Possible Major Bug: Reinvites with Multiple Switches

Using Asterisk 1.2.18
I am trying to determine why we are experiencing a call problem with Asterisk and our SIP provider ThinkTel.

We are connected in the following manner:
––––

We have canreinvite=yes and nat=no on all ends.

The issue that we are having a problem with is that asterisk has MISSING data execution to properly transfer the RTP Streams to the proper destinations. Because of this, it is keeping the Gateway data coming into the Asterisk Box.

Here is the breakdown of the callpath

 UA         Asterisk        OpenSER       MetaSwitch      VoIP2PSTNGateway
> 102 Invite >>
<<<<< Trying  <
               > Invite 102 >>>
               <<<<<<<< 183 Session Progress <
<<<<<< 183 SP <
               <<<<<<<<<<< 200 OK 102 INVITE <
               > ACK >>>>>>>>>>
<< 200 OK 102 <
               > 103 Invite >>>
> 102 ACK >>>>>
<< 102 Invite < ;reinvite rtp stream to Gateway
>> RTP STREAM to VOIP 2 PSTN >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>
               <<<<<<<<<<< 200 OK 103 INVITE <
               > 103 ACK >>>>>>
**** Missing Step here between Asterisk and OpenSER/Metaswitch***
> 200 OK 102 >>
<<<<< 102 ACK <

Does anyone know what just happened?
Answer: Asterisk just left 102 Call Sequence from Asterisk to OpenSER up in limbo.

How you ask? Well where the Missing step is, it should have shut down the 102 INVITE sequence between Asterisk to OpenSER to MetaSwitch to Gateway.

Not sure who is currently at fault (Asterisk or OpenSER/MetaSwitch) but would it not be logical to keep the INVITE 102 running on both ends instead of increasing the value to 103?

Attached below is my SIP and RTP Stream data for you guys to analyze. I hope this helps to bring about a fix to this scenario.

I hope someone can offer a patch for this issue too.


pbx*CLI> sip debug
SIP Debugging enabled
<-- SIP read from xxx.xxx.81.252:5062:
INVITE sip:12025551212@pbx.mydomain.com SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.81.252:5062;branch=z9hG4bK-417ce004
From: Anonymous <sip:1003@pbx.mydomain.com>;tag=c58e1bc7cf2df763o0
To: <sip:12025551212@pbx.mydomain.com>
Call-ID: 72786e97-e96342d3@localhost
CSeq: 101 INVITE
Max-Forwards: 70
Contact: Anonymous <sip:1003@xxx.xxx.81.252:5062>
Expires: 240
User-Agent: WRTP54G-3.1.22
Content-Length: 290
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces
Content-Type: application/sdp

v=0
o=- 7669247 7669247 IN IP4 xxx.xxx.81.252
s=-
c=IN IP4 xxx.xxx.81.252
t=0 0
m=audio 16448 RTP/AVP 18 0 8 100 101
a=rtpmap:18 G729a/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:100 NSE/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:40
a=sendrecv

--- (14 headers 14 lines) ---
Using INVITE request as basis request - 72786e97-e96342d3@localhost
Sending to xxx.xxx.81.252 : 5062 (non-NAT)
Reliably Transmitting (no NAT) to xxx.xxx.81.252:5062:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP xxx.xxx.81.252:5062;branch=z9hG4bK-417ce004;received=xxx.xxx.81.252
From: Anonymous <sip:1003@pbx.mydomain.com>;tag=c58e1bc7cf2df763o0
To: <sip:12025551212@pbx.mydomain.com>;tag=as6ce39735
Call-ID: 72786e97-e96342d3@localhost
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="12408c6a"
Content-Length: 0


---
Scheduling destruction of call '72786e97-e96342d3@localhost' in 15000 ms
Found user '1003'
<-- SIP read from xxx.xxx.81.252:5062:
ACK sip:12025551212@pbx.mydomain.com SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.81.252:5062;branch=z9hG4bK-417ce004
From: Anonymous <sip:1003@pbx.mydomain.com>;tag=c58e1bc7cf2df763o0
To: <sip:12025551212@pbx.mydomain.com>;tag=as6ce39735
Call-ID: 72786e97-e96342d3@localhost
CSeq: 101 ACK
Max-Forwards: 70
Contact: Anonymous <sip:1003@xxx.xxx.81.252:5062>
User-Agent: WRTP54G-3.1.22
Content-Length: 0


--- (10 headers 0 lines) ---
<-- SIP read from xxx.xxx.81.252:5062:
INVITE sip:12025551212@pbx.mydomain.com SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.81.252:5062;branch=z9hG4bK-5a110c5c
From: Anonymous <sip:1003@pbx.mydomain.com>;tag=c58e1bc7cf2df763o0
To: <sip:12025551212@pbx.mydomain.com>
Call-ID: 72786e97-e96342d3@localhost
CSeq: 102 INVITE
Max-Forwards: 70
Proxy-Authorization: Digest 

username="1003",realm="asterisk",nonce="12408c6a",uri="sip:12025551212@pbx.mydomain.com",algorithm=MD5,response="837

6b57939a38f5c50bc31f99feddc7e"
Contact: Anonymous <sip:1003@xxx.xxx.81.252:5062>
Expires: 240
User-Agent: WRTP54G-3.1.22
Content-Length: 290
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces
Content-Type: application/sdp

v=0
o=- 7669247 7669247 IN IP4 xxx.xxx.81.252
s=-
c=IN IP4 xxx.xxx.81.252
t=0 0
m=audio 16448 RTP/AVP 18 0 8 100 101
a=rtpmap:18 G729a/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:100 NSE/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:40
a=sendrecv

--- (15 headers 14 lines) ---
Using INVITE request as basis request - 72786e97-e96342d3@localhost
Sending to xxx.xxx.81.252 : 5062 (non-NAT)
Found user '1003'
Found RTP audio format 18
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 100
Found RTP audio format 101
Peer audio RTP is at port xxx.xxx.81.252:16448
Found description format G729a
Found description format PCMU
Found description format PCMA
Found description format NSE
Found description format telephone-event
Capabilities: us - 0x10c (ulaw|alaw|g729), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x10c 

(ulaw|alaw|g729)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for 12025551212 in from-internal (domain pbx.mydomain.com)
list_route: hop: <sip:1003@xxx.xxx.81.252:5062>
Transmitting (no NAT) to xxx.xxx.81.252:5062:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP xxx.xxx.81.252:5062;branch=z9hG4bK-5a110c5c;received=xxx.xxx.81.252
From: Anonymous <sip:1003@pbx.mydomain.com>;tag=c58e1bc7cf2df763o0
To: <sip:12025551212@pbx.mydomain.com>
Call-ID: 72786e97-e96342d3@localhost
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:12025551212@xxx.xxx.91.14>
Content-Length: 0


---
We're at xxx.xxx.91.14 port 17682
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
14 headers, 11 lines
Reliably Transmitting (no NAT) to xxx.xxx.161.67:5060:
INVITE sip:12025551212@xxx.xxx.161.67 SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.91.14:5060;branch=z9hG4bK4d616a04;rport
From: "Cyberglobe" <sip:15143423883@xxx.xxx.91.14>;tag=as6f130748
To: <sip:12025551212@xxx.xxx.161.67>
Contact: <sip:15143423883@xxx.xxx.91.14>
Call-ID: 0ea661a50aecf44b6fec583b5c7c5688@xxx.xxx.91.14
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "Cyberglobe" <sip:15143423883@xxx.xxx.91.14>;privacy=off;screen=no
Date: Fri, 15 Jun 2007 00:16:11 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 241

v=0
o=root 24191 24191 IN IP4 xxx.xxx.91.14
s=session
c=IN IP4 xxx.xxx.91.14
t=0 0
m=audio 17682 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 - - - -

---
<-- SIP read from xxx.xxx.161.67:5060:
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP xxx.xxx.91.14:5060;branch=z9hG4bK4d616a04;rport=5060
From: "Cyberglobe" <sip:15143423883@xxx.xxx.91.14>;tag=as6f130748
To: <sip:12025551212@xxx.xxx.161.67>
Call-ID: 0ea661a50aecf44b6fec583b5c7c5688@xxx.xxx.91.14
CSeq: 102 INVITE
Server: OpenSer (1.0.0-tls (x86_64/linux))
Content-Length: 0
Warning: 392 xxx.xxx.161.67:5060 "Noisy feedback tells:  pid=31963 req_src_ip=xxx.xxx.91.14 req_src_port=5060 

in_uri=sip:12025551212@xxx.xxx.161.67 out_uri=sip:12025551212@xxx.xxx.161.101:5060 via_cnt==1"


--- (9 headers 0 lines) ---
<-- SIP read from xxx.xxx.161.67:5060:
SIP/2.0 183 Session Progress
Call-ID: 0ea661a50aecf44b6fec583b5c7c5688@xxx.xxx.91.14
CSeq: 102 INVITE
From: "Cyberglobe" <sip:15143423883@xxx.xxx.91.14>;tag=as6f130748
To: <sip:12025551212@xxx.xxx.161.67>;tag=xxx.xxx.161.101+1+48b992+fb2d5c60
Via: SIP/2.0/UDP xxx.xxx.91.14:5060;branch=z9hG4bK4d616a04;rport=5060
Server: DC-SIP/2.0
Organization: MetaSwitch
Supported: 100rel
Record-Route: <sip:12025551212@xxx.xxx.161.67;ftag=as6f130748;lr=on>
Contact: <sip:12025551212@xxx.xxx.161.101:5060>
Content-Length: 185
Content-Type: application/sdp

v=0
o=- 750565946 750565946 IN IP4 xxx.xxx.161.105
s=-
c=IN IP4 xxx.xxx.161.99
t=0 0
m=audio 50156 RTP/AVP 18 0 101
a=rtpmap:101 telephone-event/8000
a=ptime:20
a=nortpproxy:yes

--- (13 headers 9 lines) ---
Found RTP audio format 18
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port xxx.xxx.161.99:50156
Found description format telephone-event
Capabilities: us - 0x100 (g729), peer - audio=0x104 (ulaw|g729)/video=0x0 (nothing), combined - 0x100 (g729)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
We're at xxx.xxx.91.14 port 11854
Adding codec 0x100 (g729) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Transmitting (no NAT) to xxx.xxx.81.252:5062:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP xxx.xxx.81.252:5062;branch=z9hG4bK-5a110c5c;received=xxx.xxx.81.252
From: Anonymous <sip:1003@pbx.mydomain.com>;tag=c58e1bc7cf2df763o0
To: <sip:12025551212@pbx.mydomain.com>;tag=as72b45889
Call-ID: 72786e97-e96342d3@localhost
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:12025551212@xxx.xxx.91.14>
Content-Type: application/sdp
Content-Length: 289

v=0
o=root 24191 24191 IN IP4 xxx.xxx.91.14
s=session
c=IN IP4 xxx.xxx.91.14
t=0 0
m=audio 11854 RTP/AVP 18 0 8 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
Got RTP packet from xxx.xxx.161.99:50156 (type 18, seq 0, ts 80, len 20)
Sent RTP packet to xxx.xxx.81.252:16448 (type 18, seq 60635, ts 0, len 20)
Got RTP packet from xxx.xxx.161.99:50156 (type 18, seq 1, ts 240, len 20)
Sent RTP packet to xxx.xxx.81.252:16448 (type 18, seq 60636, ts 160, len 20)
Got RTP packet from xxx.xxx.161.99:50156 (type 18, seq 2, ts 400, len 20)
Sent RTP packet to xxx.xxx.81.252:16448 (type 18, seq 60637, ts 320, len 20)
Got RTP packet from xxx.xxx.161.99:50156 (type 18, seq 3, ts 560, len 20)
Sent RTP packet to xxx.xxx.81.252:16448 (type 18, seq 60638, ts 480, len 20)
Got RTP packet from xxx.xxx.81.252:16448 (type 18, seq 10867, ts 386750631, len 40)
Sent RTP packet to xxx.xxx.161.99:50156 (type 18, seq 13164, ts 64, len 20)
Sent RTP packet to xxx.xxx.161.99:50156 (type 18, seq 13165, ts 224, len 20)
Got RTP packet from xxx.xxx.161.99:50156 (type 18, seq 4, ts 720, len 20)
Sent RTP packet to xxx.xxx.81.252:16448 (type 18, seq 60639, ts 640, len 20)
Got RTP packet from xxx.xxx.161.99:50156 (type 18, seq 5, ts 880, len 20)
Sent RTP packet to xxx.xxx.81.252:16448 (type 18, seq 60640, ts 800, len 20)
Got RTP packet from xxx.xxx.81.252:16448 (type 18, seq 10868, ts 386750871, len 40)
Sent RTP packet to xxx.xxx.161.99:50156 (type 18, seq 13166, ts 304, len 20)
Sent RTP packet to xxx.xxx.161.99:50156 (type 18, seq 13167, ts 464, len 20)
Got RTP packet from xxx.xxx.161.99:50156 (type 18, seq 6, ts 1040, len 20)
Sent RTP packet to xxx.xxx.81.252:16448 (type 18, seq 60641, ts 960, len 20)
....
Got RTP packet from xxx.xxx.81.252:16448 (type 18, seq 11035, ts 386804311, len 40)
Sent RTP packet to xxx.xxx.161.99:50156 (type 18, seq 13500, ts 53744, len 20)
Sent RTP packet to xxx.xxx.161.99:50156 (type 18, seq 13501, ts 53904, len 20)
Got RTP packet from xxx.xxx.161.99:50156 (type 18, seq 340, ts 54480, len 20)
Sent RTP packet to xxx.xxx.81.252:16448 (type 18, seq 60972, ts 54400, len 20)
Got RTP packet from xxx.xxx.161.99:50156 (type 18, seq 341, ts 54640, len 20)
Sent RTP packet to xxx.xxx.81.252:16448 (type 18, seq 60973, ts 54560, len 20)
Got RTP packet from xxx.xxx.81.252:16448 (type 18, seq 11036, ts 386804631, len 40)
Sent RTP packet to xxx.xxx.161.99:50156 (type 18, seq 13502, ts 54064, len 20)
Sent RTP packet to xxx.xxx.161.99:50156 (type 18, seq 13503, ts 54224, len 20)
<-- SIP read from xxx.xxx.161.67:5060:
SIP/2.0 200 OK
Call-ID: 0ea661a50aecf44b6fec583b5c7c5688@xxx.xxx.91.14
CSeq: 102 INVITE
From: "Cyberglobe" <sip:15143423883@xxx.xxx.91.14>;tag=as6f130748
To: <sip:12025551212@xxx.xxx.161.67>;tag=xxx.xxx.161.101+1+48b992+fb2d5c60
Via: SIP/2.0/UDP xxx.xxx.91.14:5060;branch=z9hG4bK4d616a04;rport=5060
Server: DC-SIP/2.0
Organization: MetaSwitch
Allow-Events: message-summary
Allow-Events: refer
Allow-Events: dialog
Allow-Events: line-seize
Allow-Events: presence
Supported: 100rel
Allow: INVITE
Allow: ACK
Allow: CANCEL
Allow: BYE
Allow: REGISTER
Allow: OPTIONS
Allow: PRACK
Allow: UPDATE
Allow: SUBSCRIBE
Allow: NOTIFY
Allow: REFER
Accept-Encoding: identity
Accept: application/sdp
Accept: application/simple-message-summary
Accept: message/sipfrag
Accept: application/isup
Accept: application/x-simple-call-service-info
Accept: multipart/mixed
Record-Route: <sip:12025551212@xxx.xxx.161.67;ftag=as6f130748;lr=on>
Contact: <sip:12025551212@xxx.xxx.161.101:5060>
Content-Length: 185
Content-Type: application/sdp

v=0
o=- 750565946 750565946 IN IP4 xxx.xxx.161.105
s=-
c=IN IP4 xxx.xxx.161.99
t=0 0
m=audio 50156 RTP/AVP 18 0 101
a=rtpmap:101 telephone-event/8000
a=ptime:20
a=nortpproxy:yes

--- (36 headers 9 lines) ---
Found RTP audio format 18
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port xxx.xxx.161.99:50156
Found description format telephone-event
Capabilities: us - 0x100 (g729), peer - audio=0x104 (ulaw|g729)/video=0x0 (nothing), combined - 0x100 (g729)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
list_route: hop: <sip:12025551212@xxx.xxx.161.67;ftag=as6f130748;lr=on>
set_destination: Parsing <sip:12025551212@xxx.xxx.161.67;ftag=as6f130748;lr=on> for address/port to send to
set_destination: set destination to xxx.xxx.161.67, port 5060
Transmitting (no NAT) to xxx.xxx.161.67:5060:
ACK sip:12025551212@xxx.xxx.161.101:5060 SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.91.14:5060;branch=z9hG4bK319b0cd7;rport
Route: <sip:12025551212@xxx.xxx.161.67;ftag=as6f130748;lr=on>
From: "Cyberglobe" <sip:15143423883@xxx.xxx.91.14>;tag=as6f130748
To: <sip:12025551212@xxx.xxx.161.67>;tag=xxx.xxx.161.101+1+48b992+fb2d5c60
Contact: <sip:15143423883@xxx.xxx.91.14>
Call-ID: 0ea661a50aecf44b6fec583b5c7c5688@xxx.xxx.91.14
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "Cyberglobe" <sip:15143423883@xxx.xxx.91.14>;privacy=off;screen=no
Content-Length: 0


---
We're at xxx.xxx.91.14 port 11854
Adding codec 0x100 (g729) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to xxx.xxx.81.252:5062:
SIP/2.0 200 OK
Via: SIP/2.0/UDP xxx.xxx.81.252:5062;branch=z9hG4bK-5a110c5c;received=xxx.xxx.81.252
From: Anonymous <sip:1003@pbx.mydomain.com>;tag=c58e1bc7cf2df763o0
To: <sip:12025551212@pbx.mydomain.com>;tag=as72b45889
Call-ID: 72786e97-e96342d3@localhost
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:12025551212@xxx.xxx.91.14>
Content-Type: application/sdp
Content-Length: 289

v=0
o=root 24191 24192 IN IP4 xxx.xxx.91.14
s=session
c=IN IP4 xxx.xxx.91.14
t=0 0
m=audio 11854 RTP/AVP 18 0 8 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
set_destination: Parsing <sip:12025551212@xxx.xxx.161.67;ftag=as6f130748;lr=on> for address/port to send to
set_destination: set destination to xxx.xxx.161.67, port 5060
We're at xxx.xxx.91.14 port 17682
Adding codec 0x100 (g729) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
15 headers, 13 lines
Reliably Transmitting (no NAT) to xxx.xxx.161.67:5060:
INVITE sip:12025551212@xxx.xxx.161.101:5060 SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.91.14:5060;branch=z9hG4bK50ef08c0;rport
Route: <sip:12025551212@xxx.xxx.161.67;ftag=as6f130748;lr=on>
From: "Cyberglobe" <sip:15143423883@xxx.xxx.91.14>;tag=as6f130748
To: <sip:12025551212@xxx.xxx.161.67>;tag=xxx.xxx.161.101+1+48b992+fb2d5c60
Contact: <sip:15143423883@xxx.xxx.91.14>
Call-ID: 0ea661a50aecf44b6fec583b5c7c5688@xxx.xxx.91.14
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "Cyberglobe" <sip:15143423883@xxx.xxx.91.14>;privacy=off;screen=no
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
X-asterisk-info: SIP re-invite (RTP bridge)
Content-Type: application/sdp
Content-Length: 291

v=0
o=root 24191 24192 IN IP4 xxx.xxx.81.252
s=session
c=IN IP4 xxx.xxx.81.252
t=0 0
m=audio 16448 RTP/AVP 18 0 8 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
Got RTP packet from xxx.xxx.161.99:50156 (type 18, seq 342, ts 54800, len 20)
Sent RTP packet to xxx.xxx.81.252:16448 (type 18, seq 60974, ts 54720, len 20)
Got RTP packet from xxx.xxx.161.99:50156 (type 18, seq 343, ts 54960, len 20)
Sent RTP packet to xxx.xxx.81.252:16448 (type 18, seq 60975, ts 54880, len 20)
Got RTP packet from xxx.xxx.81.252:16448 (type 18, seq 11037, ts 386804951, len 40)
Sent RTP packet to xxx.xxx.161.99:50156 (type 18, seq 13504, ts 54384, len 20)
Sent RTP packet to xxx.xxx.161.99:50156 (type 18, seq 13505, ts 54544, len 20)
Got RTP packet from xxx.xxx.161.99:50156 (type 18, seq 344, ts 55120, len 20)
Sent RTP packet to xxx.xxx.81.252:16448 (type 18, seq 60976, ts 55040, len 20)
Got RTP packet from xxx.xxx.161.99:50156 (type 18, seq 345, ts 55280, len 20)
Sent RTP packet to xxx.xxx.81.252:16448 (type 18, seq 60977, ts 55200, len 20)
Got RTP packet from xxx.xxx.81.252:16448 (type 18, seq 11038, ts 386805271, len 40)
Sent RTP packet to xxx.xxx.161.99:50156 (type 18, seq 13506, ts 54704, len 20)
Sent RTP packet to xxx.xxx.161.99:50156 (type 18, seq 13507, ts 54864, len 20)
Got RTP packet from xxx.xxx.161.99:50156 (type 18, seq 346, ts 55440, len 20)
Sent RTP packet to xxx.xxx.81.252:16448 (type 18, seq 60978, ts 55360, len 20)
<-- SIP read from xxx.xxx.81.252:5062:
ACK sip:12025551212@xxx.xxx.91.14 SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.81.252:5062;branch=z9hG4bK-f5e73b32
From: Anonymous <sip:1003@pbx.mydomain.com>;tag=c58e1bc7cf2df763o0
To: <sip:12025551212@pbx.mydomain.com>;tag=as72b45889
Call-ID: 72786e97-e96342d3@localhost
CSeq: 102 ACK
Max-Forwards: 70
Proxy-Authorization: Digest 

username="1003",realm="asterisk",nonce="12408c6a",uri="sip:12025551212@xxx.xxx.91.14",algorithm=MD5,response="ce942656d75

19ac7ebb49a4fd02584fb"
Contact: Anonymous <sip:1003@xxx.xxx.81.252:5062>
User-Agent: WRTP54G-3.1.22
Content-Length: 0


--- (11 headers 0 lines) ---
set_destination: Parsing <sip:1003@xxx.xxx.81.252:5062> for address/port to send to
set_destination: set destination to xxx.xxx.81.252, port 5062
We're at xxx.xxx.91.14 port 11854
Adding codec 0x100 (g729) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 12 lines
Reliably Transmitting (no NAT) to xxx.xxx.81.252:5062:
INVITE sip:1003@xxx.xxx.81.252:5062 SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.91.14:5060;branch=z9hG4bK6b3b945d;rport
From: <sip:12025551212@pbx.mydomain.com>;tag=as72b45889
To: Anonymous <sip:1003@pbx.mydomain.com>;tag=c58e1bc7cf2df763o0
Contact: <sip:12025551212@xxx.xxx.91.14>
Call-ID: 72786e97-e96342d3@localhost
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
X-asterisk-info: SIP re-invite (RTP bridge)
Content-Type: application/sdp
Content-Length: 265

v=0
o=root 24191 24193 IN IP4 xxx.xxx.161.99
s=session
c=IN IP4 xxx.xxx.161.99
t=0 0
m=audio 50156 RTP/AVP 18 0 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
Got RTP packet from xxx.xxx.161.99:50156 (type 18, seq 347, ts 55600, len 20)
Sent RTP packet to xxx.xxx.81.252:16448 (type 18, seq 60979, ts 55520, len 20)
Got RTP packet from xxx.xxx.81.252:16448 (type 18, seq 11039, ts 386805591, len 30)
Sent RTP packet to xxx.xxx.161.99:50156 (type 18, seq 13508, ts 55024, len 20)
Got RTP packet from xxx.xxx.161.99:50156 (type 18, seq 348, ts 55760, len 20)
Sent RTP packet to xxx.xxx.81.252:16448 (type 18, seq 60980, ts 55680, len 20)
Got RTP packet from xxx.xxx.81.252:16448 (type 18, seq 11040, ts 386805591, len 40)
Sent RTP packet to xxx.xxx.161.99:50156 (type 18, seq 13509, ts 55184, len 20)
Sent RTP packet to xxx.xxx.161.99:50156 (type 18, seq 13510, ts 55344, len 20)
Got RTP packet from xxx.xxx.161.99:50156 (type 18, seq 349, ts 55920, len 20)
Sent RTP packet to xxx.xxx.81.252:16448 (type 18, seq 60981, ts 55840, len 20)
<-- SIP read from xxx.xxx.161.67:5060:
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP xxx.xxx.91.14:5060;branch=z9hG4bK50ef08c0;rport=5060
From: "Cyberglobe" <sip:15143423883@xxx.xxx.91.14>;tag=as6f130748
To: <sip:12025551212@xxx.xxx.161.67>;tag=xxx.xxx.161.101+1+48b992+fb2d5c60
Call-ID: 0ea661a50aecf44b6fec583b5c7c5688@xxx.xxx.91.14
CSeq: 103 INVITE
Server: OpenSer (1.0.0-tls (x86_64/linux))
Content-Length: 0
Warning: 392 xxx.xxx.161.67:5060 "Noisy feedback tells:  pid=31961 req_src_ip=xxx.xxx.91.14 req_src_port=5060 

in_uri=sip:12025551212@xxx.xxx.161.101:5060 out_uri=sip:12025551212@xxx.xxx.161.101:5060 via_cnt==1"


--- (9 headers 0 lines) ---
<-- SIP read from xxx.xxx.161.67:5060:
SIP/2.0 200 OK
Call-ID: 0ea661a50aecf44b6fec583b5c7c5688@xxx.xxx.91.14
CSeq: 103 INVITE
From: "Cyberglobe" <sip:15143423883@xxx.xxx.91.14>;tag=as6f130748
To: <sip:12025551212@xxx.xxx.161.67>;tag=xxx.xxx.161.101+1+48b992+fb2d5c60
Via: SIP/2.0/UDP xxx.xxx.91.14:5060;branch=z9hG4bK50ef08c0;rport=5060
Server: DC-SIP/2.0
Organization: MetaSwitch
Allow-Events: message-summary
Allow-Events: refer
Allow-Events: dialog
Allow-Events: line-seize
Allow-Events: presence
Supported: 100rel
Allow: INVITE
Allow: ACK
Allow: CANCEL
Allow: BYE
Allow: REGISTER
Allow: OPTIONS
Allow: PRACK
Allow: UPDATE
Allow: SUBSCRIBE
Allow: NOTIFY
Allow: REFER
Accept-Encoding: identity
Accept: application/sdp
Accept: application/simple-message-summary
Accept: message/sipfrag
Accept: application/isup
Accept: application/x-simple-call-service-info
Accept: multipart/mixed
Record-Route: <sip:12025551212@xxx.xxx.161.67;ftag=as6f130748;lr=on>
Contact: <sip:12025551212@xxx.xxx.161.101:5060>
Content-Length: 185
Content-Type: application/sdp

v=0
o=- 750565946 750565947 IN IP4 xxx.xxx.161.105
s=-
c=IN IP4 xxx.xxx.161.99
t=0 0
m=audio 50156 RTP/AVP 18 101 0
a=rtpmap:101 telephone-event/8000
a=ptime:20
a=nortpproxy:yes

--- (36 headers 9 lines) ---
Found RTP audio format 18
Found RTP audio format 101
Found RTP audio format 0
Peer audio RTP is at port xxx.xxx.161.99:50156
Found description format telephone-event
Capabilities: us - 0x100 (g729), peer - audio=0x104 (ulaw|g729)/video=0x0 (nothing), combined - 0x100 (g729)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Got RTP packet from xxx.xxx.81.252:16448 (type 18, seq 11041, ts 386805911, len 40)
set_destination: Parsing <sip:12025551212@xxx.xxx.161.67;ftag=as6f130748;lr=on> for address/port to send to
set_destination: set destination to xxx.xxx.161.67, port 5060
Transmitting (no NAT) to xxx.xxx.161.67:5060:
ACK sip:12025551212@xxx.xxx.161.101:5060 SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.91.14:5060;branch=z9hG4bK138f4a66;rport
Route: <sip:12025551212@xxx.xxx.161.67;ftag=as6f130748;lr=on>
From: "Cyberglobe" <sip:15143423883@xxx.xxx.91.14>;tag=as6f130748
To: <sip:12025551212@xxx.xxx.161.67>;tag=xxx.xxx.161.101+1+48b992+fb2d5c60
Contact: <sip:15143423883@xxx.xxx.91.14>
Call-ID: 0ea661a50aecf44b6fec583b5c7c5688@xxx.xxx.91.14
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "Cyberglobe" <sip:15143423883@xxx.xxx.91.14>;privacy=off;screen=no
Content-Length: 0


---
<-- SIP read from xxx.xxx.81.252:5062:
SIP/2.0 200 OK
To: Anonymous <sip:1003@pbx.mydomain.com>;tag=c58e1bc7cf2df763o0
From: <sip:12025551212@pbx.mydomain.com>;tag=as72b45889
Call-ID: 72786e97-e96342d3@localhost
CSeq: 102 INVITE
Via: SIP/2.0/UDP xxx.xxx.91.14:5060;branch=z9hG4bK6b3b945d
Contact: Anonymous <sip:1003@xxx.xxx.81.252:5062>
Server: WRTP54G-3.1.22
Content-Length: 269
Content-Type: application/sdp

v=0
o=- 7670124 7670124 IN IP4 xxx.xxx.81.252
s=-
c=IN IP4 xxx.xxx.81.252
t=0 0
m=audio 16448 RTP/AVP 18 100 101
a=rtpmap:18 G729a/8000
a=rtpmap:100 NSE/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:40
a=sendrecv
a=silenceSupp:off - - - -

--- (10 headers 13 lines) ---
Found RTP audio format 18
Found RTP audio format 100
Found RTP audio format 101
Peer audio RTP is at port xxx.xxx.81.252:16448
Found description format G729a
Found description format NSE
Found description format telephone-event
Capabilities: us - 0x10c (ulaw|alaw|g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
list_route: hop: <sip:1003@xxx.xxx.81.252:5062>
set_destination: Parsing <sip:1003@xxx.xxx.81.252:5062> for address/port to send to
set_destination: set destination to xxx.xxx.81.252, port 5062
Transmitting (no NAT) to xxx.xxx.81.252:5062:
ACK sip:1003@xxx.xxx.81.252:5062 SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.91.14:5060;branch=z9hG4bK131484a2;rport
From: <sip:12025551212@pbx.mydomain.com>;tag=as72b45889
To: Anonymous <sip:1003@pbx.mydomain.com>;tag=c58e1bc7cf2df763o0
Contact: <sip:12025551212@xxx.xxx.91.14>
Call-ID: 72786e97-e96342d3@localhost
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
Got RTP packet from xxx.xxx.161.99:50156 (type 18, seq 354, ts 56720, len 20)
Sent RTP packet to xxx.xxx.81.252:16448 (type 18, seq 60984, ts 56640, len 20)
...
Got RTP packet from xxx.xxx.161.99:50156 (type 18, seq 480, ts 76880, len 20)
Sent RTP packet to xxx.xxx.81.252:16448 (type 18, seq 61109, ts 76800, len 20)
<-- SIP read from xxx.xxx.161.67:5060:
BYE sip:15143423883@xxx.xxx.91.14 SIP/2.0
Record-Route: <sip:15143423883@xxx.xxx.161.67;ftag=xxx.xxx.161.101+1+48b992+fb2d5c60;lr=on>
Via: SIP/2.0/UDP xxx.xxx.161.67;branch=z9hG4bKbc1e.de2ea8b6.0
Via: SIP/2.0/UDP xxx.xxx.161.101:5060;rport=5060;branch=z9hG4bK-9d1102726719941f9f08f3ad31d32e0d-xxx.xxx.161.101-1
Allow-Events: message-summary
Allow-Events: refer
Allow-Events: dialog
Allow-Events: line-seize
Allow-Events: presence
Max-Forwards: 69
Call-ID: 0ea661a50aecf44b6fec583b5c7c5688@xxx.xxx.91.14
From: <sip:12025551212@xxx.xxx.161.67>;tag=xxx.xxx.161.101+1+48b992+fb2d5c60
To: "Cyberglobe" <sip:15143423883@xxx.xxx.91.14>;tag=as6f130748
CSeq: 273359792 BYE
Organization: MetaSwitch
Supported: 100rel
Content-Length: 0


--- (17 headers 0 lines) ---
Sending to xxx.xxx.161.67 : 5060 (non-NAT)
Transmitting (no NAT) to xxx.xxx.161.67:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP xxx.xxx.161.67;branch=z9hG4bKbc1e.de2ea8b6.0;received=xxx.xxx.161.67
Via: SIP/2.0/UDP xxx.xxx.161.101:5060;rport=5060;branch=z9hG4bK-9d1102726719941f9f08f3ad31d32e0d-xxx.xxx.161.101-1
Record-Route: <sip:15143423883@xxx.xxx.161.67;ftag=xxx.xxx.161.101+1+48b992+fb2d5c60;lr=on>
From: <sip:12025551212@xxx.xxx.161.67>;tag=xxx.xxx.161.101+1+48b992+fb2d5c60
To: "Cyberglobe" <sip:15143423883@xxx.xxx.91.14>;tag=as6f130748
Call-ID: 0ea661a50aecf44b6fec583b5c7c5688@xxx.xxx.91.14
CSeq: 273359792 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:15143423883@xxx.xxx.91.14>
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing


---
set_destination: Parsing <sip:1003@xxx.xxx.81.252:5062> for address/port to send to
set_destination: set destination to xxx.xxx.81.252, port 5062
We're at xxx.xxx.91.14 port 11854
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 11 lines
Reliably Transmitting (no NAT) to xxx.xxx.81.252:5062:
INVITE sip:1003@xxx.xxx.81.252:5062 SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.91.14:5060;branch=z9hG4bK20bd7956;rport
From: <sip:12025551212@pbx.mydomain.com>;tag=as72b45889
To: Anonymous <sip:1003@pbx.mydomain.com>;tag=c58e1bc7cf2df763o0
Contact: <sip:12025551212@xxx.xxx.91.14>
Call-ID: 72786e97-e96342d3@localhost
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
X-asterisk-info: SIP re-invite (RTP bridge)
Content-Type: application/sdp
Content-Length: 241

v=0
o=root 24191 24194 IN IP4 xxx.xxx.91.14
s=session
c=IN IP4 xxx.xxx.91.14
t=0 0
m=audio 11854 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 - - - -

---
Destroying call '0ea661a50aecf44b6fec583b5c7c5688@xxx.xxx.91.14'
Got RTP packet from xxx.xxx.81.252:16448 (type 18, seq 11108, ts 386827351, len 20)
<-- SIP read from xxx.xxx.81.252:5062:
SIP/2.0 200 OK
To: Anonymous <sip:1003@pbx.mydomain.com>;tag=c58e1bc7cf2df763o0
From: <sip:12025551212@pbx.mydomain.com>;tag=as72b45889
Call-ID: 72786e97-e96342d3@localhost
CSeq: 103 INVITE
Via: SIP/2.0/UDP xxx.xxx.91.14:5060;branch=z9hG4bK20bd7956
Contact: Anonymous <sip:1003@xxx.xxx.81.252:5062>
Server: WRTP54G-3.1.22
Content-Length: 269
Content-Type: application/sdp

v=0
o=- 7670390 7670390 IN IP4 xxx.xxx.81.252
s=-
c=IN IP4 xxx.xxx.81.252
t=0 0
m=audio 16448 RTP/AVP 18 100 101
a=rtpmap:18 G729a/8000
a=rtpmap:100 NSE/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:40
a=sendrecv
a=silenceSupp:off - - - -

--- (10 headers 13 lines) ---
Found RTP audio format 18
Found RTP audio format 100
Found RTP audio format 101
Peer audio RTP is at port xxx.xxx.81.252:16448
Found description format G729a
Found description format NSE
Found description format telephone-event
Capabilities: us - 0x10c (ulaw|alaw|g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
set_destination: Parsing <sip:1003@xxx.xxx.81.252:5062> for address/port to send to
set_destination: set destination to xxx.xxx.81.252, port 5062
Transmitting (no NAT) to xxx.xxx.81.252:5062:
ACK sip:1003@xxx.xxx.81.252:5062 SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.91.14:5060;branch=z9hG4bK64706c88;rport
From: <sip:12025551212@pbx.mydomain.com>;tag=as72b45889
To: Anonymous <sip:1003@pbx.mydomain.com>;tag=c58e1bc7cf2df763o0
Contact: <sip:12025551212@xxx.xxx.91.14>
Call-ID: 72786e97-e96342d3@localhost
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
Got RTP packet from xxx.xxx.81.252:16448 (type 18, seq 11110, ts 386827671, len 40)
...
Got RTP packet from xxx.xxx.81.252:16448 (type 18, seq 11130, ts 386834071, len 40)
12 headers, 0 lines
Reliably Transmitting (no NAT) to 64.15.69.138:5060:
OPTIONS sip:64.15.69.138 SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.91.14:5060;branch=z9hG4bK6245178a;rport
From: "Unknown" <sip:Unknown@xxx.xxx.91.14>;tag=as2283c3c0
To: <sip:64.15.69.138>
Contact: <sip:Unknown@xxx.xxx.91.14>
Call-ID: 1a39d77e7df31e7a369a35573426f7b7@xxx.xxx.91.14
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 15 Jun 2007 00:16:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
12 headers, 0 lines
Reliably Transmitting (no NAT) to 64.15.69.138:5060:
OPTIONS sip:64.15.69.138 SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.91.14:5060;branch=z9hG4bK4dfc63f7;rport
From: "Unknown" <sip:Unknown@xxx.xxx.91.14>;tag=as378b54d8
To: <sip:64.15.69.138>
Contact: <sip:Unknown@xxx.xxx.91.14>
Call-ID: 468afaf64579452c7c57e7060c642441@xxx.xxx.91.14
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 15 Jun 2007 00:16:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
Got RTP packet from xxx.xxx.81.252:16448 (type 18, seq 11131, ts 386834391, len 40)
<-- SIP read from 64.15.69.138:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP xxx.xxx.91.14:5060;branch=z9hG4bK6245178a;received=xxx.xxx.91.14;rport=5060
From: "Unknown" <sip:Unknown@xxx.xxx.91.14>;tag=as2283c3c0
To: <sip:64.15.69.138>;tag=as48d69f99
Call-ID: 1a39d77e7df31e7a369a35573426f7b7@xxx.xxx.91.14
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Accept: application/sdp
Content-Length: 0


--- (10 headers 0 lines) ---
Destroying call '1a39d77e7df31e7a369a35573426f7b7@xxx.xxx.91.14'
Got RTP packet from xxx.xxx.81.252:16448 (type 18, seq 11132, ts 386834711, len 40)
<-- SIP read from 64.15.69.138:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP xxx.xxx.91.14:5060;branch=z9hG4bK4dfc63f7;received=xxx.xxx.91.14;rport=5060
From: "Unknown" <sip:Unknown@xxx.xxx.91.14>;tag=as378b54d8
To: <sip:64.15.69.138>;tag=as7f127957
Call-ID: 468afaf64579452c7c57e7060c642441@xxx.xxx.91.14
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Accept: application/sdp
Content-Length: 0


--- (10 headers 0 lines) ---
Destroying call '468afaf64579452c7c57e7060c642441@xxx.xxx.91.14'
Got RTP packet from xxx.xxx.81.252:16448 (type 18, seq 11133, ts 386835031, len 40)
....
Got RTP packet from xxx.xxx.81.252:16448 (type 18, seq 11140, ts 386837271, len 40)
<-- SIP read from xxx.xxx.81.252:5062:
BYE sip:12025551212@xxx.xxx.91.14 SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.81.252:5062;branch=z9hG4bK-2c7ebf0
From: Anonymous <sip:1003@pbx.mydomain.com>;tag=c58e1bc7cf2df763o0
To: <sip:12025551212@pbx.mydomain.com>;tag=as72b45889
Call-ID: 72786e97-e96342d3@localhost
CSeq: 103 BYE
Max-Forwards: 70
Proxy-Authorization: Digest 

username="1003",realm="asterisk",nonce="12408c6a",uri="sip:12025551212@xxx.xxx.91.14",algorithm=MD5,response="e4441e301a6

3b1f45c9b8b84d98205b5"
User-Agent: WRTP54G-3.1.22
Content-Length: 0


--- (10 headers 0 lines) ---
Sending to xxx.xxx.81.252 : 5062 (non-NAT)
Transmitting (no NAT) to xxx.xxx.81.252:5062:
SIP/2.0 200 OK
Via: SIP/2.0/UDP xxx.xxx.81.252:5062;branch=z9hG4bK-2c7ebf0;received=xxx.xxx.81.252
From: Anonymous <sip:1003@pbx.mydomain.com>;tag=c58e1bc7cf2df763o0
To: <sip:12025551212@pbx.mydomain.com>;tag=as72b45889
Call-ID: 72786e97-e96342d3@localhost
CSeq: 103 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:12025551212@xxx.xxx.91.14>
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing



******* WRTP54G DEBUG INFO  Debug is in Reverse Order.
06-14-2007	20:16:29	Local2.Debug	192.168.15.1	DLG Terminated
06-14-2007	20:16:29	Local7.Debug	192.168.15.1	SIP/2.0 200 OK<013><010>Via: SIP/2.0/UDP 

xxx.xxx.81.252:5062;branch=z9hG4bK-2c7ebf0;received=xxx.xxx.81.252<013><010>From: Anonymous 

<sip:1003@pbx.mydomain.com>;tag=c58e1bc7cf2df763o0<013><010>To: 

<sip:12025551212@pbx.mydomain.com>;tag=as72b45889<013><010>Call-ID: 72786e97-e96342d3@localhost<013><010>CSeq: 103 

BYE<013><010>User-Agent: Asterisk PBX<013><010>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, 

NOTIFY<013><010>Contact: <sip:12025551212@xxx.xxx.91.14><013><010>Content-Length: 0<013><010>X-Asterisk-HangupCause: 

Normal Clearing
06-14-2007	20:16:29	Local0.Info	192.168.15.1	[0:5062]<<xxx.xxx.91.14:5060
06-14-2007	20:16:29	Local7.Debug	192.168.15.1	BYE sip:12025551212@xxx.xxx.91.14 SIP/2.0<013><010>Via: 

SIP/2.0/UDP xxx.xxx.81.252:5062;branch=z9hG4bK-2c7ebf0<013><010>From: Anonymous 

<sip:1003@pbx.mydomain.com>;tag=c58e1bc7cf2df763o0<013><010>To: 

<sip:12025551212@pbx.mydomain.com>;tag=as72b45889<013><010>Call-ID: 72786e97-e96342d3@localhost<013><010>CSeq: 103 

BYE<013><010>Max-Forwards: 70<013><010>Proxy-Authorization: Digest 

username="1003",realm="asterisk",nonce="12408c6a",uri="sip:12025551212@xxx.xxx.91.14",algorithm=MD5,response="e4441e301a6

3b1f45c9b8b84d98205b5"<013><010>User-Agent: WRTP54G-3.1.22<013><010>Content-Length: 0
06-14-2007	20:16:29	Local0.Info	192.168.15.1	[0:5062]->xxx.xxx.91.14:5060
06-14-2007	20:16:27	Local7.Debug	192.168.15.1	ACK sip:1003@xxx.xxx.81.252:5062 SIP/2.0<013><010>Via: 

SIP/2.0/UDP xxx.xxx.91.14:5060;branch=z9hG4bK64706c88;rport<013><010>From: 

<sip:12025551212@pbx.mydomain.com>;tag=as72b45889<013><010>To: Anonymous 

<sip:1003@pbx.mydomain.com>;tag=c58e1bc7cf2df763o0<013><010>Contact: 

<sip:12025551212@xxx.xxx.91.14><013><010>Call-ID: 72786e97-e96342d3@localhost<013><010>CSeq: 103 ACK<013><010>User-Agent: 

Asterisk PBX<013><010>Max-Forwards: 70<013><010>Content-Length: 0
06-14-2007	20:16:27	Local0.Info	192.168.15.1	[0:5062]<<xxx.xxx.91.14:5060
06-14-2007	20:16:27	Local7.Debug	192.168.15.1	SIP/2.0 200 OK<013><010>To: Anonymous 

<sip:1003@pbx.mydomain.com>;tag=c58e1bc7cf2df763o0<013><010>From: 

<sip:12025551212@pbx.mydomain.com>;tag=as72b45889<013><010>Call-ID: 72786e97-e96342d3@localhost<013><010>CSeq: 103 

INVITE<013><010>Via: SIP/2.0/UDP xxx.xxx.91.14:5060;branch=z9hG4bK20bd7956<013><010>Contact: Anonymous 

<sip:1003@xxx.xxx.81.252:5062><013><010>Server: WRTP54G-3.1.22<013><010>Content-Length: 269<013><010>Content-Type: 

application/sdp<013><010><013><010>v=0<013><010>o=- 7670390 7670390 IN IP4 xxx.xxx.81.252<013><010>s=-<013><010>c=IN IP4 

xxx.xxx.81.252<013><010>t=0 0<013><010>m=audio 16448 RTP/AVP 18 100 101<013><010>a=rtpmap:18 

G729a/8000<013><010>a=rtpmap:100 NSE/8000<013><010>a=rtpmap:101 telephone-event/8000<013><010>a=fmtp:101 

0-15<013><010>a=ptime:40<013><010>a=sendrecv<013><010>a=silenceSupp:off - - - -
06-14-2007	20:16:27	Local0.Info	192.168.15.1	[0:5062]->xxx.xxx.91.14:5060
06-14-2007	20:16:27	Local2.Debug	192.168.15.1	CC:Remote Resume
06-14-2007	20:16:27	Local2.Debug	192.168.15.1	[0:0]RTP Tx Up (pt=18->d8fc5b0e:11854)
06-14-2007	20:16:27	Local2.Debug	192.168.15.1	[0:0]RTP Tx Dn
06-14-2007	20:16:27	Local7.Debug	192.168.15.1	INVITE sip:1003@xxx.xxx.81.252:5062 SIP/2.0<013><010>Via: 

SIP/2.0/UDP xxx.xxx.91.14:5060;branch=z9hG4bK20bd7956;rport<013><010>From: 

<sip:12025551212@pbx.mydomain.com>;tag=as72b45889<013><010>To: Anonymous 

<sip:1003@pbx.mydomain.com>;tag=c58e1bc7cf2df763o0<013><010>Contact: 

<sip:12025551212@xxx.xxx.91.14><013><010>Call-ID: 72786e97-e96342d3@localhost<013><010>CSeq: 103 

INVITE<013><010>User-Agent: Asterisk PBX<013><010>Max-Forwards: 70<013><010>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, 

REFER, SUBSCRIBE, NOTIFY<013><010>X-asterisk-info: SIP re-invite (RTP bridge)<013><010>Content-Type: 

application/sdp<013><010>Content-Length: 241<013><010><013><010>v=0<013><010>o=root 24191 24194 IN IP4 

xxx.xxx.91.14<013><010>s=session<013><010>c=IN IP4 xxx.xxx.91.14<013><010>t=0 0<013><010>m=audio 11854 RTP/AVP 18 

101<013><010>a=rtpmap:18 G729/8000<013><010>a=fmtp:18 annexb=no<013><010>a=rtpmap:101 

telephone-event/8000<013><010>a=fmtp:101 0-16<013><010>a=silenceSupp:off - - - -
06-14-2007	20:16:27	Local0.Info	192.168.15.1	[0:5062]<<xxx.xxx.91.14:5060
06-14-2007	20:16:25	Local7.Debug	192.168.15.1	ACK sip:1003@xxx.xxx.81.252:5062 SIP/2.0<013><010>Via: 

SIP/2.0/UDP xxx.xxx.91.14:5060;branch=z9hG4bK131484a2;rport<013><010>From: 

<sip:12025551212@pbx.mydomain.com>;tag=as72b45889<013><010>To: Anonymous 

<sip:1003@pbx.mydomain.com>;tag=c58e1bc7cf2df763o0<013><010>Contact: 

<sip:12025551212@xxx.xxx.91.14><013><010>Call-ID: 72786e97-e96342d3@localhost<013><010>CSeq: 102 ACK<013><010>User-Agent: 

Asterisk PBX<013><010>Max-Forwards: 70<013><010>Content-Length: 0
06-14-2007	20:16:25	Local0.Info	192.168.15.1	[0:5062]<<xxx.xxx.91.14:5060
06-14-2007	20:16:25	Local7.Debug	192.168.15.1	SIP/2.0 200 OK<013><010>To: Anonymous 

<sip:1003@pbx.mydomain.com>;tag=c58e1bc7cf2df763o0<013><010>From: 

<sip:12025551212@pbx.mydomain.com>;tag=as72b45889<013><010>Call-ID: 72786e97-e96342d3@localhost<013><010>CSeq: 102 

INVITE<013><010>Via: SIP/2.0/UDP xxx.xxx.91.14:5060;branch=z9hG4bK6b3b945d<013><010>Contact: Anonymous 

<sip:1003@xxx.xxx.81.252:5062><013><010>Server: WRTP54G-3.1.22<013><010>Content-Length: 269<013><010>Content-Type: 

application/sdp<013><010><013><010>v=0<013><010>o=- 7670124 7670124 IN IP4 xxx.xxx.81.252<013><010>s=-<013><010>c=IN IP4 

xxx.xxx.81.252<013><010>t=0 0<013><010>m=audio 16448 RTP/AVP 18 100 101<013><010>a=rtpmap:18 

G729a/8000<013><010>a=rtpmap:100 NSE/8000<013><010>a=rtpmap:101 telephone-event/8000<013><010>a=fmtp:101 

0-15<013><010>a=ptime:40<013><010>a=sendrecv<013><010>a=silenceSupp:off - - - -
06-14-2007	20:16:25	Local0.Info	192.168.15.1	[0:5062]->xxx.xxx.91.14:5060
06-14-2007	20:16:25	Local2.Debug	192.168.15.1	CC:Remote Resume
06-14-2007	20:16:25	Local2.Debug	192.168.15.1	[0:0]RTP Tx Up (pt=18->9f12a163:50156)
06-14-2007	20:16:25	Local2.Debug	192.168.15.1	[0:0]RTP Tx Dn
06-14-2007	20:16:25	Local7.Debug	192.168.15.1	INVITE sip:1003@xxx.xxx.81.252:5062 SIP/2.0<013><010>Via: 

SIP/2.0/UDP xxx.xxx.91.14:5060;branch=z9hG4bK6b3b945d;rport<013><010>From: 

<sip:12025551212@pbx.mydomain.com>;tag=as72b45889<013><010>To: Anonymous 

<sip:1003@pbx.mydomain.com>;tag=c58e1bc7cf2df763o0<013><010>Contact: 

<sip:12025551212@xxx.xxx.91.14><013><010>Call-ID: 72786e97-e96342d3@localhost<013><010>CSeq: 102 

INVITE<013><010>User-Agent: Asterisk PBX<013><010>Max-Forwards: 70<013><010>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, 

REFER, SUBSCRIBE, NOTIFY<013><010>X-asterisk-info: SIP re-invite (RTP bridge)<013><010>Content-Type: 

application/sdp<013><010>Content-Length: 265<013><010><013><010>v=0<013><010>o=root 24191 24193 IN IP4 

xxx.xxx.161.99<013><010>s=session<013><010>c=IN IP4 xxx.xxx.161.99<013><010>t=0 0<013><010>m=audio 50156 RTP/AVP 18 0 

101<013><010>a=rtpmap:18 G729/8000<013><010>a=fmtp:18 annexb=no<013><010>a=rtpmap:0 PCMU/8000<013><010>a=rtpmap:101 

telephone-event/8000<013><010>a=fmtp:101 0-16<013><010>a=silenceSupp:off - - - -
06-14-2007	20:16:25	Local0.Info	192.168.15.1	[0:5062]<<xxx.xxx.91.14:5060
06-14-2007	20:16:25	Local2.Debug	192.168.15.1	CC:Connected
06-14-2007	20:16:25	Local2.Debug	192.168.15.1	CC:Remote Resume
06-14-2007	20:16:25	Local2.Debug	192.168.15.1	[0:0]RTP Tx Up (pt=18->d8fc5b0e:11854)
06-14-2007	20:16:25	Local2.Debug	192.168.15.1	[0:0]RTP Tx Dn
06-14-2007	20:16:25	Local7.Debug	192.168.15.1	ACK sip:12025551212@xxx.xxx.91.14 SIP/2.0<013><010>Via: 

SIP/2.0/UDP xxx.xxx.81.252:5062;branch=z9hG4bK-f5e73b32<013><010>From: Anonymous 

<sip:1003@pbx.mydomain.com>;tag=c58e1bc7cf2df763o0<013><010>To: 

<sip:12025551212@pbx.mydomain.com>;tag=as72b45889<013><010>Call-ID: 72786e97-e96342d3@localhost<013><010>CSeq: 102 

ACK<013><010>Max-Forwards: 70<013><010>Proxy-Authorization: Digest 

username="1003",realm="asterisk",nonce="12408c6a",uri="sip:12025551212@xxx.xxx.91.14",algorithm=MD5,response="ce942656d75

19ac7ebb49a4fd02584fb"<013><010>Contact: Anonymous <sip:1003@xxx.xxx.81.252:5062><013><010>User-Agent: 

WRTP54G-3.1.22<013><010>Content-Length: 0
06-14-2007	20:16:25	Local0.Info	192.168.15.1	[0:5062]->xxx.xxx.91.14:5060
06-14-2007	20:16:25	Local7.Debug	192.168.15.1	SIP/2.0 200 OK<013><010>Via: SIP/2.0/UDP 

xxx.xxx.81.252:5062;branch=z9hG4bK-5a110c5c;received=xxx.xxx.81.252<013><010>From: Anonymous 

<sip:1003@pbx.mydomain.com>;tag=c58e1bc7cf2df763o0<013><010>To: 

<sip:12025551212@pbx.mydomain.com>;tag=as72b45889<013><010>Call-ID: 72786e97-e96342d3@localhost<013><010>CSeq: 102 

INVITE<013><010>User-Agent: Asterisk PBX<013><010>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, 

NOTIFY<013><010>Contact: <sip:12025551212@xxx.xxx.91.14><013><010>Content-Type: application/sdp<013><010>Content-Length: 

289<013><010><013><010>v=0<013><010>o=root 24191 24192 IN IP4 xxx.xxx.91.14<013><010>s=session<013><010>c=IN IP4 

xxx.xxx.91.14<013><010>t=0 0<013><010>m=audio 11854 RTP/AVP 18 0 8 101<013><010>a=rtpmap:18 G729/8000<013><010>a=fmtp:18 

annexb=no<013><010>a=rtpmap:0 PCMU/8000<013><010>a=rtpmap:8 PCMA/8000<013><010>a=rtpmap:101 

telephone-event/8000<013><010>a=fmtp:101 0-16<013><010>a=silenceSupp:off - - - -
06-14-2007	20:16:25	Local0.Info	192.168.15.1	[0:5062]<<xxx.xxx.91.14:5060
06-14-2007	20:16:18	Local2.Debug	192.168.15.1	CC:CallProgress
06-14-2007	20:16:18	Local2.Debug	192.168.15.1	[0:0]RTCP Tx Up
06-14-2007	20:16:18	Local2.Debug	192.168.15.1	[0:0]RTP Tx Up (pt=18->d8fc5b0e:11854)
06-14-2007	20:16:18	Local7.Debug	192.168.15.1	SIP/2.0 183 Session Progress<013><010>Via: SIP/2.0/UDP 

xxx.xxx.81.252:5062;branch=z9hG4bK-5a110c5c;received=xxx.xxx.81.252<013><010>From: Anonymous 

<sip:1003@pbx.mydomain.com>;tag=c58e1bc7cf2df763o0<013><010>To: 

<sip:12025551212@pbx.mydomain.com>;tag=as72b45889<013><010>Call-ID: 72786e97-e96342d3@localhost<013><010>CSeq: 102 

INVITE<013><010>User-Agent: Asterisk PBX<013><010>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, 

NOTIFY<013><010>Contact: <sip:12025551212@xxx.xxx.91.14><013><010>Content-Type: application/sdp<013><010>Content-Length: 

289<013><010><013><010>v=0<013><010>o=root 24191 24191 IN IP4 xxx.xxx.91.14<013><010>s=session<013><010>c=IN IP4 

xxx.xxx.91.14<013><010>t=0 0<013><010>m=audio 11854 RTP/AVP 18 0 8 101<013><010>a=rtpmap:18 G729/8000<013><010>a=fmtp:18 

annexb=no<013><010>a=rtpmap:0 PCMU/8000<013><010>a=rtpmap:8 PCMA/8000<013><010>a=rtpmap:101 

telephone-event/8000<013><010>a=fmtp:101 0-16<013><010>a=silenceSupp:off - - - -
06-14-2007	20:16:18	Local0.Info	192.168.15.1	[0:5062]<<xxx.xxx.91.14:5060
06-14-2007	20:16:16	Local7.Debug	192.168.15.1	SIP/2.0 100 Trying<013><010>Via: SIP/2.0/UDP 

xxx.xxx.81.252:5062;branch=z9hG4bK-5a110c5c;received=xxx.xxx.81.252<013><010>From: Anonymous 

<sip:1003@pbx.mydomain.com>;tag=c58e1bc7cf2df763o0<013><010>To: 

<sip:12025551212@pbx.mydomain.com><013><010>Call-ID: 72786e97-e96342d3@localhost<013><010>CSeq: 102 

INVITE<013><010>User-Agent: Asterisk PBX<013><010>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, 

NOTIFY<013><010>Contact: <sip:12025551212@xxx.xxx.91.14><013><010>Content-Length: 0
06-14-2007	20:16:16	Local0.Info	192.168.15.1	[0:5062]<<xxx.xxx.91.14:5060
06-14-2007	20:16:16	Local7.Debug	192.168.15.1	INVITE sip:12025551212@pbx.mydomain.com 

SIP/2.0<013><010>Via: SIP/2.0/UDP xxx.xxx.81.252:5062;branch=z9hG4bK-5a110c5c<013><010>From: Anonymous 

<sip:1003@pbx.mydomain.com>;tag=c58e1bc7cf2df763o0<013><010>To: 

<sip:12025551212@pbx.mydomain.com><013><010>Call-ID: 72786e97-e96342d3@localhost<013><010>CSeq: 102 

INVITE<013><010>Max-Forwards: 70<013><010>Proxy-Authorization: Digest 

username="1003",realm="asterisk",nonce="12408c6a",uri="sip:12025551212@pbx.mydomain.com",algorithm=MD5,response="837

6b57939a38f5c50bc31f99feddc7e"<013><010>Contact: Anonymous <sip:1003@xxx.xxx.81.252:5062><013><010>Expires: 

240<013><010>User-Agent: WRTP54G-3.1.22<013><010>Content-Length: 290<013><010>Allow: ACK, BYE, CANCEL, INFO, INVITE, 

NOTIFY, OPTIONS, REFER<013><010>Supported: x-sipura, replaces<013><010>Content-Type: 

application/sdp<013><010><013><010>v=0<013><010>o=- 7669247 7669247 IN IP4 xxx.xxx.81.252<013><010>s=-<013><010>c=IN IP4 

xxx.xxx.81.252<013><010>t=0 0<013><010>m=audio 16448 RTP/AVP 18 0 8 100 101<013><010>a=rtpmap:18 

G729a/8000<013><010>a=rtpmap:0 PCMU/8000<013><010>a=rtpmap:8 PCMA/8000<013><010>a=rtpmap:100 

NSE/8000<013><010>a=rtpmap:101 telephone-event/8000<013><010>a=fmtp:101 0-15<013><010>a=ptime:40<013><010>a=sendrecv
06-14-2007	20:16:16	Local0.Info	192.168.15.1	[0:5062]->xxx.xxx.91.14:5060
06-14-2007	20:16:16	Local7.Debug	192.168.15.1	ACK sip:12025551212@pbx.mydomain.com 

SIP/2.0<013><010>Via: SIP/2.0/UDP xxx.xxx.81.252:5062;branch=z9hG4bK-417ce004<013><010>From: Anonymous 

<sip:1003@pbx.mydomain.com>;tag=c58e1bc7cf2df763o0<013><010>To: 

<sip:12025551212@pbx.mydomain.com>;tag=as6ce39735<013><010>Call-ID: 72786e97-e96342d3@localhost<013><010>CSeq: 101 

ACK<013><010>Max-Forwards: 70<013><010>Contact: Anonymous <sip:1003@xxx.xxx.81.252:5062><013><010>User-Agent: 

WRTP54G-3.1.22<013><010>Content-Length: 0
06-14-2007	20:16:16	Local0.Info	192.168.15.1	[0:5062]->xxx.xxx.91.14:5060
06-14-2007	20:16:16	Local7.Debug	192.168.15.1	SIP/2.0 407 Proxy Authentication Required<013><010>Via: 

SIP/2.0/UDP xxx.xxx.81.252:5062;branch=z9hG4bK-417ce004;received=xxx.xxx.81.252<013><010>From: Anonymous 

<sip:1003@pbx.mydomain.com>;tag=c58e1bc7cf2df763o0<013><010>To: 

<sip:12025551212@pbx.mydomain.com>;tag=as6ce39735<013><010>Call-ID: 72786e97-e96342d3@localhost<013><010>CSeq: 101 

INVITE<013><010>User-Agent: Asterisk PBX<013><010>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, 

NOTIFY<013><010>Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="12408c6a"<013><010>Content-Length: 0
06-14-2007	20:16:16	Local0.Info	192.168.15.1	[0:5062]<<xxx.xxx.91.14:5060
06-14-2007	20:16:16	Local7.Debug	192.168.15.1	INVITE sip:12025551212@pbx.mydomain.com 

SIP/2.0<013><010>Via: SIP/2.0/UDP xxx.xxx.81.252:5062;branch=z9hG4bK-417ce004<013><010>From: Anonymous 

<sip:1003@pbx.mydomain.com>;tag=c58e1bc7cf2df763o0<013><010>To: 

<sip:12025551212@pbx.mydomain.com><013><010>Call-ID: 72786e97-e96342d3@localhost<013><010>CSeq: 101 

INVITE<013><010>Max-Forwards: 70<013><010>Contact: Anonymous <sip:1003@xxx.xxx.81.252:5062><013><010>Expires: 

240<013><010>User-Agent: WRTP54G-3.1.22<013><010>Content-Length: 290<013><010>Allow: ACK, BYE, CANCEL, INFO, INVITE, 

NOTIFY, OPTIONS, REFER<013><010>Supported: x-sipura, replaces<013><010>Content-Type: 

application/sdp<013><010><013><010>v=0<013><010>o=- 7669247 7669247 IN IP4 xxx.xxx.81.252<013><010>s=-<013><010>c=IN IP4 

xxx.xxx.81.252<013><010>t=0 0<013><010>m=audio 16448 RTP/AVP 18 0 8 100 101<013><010>a=rtpmap:18 

G729a/8000<013><010>a=rtpmap:0 PCMU/8000<013><010>a=rtpmap:8 PCMA/8000<013><010>a=rtpmap:100 

NSE/8000<013><010>a=rtpmap:101 telephone-event/8000<013><010>a=fmtp:101 0-15<013><010>a=ptime:40<013><010>a=sendrecv
06-14-2007	20:16:16	Local0.Info	192.168.15.1	[0:5062]->xxx.xxx.91.14:5060
06-14-2007	20:16:16	Local3.Debug	192.168.15.1	RSE_DEBUG: reference domain:pbx.mydomain.com
06-14-2007	20:16:16	Local2.Debug	192.168.15.1	[0:0]RTP Rx Up
06-14-2007	20:16:16	Local2.Debug	192.168.15.1	[0:0]AUD ALLOC CALL (port=16448)
06-14-2007	20:16:16	Local2.Debug	192.168.15.1	Calling:12025551212@pbx.mydomain.com:0

Can anyone confirm this as well?
Anyone know of any possible fix for this?
Does asterisk think that the UAs should support Symmetric RTP to fix this scenario?
Anyone?