Scenario:

Dial a tollfree number from a Polycom 430 connected to my * (phone to * is
RFC2833 DTMF). Call goes out PRI, into an ITSP's gateway that sends it back
to the same * over SIP (using INFO DTMF).

I dial the number and hear the announcements just fine. I dial 1234 (not a
valid extension, but one that should result in me going to the 'i'
extension). None of the DTMF digits I type appear to be used to match
against an extension by *. The SIP debug show the INFO packets coming in
from the ITSP and being ACK'd by my *, and the CLI output shows that the
DTMF relay events are firing.

After 10 seconds, I get sent to the 't' extension, and after 3 cycles of
this, I get hung up on (as designed).


Masks:

A.B.C.D: IP of the ITSP's SIP system
E.F.G.H: IP of the ITSP's media gateway
1888XXXXXXX: the tollfree number I'm dialing
1416YYYYYYY: the CLID I present when dialing out


SIP settings:

pbxtel-01*CLI> sip show settings
pbxtel-01*CLI>

Global Settings:
----------------
  SIP Port:               5060            
  Bindaddress:            0.0.0.0
  Videosupport:           No
  AutoCreatePeer:         No
  Allow unknown access:   Yes
  Allow subscriptions:    Yes
  Allow overlap dialing:  Yes
  Promsic. redir:         No
  SIP domain support:     No
  Call to non-local dom.: Yes
  URI user is phone no:   No
  Our auth realm          asterisk
  Realm. auth:            No
  Always auth rejects:    No
  Call limit peers only:  Yes
  Direct RTP setup:       No
  User Agent:             My PBX
  MWI checking interval:  10 secs
  Reg. context:           (not set)
  Caller ID:              asterisk
  From: Domain:
  Record SIP history:     Off
  Call Events:            Off
  IP ToS SIP:             none
  IP ToS RTP audio:       none
  IP ToS RTP video:       none
  T38 fax pt UDPTL:       No
  RFC2833 Compensation:   No
  SIP realtime:           Disabled

Global Signalling Settings:
---------------------------
  Codecs:                 0x4 (ulaw)
  Codec Order:            ulaw:20
  T1 minimum:             100
  Relax DTMF:             No
  Compact SIP headers:    No
  RTP Keepalive:          0 (Disabled)
  RTP Timeout:            0 (Disabled)
  RTP Hold Timeout:       0 (Disabled)
  MWI NOTIFY mime type:   application/simple-message-summary
  DNS SRV lookup:         No
  Pedantic SIP support:   No
  Reg. min duration       60 secs
  Reg. max duration:      3600 secs
  Reg. default duration:  120 secs
  Outbound reg. timeout:  20 secs
  Outbound reg. attempts: 0
  Notify ringing state:   Yes
  Notify hold state:      No
  SIP Transfer mode:      open
  Max Call Bitrate:       384 kbps
  Auto-Framing:           No

Default Settings:
-----------------
  Context:                from-internal-unknown
  Nat:                    No
  DTMF:                   info
  Qualify:                0
  Use ClientCode:         No
  Progress inband:        Never
  Language:               (Defaults to English)
  MOH Interpret:          default
  MOH Suggest:
  Voice Mail Extension:   Voicemail

----
pbxtel-01*CLI> sip show peer myupstream
pbxtel-01*CLI>

  * Name       : myupstream
  Secret       : <Not set>
  MD5Secret    : <Not set>
  Context      : from-myupstream
  Subscr.Cont. : <Not set>
  Language     :
  AMA flags    : Unknown
  Transfer mode: open
  CallingPres  : Presentation Allowed, Not Screened
  Callgroup    :
  Pickupgroup  :
  Mailbox      :
  VM Extension : Voicemail
  LastMsgsSent : 32767/65535
  Call limit   : 0
  Dynamic      : No
  Callerid     : "" <>
  MaxCallBR    : 384 kbps
  Expire       : -1
  Insecure     : no
  Nat          : No
  ACL          : No
  T38 pt UDPTL : No
  CanReinvite  : Yes
  PromiscRedir : No
  User=Phone   : No
  Video Support: No
  Trust RPID   : No
  Send RPID    : No
  Subscriptions: Yes
  Overlap dial : Yes
  DTMFmode     : info
  LastMsg      : 0
  ToHost       : A.B.C.D
  Addr->IP     : A.B.C.D Port 5060
  Defaddr->IP  : 0.0.0.0 Port 0
  Def. Username:
  SIP Options  : (none)
  Codecs       : 0x4 (ulaw)
  Codec Order  : (ulaw:20)
  Auto-Framing:  No
  Status       : Unmonitored
  Useragent    :
  Reg. Contact :

pbxtel-01*CLI>


SIP config:

[myupstream]
type=peer
host=A.B.C.D
nat=never
context=from-myupstream


Dialplan:

[from-myupstream]

; conference bridge
exten   => 1888XXXXXXX,1,Set(IVR_CONTEXT=ivr-meetme)
exten   => 1888XXXXXXX,n,Ringing
exten   => 1888XXXXXXX,n,Macro(pstncid)
exten   => 1888XXXXXXX,n,Wait(2)
exten   => 1888XXXXXXX,n,Answer
exten   => 1888XXXXXXX,n,Goto(${IVR_CONTEXT}|s|1)
exten   => 1888XXXXXXX,n,Hangup

[ivr-meetme]
exten   => s,1,Set(TIMEOUT(digit)=2)
exten   => s,n,Set(ATTEMPTS=1)
exten   => s,n,Set(MAX_ATTEMPTS=3)

exten   => s,n,Background(welcome_to_conf_bridge)

exten   => s,n(check-attempts),GotoIf($["${ATTEMPTS}" <= "${MAX_ATTEMPTS}"]?top)
exten   => s,n,Playback(sorry-youre-having-problems)
exten   => s,n,Playback(please-try-again-later)
exten   => s,n,Hangup

exten   => s,n(top),Background(enter_conf_number)
exten   => s,n,WaitExten(10)

exten   => i,1,Set(ATTEMPTS=${MATH(${ATTEMPTS} + 1,int)})
exten   => i,n,Playback(conf-invalid)
exten   => i,n,Goto(s|check-attempts)

exten   => t,1,Set(ATTEMPTS=${MATH(${ATTEMPTS} + 1,int)})
exten   => t,n,Goto(s|check-attempts)

exten   => #,1,MeetMe(,oIcsMXaA)

exten   => 8011,1,MeetMe(8011,oIcsMXw)
exten   => 8012,1,MeetMe(8012,oIcsMXw)
exten   => 8013,1,MeetMe(8013,oIcsMXw)
exten   => 8014,1,MeetMe(8014,oIcsMXw)
exten   => 8015,1,MeetMe(8015,oIcsMXw)

exten   => 0,1,Goto(s|1)


CLI output:

pbxtel-01*CLI> core set verbose 3
Verbosity was 0 and is now 3
pbxtel-01*CLI> sip debug peer myupstream
SIP Debugging Enabled for IP: A.B.C.D:5060
    -- Executing [91888XXXXXXX@from-internal-admin:1] Macro("SIP/427-9fda1e30", "dialout|from-internal-admin|Zap/R3/1888XXXXXXX|1888XXXXXXX|0|0") in new stack
    -- Executing [s@macro-dialout:1] GotoIf("SIP/427-9fda1e30", "1?checkrecorddn") in new stack
    -- Goto (macro-dialout,s,4)
    -- Executing [s@macro-dialout:4] GotoIf("SIP/427-9fda1e30", "0?recorddn") in new stack
    -- Executing [s@macro-dialout:5] GotoIf("SIP/427-9fda1e30", "0?recorddn") in new stack
    -- Executing [s@macro-dialout:6] Goto("SIP/427-9fda1e30", "checkaccountcode") in new stack
    -- Goto (macro-dialout,s,8)
    -- Executing [s@macro-dialout:8] GotoIf("SIP/427-9fda1e30", "1?checkblockcid") in new stack
    -- Goto (macro-dialout,s,10)
    -- Executing [s@macro-dialout:10] GotoIf("SIP/427-9fda1e30", "0?blockclid") in new stack
    -- Executing [s@macro-dialout:11] GotoIf("SIP/427-9fda1e30", "0?blockclid") in new stack
    -- Executing [s@macro-dialout:12] GotoIf("SIP/427-9fda1e30", "0?blockclid") in new stack
    -- Executing [s@macro-dialout:13] GotoIf("SIP/427-9fda1e30", "0?getclid-ld:getclid-local") in new stack
    -- Goto (macro-dialout,s,15)
    -- Executing [s@macro-dialout:15] GotoIf("SIP/427-9fda1e30", "1?setclid-local:blockclid") in new stack
    -- Goto (macro-dialout,s,18)
    -- Executing [s@macro-dialout:18] Set("SIP/427-9fda1e30", "CALLERID(all)=ME <416-YYY-YYYY>") in new stack
    -- Executing [s@macro-dialout:19] Goto("SIP/427-9fda1e30", "checkstuffed") in new stack
    -- Goto (macro-dialout,s,22)
    -- Executing [s@macro-dialout:22] GotoIf("SIP/427-9fda1e30", "1?checkqmoutqdial") in new stack
    -- Goto (macro-dialout,s,25)
    -- Executing [s@macro-dialout:25] GotoIf("SIP/427-9fda1e30", "1?dialnostuff") in new stack
    -- Goto (macro-dialout,s,28)
    -- Executing [s@macro-dialout:28] Dial("SIP/427-9fda1e30", "Zap/R3/1888XXXXXXX") in new stack
    -- Requested transfer capability: 0x00 - SPEECH
    -- Called R3/1888XXXXXXX
    -- Zap/85-1 is proceeding passing it to SIP/427-9fda1e30
pbxtel-01*CLI>
<--- SIP read from A.B.C.D:5060 --->
INVITE sip:1888XXXXXXX@10.1.1.16:5060 SIP/2.0
Via: SIP/2.0/UDP A.B.C.D:5060;rport;branch=z9hG4bK1da3823e556c2fbe119b7d12a3c51d33
From: "1416YYYYYYY" <sip:1416YYYYYYY@A.B.C.D:5060;user=phone>;tag=GR52RWG346-34
To: "1888XXXXXXX" <sip:1888XXXXXXX@10.1.1.16:5060>
Call-ID: 110832659@A.B.C.D
CSeq: 1 INVITE
Contact: <sip:A.B.C.D:5060>
Remote-Party-ID: "1416YYYYYYY" <sip:1416YYYYYYY@A.B.C.D:5060>;party=calling;screen=yes;Privacy=off
max-forwards: 70
Allow: ACK, NOTIFY, OPTIONS, REFER, INFO, BYE, CANCEL, INVITE
Content-Type: application/sdp
Content-Length:   449

v=0
o=Clarent 110943 110944 IN IP4 E.F.G.H
s=Clarent C5CM
c=IN IP4 E.F.G.H
t=0 0
m=audio 4570 RTP/AVP 18 0 4 8 96 100 102 103 104 105
a=rtpmap:18 G729/8000
a=ptime:20
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
a=rtpmap:100 AC100/8000
a=rtpmap:102 AC102/8000
a=rtpmap:103 AC103/8000
a=rtpmap:104 AC104/8000
a=rtpmap:105 AC105/8000

<------------->
--- (12 headers 19 lines) ---
Sending to A.B.C.D : 5060 (NAT)
Using INVITE request as basis request - 110832659@A.B.C.D
Found peer 'myupstream'
Found RTP audio format 18
Found RTP audio format 0
Found RTP audio format 4
Found RTP audio format 8
Found RTP audio format 96
Found RTP audio format 100
Found RTP audio format 102
Found RTP audio format 103
Found RTP audio format 104
Found RTP audio format 105
Peer audio RTP is at port E.F.G.H:4570
Found description format G729 for ID 18
Found description format PCMU for ID 0
Found description format G723 for ID 4
Found description format PCMA for ID 8
Found description format telephone-event for ID 96
Found description format AC100 for ID 100
Found description format AC102 for ID 102
Found description format AC103 for ID 103
Found description format AC104 for ID 104
Found description format AC105 for ID 105
Capabilities: us - 0x4 (ulaw), peer - audio=0x10010d (g723|ulaw|alaw|g729|h263p)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x1 (telephone-event), combined - 0x0 (nothing)
Peer audio RTP is at port E.F.G.H:4570
Looking for 1888XXXXXXX in from-myupstream (domain 10.1.1.16)
list_route: hop: <sip:A.B.C.D:5060>
pbxtel-01*CLI>
<--- Transmitting (no NAT) to A.B.C.D:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP A.B.C.D:5060;rport;branch=z9hG4bK1da3823e556c2fbe119b7d12a3c51d33;received=A.B.C.D
From: "1416YYYYYYY" <sip:1416YYYYYYY@A.B.C.D:5060;user=phone>;tag=GR52RWG346-34
To: "1888XXXXXXX" <sip:1888XXXXXXX@10.1.1.16:5060>
Call-ID: 110832659@A.B.C.D
CSeq: 1 INVITE
User-Agent: Comwave PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:1888XXXXXXX@10.1.1.16>
Content-Length: 0


<------------>
    -- Executing [1888XXXXXXX@from-myupstream:1] Set("SIP/5060-a02541c0", "IVR_CONTEXT=ivr-meetme") in new stack
    -- Executing [1888XXXXXXX@from-myupstream:2] Ringing("SIP/5060-a02541c0", "") in new stack
pbxtel-01*CLI>
<--- Transmitting (no NAT) to A.B.C.D:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP A.B.C.D:5060;rport;branch=z9hG4bK1da3823e556c2fbe119b7d12a3c51d33;received=A.B.C.D
From: "1416YYYYYYY" <sip:1416YYYYYYY@A.B.C.D:5060;user=phone>;tag=GR52RWG346-34
To: "1888XXXXXXX" <sip:1888XXXXXXX@10.1.1.16:5060>;tag=as5d446137
Call-ID: 110832659@A.B.C.D
CSeq: 1 INVITE
User-Agent: Comwave PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:1888XXXXXXX@10.1.1.16>
Content-Length: 0


<------------>
    -- Executing [1888XXXXXXX@from-myupstream:3] Macro("SIP/5060-a02541c0", "pstncid") in new stack
    -- Executing [s@macro-pstncid:1] GotoIf("SIP/5060-a02541c0", "0?nocid") in new stack
    -- Executing [s@macro-pstncid:2] GotoIf("SIP/5060-a02541c0", "1?nocid") in new stack
    -- Goto (macro-pstncid,s,8)
    -- Executing [s@macro-pstncid:8] NoOp("SIP/5060-a02541c0", "") in new stack
    -- Executing [1888XXXXXXX@from-myupstream:4] Wait("SIP/5060-a02541c0", "2") in new stack
    -- Zap/85-1 is ringing
    -- Executing [1888XXXXXXX@from-myupstream:5] Answer("SIP/5060-a02541c0", "") in new stack
Audio is at 10.1.1.16 port 60746
Adding codec 0x4 (ulaw) to SDP
pbxtel-01*CLI>
<--- Reliably Transmitting (no NAT) to A.B.C.D:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP A.B.C.D:5060;rport;branch=z9hG4bK1da3823e556c2fbe119b7d12a3c51d33;received=A.B.C.D
From: "1416YYYYYYY" <sip:1416YYYYYYY@A.B.C.D:5060;user=phone>;tag=GR52RWG346-34
To: "1888XXXXXXX" <sip:1888XXXXXXX@10.1.1.16:5060>;tag=as5d446137
Call-ID: 110832659@A.B.C.D
CSeq: 1 INVITE
User-Agent: Comwave PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:1888XXXXXXX@10.1.1.16>
Content-Type: application/sdp
Content-Length: 176

v=0
o=root 6380 6380 IN IP4 10.1.1.16
s=session
c=IN IP4 10.1.1.16
t=0 0
m=audio 60746 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
    -- Executing [1888XXXXXXX@from-myupstream:6] Goto("SIP/5060-a02541c0", "ivr-meetme|s|1") in new stack
    -- Goto (ivr-meetme,s,1)
    -- Executing [s@ivr-meetme:1] Set("SIP/5060-a02541c0", "TIMEOUT(digit)=2") in new stack
    -- Digit timeout set to 2
    -- Executing [s@ivr-meetme:2] Set("SIP/5060-a02541c0", "ATTEMPTS=1") in new stack
    -- Executing [s@ivr-meetme:3] Set("SIP/5060-a02541c0", "MAX_ATTEMPTS=3") in new stack
    -- Executing [s@ivr-meetme:4] BackGround("SIP/5060-a02541c0", "welcome_to_conf_bridge") in new stack
    -- <SIP/5060-a02541c0> Playing 'welcome_to_conf_bridge' (language 'en')
pbxtel-01*CLI>
<--- SIP read from A.B.C.D:5060 --->
ACK sip:1888XXXXXXX@10.1.1.16 SIP/2.0
Via: SIP/2.0/UDP A.B.C.D:5060;rport;branch=z9hG4bK54430b8fc9db6214771f31bb733d5aed
From: "1416YYYYYYY" <sip:1416YYYYYYY@A.B.C.D:5060;user=phone>;tag=GR52RWG346-34
To: "1888XXXXXXX" <sip:1888XXXXXXX@10.1.1.16:5060>;tag=as5d446137
Call-ID: 110832659@A.B.C.D
CSeq: 1 ACK
Contact: <sip:A.B.C.D:5060>
Remote-Party-ID: "1416YYYYYYY" <sip:1416YYYYYYY@A.B.C.D:5060>;party=calling;screen=yes;Privacy=off
max-forwards: 70
Allow: ACK, NOTIFY, OPTIONS, REFER, INFO, BYE, CANCEL, INVITE


<------------->
--- (10 headers 0 lines) ---
    -- Zap/85-1 answered SIP/427-9fda1e30
    -- Executing [s@ivr-meetme:5] GotoIf("SIP/5060-a02541c0", "1?top") in new stack
    -- Goto (ivr-meetme,s,9)
    -- Executing [s@ivr-meetme:9] BackGround("SIP/5060-a02541c0", "enter_conf_number") in new stack
    -- <SIP/5060-a02541c0> Playing 'enter_conf_number' (language 'en')
pbxtel-01*CLI>
<--- SIP read from A.B.C.D:5060 --->
INFO sip:1888XXXXXXX@10.1.1.16 SIP/2.0
Via: SIP/2.0/UDP A.B.C.D:5060;rport;branch=z9hG4bKb6a8c185a2c310e96787cd71a4659057
From: "1416YYYYYYY" <sip:1416YYYYYYY@A.B.C.D:5060;user=phone>;tag=GR52RWG346-34
To: "1888XXXXXXX" <sip:1888XXXXXXX@10.1.1.16:5060>;tag=as5d446137
Call-ID: 110832659@A.B.C.D
CSeq: 2 INFO
Contact: "1416YYYYYYY" <sip:1416YYYYYYY@A.B.C.D:5060;user=phone>;tag=GR52RWG346-34
Remote-Party-ID: "1416YYYYYYY" <sip:1416YYYYYYY@A.B.C.D:5060>;party=calling;screen=yes;Privacy=off
max-forwards: 70
Allow: ACK, NOTIFY, OPTIONS, REFER, INFO, BYE, CANCEL, INVITE
Content-Type: application/dtmf-relay
Content-Length:    24

signal=1
duration=100

<------------->
--- (12 headers 2 lines) ---
Receiving INFO!
* DTMF-relay event received: 1

<--- Transmitting (no NAT) to A.B.C.D:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP A.B.C.D:5060;rport;branch=z9hG4bKb6a8c185a2c310e96787cd71a4659057;received=A.B.C.D
rom: "1416YYYYYYY" <sip:1416YYYYYYY@A.B.C.D:5060;user=phone>;tag=GR52RWG346-34
To: "1888XXXXXXX" <sip:1888XXXXXXX@10.1.1.16:5060>;tag=as5d446137
Call-ID: 110832659@A.B.C.D
CSeq: 2 INFO
User-Agent: Comwave PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:1888XXXXXXX@10.1.1.16>
Content-Length: 0


<------------>
pbxtel-01*CLI>
<--- SIP read from A.B.C.D:5060 --->
INFO sip:1888XXXXXXX@10.1.1.16 SIP/2.0
Via: SIP/2.0/UDP A.B.C.D:5060;rport;branch=z9hG4bK08fe093368e1970f9e3b696a1e100c5b
From: "1416YYYYYYY" <sip:1416YYYYYYY@A.B.C.D:5060;user=phone>;tag=GR52RWG346-34
To: "1888XXXXXXX" <sip:1888XXXXXXX@10.1.1.16:5060>;tag=as5d446137
Call-ID: 110832659@A.B.C.D
CSeq: 3 INFO
Contact: "1416YYYYYYY" <sip:1416YYYYYYY@A.B.C.D:5060;user=phone>;tag=GR52RWG346-34
Remote-Party-ID: "1416YYYYYYY" <sip:1416YYYYYYY@A.B.C.D:5060>;party=calling;screen=yes;Privacy=off
max-forwards: 70
Allow: ACK, NOTIFY, OPTIONS, REFER, INFO, BYE, CANCEL, INVITE
Content-Type: application/dtmf-relay
Content-Length:    24

signal=2
duration=140

<------------->
--- (12 headers 2 lines) ---
Receiving INFO!
* DTMF-relay event received: 2

<--- Transmitting (no NAT) to A.B.C.D:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP A.B.C.D:5060;rport;branch=z9hG4bK08fe093368e1970f9e3b696a1e100c5b;received=A.B.C.D
rom: "1416YYYYYYY" <sip:1416YYYYYYY@A.B.C.D:5060;user=phone>;tag=GR52RWG346-34
To: "1888XXXXXXX" <sip:1888XXXXXXX@10.1.1.16:5060>;tag=as5d446137
Call-ID: 110832659@A.B.C.D
CSeq: 3 INFO
User-Agent: Comwave PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:1888XXXXXXX@10.1.1.16>
Content-Length: 0


<------------>
pbxtel-01*CLI>
<--- SIP read from A.B.C.D:5060 --->
INFO sip:1888XXXXXXX@10.1.1.16 SIP/2.0
Via: SIP/2.0/UDP A.B.C.D:5060;rport;branch=z9hG4bKc79e31e7d8556af1dc62a79b44e01789
From: "1416YYYYYYY" <sip:1416YYYYYYY@A.B.C.D:5060;user=phone>;tag=GR52RWG346-34
To: "1888XXXXXXX" <sip:1888XXXXXXX@10.1.1.16:5060>;tag=as5d446137
Call-ID: 110832659@A.B.C.D
CSeq: 4 INFO
Contact: "1416YYYYYYY" <sip:1416YYYYYYY@A.B.C.D:5060;user=phone>;tag=GR52RWG346-34
Remote-Party-ID: "1416YYYYYYY" <sip:1416YYYYYYY@A.B.C.D:5060>;party=calling;screen=yes;Privacy=off
max-forwards: 70
Allow: ACK, NOTIFY, OPTIONS, REFER, INFO, BYE, CANCEL, INVITE
Content-Type: application/dtmf-relay
Content-Length:    24

signal=3
duration=160

<------------->
--- (12 headers 2 lines) ---
Receiving INFO!
* DTMF-relay event received: 3
pbxtel-01*CLI>
<--- Transmitting (no NAT) to A.B.C.D:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP A.B.C.D:5060;rport;branch=z9hG4bKc79e31e7d8556af1dc62a79b44e01789;received=A.B.C.D
From: "1416YYYYYYY" <sip:1416YYYYYYY@A.B.C.D:5060;user=phone>;tag=GR52RWG346-34
To: "1888XXXXXXX" <sip:1888XXXXXXX@10.1.1.16:5060>;tag=as5d446137
Call-ID: 110832659@A.B.C.D
CSeq: 4 INFO
User-Agent: Comwave PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:1888XXXXXXX@10.1.1.16>
Content-Length: 0


<------------>
pbxtel-01*CLI>
<--- SIP read from A.B.C.D:5060 --->
INFO sip:1888XXXXXXX@10.1.1.16 SIP/2.0
Via: SIP/2.0/UDP A.B.C.D:5060;rport;branch=z9hG4bK1ab1a5e49a40b04f4d384620b782dd0e
From: "1416YYYYYYY" <sip:1416YYYYYYY@A.B.C.D:5060;user=phone>;tag=GR52RWG346-34
To: "1888XXXXXXX" <sip:1888XXXXXXX@10.1.1.16:5060>;tag=as5d446137
Call-ID: 110832659@A.B.C.D
CSeq: 5 INFO
Contact: "1416YYYYYYY" <sip:1416YYYYYYY@A.B.C.D:5060;user=phone>;tag=GR52RWG346-34
Remote-Party-ID: "1416YYYYYYY" <sip:1416YYYYYYY@A.B.C.D:5060>;party=calling;screen=yes;Privacy=off
max-forwards: 70
Allow: ACK, NOTIFY, OPTIONS, REFER, INFO, BYE, CANCEL, INVITE
Content-Type: application/dtmf-relay
Content-Length:    24

signal=4
duration=160

<------------->
--- (12 headers 2 lines) ---
Receiving INFO!
* DTMF-relay event received: 4
pbxtel-01*CLI>
<--- Transmitting (no NAT) to A.B.C.D:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP A.B.C.D:5060;rport;branch=z9hG4bK1ab1a5e49a40b04f4d384620b782dd0e;received=A.B.C.D
From: "1416YYYYYYY" <sip:1416YYYYYYY@A.B.C.D:5060;user=phone>;tag=GR52RWG346-34
To: "1888XXXXXXX" <sip:1888XXXXXXX@10.1.1.16:5060>;tag=as5d446137
Call-ID: 110832659@A.B.C.D
CSeq: 5 INFO
User-Agent: Comwave PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:1888XXXXXXX@10.1.1.16>
Content-Length: 0


<------------>
    -- Executing [s@ivr-meetme:10] WaitExten("SIP/5060-a02541c0", "10") in new stack
    -- Timeout on SIP/5060-a02541c0, going to 't'
    -- Executing [t@ivr-meetme:1] Set("SIP/5060-a02541c0", "ATTEMPTS=2") in new stack
    -- Executing [t@ivr-meetme:2] Goto("SIP/5060-a02541c0", "s|check-attempts") in new stack
    -- Goto (ivr-meetme,s,5)
    -- Executing [s@ivr-meetme:5] GotoIf("SIP/5060-a02541c0", "1?top") in new stack
    -- Goto (ivr-meetme,s,9)
    -- Executing [s@ivr-meetme:9] BackGround("SIP/5060-a02541c0", "enter_conf_number") in new stack
    -- <SIP/5060-a02541c0> Playing 'enter_conf_number' (language 'en')
  == Spawn extension (macro-dialout, s, 28) exited non-zero on 'SIP/427-9fda1e30' in macro 'dialout'
  == Spawn extension (macro-dialout, s, 28) exited non-zero on 'SIP/427-9fda1e30'
    -- Executing [h@macro-dialout:1] Macro("SIP/427-9fda1e30", "loghangupcause") in new stack
    -- Executing [s@macro-loghangupcause:1] GotoIf("SIP/427-9fda1e30", "0?nocause") in new stack
    -- Executing [s@macro-loghangupcause:2] Set("SIP/427-9fda1e30", "CDR(userfield)=Hangupcause:16") in new stack
    -- Executing [s@macro-loghangupcause:3] Goto("SIP/427-9fda1e30", "end") in new stack
    -- Goto (macro-loghangupcause,s,5)
    -- Executing [s@macro-loghangupcause:5] NoOp("SIP/427-9fda1e30", "") in new stack
pbxtel-01*CLI>
<--- SIP read from A.B.C.D:5060 --->
BYE sip:1888XXXXXXX@10.1.1.16 SIP/2.0
Via: SIP/2.0/UDP A.B.C.D:5060;rport;branch=z9hG4bKac94a21435e4567d76faf530e709e029
From: "1416YYYYYYY" <sip:1416YYYYYYY@A.B.C.D:5060;user=phone>;tag=GR52RWG346-34
To: "1888XXXXXXX" <sip:1888XXXXXXX@10.1.1.16:5060>;tag=as5d446137
Call-ID: 110832659@A.B.C.D
CSeq: 6 BYE
Remote-Party-ID: "1416YYYYYYY" <sip:1416YYYYYYY@A.B.C.D:5060>;party=calling;screen=yes;Privacy=off
max-forwards: 70
Allow: ACK, NOTIFY, OPTIONS, REFER, INFO, BYE, CANCEL, INVITE


<------------->
--- (9 headers 0 lines) ---
Sending to A.B.C.D : 5060 (NAT)

<--- Transmitting (NAT) to A.B.C.D:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP A.B.C.D:5060;rport;branch=z9hG4bKac94a21435e4567d76faf530e709e029;received=A.B.C.D
From: "1416YYYYYYY" <sip:1416YYYYYYY@A.B.C.D:5060;user=phone>;tag=GR52RWG346-34
To: "1888XXXXXXX" <sip:1888XXXXXXX@10.1.1.16:5060>;tag=as5d446137
Call-ID: 110832659@A.B.C.D
CSeq: 6 BYE
User-Agent: Comwave PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:1888XXXXXXX@10.1.1.16>
Content-Length: 0


<------------>
  == Spawn extension (ivr-meetme, s, 9) exited non-zero on 'SIP/5060-a02541c0'
Really destroying SIP dialog '110832659@A.B.C.D' Method: BYE
pbxtel-01*CLI> sip no debug
SIP Debugging Disabled
pbxtel-01*CLI> core set verbose 0
Verbosity is now OFF
pbxtel-01*CLI>