take a look through your router’s possible config options and turn off any inspection/rewrite type options.

Let me know if this helps.

  • Darren

I’m using a Cisco WRT610N It does have an option for SIP ALG but it is disabled. I’ve tried it both enabled and disabled.

let me know when you want to do a sip trace.

Replaced the Cisco Router with a D-Link Router and got the same result.

Would you be willing to directly connect your FreePBX box to the IP without a firewall in front? Just for testing…

I have seen this exact error on 2 completely different builds of asterisk 1.6.0.17 with each box at a different site. One was on a VPS residing on a hosting provider network. I built the box using CentOS 5.4 box w/Asterisk 1.6.0.17 and FreePBX 2.6 ‘from scratch’. After getting the issue I wrote it off as some error I made compiling things.

This week I tried upgrading my AsteriskNOW 1.5 box from Asterisk14 to Asterisk16 using:

yum shell
install asterisk16-core asterisk16
remove asterisk14 asterisk14-core
ts solve
ts run
remove asterisk14-core
ts solve
ts run

This box is at my office, and in BOTH locations I tried using public IPs with NO firewalling. Same issue. Outgoing works, incoming rings my phone, but when I pick up the calling party gets ‘You have reached a number that is disconnected…’. Subsequently, my internal phone gets rung twice after the initial call, but picking up results in dead air.

Anyway, I thought you may want to know that the person who started this thread isn’t the only person experiencing the problem. Maybe we did something wrong! I dunno :slight_smile: Hope you all can come up with a fix or point out where we were stupid. I’ll check my dialplan to see why it may be calling externally.

Thanks for the help all.

Just to continue the thread. I did put the asterisk box outside the firewall with the same result. Interesting that callguy didn’t see the problem until he upgraded from 1.4 to 1.6. I’ll try to build a 1.4 box tonight and try it.

Setting session-timers=refuse under trunk details seems to solve the issue

I think 1.6 is the issue - it looks like session timers default to on. mbrevada helped test this. He said he added session-timers=refuse to sip.conf for the trunk configurations (in FreePBX go to Trunk Settings and add that line).

I am checking as to why we even tolerate these at all on our end - I was pretty sure I don’t allow session timers on the servers. I’ll check into that. May also update SIPStation module so that this is the default behavior.

In the meantime, this appears to fix the issue.