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---   onNewCallCreated b7504400: ooh323c_11
+++   onNewCallCreated ooh323c_11
---   ooh323_onReceivedSetup ooh323c_11
---   ooh323_alloc
+++   ooh323_alloc
---   find_user: 9001112233 [mypbx], 10.20.182.78
+++   find_user
        Adding capabilities to call(incoming, ooh323c_11)
        Adding g729A capability to call(incoming, ooh323c_11)
        Adding g729 capability to call(incoming, ooh323c_11)
        Adding g729B capability to call(incoming, ooh323c_11)
---   configure_local_rtp
+++   configure_local_rtp
---   ooh323_new - uplink-h323, 0
+++   h323_new
+++   ooh323_onReceivedSetup - Determined context from_uplink, extension 1224499
    -- Executing [1224499@from_uplink:1] Answer("OOH323/uplink-h323-10", "") in new stack
--- ooh323_answer
+++ ooh323_answer
----- ooh323_indicate -1 on call ooh323c_11
++++  ooh323_indicate -1 on ooh323c_11
    -- Executing [1224499@from_uplink:2] Dial("OOH323/uplink-h323-10", "SIP/testuser") in new stack
  == Using SIP RTP CoS mark 5
Audio is at 15646
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 90011122336.2:55726:
INVITE sip:testuser@172.16.0.137:5060 SIP/2.0
Via: SIP/2.0/UDP 10.20.182.103:5060;branch=z9hG4bK714357be;rport
Max-Forwards: 70
From: "9001112233 [mypbx]" <sip:9001112233@10.20.182.103>;tag=as0e6552ec
To: <sip:testuser@172.16.0.137:5060>
Contact: <sip:9001112233@10.20.182.103:5060>
Call-ID: 7af4263648279b276678d03372793143@10.20.182.103:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.15.1
Date: Tue, 23 Oct 2012 17:38:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 291

v=0
o=root 1323698518 1323698518 IN IP4 10.20.182.103
s=Asterisk PBX 1.8.15.1
c=IN IP4 10.20.182.103
t=0 0
m=audio 15646 RTP/AVP 18 101
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

---
    -- Called SIP/testuser
----- ooh323_indicate 22 on call ooh323c_11
++++  ooh323_indicate 22 on ooh323c_11
----- ooh323_indicate 22 on call ooh323c_11
++++  ooh323_indicate 22 on ooh323c_11

<--- SIP read from UDP:90011122336.2:55726 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.20.182.103:5060;branch=z9hG4bK714357be;rport
From: "9001112233 [mypbx]" <sip:9001112233@10.20.182.103>;tag=as0e6552ec
To: <sip:testuser@172.16.0.137:5060>;tag=1c2029375717
Call-ID: 7af4263648279b276678d03372793143@10.20.182.103:5060
CSeq: 102 INVITE
Supported: em,timer,replaces,path,early-session,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Audiocodes-Sip-Gateway-MP-114 FXS/v.6.40A.011.008
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---

<--- SIP read from UDP:90011122336.2:55726 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.20.182.103:5060;branch=z9hG4bK714357be;rport
From: "9001112233 [mypbx]" <sip:9001112233@10.20.182.103>;tag=as0e6552ec
To: <sip:testuser@172.16.0.137:5060>;tag=1c2029375717
Call-ID: 7af4263648279b276678d03372793143@10.20.182.103:5060
CSeq: 102 INVITE
Contact: <sip:testuser@172.16.0.137:5060>
Supported: em,timer,replaces,path,early-session,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Audiocodes-Sip-Gateway-MP-114 FXS/v.6.40A.011.008
Content-Type: application/sdp
Content-Length: 256

v=0
o=AudiocodesGW 2029396522 2029396385 IN IP4 172.16.0.137
s=Phone-Call
c=IN IP4 172.16.0.137
t=0 0
m=audio 6030 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
<------------->
--- (12 headers 12 lines) ---
list_route: hop: <sip:testuser@172.16.0.137:5060>
Found RTP audio format 18
Found RTP audio format 101
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - 0x80000008010e (gsm|ulaw|alaw|g729|h263|testlaw), peer - audio=0x100 (g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x100 (g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.16.0.137:6030
    -- SIP/testuser-0000000a is making progress passing it to OOH323/uplink-h323-10
----- ooh323_indicate 14 on call ooh323c_11
++++  ooh323_indicate 14 on ooh323c_11
---   onCallEstablished ooh323c_11
---   find_call
+++   find_call
+++   onCallEstablished ooh323c_11
---   onCallCleared ooh323c_11
---   find_call
+++   find_call
+++   onCallCleared
Scheduling destruction of SIP dialog '7af4263648279b276678d03372793143@10.20.182.103:5060' in 32000 ms (Method: INVITE)
Reliably Transmitting (NAT) to 90011122336.2:55726:
CANCEL sip:testuser@172.16.0.137:5060 SIP/2.0
Via: SIP/2.0/UDP 10.20.182.103:5060;branch=z9hG4bK714357be;rport
Max-Forwards: 70
From: "9001112233 [mypbx]" <sip:9001112233@10.20.182.103>;tag=as0e6552ec
To: <sip:testuser@172.16.0.137:5060>
Call-ID: 7af4263648279b276678d03372793143@10.20.182.103:5060
CSeq: 102 CANCEL
User-Agent: Asterisk PBX 1.8.15.1
Content-Length: 0


---
Scheduling destruction of SIP dialog '7af4263648279b276678d03372793143@10.20.182.103:5060' in 32000 ms (Method: INVITE)
  == Spawn extension (from_uplink, 1224499, 2) exited non-zero on 'OOH323/uplink-h323-10'
---   ooh323_hangup
+++   ooh323_hangup

<--- SIP read from UDP:90011122336.2:55726 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 10.20.182.103:5060;branch=z9hG4bK714357be;rport
From: "9001112233 [mypbx]" <sip:9001112233@10.20.182.103>;tag=as0e6552ec
To: <sip:testuser@172.16.0.137:5060>;tag=1c2029375717
Call-ID: 7af4263648279b276678d03372793143@10.20.182.103:5060
CSeq: 102 INVITE
Supported: em,timer,replaces,path,early-session,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Audiocodes-Sip-Gateway-MP-114 FXS/v.6.40A.011.008
Reason: SIP ;cause=487 ;text="487 Request Terminated"
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Transmitting (NAT) to 90011122336.2:55726:
ACK sip:testuser@172.16.0.137:5060 SIP/2.0
Via: SIP/2.0/UDP 10.20.182.103:5060;branch=z9hG4bK714357be;rport
Max-Forwards: 70
From: "9001112233 [mypbx]" <sip:9001112233@10.20.182.103>;tag=as0e6552ec
To: <sip:testuser@172.16.0.137:5060>;tag=1c2029375717
Contact: <sip:9001112233@10.20.182.103:5060>
Call-ID: 7af4263648279b276678d03372793143@10.20.182.103:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.15.1
Content-Length: 0


---
Scheduling destruction of SIP dialog '7af4263648279b276678d03372793143@10.20.182.103:5060' in 32000 ms (Method: INVITE)

<--- SIP read from UDP:90011122336.2:55726 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.20.182.103:5060;branch=z9hG4bK714357be;rport
From: "9001112233 [mypbx]" <sip:9001112233@10.20.182.103>;tag=as0e6552ec
To: <sip:testuser@172.16.0.137:5060>;tag=1c2029375717
Call-ID: 7af4263648279b276678d03372793143@10.20.182.103:5060
CSeq: 102 CANCEL
Contact: <sip:testuser@172.16.0.137:5060>
Supported: em,timer,replaces,path,early-session,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Audiocodes-Sip-Gateway-MP-114 FXS/v.6.40A.011.008
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
---   ooh323_destroy
 Destroying uplink-h323
 Destroying ooh323c_11
---   find_user: (null), 10.20.182.78
+++   find_user
+++   ooh323_destroy