voip-dev-01-01-1002*CLI> 
<--- SIP read from UDP:INCOMING-SIP-SERVER:5060 --->
INVITE sip:123@SERVER SIP/2.0
Via: SIP/2.0/UDP INCOMING-SIP-SERVER:5060;branch=z9hG4bK25a2f7f9;rport
From: "CALLER-A" <sip:CALLER-A@INCOMING-SIP-SERVER>;tag=as090e6b68
To: <sip:123@SERVER>
Contact: <sip:CALLER-A@INCOMING-SIP-SERVER>
Call-ID: 28d5c5603cec773d7af79252766caaa5@INCOMING-SIP-SERVER
CSeq: 102 INVITE
User-Agent: silver.sip
Max-Forwards: 70
Remote-Party-ID: "CALLER-A" <sip:CALLER-A@INCOMING-SIP-SERVER>;privacy=off;screen=no
Date: Wed, 26 May 2010 09:12:03 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
P-Asserted-Identity: <sip:0043720551111@INCOMING-SIP-SERVER>
Content-Type: application/sdp
Content-Length: 269

v=0
o=root 13639 13639 IN IP4 INCOMING-SIP-SERVER
s=session
c=IN IP4 INCOMING-SIP-SERVER
t=0 0
m=audio 11452 RTP/AVP 8 0 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

<------------->
--- (15 headers 12 lines) ---
  == Using SIP RTP CoS mark 5
  == Using UDPTL CoS mark 5
Sending to INCOMING-SIP-SERVER : 5060 (NAT)
Using INVITE request as basis request - 28d5c5603cec773d7af79252766caaa5@INCOMING-SIP-SERVER
Found peer 'multi-00' for 'CALLER-A' from INCOMING-SIP-SERVER:5060
Found RTP audio format 8 
Found RTP audio format 0 
Found RTP audio format 3 
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format GSM for ID 3
Found audio description format telephone-event for ID 101
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port INCOMING-SIP-SERVER:11452
Looking for 123 in in_world (domain SERVER)
list_route: hop: <sip:CALLER-A@INCOMING-SIP-SERVER>
voip-dev-01-01-1002*CLI> 
<--- Transmitting (no NAT) to INCOMING-SIP-SERVER:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP INCOMING-SIP-SERVER:5060;branch=z9hG4bK25a2f7f9;rport;received=INCOMING-SIP-SERVER
From: "CALLER-A" <sip:CALLER-A@INCOMING-SIP-SERVER>;tag=as090e6b68
To: <sip:123@SERVER>
Call-ID: 28d5c5603cec773d7af79252766caaa5@INCOMING-SIP-SERVER
CSeq: 102 INVITE
Server: PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:123@SERVER>
Content-Length: 0

<------------>
    -- Executing [123@in_world:1] Dial("SIP/multi-00-00000008", "SIP/phonea,30") in new stack
  == Using SIP RTP CoS mark 5
  == Using UDPTL CoS mark 5
Audio is at SERVER port 15022
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 86.59.13.51:6281:
INVITE sip:phonea@86.59.13.51:1198 SIP/2.0
Via: SIP/2.0/UDP SERVER:5060;branch=z9hG4bK55386b89;rport
Max-Forwards: 70
From: "CALLER-A" <sip:CALLER-A@SERVER>;tag=as27c0cc40
To: <sip:phonea@86.59.13.51:1198>
Contact: <sip:CALLER-A@SERVER>
Call-ID: 6cdd9fcf6e250541296140715aeb1603@SERVER
CSeq: 102 INVITE
User-Agent: PBX
Date: Wed, 26 May 2010 09:12:03 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 261

v=0
o=root 964482548 964482548 IN IP4 SERVER
s=Asterisk PBX 1.6.2.7
c=IN IP4 SERVER
t=0 0
m=audio 15022 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
    -- Called phoneaLI> 
voip-dev-01-01-1002*CLI> 
<--- SIP read from UDP:86.59.13.51:6281 --->
SIP/2.0 100 Trying
To: <sip:phonea@86.59.13.51:1198>
From: "CALLER-A" <sip:CALLER-A@SERVER>;tag=as27c0cc40
Call-ID: 6cdd9fcf6e250541296140715aeb1603@SERVER
CSeq: 102 INVITE
Via: SIP/2.0/UDP SERVER:5060;branch=z9hG4bK55386b89
Server: Cisco/SPA525G-7.4.3
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
voip-dev-01-01-1002*CLI> 
<--- SIP read from UDP:86.59.13.51:6281 --->
SIP/2.0 180 Ringing
To: <sip:phonea@86.59.13.51:1198>;tag=7f6a780daa1a7f07i1
From: "CALLER-A" <sip:CALLER-A@SERVER>;tag=as27c0cc40
Call-ID: 6cdd9fcf6e250541296140715aeb1603@SERVER
CSeq: 102 INVITE
Via: SIP/2.0/UDP SERVER:5060;branch=z9hG4bK55386b89
Contact: <sip:phonea@86.59.13.51:1198>
Server: Cisco/SPA525G-7.4.3
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
    -- SIP/phonea-00000009 is ringing
voip-dev-01-01-1002*CLI> 
<--- Transmitting (no NAT) to INCOMING-SIP-SERVER:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP INCOMING-SIP-SERVER:5060;branch=z9hG4bK25a2f7f9;rport;received=INCOMING-SIP-SERVER
From: "CALLER-A" <sip:CALLER-A@INCOMING-SIP-SERVER>;tag=as090e6b68
To: <sip:123@SERVER>;tag=as72544fa9
Call-ID: 28d5c5603cec773d7af79252766caaa5@INCOMING-SIP-SERVER
CSeq: 102 INVITE
Server: PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:123@SERVER>
Content-Length: 0

<------------>
voip-dev-01-01-1002*CLI> 
<--- SIP read from UDP:86.59.13.51:6281 --->
SIP/2.0 200 OK
To: <sip:phonea@86.59.13.51:1198>;tag=7f6a780daa1a7f07i1
From: "CALLER-A" <sip:CALLER-A@SERVER>;tag=as27c0cc40
Call-ID: 6cdd9fcf6e250541296140715aeb1603@SERVER
CSeq: 102 INVITE
Via: SIP/2.0/UDP SERVER:5060;branch=z9hG4bK55386b89
Contact: <sip:phonea@86.59.13.51:1198>
Server: Cisco/SPA525G-7.4.3
Content-Length: 205
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE
Supported: replaces
Content-Type: application/sdp

v=0
o=- 437553 437553 IN IP4 192.168.2.170
s=-
c=IN IP4 86.59.13.51
t=0 0
m=audio 5332 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv

<------------->
--- (12 headers 11 lines) ---
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 86.59.13.51:5332
list_route: hop: <sip:phonea@86.59.13.51:1198>
set_destination: Parsing <sip:phonea@86.59.13.51:1198> for address/port to send to
set_destination: set destination to 86.59.13.51, port 1198
Transmitting (NAT) to 86.59.13.51:6281:
ACK sip:phonea@86.59.13.51:1198 SIP/2.0
Via: SIP/2.0/UDP SERVER:5060;branch=z9hG4bK208c4df1;rport
Max-Forwards: 70
From: "CALLER-A" <sip:CALLER-A@SERVER>;tag=as27c0cc40
To: <sip:phonea@86.59.13.51:1198>;tag=7f6a780daa1a7f07i1
Contact: <sip:CALLER-A@SERVER>
Call-ID: 6cdd9fcf6e250541296140715aeb1603@SERVER
CSeq: 102 ACK
User-Agent: PBX
Content-Length: 0

---
    -- SIP/phonea-00000009 answered SIP/multi-00-00000008
Audio is at SERVER port 12918
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
voip-dev-01-01-1002*CLI> 
<--- Reliably Transmitting (no NAT) to INCOMING-SIP-SERVER:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP INCOMING-SIP-SERVER:5060;branch=z9hG4bK25a2f7f9;rport;received=INCOMING-SIP-SERVER
From: "CALLER-A" <sip:CALLER-A@INCOMING-SIP-SERVER>;tag=as090e6b68
To: <sip:123@SERVER>;tag=as72544fa9
Call-ID: 28d5c5603cec773d7af79252766caaa5@INCOMING-SIP-SERVER
CSeq: 102 INVITE
Server: PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:123@SERVER>
Content-Type: application/sdp
Content-Length: 308

v=0
o=root 154827119 154827119 IN IP4 SERVER
s=Asterisk PBX 1.6.2.7
c=IN IP4 SERVER
t=0 0
m=audio 12918 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
voip-dev-01-01-1002*CLI> 
<--- SIP read from UDP:INCOMING-SIP-SERVER:5060 --->
ACK sip:123@SERVER SIP/2.0
Via: SIP/2.0/UDP INCOMING-SIP-SERVER:5060;branch=z9hG4bK75e698ab;rport
From: "CALLER-A" <sip:CALLER-A@INCOMING-SIP-SERVER>;tag=as090e6b68
To: <sip:123@SERVER>;tag=as72544fa9
Contact: <sip:CALLER-A@INCOMING-SIP-SERVER>
Call-ID: 28d5c5603cec773d7af79252766caaa5@INCOMING-SIP-SERVER
CSeq: 102 ACK
User-Agent: silver.sip
Max-Forwards: 70
Remote-Party-ID: "CALLER-A" <sip:CALLER-A@INCOMING-SIP-SERVER>;privacy=off;screen=no
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
    -- Packet2Packet bridging SIP/multi-00-00000008 and SIP/phonea-00000009
voip-dev-01-01-1002*CLI> 
<--- SIP read from UDP:86.59.13.51:6281 --->
INVITE sip:CALLER-A@SERVER SIP/2.0
Via: SIP/2.0/UDP 86.59.13.51:1198;branch=z9hG4bK-fa175a61
From: <sip:phonea@86.59.13.51:1255>;tag=7f6a780daa1a7f07i1
To: "CALLER-A" <sip:CALLER-A@SERVER>;tag=as27c0cc40
Call-ID: 6cdd9fcf6e250541296140715aeb1603@SERVER
CSeq: 101 INVITE
Max-Forwards: 70
Contact: <sip:phonea@86.59.13.51:1198>
Expires: 30
User-Agent: Cisco/SPA525G-7.4.3
Content-Length: 226
Content-Type: application/sdp

v=0
o=- 437553 437554 IN IP4 192.168.2.170
s=-
c=IN IP4 0.0.0.0
t=0 0
m=audio 16412 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendonly

<------------->
--- (12 headers 12 lines) ---
Sending to 86.59.13.51 : 6281 (NAT)
Found RTP audio format 8 
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x8 (alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 0.0.0.0:16412

<--- Transmitting (NAT) to 86.59.13.51:6281 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 86.59.13.51:1198;branch=z9hG4bK-fa175a61;received=86.59.13.51
From: <sip:phonea@86.59.13.51:1255>;tag=7f6a780daa1a7f07i1
To: "CALLER-A" <sip:CALLER-A@SERVER>;tag=as27c0cc40
Call-ID: 6cdd9fcf6e250541296140715aeb1603@SERVER
CSeq: 101 INVITE
Server: PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:CALLER-A@SERVER>
Content-Length: 0

<------------>
Audio is at SERVER port 15022
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to 86.59.13.51:6281 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 86.59.13.51:1198;branch=z9hG4bK-fa175a61;received=86.59.13.51
From: <sip:phonea@86.59.13.51:1255>;tag=7f6a780daa1a7f07i1
To: "CALLER-A" <sip:CALLER-A@SERVER>;tag=as27c0cc40
Call-ID: 6cdd9fcf6e250541296140715aeb1603@SERVER
CSeq: 101 INVITE
Server: PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:CALLER-A@SERVER>
Content-Type: application/sdp
Content-Length: 261

v=0
o=root 964482548 964482549 IN IP4 SERVER
s=Asterisk PBX 1.6.2.7
c=IN IP4 SERVER
t=0 0
m=audio 15022 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=recvonly

<------------>
    -- Started music on hold, class 'default', on SIP/multi-00-00000008
voip-dev-01-01-1002*CLI> 
<--- SIP read from UDP:86.59.13.51:6281 --->
ACK sip:CALLER-A@SERVER SIP/2.0
Via: SIP/2.0/UDP 86.59.13.51:1198;branch=z9hG4bK-3fd0e1b3
From: <sip:phonea@86.59.13.51:1255>;tag=7f6a780daa1a7f07i1
To: "CALLER-A" <sip:CALLER-A@SERVER>;tag=as27c0cc40
Call-ID: 6cdd9fcf6e250541296140715aeb1603@SERVER
CSeq: 101 ACK
Max-Forwards: 70
Contact: <sip:phonea@86.59.13.51:1198>
User-Agent: Cisco/SPA525G-7.4.3
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
voip-dev-01-01-1002*CLI> 
<--- SIP read from UDP:86.59.13.51:6281 --->
INVITE sip:456@voip-dev-01-01-1002.sil.at SIP/2.0
Via: SIP/2.0/UDP 86.59.13.51:1198;branch=z9hG4bK-e9b5054c
From: <sip:phonea@voip-dev-01-01-1002.sil.at>;tag=8f2eca195fdd761bo1
To: <sip:456@voip-dev-01-01-1002.sil.at>
Call-ID: b168071-7e2573a3@192.168.2.170
CSeq: 101 INVITE
Max-Forwards: 70
Contact: <sip:phonea@86.59.13.51:1198>
Expires: 240
User-Agent: Cisco/SPA525G-7.4.3
Content-Length: 335
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE
Supported: replaces
Content-Type: application/sdp

v=0
o=- 437985 437985 IN IP4 192.168.2.170
s=-
c=IN IP4 86.59.13.51
t=0 0
m=audio 5358 RTP/AVP 8 0 2 9 104 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:9 G722/8000
a=rtpmap:104 L16/16000
a=rtpmap:18 G729a/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv

<------------->
--- (14 headers 16 lines) ---
  == Using SIP RTP CoS mark 5
  == Using UDPTL CoS mark 5
Sending to 86.59.13.51 : 6281 (NAT)
Using INVITE request as basis request - b168071-7e2573a3@192.168.2.170
Found peer 'phonea' for 'phonea' from 86.59.13.51:6281

<--- Reliably Transmitting (NAT) to 86.59.13.51:6281 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 86.59.13.51:1198;branch=z9hG4bK-e9b5054c;received=86.59.13.51
From: <sip:phonea@voip-dev-01-01-1002.sil.at>;tag=8f2eca195fdd761bo1
To: <sip:456@voip-dev-01-01-1002.sil.at>;tag=as708e351f
Call-ID: b168071-7e2573a3@192.168.2.170
CSeq: 101 INVITE
Server: PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="pbx.sil.at", nonce="57c31634"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog 'b168071-7e2573a3@192.168.2.170' in 6400 ms (Method: INVITE)
voip-dev-01-01-1002*CLI> 
<--- SIP read from UDP:86.59.13.51:6281 --->
ACK sip:456@voip-dev-01-01-1002.sil.at SIP/2.0
Via: SIP/2.0/UDP 86.59.13.51:1198;branch=z9hG4bK-e9b5054c
From: <sip:phonea@voip-dev-01-01-1002.sil.at>;tag=8f2eca195fdd761bo1
To: <sip:456@voip-dev-01-01-1002.sil.at>;tag=as708e351f
Call-ID: b168071-7e2573a3@192.168.2.170
CSeq: 101 ACK
Max-Forwards: 70
Contact: <sip:phonea@86.59.13.51:1198>
User-Agent: Cisco/SPA525G-7.4.3
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
voip-dev-01-01-1002*CLI> 
<--- SIP read from UDP:86.59.13.51:6281 --->
INVITE sip:456@voip-dev-01-01-1002.sil.at SIP/2.0
Via: SIP/2.0/UDP 86.59.13.51:1198;branch=z9hG4bK-503b3e64
From: <sip:phonea@voip-dev-01-01-1002.sil.at>;tag=8f2eca195fdd761bo1
To: <sip:456@voip-dev-01-01-1002.sil.at>
Call-ID: b168071-7e2573a3@192.168.2.170
CSeq: 102 INVITE
Max-Forwards: 70
Authorization: Digest username="phonea",realm="pbx.sil.at",nonce="57c31634",uri="sip:456@voip-dev-01-01-1002.sil.at",algorithm=MD5,response="6d347ac90268c3967c9d54ce5e278dd6"
Contact: <sip:phonea@86.59.13.51:1198>
Expires: 240
User-Agent: Cisco/SPA525G-7.4.3
Content-Length: 335
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE
Supported: replaces
Content-Type: application/sdp

v=0
o=- 437985 437985 IN IP4 192.168.2.170
s=-
c=IN IP4 86.59.13.51
t=0 0
m=audio 5358 RTP/AVP 8 0 2 9 104 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:9 G722/8000
a=rtpmap:104 L16/16000
a=rtpmap:18 G729a/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv

<------------->
--- (15 headers 16 lines) ---
Sending to 86.59.13.51 : 6281 (NAT)
Using INVITE request as basis request - b168071-7e2573a3@192.168.2.170
Found peer 'phonea' for 'phonea' from 86.59.13.51:6281
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 2
Found RTP audio format 9
Found RTP audio format 104
Found RTP audio format 18
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format G726-32 for ID 2
Found audio description format G722 for ID 9
Found audio description format L16 for ID 104
Found audio description format G729a for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - 0x8 (alaw), peer - audio=0x40190c (ulaw|alaw|g726|g729|g722|mpeg4)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 86.59.13.51:5358
Looking for 456 in outcust (domain voip-dev-01-01-1002.sil.at)
list_route: hop: <sip:phonea@86.59.13.51:1198>

<--- Transmitting (NAT) to 86.59.13.51:6281 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 86.59.13.51:1198;branch=z9hG4bK-503b3e64;received=86.59.13.51
From: <sip:phonea@voip-dev-01-01-1002.sil.at>;tag=8f2eca195fdd761bo1
To: <sip:456@voip-dev-01-01-1002.sil.at>
Call-ID: b168071-7e2573a3@192.168.2.170
CSeq: 102 INVITE
Server: PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:456@SERVER>
Content-Length: 0

<------------>
    -- Executing [456@outcust:1] Answer("SIP/phonea-0000000a", "") in new stack
Audio is at SERVER port 16128
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to 86.59.13.51:6281 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 86.59.13.51:1198;branch=z9hG4bK-503b3e64;received=86.59.13.51
From: <sip:phonea@voip-dev-01-01-1002.sil.at>;tag=8f2eca195fdd761bo1
To: <sip:456@voip-dev-01-01-1002.sil.at>;tag=as5aebfb1b
Call-ID: b168071-7e2573a3@192.168.2.170
CSeq: 102 INVITE
Server: PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:456@SERVER>
Content-Type: application/sdp
Content-Length: 261

v=0
o=root 535623116 535623116 IN IP4 SERVER
s=Asterisk PBX 1.6.2.7
c=IN IP4 SERVER
t=0 0
m=audio 16128 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
voip-dev-01-01-1002*CLI> 
<--- SIP read from UDP:86.59.13.51:6281 --->
ACK sip:456@SERVER SIP/2.0
Via: SIP/2.0/UDP 86.59.13.51:1198;branch=z9hG4bK-6611657
From: <sip:phonea@voip-dev-01-01-1002.sil.at>;tag=8f2eca195fdd761bo1
To: <sip:456@voip-dev-01-01-1002.sil.at>;tag=as5aebfb1b
Call-ID: b168071-7e2573a3@192.168.2.170
CSeq: 102 ACK
Max-Forwards: 70
Authorization: Digest username="phonea",realm="pbx.sil.at",nonce="57c31634",uri="sip:456@voip-dev-01-01-1002.sil.at",algorithm=MD5,response="6d347ac90268c3967c9d54ce5e278dd6"
Contact: <sip:phonea@86.59.13.51:1198>
User-Agent: Cisco/SPA525G-7.4.3
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
    -- Executing [456@outcust:2] Dial("SIP/phonea-0000000a", "SIP/phoneb,30") in new stack
  == Using SIP RTP CoS mark 5
  == Using UDPTL CoS mark 5
Audio is at SERVER port 11020
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 86.59.13.51:4927:
INVITE sip:phoneb@86.59.13.51:6206 SIP/2.0
Via: SIP/2.0/UDP SERVER:5060;branch=z9hG4bK2ecaae4a;rport
Max-Forwards: 70
From: "Stefan Schmidt" <sip:phonea@SERVER>;tag=as6e9f5035
To: <sip:phoneb@86.59.13.51:6206>
Contact: <sip:phonea@SERVER>
Call-ID: 5c5747b31d3ce3f77d0ca27b2fd9ca9e@SERVER
CSeq: 102 INVITE
User-Agent: PBX
Date: Wed, 26 May 2010 09:12:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 263

v=0
o=root 2072553636 2072553636 IN IP4 SERVER
s=Asterisk PBX 1.6.2.7
c=IN IP4 SERVER
t=0 0
m=audio 11020 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
    -- Called phoneb
voip-dev-01-01-1002*CLI> 
<--- SIP read from UDP:86.59.13.51:4927 --->
SIP/2.0 100 Trying
To: <sip:phoneb@86.59.13.51:6206>
From: "Stefan Schmidt" <sip:phonea@SERVER>;tag=as6e9f5035
Call-ID: 5c5747b31d3ce3f77d0ca27b2fd9ca9e@SERVER
CSeq: 102 INVITE
Via: SIP/2.0/UDP SERVER:5060;branch=z9hG4bK2ecaae4a
Server: Cisco/SPA525G-7.4.4
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
voip-dev-01-01-1002*CLI> 
<--- SIP read from UDP:86.59.13.51:4927 --->
SIP/2.0 180 Ringing
To: <sip:phoneb@86.59.13.51:6206>;tag=76eab77be6b78e2ai2
From: "Stefan Schmidt" <sip:phonea@SERVER>;tag=as6e9f5035
Call-ID: 5c5747b31d3ce3f77d0ca27b2fd9ca9e@SERVER
CSeq: 102 INVITE
Via: SIP/2.0/UDP SERVER:5060;branch=z9hG4bK2ecaae4a
Contact: <sip:phoneb@86.59.13.51:6206>
Server: Cisco/SPA525G-7.4.4
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
    -- SIP/phoneb-0000000b is ringing
voip-dev-01-01-1002*CLI> 
<--- SIP read from UDP:86.59.13.51:4927 --->
SIP/2.0 200 OK
To: <sip:phoneb@86.59.13.51:6206>;tag=76eab77be6b78e2ai2
From: "Stefan Schmidt" <sip:phonea@SERVER>;tag=as6e9f5035
Call-ID: 5c5747b31d3ce3f77d0ca27b2fd9ca9e@SERVER
CSeq: 102 INVITE
Via: SIP/2.0/UDP SERVER:5060;branch=z9hG4bK2ecaae4a
Contact: <sip:phoneb@86.59.13.51:6206>
Server: Cisco/SPA525G-7.4.4
Content-Length: 204
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE
Supported: replaces
Content-Type: application/sdp

v=0
o=- 238826 238826 IN IP4 192.168.2.77
s=-
c=IN IP4 86.59.13.51
t=0 0
m=audio 6750 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv

<------------->
--- (12 headers 11 lines) ---
Found RTP audio format 8 
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 86.59.13.51:6750
list_route: hop: <sip:phoneb@86.59.13.51:6206>
set_destination: Parsing <sip:phoneb@86.59.13.51:6206> for address/port to send to
set_destination: set destination to 86.59.13.51, port 6206
Transmitting (NAT) to 86.59.13.51:4927:
ACK sip:phoneb@86.59.13.51:6206 SIP/2.0
Via: SIP/2.0/UDP SERVER:5060;branch=z9hG4bK5b1e33ba;rport
Max-Forwards: 70
From: "Stefan Schmidt" <sip:phonea@SERVER>;tag=as6e9f5035
To: <sip:phoneb@86.59.13.51:6206>;tag=76eab77be6b78e2ai2
Contact: <sip:phonea@SERVER>
Call-ID: 5c5747b31d3ce3f77d0ca27b2fd9ca9e@SERVER
CSeq: 102 ACK
User-Agent: PBX
Content-Length: 0

---
    -- SIP/phoneb-0000000b answered SIP/phonea-0000000a
    -- Packet2Packet bridging SIP/phonea-0000000a and SIP/phoneb-0000000b
voip-dev-01-01-1002*CLI> 
<--- SIP read from UDP:86.59.13.51:6281 --->
REFER sip:CALLER-A@SERVER SIP/2.0
Via: SIP/2.0/UDP 86.59.13.51:1198;branch=z9hG4bK-d6fe67f2
From: <sip:phonea@86.59.13.51:1255>;tag=7f6a780daa1a7f07i1
To: "CALLER-A" <sip:CALLER-A@SERVER>;tag=as27c0cc40
Referred-By: <sip:phonea@voip-dev-01-01-1002.sil.at>
Call-ID: 6cdd9fcf6e250541296140715aeb1603@SERVER
CSeq: 102 REFER
Max-Forwards: 70
Contact: <sip:phonea@86.59.13.51:1198>
Refer-To: <sip:456@voip-dev-01-01-1002.sil.at?Replaces=b168071-7e2573a3%40192.168.2.170%3Bfrom-tag%3D8f2eca195fdd761bo1%3Bto-tag%3Das5aebfb1b>
User-Agent: Cisco/SPA525G-7.4.3
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Call 6cdd9fcf6e250541296140715aeb1603@SERVER got a SIP call transfer from caller: (REFER)!
SIP transfer to extension 456@outcust by phonea@voip-dev-01-01-1002.sil.at

<--- Transmitting (NAT) to 86.59.13.51:6281 --->
SIP/2.0 202 Accepted
Via: SIP/2.0/UDP 86.59.13.51:1198;branch=z9hG4bK-d6fe67f2;received=86.59.13.51
From: <sip:phonea@86.59.13.51:1255>;tag=7f6a780daa1a7f07i1
To: "CALLER-A" <sip:CALLER-A@SERVER>;tag=as27c0cc40
Call-ID: 6cdd9fcf6e250541296140715aeb1603@SERVER
CSeq: 102 REFER
Server: PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:CALLER-A@SERVER>
Content-Length: 0

<------------>
    -- Stopped music on hold on SIP/multi-00-00000008
set_destination: Parsing <sip:phonea@86.59.13.51:1198> for address/port to send to
set_destination: set destination to 86.59.13.51, port 1198
Reliably Transmitting (NAT) to 86.59.13.51:6281:
NOTIFY sip:phonea@86.59.13.51:1198 SIP/2.0
Via: SIP/2.0/UDP SERVER:5060;branch=z9hG4bK6d705e90;rport
Max-Forwards: 70
From: "CALLER-A" <sip:CALLER-A@SERVER>;tag=as27c0cc40
To: <sip:phonea@86.59.13.51:1255>;tag=7f6a780daa1a7f07i1
Contact: <sip:CALLER-A@SERVER>
Call-ID: 6cdd9fcf6e250541296140715aeb1603@SERVER
CSeq: 103 NOTIFY
User-Agent: PBX
Event: refer;id=102
Subscription-state: terminated;reason=noresource
Content-Type: message/sipfrag;version=2.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 16

SIP/2.0 200 OK

---
Scheduling destruction of SIP dialog 'b168071-7e2573a3@192.168.2.170' in 6400 ms (Method: ACK)
set_destination: Parsing <sip:phonea@86.59.13.51:1198> for address/port to send to
set_destination: set destination to 86.59.13.51, port 1198
Reliably Transmitting (NAT) to 86.59.13.51:6281:
BYE sip:phonea@86.59.13.51:1198 SIP/2.0
Via: SIP/2.0/UDP SERVER:5060;branch=z9hG4bK157ea38d;rport
Max-Forwards: 70
From: <sip:456@voip-dev-01-01-1002.sil.at>;tag=as5aebfb1b
To: <sip:phonea@voip-dev-01-01-1002.sil.at>;tag=8f2eca195fdd761bo1
Call-ID: b168071-7e2573a3@192.168.2.170
CSeq: 102 BYE
User-Agent: PBX
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

---
    -- Packet2Packet bridging SIP/multi-00-00000008 and SIP/phoneb-0000000b
Scheduling destruction of SIP dialog '6cdd9fcf6e250541296140715aeb1603@SERVER' in 6400 ms (Method: REFER)
  == Spawn extension (in_world, 123, 1) exited non-zero on 'SIP/phonea-0000000a<ZOMBIE>'
voip-dev-01-01-1002*CLI> 
<--- SIP read from UDP:86.59.13.51:6281 --->
SIP/2.0 200 OK
To: <sip:phonea@86.59.13.51:1255>;tag=7f6a780daa1a7f07i1
From: "CALLER-A" <sip:CALLER-A@SERVER>;tag=as27c0cc40
Call-ID: 6cdd9fcf6e250541296140715aeb1603@SERVER
CSeq: 103 NOTIFY
Via: SIP/2.0/UDP SERVER:5060;branch=z9hG4bK6d705e90
Server: Cisco/SPA525G-7.4.3
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
SIP Response message for INCOMING dialog NOTIFY arrived

<--- SIP read from UDP:86.59.13.51:6281 --->
SIP/2.0 200 OK
To: <sip:phonea@voip-dev-01-01-1002.sil.at>;tag=8f2eca195fdd761bo1
From: <sip:456@voip-dev-01-01-1002.sil.at>;tag=as5aebfb1b
Call-ID: b168071-7e2573a3@192.168.2.170
CSeq: 102 BYE
Via: SIP/2.0/UDP SERVER:5060;branch=z9hG4bK157ea38d
Server: Cisco/SPA525G-7.4.3
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog 'b168071-7e2573a3@192.168.2.170' Method: ACK
voip-dev-01-01-1002*CLI> 
<--- SIP read from UDP:INCOMING-SIP-SERVER:5060 --->
BYE sip:123@SERVER SIP/2.0
Via: SIP/2.0/UDP INCOMING-SIP-SERVER:5060;branch=z9hG4bK3c477237;rport
From: "CALLER-A" <sip:CALLER-A@INCOMING-SIP-SERVER>;tag=as090e6b68
To: <sip:123@SERVER>;tag=as72544fa9
Call-ID: 28d5c5603cec773d7af79252766caaa5@INCOMING-SIP-SERVER
CSeq: 103 BYE
User-Agent: silver.sip
Max-Forwards: 70
Remote-Party-ID: "CALLER-A" <sip:CALLER-A@INCOMING-SIP-SERVER>;privacy=off;screen=no
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Sending to INCOMING-SIP-SERVER : 5060 (no NAT)

<--- Transmitting (no NAT) to INCOMING-SIP-SERVER:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP INCOMING-SIP-SERVER:5060;branch=z9hG4bK3c477237;rport;received=INCOMING-SIP-SERVER
From: "CALLER-A" <sip:CALLER-A@INCOMING-SIP-SERVER>;tag=as090e6b68
To: <sip:123@SERVER>;tag=as72544fa9
Call-ID: 28d5c5603cec773d7af79252766caaa5@INCOMING-SIP-SERVER
CSeq: 103 BYE
Server: PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog '5c5747b31d3ce3f77d0ca27b2fd9ca9e@SERVER' in 6400 ms (Method: INVITE)
set_destination: Parsing <sip:phoneb@86.59.13.51:6206> for address/port to send to
set_destination: set destination to 86.59.13.51, port 6206
Reliably Transmitting (NAT) to 86.59.13.51:4927:
BYE sip:phoneb@86.59.13.51:6206 SIP/2.0
Via: SIP/2.0/UDP SERVER:5060;branch=z9hG4bK0654e426;rport
Max-Forwards: 70
From: "Stefan Schmidt" <sip:phonea@SERVER>;tag=as6e9f5035
To: <sip:phoneb@86.59.13.51:6206>;tag=76eab77be6b78e2ai2
Call-ID: 5c5747b31d3ce3f77d0ca27b2fd9ca9e@SERVER
CSeq: 103 BYE
User-Agent: PBX
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

---
  == Spawn extension (outcust, 456, 2) exited non-zero on 'SIP/multi-00-00000008'
voip-dev-01-01-1002*CLI> 
<--- SIP read from UDP:86.59.13.51:4927 --->
SIP/2.0 200 OK
To: <sip:phoneb@86.59.13.51:6206>;tag=76eab77be6b78e2ai2
From: "Stefan Schmidt" <sip:phonea@SERVER>;tag=as6e9f5035
Call-ID: 5c5747b31d3ce3f77d0ca27b2fd9ca9e@SERVER
CSeq: 103 BYE
Via: SIP/2.0/UDP SERVER:5060;branch=z9hG4bK0654e426
Server: Cisco/SPA525G-7.4.4
Content-Length: 0