Cisco 8961 Configuration Troubles

So here’s a little backstory - I have 3 Cisco 6900 (6921 and 6941) working awesome with our system (PBX In a flash) - I decided to purchase a Cisco 8961 to test out, assuming since it was the same “family” of the new 9.x firmware that it would be relatively simple (in Cisco terms) to get working. Not so much.

I unpacked it, set up to point to my update server - updated the firmware to the same relative revision (in my case 9.3.1.x) as the 6900 phones. That went fine. I then copied a 6941 xml config file, editing the bits inside to point to the right extension, etc…and loaded that up. The phone appears to register, the lines show up on the buttons as expected. I can place calls to other extensions or external phones with no problem, and can hear audio both ways. HOWEVER…

When I place a call INBOUND to the 8961 from any phone, it will not ring but instead the caller hears the congestion tones. I have a sip trace, and can post a copy of the xml config I am using, but I am not sure the bet way to insert them here. Below is my first attempt at the sip trace. In sip show peers, the phone doesnt really show up, as you will see:


SIP SHOW PEERS INFO:
8961 is extension 206 and 906:

pbxCLI> sip show peers
Name/username Host Dyn Forcerport ACL Port Status
100 (Unspecified) D A 0 UNKNOWN
1001 (Unspecified) D A 0 UNKNOWN
201/201 10.10.50.2 D A 5060 Unmonitored
202 (Unspecified) D A 0 Unmonitored
206/206 (Unspecified) D N A 0 Unmonitored
207/207 10.10.50.4 D A 5060 Unmonitored
208/208 10.10.50.6 D A 5060 Unmonitored
211/211 10.10.50.10 D N A 5060 OK (92 ms)
213/213 10.10.50.13 D N A 35420 Unmonitored
214/214 10.10.50.11 D N A 35851 Unmonitored
215/215 10.10.50.12 D N A 1041 Unmonitored
3001 (Unspecified) D A 0 UNKNOWN
301/301 10.10.50.4 D A 5060 Unmonitored
3401/3401 10.10.81.12 D A 5060 Unmonitored
3402/3402 10.10.81.14 D A 5060 OK (35 ms)
901/901 10.10.50.2 D A 5060 Unmonitored
902 (Unspecified) D A 0 Unmonitored
906/906 (Unspecified) D N A 0 Unmonitored
907/907 10.10.50.4 D A 5060 Unmonitored
908/908 10.10.50.6 D A 5060 Unmonitored
913/913 10.10.50.13 D N A 35420 Unmonitored
914/914 10.10.50.11 D N A 35851 Unmonitored
915/915 10.10.50.12 D N A 1041 Unmonitored
9401/9401 10.10.81.12 D A 5060 Unmonitored
9402/9402 10.10.81.14 D A 5060 OK (31 ms)
vitel-inbound/bravonoj 66.241.96.164 N 5060 Unmonitored
vitel-outbound/bravonoj 64.2.142.215 N 5060 Unmonitored
27 sip peers [Monitored: 3 online, 3 offline Unmonitored: 17 online, 4 offline]
> doing dnsmgr_lookup for ‘inbound24.vitelity.net
> doing dnsmgr_lookup for 'inbound24.vitelity.net
pbx
CLI>




pbx*CLI> sip show tcp
Address Transport Type
10.10.50.11:35851 TCP Server
10.10.50.14:53123 TCP Server
10.10.50.13:35420 TCP Server
10.10.50.12:1041 TCP Server




--------------BELOW IS THE SIP TRACE/SIP DEBUG FROM THE TIME OF THE CALL. ---------------------------------
--------------211 is the phone that I used to call the 8961 (at ext. 206) ---------------------------------




<— SIP read from UDP:10.10.50.10:5060 —>
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.10.50.10:5060;branch=z9hG4bK9bde9cd1
Max-Forwards: 70
To: sip:[email protected]
From: sip:[email protected];tag=3799445153
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: sip:[email protected]:5060
Allow: INVITE,ACK,CANCEL,BYE,OPTIONS,NOTIFY
Content-Type: application/sdp
User-Agent: Panasonic_KX-TGP500B04/12.02 (0080f0b6e9e4)
Content-Length: 313

v=0
o=- 1399056903 1399056903 IN IP4 10.10.50.10
s=-
c=IN IP4 10.10.50.10
t=0 0
m=audio 16048 RTP/AVP 9 8 2 18 0 101
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=ptime:20
<------------->
— (12 headers 15 lines) —
Sending to 10.10.50.10:5060 (NAT)
Using INVITE request as basis request - [email protected]
Found peer ‘211’ for ‘211’ from 10.10.50.10:5060

<— Reliably Transmitting (NAT) to 10.10.50.10:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.10.50.10:5060;branch=z9hG4bK9bde9cd1;received=10.10.50.10;rport=5060
From: sip:[email protected];tag=3799445153
To: sip:[email protected];tag=as0f115b43
Call-ID: [email protected]
CSeq: 1 INVITE
Server: FPBX-2.9.0(1.8.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="0fc14ec5"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘[email protected]’ in 6400 ms (Method: INVITE)

<— SIP read from UDP:10.10.50.10:5060 —>
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.10.50.10:5060;branch=z9hG4bK9bde9cd1
Max-Forwards: 70
To: sip:[email protected];tag=as0f115b43
From: sip:[email protected];tag=3799445153
Call-ID: [email protected]
CSeq: 1 ACK
Content-Length: 0

<------------->
— (8 headers 0 lines) —

<— SIP read from UDP:10.10.50.10:5060 —>
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.10.50.10:5060;branch=z9hG4bK8afa4fbf
Max-Forwards: 70
To: sip:[email protected]
From: sip:[email protected];tag=3799445153
Call-ID: [email protected]
CSeq: 2 INVITE
Contact: sip:[email protected]:5060
Authorization: Digest realm=“asterisk”, nonce=“0fc14ec5”, algorithm=MD5, uri=“sip:[email protected]:5060”, username=“211”, response="bb36ef5424afe4c018f9724105a7bfdb"
Allow: INVITE,ACK,CANCEL,BYE,OPTIONS,NOTIFY
Content-Type: application/sdp
User-Agent: Panasonic_KX-TGP500B04/12.02 (0080f0b6e9e4)
Content-Length: 313

v=0
o=- 1399056903 1399056903 IN IP4 10.10.50.10
s=-
c=IN IP4 10.10.50.10
t=0 0
m=audio 16048 RTP/AVP 9 8 2 18 0 101
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=ptime:20
<------------->
— (13 headers 15 lines) —
Sending to 10.10.50.10:5060 (NAT)
Using INVITE request as basis request - [email protected]
Found peer ‘211’ for ‘211’ from 10.10.50.10:5060
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Found RTP audio format 9
Found RTP audio format 8
Found RTP audio format 2
Found RTP audio format 18
Found RTP audio format 0
Found RTP audio format 101
Found audio description format G722 for ID 9
Found audio description format PCMA for ID 8
Found audio description format G726-32 for ID 2
Found audio description format G729 for ID 18
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x28000e (gsm|ulaw|alaw|h263|h264), peer - audio=0x190c (ulaw|alaw|g726|g729|g722)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 10.10.50.10:16048
Peer doesn’t provide video
Looking for 206 in from-internal (domain 10.10.60.6)
list_route: hop: sip:[email protected]:5060

<— Transmitting (NAT) to 10.10.50.10:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.10.50.10:5060;branch=z9hG4bK8afa4fbf;received=10.10.50.10;rport=5060
From: sip:[email protected];tag=3799445153
To: sip:[email protected]
Call-ID: [email protected]
CSeq: 2 INVITE
Server: FPBX-2.9.0(1.8.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:[email protected]:5060
Content-Length: 0

<------------>
– Executing [[email protected]:1] ExecIf(“SIP/211-0000013d”, “0?Set(__RINGTIMER=0)”) in new stack
– Executing [[email protected]:2] Macro(“SIP/211-0000013d”, “exten-vm,novm,206,0,0,0”) in new stack
– Executing [[email protected]:1] Macro(“SIP/211-0000013d”, “user-callerid,”) in new stack
– Executing [[email protected]:1] Set(“SIP/211-0000013d”, “AMPUSER=211”) in new stack
– Executing [[email protected]:2] GotoIf(“SIP/211-0000013d”, “0?report”) in new stack
– Executing [[email protected]:3] ExecIf(“SIP/211-0000013d”, “1?Set(REALCALLERIDNUM=211)”) in new stack
– Executing [[email protected]:4] Set(“SIP/211-0000013d”, “AMPUSER=211”) in new stack
– Executing [[email protected]:5] Set(“SIP/211-0000013d”, “AMPUSERCIDNAME=Cordless 1”) in new stack
– Executing [[email protected]:6] GotoIf(“SIP/211-0000013d”, “0?report”) in new stack
– Executing [[email protected]:7] Set(“SIP/211-0000013d”, “AMPUSERCID=211”) in new stack
– Executing [[email protected]:8] Set(“SIP/211-0000013d”, “CALLERID(all)=“Cordless 1” <211>”) in new stack
– Executing [[email protected]:9] GotoIf(“SIP/211-0000013d”, “0?limit”) in new stack
– Executing [[email protected]:10] ExecIf(“SIP/211-0000013d”, “0?Set(GROUP(concurrency_limit)=211)”) in new stack
– Executing [[email protected]:11] ExecIf(“SIP/211-0000013d”, “0?Set(CHANNEL(language)=)”) in new stack
– Executing [[email protected]:12] GotoIf(“SIP/211-0000013d”, “0?continue”) in new stack
– Executing [[email protected]:13] Set(“SIP/211-0000013d”, “__TTL=64”) in new stack
– Executing [[email protected]:14] GotoIf(“SIP/211-0000013d”, “1?continue”) in new stack
– Goto (macro-user-callerid,s,25)
– Executing [[email protected]:25] Set(“SIP/211-0000013d”, “CALLERID(number)=211”) in new stack
– Executing [[email protected]:26] Set(“SIP/211-0000013d”, “CALLERID(name)=Cordless 1”) in new stack
– Executing [[email protected]:27] Set(“SIP/211-0000013d”, “CHANNEL(language)=en”) in new stack
– Executing [[email protected]:2] Set(“SIP/211-0000013d”, “RingGroupMethod=none”) in new stack
– Executing [[email protected]:3] Set(“SIP/211-0000013d”, “__EXTTOCALL=206”) in new stack
– Executing [[email protected]:4] Set(“SIP/211-0000013d”, “__PICKUPMARK=206”) in new stack
– Executing [[email protected]:5] Set(“SIP/211-0000013d”, “RT=”) in new stack
– Executing [[email protected]:6] Macro(“SIP/211-0000013d”, “record-enable,206,IN”) in new stack
– Executing [[email protected]:1] GotoIf(“SIP/211-0000013d”, “1?check”) in new stack
– Goto (macro-record-enable,s,4)
– Executing [[email protected]:4] ExecIf(“SIP/211-0000013d”, “0?MacroExit()”) in new stack
– Executing [[email protected]:5] GotoIf(“SIP/211-0000013d”, “0?Group:OUT”) in new stack
– Goto (macro-record-enable,s,14)
– Executing [[email protected]:14] GotoIf(“SIP/211-0000013d”, “1?IN”) in new stack
– Goto (macro-record-enable,s,18)
– Executing [[email protected]:18] ExecIf(“SIP/211-0000013d”, “1?MacroExit()”) in new stack
– Executing [[email protected]:7] Macro(“SIP/211-0000013d”, “dial-one,tr,206”) in new stack
– Executing [[email protected]:1] Set(“SIP/211-0000013d”, “DEXTEN=206”) in new stack
– Executing [[email protected]:2] Set(“SIP/211-0000013d”, “DIALSTATUS_CW=”) in new stack
– Executing [[email protected]:3] GosubIf(“SIP/211-0000013d”, “0?screen,1”) in new stack
– Executing [[email protected]:4] GosubIf(“SIP/211-0000013d”, “0?cf,1”) in new stack
– Executing [[email protected]:5] GotoIf(“SIP/211-0000013d”, “1?skip1”) in new stack
– Goto (macro-dial-one,s,8)
– Executing [[email protected]:8] GotoIf(“SIP/211-0000013d”, “0?nodial”) in new stack
– Executing [[email protected]:9] GotoIf(“SIP/211-0000013d”, “0?continue”) in new stack
– Executing [[email protected]:10] Set(“SIP/211-0000013d”, “EXTHASCW=ENABLED”) in new stack
– Executing [[email protected]:11] GotoIf(“SIP/211-0000013d”, “0?next1:cwinusebusy”) in new stack
– Goto (macro-dial-one,s,23)
– Executing [[email protected]:23] GotoIf(“SIP/211-0000013d”, “1?next3:continue”) in new stack
– Goto (macro-dial-one,s,24)
– Executing [[email protected]:24] ExecIf(“SIP/211-0000013d”, “0?Set(DIALSTATUS_CW=BUSY)”) in new stack
– Executing [[email protected]:25] GotoIf(“SIP/211-0000013d”, “0?nodial”) in new stack
– Executing [[email protected]:26] GosubIf(“SIP/211-0000013d”, “1?dstring,1:dlocal,1”) in new stack
– Executing [[email protected]:1] Set(“SIP/211-0000013d”, “DSTRING=”) in new stack
– Executing [[email protected]:2] Set(“SIP/211-0000013d”, “DEVICES=206”) in new stack
– Executing [[email protected]:3] ExecIf(“SIP/211-0000013d”, “0?Return()”) in new stack
– Executing [[email protected]:4] ExecIf(“SIP/211-0000013d”, “0?Set(DEVICES=06)”) in new stack
– Executing [[email protected]:5] Set(“SIP/211-0000013d”, “LOOPCNT=1”) in new stack
– Executing [[email protected]:6] Set(“SIP/211-0000013d”, “ITER=1”) in new stack
– Executing [[email protected]:7] Set(“SIP/211-0000013d”, “THISDIAL=SIP/206”) in new stack
– Executing [[email protected]:8] GosubIf(“SIP/211-0000013d”, “1?zap2dahdi,1”) in new stack
– Executing [[email protected]:1] ExecIf(“SIP/211-0000013d”, “0?Return()”) in new stack
– Executing [[email protected]:2] Set(“SIP/211-0000013d”, “NEWDIAL=”) in new stack
– Executing [[email protected]:3] Set(“SIP/211-0000013d”, “LOOPCNT2=1”) in new stack
– Executing [[email protected]:4] Set(“SIP/211-0000013d”, “ITER2=1”) in new stack
– Executing [[email protected]:5] Set(“SIP/211-0000013d”, “THISPART2=SIP/206”) in new stack
– Executing [[email protected]:6] ExecIf(“SIP/211-0000013d”, “0?Set(THISPART2=DAHDI/206)”) in new stack
– Executing [[email protected]:7] Set(“SIP/211-0000013d”, “NEWDIAL=SIP/206&”) in new stack
– Executing [[email protected]:8] Set(“SIP/211-0000013d”, “ITER2=2”) in new stack
– Executing [[email protected]:9] GotoIf(“SIP/211-0000013d”, “0?begin2”) in new stack
– Executing [[email protected]:10] Set(“SIP/211-0000013d”, “THISDIAL=SIP/206”) in new stack
– Executing [[email protected]:11] Return(“SIP/211-0000013d”, “”) in new stack
– Executing [[email protected]:9] Set(“SIP/211-0000013d”, “DSTRING=SIP/206&”) in new stack
– Executing [[email protected]:10] Set(“SIP/211-0000013d”, “ITER=2”) in new stack
– Executing [[email protected]:11] GotoIf(“SIP/211-0000013d”, “0?begin”) in new stack
– Executing [[email protected]:12] Set(“SIP/211-0000013d”, “DSTRING=SIP/206”) in new stack
– Executing [[email protected]:13] Return(“SIP/211-0000013d”, “”) in new stack
– Executing [[email protected]:27] GotoIf(“SIP/211-0000013d”, “0?nodial”) in new stack
– Executing [[email protected]:28] GotoIf(“SIP/211-0000013d”, “0?skiptrace”) in new stack
– Executing [[email protected]:29] GosubIf(“SIP/211-0000013d”, “1?ctset,1:ctclear,1”) in new stack
– Executing [[email protected]:1] Set(“SIP/211-0000013d”, “DB(CALLTRACE/206)=211”) in new stack
– Executing [[email protected]:2] Return(“SIP/211-0000013d”, “”) in new stack
– Executing [[email protected]:30] Set(“SIP/211-0000013d”, “D_OPTIONS=tr”) in new stack
– Executing [[email protected]:31] ExecIf(“SIP/211-0000013d”, “0?SIPAddHeader(Alert-Info: )”) in new stack
– Executing [[email protected]:32] ExecIf(“SIP/211-0000013d”, “0?SIPAddHeader()”) in new stack
– Executing [[email protected]:33] ExecIf(“SIP/211-0000013d”, “0?Set(CHANNEL(musicclass)=)”) in new stack
– Executing [[email protected]:34] GosubIf(“SIP/211-0000013d”, “0?qwait,1”) in new stack
– Executing [[email protected]:35] Set(“SIP/211-0000013d”, “__CWIGNORE=”) in new stack
– Executing [[email protected]:36] Set(“SIP/211-0000013d”, “__KEEPCID=TRUE”) in new stack
– Executing [[email protected]:37] GotoIf(“SIP/211-0000013d”, “0?usegoto,1”) in new stack
– Executing [[email protected]:38] GotoIf(“SIP/211-0000013d”, “0?godial”) in new stack
– Executing [[email protected]:39] Set(“SIP/211-0000013d”, “CONNECTEDLINE(name,i)=Living Room”) in new stack
– Executing [[email protected]:40] Set(“SIP/211-0000013d”, “CONNECTEDLINE(num)=206”) in new stack
– Executing [[email protected]:41] Set(“SIP/211-0000013d”, “D_OPTIONS=trI”) in new stack
– Executing [[email protected]:42] Dial(“SIP/211-0000013d”, “SIP/206,trI”) in new stack
Really destroying SIP dialog ‘[email protected][::1]:5060’ Method: INVITE
== Everyone is busy/congested at this time (1:0/0/1)
– Executing [[email protected]:43] ExecIf(“SIP/211-0000013d”, “0?Set(DIALSTATUS=)”) in new stack
– Executing [[email protected]:44] GosubIf(“SIP/211-0000013d”, “0?s-CHANUNAVAIL,1”) in new stack
– Executing [[email protected]:45] MacroExit(“SIP/211-0000013d”, “”) in new stack
– Executing [[email protected]:8] Set(“SIP/211-0000013d”, “SV_DIALSTATUS=CHANUNAVAIL”) in new stack
– Executing [[email protected]:9] GosubIf(“SIP/211-0000013d”, “0?docfu,1”) in new stack
– Executing [[email protected]:10] GosubIf(“SIP/211-0000013d”, “0?docfb,1”) in new stack
– Executing [[email protected]:11] Set(“SIP/211-0000013d”, “DIALSTATUS=CHANUNAVAIL”) in new stack
– Executing [[email protected]:12] ExecIf(“SIP/211-0000013d”, “0?MacroExit()”) in new stack
– Executing [[email protected]:13] GotoIf(“SIP/211-0000013d”, “1?s-CHANUNAVAIL,1”) in new stack
– Goto (macro-exten-vm,s-CHANUNAVAIL,1)
– Executing [[email protected]:1] GotoIf(“SIP/211-0000013d”, “0?exit,1”) in new stack
– Executing [[email protected]:2] PlayTones(“SIP/211-0000013d”, “congestion”) in new stack
– Executing [[email protected]:3] Congestion(“SIP/211-0000013d”, “10”) in new stack

<— Reliably Transmitting (NAT) to 10.10.50.10:5060 —>
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 10.10.50.10:5060;branch=z9hG4bK8afa4fbf;received=10.10.50.10;rport=5060
From: sip:[email protected];tag=3799445153
To: sip:[email protected];tag=as2f8b2192
Call-ID: [email protected]
CSeq: 2 INVITE
Server: FPBX-2.9.0(1.8.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-Asterisk-HangupCause: Unknown
X-Asterisk-HangupCauseCode: 20
Content-Length: 0

<------------>
== Spawn extension (macro-exten-vm, s-CHANUNAVAIL, 3) exited non-zero on ‘SIP/211-0000013d’ in macro ‘exten-vm’
== Spawn extension (from-internal, 206, 2) exited non-zero on ‘SIP/211-0000013d’
– Executing [[email protected]:1] Hangup(“SIP/211-0000013d”, “”) in new stack
== Spawn extension (from-internal, h, 1) exited non-zero on ‘SIP/211-0000013d’

<— SIP read from UDP:10.10.50.10:5060 —>
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.10.50.10:5060;branch=z9hG4bK8afa4fbf
Max-Forwards: 70
To: sip:[email protected];tag=as2f8b2192
From: sip:[email protected];tag=3799445153
Call-ID: [email protected]
CSeq: 2 ACK
Authorization: Digest realm=“asterisk”, nonce=“0fc14ec5”, algorithm=MD5, uri=“sip:[email protected]:5060”, username=“211”, response="afdc81e1df0b3ba97366574edbea51db"
Content-Length: 0

<------------->
— (9 headers 0 lines) —
Really destroying SIP dialog ‘[email protected]’ Method: ACK
Reliably Transmitting (NAT) to 10.10.50.10:5060:
OPTIONS sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.10.60.6:5060;branch=z9hG4bK32d44700;rport
Max-Forwards: 70
From: “Unknown” sip:[email protected];tag=as2af44884
To: sip:[email protected]:5060
Contact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.9.0(1.8.13.0)
Date: Fri, 02 May 2014 18:55:01 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<— SIP read from UDP:10.10.50.10:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.60.6:5060;branch=z9hG4bK32d44700;rport=5060
To: sip:21[email protected];tag=808883192
From: “Unknown” sip:[email protected];tag=as2af44884
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
Contact: sip:10.10.50.10:5060
Allow: INVITE,ACK,CANCEL,BYE,OPTIONS,NOTIFY
Content-Length: 0

<------------->
— (9 headers 0 lines) —
Really destroying SIP dialog ‘[email protected]:5060’ Method: OPTIONS
pbxCLI> sip set debug off
SIP Debugging Disabled
> doing dnsmgr_lookup for ‘inbound24.vitelity.net
> doing dnsmgr_lookup for ‘inbound24.vitelity.net
> doing dnsmgr_lookup for ‘inbound24.vitelity.net
> doing dnsmgr_lookup for 'inbound24.vitelity.net
pbx
CLI> exit
Executing last minute cleanups