<-- SIP read from 203.82.162.21:5060:
INVITE sip:100397@203.82.162.33 SIP/2.0
Via: SIP/2.0/UDP 203.82.162.21:5060;branch=z9hG4bK-c6f9b187
From: 100398 <sip:100398@203.82.162.33>;tag=c083de3f49dd73cdo0
To: <sip:100397@203.82.162.33>
Remote-Party-ID: 100398 <sip:100398@203.82.162.33>;screen=yes;party=calling
Call-ID: a551ca7-aec30885@203.82.162.21
CSeq: 101 INVITE
Max-Forwards: 70
Contact: 100398 <sip:100398@203.82.162.21:5060>
Expires: 240
User-Agent: Linksys/SPA2100-3.3.6
Content-Length: 393
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER
Supported: 100rel, x-sipura
Content-Type: application/sdp

v=0
o=- 8833 8833 IN IP4 203.82.162.21
s=-
c=IN IP4 203.82.162.21
t=0 0
m=audio 16446 RTP/AVP 8 0 2 4 18 96 97 98 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729a/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv

--- (15 headers 18 lines)---
Sending to 203.82.162.21 : 5060 (no NAT)
Using INVITE request as basis request - a551ca7-aec30885@203.82.162.21
Reliably Transmitting (no NAT) to 203.82.162.21:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 203.82.162.21:5060;branch=z9hG4bK-c6f9b187;received=203.82.162.21
From: 100398 <sip:100398@203.82.162.33>;tag=c083de3f49dd73cdo0
To: <sip:100397@203.82.162.33>;tag=as67019244
Call-ID: a551ca7-aec30885@203.82.162.21
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:100397@203.82.162.33>
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="57b57bd4"
Content-Length: 0


---
Scheduling destruction of SIP dialog 'a551ca7-aec30885@203.82.162.21' in 32000 ms (Method: INVITE)
Found user '100398'
thirtythree*CLI>
<-- SIP read from 203.82.162.21:5060:
ACK sip:100397@203.82.162.33 SIP/2.0
Via: SIP/2.0/UDP 203.82.162.21:5060;branch=z9hG4bK-c6f9b187
From: 100398 <sip:100398@203.82.162.33>;tag=c083de3f49dd73cdo0
To: <sip:100397@203.82.162.33>;tag=as67019244
Call-ID: a551ca7-aec30885@203.82.162.21
CSeq: 101 ACK
Max-Forwards: 70
Contact: 100398 <sip:100398@203.82.162.21:5060>
User-Agent: Linksys/SPA2100-3.3.6
Content-Length: 0


--- (10 headers 0 lines)---
thirtythree*CLI>
<-- SIP read from 203.82.162.21:5060:
INVITE sip:100397@203.82.162.33 SIP/2.0
Via: SIP/2.0/UDP 203.82.162.21:5060;branch=z9hG4bK-4053be13
From: 100398 <sip:100398@203.82.162.33>;tag=c083de3f49dd73cdo0
To: <sip:100397@203.82.162.33>
Remote-Party-ID: 100398 <sip:100398@203.82.162.33>;screen=yes;party=calling
Call-ID: a551ca7-aec30885@203.82.162.21
CSeq: 102 INVITE
Max-Forwards: 70
Proxy-Authorization: Digest username="100398",realm="asterisk",nonce="57b57bd4",uri="sip:100397@203.82.162.33",algorithm=MD5,response="2a6a912dfb8e0e6f7c99214e31a4bec6"
Contact: 100398 <sip:100398@203.82.162.21:5060>
Expires: 240
User-Agent: Linksys/SPA2100-3.3.6
Content-Length: 393
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER
Supported: 100rel, x-sipura
Content-Type: application/sdp

v=0
o=- 8833 8833 IN IP4 203.82.162.21
s=-
c=IN IP4 203.82.162.21
t=0 0
m=audio 16446 RTP/AVP 8 0 2 4 18 96 97 98 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729a/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv

--- (16 headers 18 lines)---
Sending to 203.82.162.21 : 5060 (no NAT)
Using INVITE request as basis request - a551ca7-aec30885@203.82.162.21
Found user '100398'
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 2
Found RTP audio format 4
Found RTP audio format 18
Found RTP audio format 96
Found RTP audio format 97
Found RTP audio format 98
Found RTP audio format 101
Peer audio RTP is at port 203.82.162.21:16446
Found description format PCMA for ID 8
Found description format PCMU for ID 0
Found description format G726-32 for ID 2
Found description format G723 for ID 4
Found description format G729a for ID 18
Found description format G726-40 for ID 96
Found description format G726-24 for ID 97
Found description format G726-16 for ID 98
Found description format telephone-event for ID 101
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xd0d (g723|ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 203.82.162.21:16446
Looking for 100397 in from-internal (domain 203.82.162.33)
list_route: hop: <sip:100398@203.82.162.21:5060>
Transmitting (no NAT) to 203.82.162.21:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 203.82.162.21:5060;branch=z9hG4bK-4053be13;received=203.82.162.21
From: 100398 <sip:100398@203.82.162.33>;tag=c083de3f49dd73cdo0
To: <sip:100397@203.82.162.33>
Call-ID: a551ca7-aec30885@203.82.162.21
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:100397@203.82.162.33>
Content-Length: 0


---
Audio is at 203.82.162.33 port 17078
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 203.82.162.20:5060:
INVITE sip:100397@203.82.162.20:5060 SIP/2.0
Via: SIP/2.0/UDP 203.82.162.33:5060;branch=z9hG4bK25ea7ac2
From: "100398" <sip:100398@203.82.162.33>;tag=as48da8ee0
To: <sip:100397@203.82.162.20:5060>
Contact: <sip:100398@203.82.162.33>
Call-ID: 7ada34970b39f974190e03eb2c7c6219@203.82.162.33
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 21 Aug 2006 01:47:55 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 252

v=0
o=root 4236 4236 IN IP4 203.82.162.21
s=session
c=IN IP4 203.82.162.21
t=0 0
m=audio 16446 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=sendrecv

---
thirtythree*CLI>
<-- SIP read from 203.82.162.20:5060:
SIP/2.0 100 Trying
To: <sip:100397@203.82.162.20:5060>
From: "100398" <sip:100398@203.82.162.33>;tag=as48da8ee0
Call-ID: 7ada34970b39f974190e03eb2c7c6219@203.82.162.33
CSeq: 102 INVITE
Via: SIP/2.0/UDP 203.82.162.33:5060;branch=z9hG4bK25ea7ac2
Server: Linksys/SPA2100-3.3.6
Content-Length: 0


--- (8 headers 0 lines)---
thirtythree*CLI>
<-- SIP read from 203.82.162.20:5060:
SIP/2.0 180 Ringing
To: <sip:100397@203.82.162.20:5060>;tag=e9fb21f55b95440ai0
From: "100398" <sip:100398@203.82.162.33>;tag=as48da8ee0
Call-ID: 7ada34970b39f974190e03eb2c7c6219@203.82.162.33
CSeq: 102 INVITE
Via: SIP/2.0/UDP 203.82.162.33:5060;branch=z9hG4bK25ea7ac2
Server: Linksys/SPA2100-3.3.6
Remote-Party-ID: 100397 <sip:100397@203.82.162.33>;screen=yes;party=called
Content-Length: 0


--- (9 headers 0 lines)---
Transmitting (no NAT) to 203.82.162.21:5060:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 203.82.162.21:5060;branch=z9hG4bK-4053be13;received=203.82.162.21
From: 100398 <sip:100398@203.82.162.33>;tag=c083de3f49dd73cdo0
To: <sip:100397@203.82.162.33>;tag=as6792a738
Call-ID: a551ca7-aec30885@203.82.162.21
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:100397@203.82.162.33>
Content-Length: 0


---
thirtythree*CLI>
<-- SIP read from 203.82.162.20:5060:
SIP/2.0 200 OK
To: <sip:100397@203.82.162.20:5060>;tag=e9fb21f55b95440ai0
From: "100398" <sip:100398@203.82.162.33>;tag=as48da8ee0
Call-ID: 7ada34970b39f974190e03eb2c7c6219@203.82.162.33
CSeq: 102 INVITE
Via: SIP/2.0/UDP 203.82.162.33:5060;branch=z9hG4bK25ea7ac2
Contact: 100397 <sip:100397@203.82.162.20:5060>
Server: Linksys/SPA2100-3.3.6
Remote-Party-ID: 100397 <sip:100397@203.82.162.33>;screen=yes;party=called
Content-Length: 206
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER
Supported: 100rel, x-sipura
Content-Type: application/sdp

v=0
o=- 21244 21244 IN IP4 203.82.162.20
s=-
c=IN IP4 203.82.162.20
t=0 0
m=audio 16394 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv

--- (13 headers 11 lines)---
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 203.82.162.20:16394
Found description format PCMA for ID 8
Found description format telephone-event for ID 101
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x8 (alaw)/video=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 203.82.162.20:16394
Peer video RTP is at port 203.82.162.20:15624
list_route: hop: <sip:100397@203.82.162.20:5060>
set_destination: Parsing <sip:100397@203.82.162.20:5060> for address/port to send to
set_destination: set destination to 203.82.162.20, port 5060
Transmitting (no NAT) to 203.82.162.20:5060:
ACK sip:100397@203.82.162.20:5060 SIP/2.0
Via: SIP/2.0/UDP 203.82.162.33:5060;branch=z9hG4bK6663f416
From: "100398" <sip:100398@203.82.162.33>;tag=as48da8ee0
To: <sip:100397@203.82.162.20:5060>;tag=e9fb21f55b95440ai0
Contact: <sip:100398@203.82.162.33>
Call-ID: 7ada34970b39f974190e03eb2c7c6219@203.82.162.33
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
Audio is at 203.82.162.33 port 19504
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 203.82.162.21:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 203.82.162.21:5060;branch=z9hG4bK-4053be13;received=203.82.162.21
From: 100398 <sip:100398@203.82.162.33>;tag=c083de3f49dd73cdo0
To: <sip:100397@203.82.162.33>;tag=as6792a738
Call-ID: a551ca7-aec30885@203.82.162.21
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:100397@203.82.162.33>
Content-Type: application/sdp
Content-Length: 252

v=0
o=root 4236 4236 IN IP4 203.82.162.20
s=session
c=IN IP4 203.82.162.20
t=0 0
m=audio 16394 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=sendrecv

---
thirtythree*CLI>
<-- SIP read from 203.82.162.21:5060:
ACK sip:100397@203.82.162.33 SIP/2.0
Via: SIP/2.0/UDP 203.82.162.21:5060;branch=z9hG4bK-37d0fa7c
From: 100398 <sip:100398@203.82.162.33>;tag=c083de3f49dd73cdo0
To: <sip:100397@203.82.162.33>;tag=as6792a738
Call-ID: a551ca7-aec30885@203.82.162.21
CSeq: 102 ACK
Max-Forwards: 70
Proxy-Authorization: Digest username="100398",realm="asterisk",nonce="57b57bd4",uri="sip:100397@203.82.162.33",algorithm=MD5,response="379738e59f357303fd394b146e2d4125"
Contact: 100398 <sip:100398@203.82.162.21:5060>
User-Agent: Linksys/SPA2100-3.3.6
Content-Length: 0


--- (11 headers 0 lines)---
thirtythree*CLI>
<-- SIP read from 203.82.162.21:5060:
INVITE sip:100397@203.82.162.33 SIP/2.0
Via: SIP/2.0/UDP 203.82.162.21:5060;branch=z9hG4bK-17764195
From: 100398 <sip:100398@203.82.162.33>;tag=c083de3f49dd73cdo0
To: <sip:100397@203.82.162.33>;tag=as6792a738
Remote-Party-ID: 100398 <sip:100398@203.82.162.33>;screen=yes;party=calling
Call-ID: a551ca7-aec30885@203.82.162.21
CSeq: 103 INVITE
Max-Forwards: 70
Proxy-Authorization: Digest username="100398",realm="asterisk",nonce="57b57bd4",uri="sip:100397@203.82.162.33",algorithm=MD5,response="2a6a912dfb8e0e6f7c99214e31a4bec6"
Contact: 100398 <sip:100398@203.82.162.21:5060>
Expires: 30
User-Agent: Linksys/SPA2100-3.3.6
Content-Length: 265
Content-Type: application/sdp

v=0
o=- 10082 10082 IN IP4 203.82.162.21
s=-
c=IN IP4 203.82.162.21
t=0 0
m=image 16446 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:200
a=T38FaxMaxDatagram:200
a=T38FaxUdpEC:t38UDPRedundancy

--- (14 headers 12 lines)---
Sending to 203.82.162.21 : 5060 (no NAT)
Got T.38 offer in SDP in dialog a551ca7-aec30885@203.82.162.21
Peer doesn't provide audio
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x0 (nothing)/video=0x0 (nothing), combined - 0x0 (nothing)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
[Aug 21 11:48:07] NOTICE[4247]: chan_sip.c:4858 process_sdp: No compatible codecs, not accepting this offer!
Transmitting (no NAT) to 203.82.162.21:5060:
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP 203.82.162.21:5060;branch=z9hG4bK-17764195;received=203.82.162.21
From: 100398 <sip:100398@203.82.162.33>;tag=c083de3f49dd73cdo0
To: <sip:100397@203.82.162.33>;tag=as6792a738
Call-ID: a551ca7-aec30885@203.82.162.21
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:100397@203.82.162.33>
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16


---
thirtythree*CLI>
<-- SIP read from 203.82.162.21:5060:
ACK sip:100397@203.82.162.33 SIP/2.0
Via: SIP/2.0/UDP 203.82.162.21:5060;branch=z9hG4bK-17764195
From: 100398 <sip:100398@203.82.162.33>;tag=c083de3f49dd73cdo0
To: <sip:100397@203.82.162.33>;tag=as6792a738
Call-ID: a551ca7-aec30885@203.82.162.21
CSeq: 103 ACK
Max-Forwards: 70
Proxy-Authorization: Digest username="100398",realm="asterisk",nonce="57b57bd4",uri="sip:100397@203.82.162.33",algorithm=MD5,response="379738e59f357303fd394b146e2d4125"
Contact: 100398 <sip:100398@203.82.162.21:5060>
User-Agent: Linksys/SPA2100-3.3.6
Content-Length: 0


--- (11 headers 0 lines)---
thirtythree*CLI>
<-- SIP read from 203.82.162.21:5060:
INVITE sip:100397@203.82.162.33 SIP/2.0
Via: SIP/2.0/UDP 203.82.162.21:5060;branch=z9hG4bK-26fb7d51
From: 100398 <sip:100398@203.82.162.33>;tag=c083de3f49dd73cdo0
To: <sip:100397@203.82.162.33>;tag=as6792a738
Remote-Party-ID: 100398 <sip:100398@203.82.162.33>;screen=yes;party=calling
Call-ID: a551ca7-aec30885@203.82.162.21
CSeq: 104 INVITE
Max-Forwards: 70
Proxy-Authorization: Digest username="100398",realm="asterisk",nonce="57b57bd4",uri="sip:100397@203.82.162.33",algorithm=MD5,response="2a6a912dfb8e0e6f7c99214e31a4bec6"
Contact: 100398 <sip:100398@203.82.162.21:5060>
Expires: 30
User-Agent: Linksys/SPA2100-3.3.6
Content-Length: 150
Content-Type: application/sdp

v=0
o=- 10083 10083 IN IP4 203.82.162.21
s=-
c=IN IP4 203.82.162.21
t=0 0
m=audio 16446 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=ptime:30
a=sendrecv

--- (14 headers 9 lines)---
Sending to 203.82.162.21 : 5060 (no NAT)
Found RTP audio format 8
Peer audio RTP is at port 203.82.162.21:16446
Found description format PCMA for ID 8
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 203.82.162.21:16446
Audio is at 203.82.162.33 port 19504
Adding codec 0x8 (alaw) to SDP
Reliably Transmitting (no NAT) to 203.82.162.21:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 203.82.162.21:5060;branch=z9hG4bK-26fb7d51;received=203.82.162.21
From: 100398 <sip:100398@203.82.162.33>;tag=c083de3f49dd73cdo0
To: <sip:100397@203.82.162.33>;tag=as6792a738
Call-ID: a551ca7-aec30885@203.82.162.21
CSeq: 104 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:100397@203.82.162.33>
Content-Type: application/sdp
Content-Length: 172

v=0
o=root 4236 4237 IN IP4 203.82.162.20
s=session
c=IN IP4 203.82.162.20
t=0 0
m=audio 16394 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=sendrecv

---
thirtythree*CLI>
<-- SIP read from 203.82.162.21:5060:
ACK sip:100397@203.82.162.33 SIP/2.0
Via: SIP/2.0/UDP 203.82.162.21:5060;branch=z9hG4bK-934314c6
From: 100398 <sip:100398@203.82.162.33>;tag=c083de3f49dd73cdo0
To: <sip:100397@203.82.162.33>;tag=as6792a738
Call-ID: a551ca7-aec30885@203.82.162.21
CSeq: 104 ACK
Max-Forwards: 70
Proxy-Authorization: Digest username="100398",realm="asterisk",nonce="57b57bd4",uri="sip:100397@203.82.162.33",algorithm=MD5,response="379738e59f357303fd394b146e2d4125"
Contact: 100398 <sip:100398@203.82.162.21:5060>
User-Agent: Linksys/SPA2100-3.3.6
Content-Length: 0


--- (11 headers 0 lines)---
thirtythree*CLI>
<-- SIP read from 203.82.162.20:5060:
INVITE sip:100398@203.82.162.33 SIP/2.0
Via: SIP/2.0/UDP 203.82.162.20:5060;branch=z9hG4bK-c6760d6c
From: <sip:100397@203.82.162.20:5060>;tag=e9fb21f55b95440ai0
To: "100398" <sip:100398@203.82.162.33>;tag=as48da8ee0
Remote-Party-ID: 100397 <sip:100397@203.82.162.33>;screen=yes;party=called
Call-ID: 7ada34970b39f974190e03eb2c7c6219@203.82.162.33
CSeq: 101 INVITE
Max-Forwards: 70
Contact: 100397 <sip:100397@203.82.162.20:5060>
Expires: 30
User-Agent: Linksys/SPA2100-3.3.6
Content-Length: 265
Content-Type: application/sdp

v=0
o=- 22663 22663 IN IP4 203.82.162.20
s=-
c=IN IP4 203.82.162.20
t=0 0
m=image 16394 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:200
a=T38FaxMaxDatagram:200
a=T38FaxUdpEC:t38UDPRedundancy

--- (13 headers 12 lines)---
Sending to 203.82.162.20 : 5060 (no NAT)
Got T.38 offer in SDP in dialog 7ada34970b39f974190e03eb2c7c6219@203.82.162.33
Peer doesn't provide audio
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x0 (nothing)/video=0x0 (nothing), combined - 0x0 (nothing)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
[Aug 21 11:48:09] NOTICE[4247]: chan_sip.c:4858 process_sdp: No compatible codecs, not accepting this offer!
Transmitting (no NAT) to 203.82.162.20:5060:
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP 203.82.162.20:5060;branch=z9hG4bK-c6760d6c;received=203.82.162.20
From: <sip:100397@203.82.162.20:5060>;tag=e9fb21f55b95440ai0
To: "100398" <sip:100398@203.82.162.33>;tag=as48da8ee0
Call-ID: 7ada34970b39f974190e03eb2c7c6219@203.82.162.33
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:100398@203.82.162.33>
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16


---
thirtythree*CLI>
<-- SIP read from 203.82.162.20:5060:
ACK sip:100398@203.82.162.33 SIP/2.0
Via: SIP/2.0/UDP 203.82.162.20:5060;branch=z9hG4bK-c6760d6c
From: <sip:100397@203.82.162.20:5060>;tag=e9fb21f55b95440ai0
To: "100398" <sip:100398@203.82.162.33>;tag=as48da8ee0
Call-ID: 7ada34970b39f974190e03eb2c7c6219@203.82.162.33
CSeq: 101 ACK
Max-Forwards: 70
Contact: 100397 <sip:100397@203.82.162.20:5060>
User-Agent: Linksys/SPA2100-3.3.6
Content-Length: 0


--- (10 headers 0 lines)---
thirtythree*CLI>
<-- SIP read from 203.82.162.20:5060:
INVITE sip:100398@203.82.162.33 SIP/2.0
Via: SIP/2.0/UDP 203.82.162.20:5060;branch=z9hG4bK-52a2fc8b
From: <sip:100397@203.82.162.20:5060>;tag=e9fb21f55b95440ai0
To: "100398" <sip:100398@203.82.162.33>;tag=as48da8ee0
Remote-Party-ID: 100397 <sip:100397@203.82.162.33>;screen=yes;party=called
Call-ID: 7ada34970b39f974190e03eb2c7c6219@203.82.162.33
CSeq: 102 INVITE
Max-Forwards: 70
Contact: 100397 <sip:100397@203.82.162.20:5060>
Expires: 30
User-Agent: Linksys/SPA2100-3.3.6
Content-Length: 150
Content-Type: application/sdp

v=0
o=- 22664 22664 IN IP4 203.82.162.20
s=-
c=IN IP4 203.82.162.20
t=0 0
m=audio 16394 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=ptime:30
a=sendrecv

--- (13 headers 9 lines)---
Sending to 203.82.162.20 : 5060 (no NAT)
Found RTP audio format 8
Peer audio RTP is at port 203.82.162.20:16394
Found description format PCMA for ID 8
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 203.82.162.20:16394
Audio is at 203.82.162.33 port 17078
Adding codec 0x8 (alaw) to SDP
Reliably Transmitting (no NAT) to 203.82.162.20:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 203.82.162.20:5060;branch=z9hG4bK-52a2fc8b;received=203.82.162.20
From: <sip:100397@203.82.162.20:5060>;tag=e9fb21f55b95440ai0
To: "100398" <sip:100398@203.82.162.33>;tag=as48da8ee0
Call-ID: 7ada34970b39f974190e03eb2c7c6219@203.82.162.33
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:100398@203.82.162.33>
Content-Type: application/sdp
Content-Length: 172

v=0
o=root 4236 4237 IN IP4 203.82.162.21
s=session
c=IN IP4 203.82.162.21
t=0 0
m=audio 16446 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=sendrecv

---
thirtythree*CLI>
<-- SIP read from 203.82.162.20:5060:
ACK sip:100398@203.82.162.33 SIP/2.0
Via: SIP/2.0/UDP 203.82.162.20:5060;branch=z9hG4bK-790a3ff9
From: <sip:100397@203.82.162.20:5060>;tag=e9fb21f55b95440ai0
To: "100398" <sip:100398@203.82.162.33>;tag=as48da8ee0
Call-ID: 7ada34970b39f974190e03eb2c7c6219@203.82.162.33
CSeq: 102 ACK
Max-Forwards: 70
Contact: 100397 <sip:100397@203.82.162.20:5060>
User-Agent: Linksys/SPA2100-3.3.6
Content-Length: 0


--- (10 headers 0 lines)---
thirtythree*CLI> sip show channels
Peer             User/ANR    Call ID      Seq (Tx/Rx)  Form  Hold     Last Message
203.82.162.20    100397      7ada34970b3  00102/00102  alaw  No       Rx: ACK
203.82.162.21    100398      a551ca7-aec  00101/00104  alaw  No       Rx: ACK
2 active SIP channels
thirtythree*CLI>
<-- SIP read from 203.82.162.21:5060:
BYE sip:100397@203.82.162.33 SIP/2.0
Via: SIP/2.0/UDP 203.82.162.21:5060;branch=z9hG4bK-9fb4fb8a
From: 100398 <sip:100398@203.82.162.33>;tag=c083de3f49dd73cdo0
To: <sip:100397@203.82.162.33>;tag=as6792a738
Call-ID: a551ca7-aec30885@203.82.162.21
CSeq: 105 BYE
Max-Forwards: 70
Proxy-Authorization: Digest username="100398",realm="asterisk",nonce="57b57bd4",uri="sip:100397@203.82.162.33",algorithm=MD5,response="4a48b14fba058f39f1ef8bd85f538a9a"
User-Agent: Linksys/SPA2100-3.3.6
Content-Length: 0


--- (10 headers 0 lines)---
Sending to 203.82.162.21 : 5060 (no NAT)
Transmitting (no NAT) to 203.82.162.21:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 203.82.162.21:5060;branch=z9hG4bK-9fb4fb8a;received=203.82.162.21
From: 100398 <sip:100398@203.82.162.33>;tag=c083de3f49dd73cdo0
To: <sip:100397@203.82.162.33>;tag=as6792a738
Call-ID: a551ca7-aec30885@203.82.162.21
CSeq: 105 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:100397@203.82.162.33>
Content-Length: 0


---
set_destination: Parsing <sip:100397@203.82.162.20:5060> for address/port to send to
set_destination: set destination to 203.82.162.20, port 5060
Reliably Transmitting (no NAT) to 203.82.162.20:5060:
BYE sip:100397@203.82.162.20:5060 SIP/2.0
Via: SIP/2.0/UDP 203.82.162.33:5060;branch=z9hG4bK1790cd99
From: "100398" <sip:100398@203.82.162.33>;tag=as48da8ee0
To: <sip:100397@203.82.162.20:5060>;tag=e9fb21f55b95440ai0
Contact: <sip:100398@203.82.162.33>
Call-ID: 7ada34970b39f974190e03eb2c7c6219@203.82.162.33
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
thirtythree*CLI>
<-- SIP read from 203.82.162.20:5060:
SIP/2.0 200 OK
To: <sip:100397@203.82.162.20:5060>;tag=e9fb21f55b95440ai0
From: "100398" <sip:100398@203.82.162.33>;tag=as48da8ee0
Call-ID: 7ada34970b39f974190e03eb2c7c6219@203.82.162.33
CSeq: 103 BYE
Via: SIP/2.0/UDP 203.82.162.33:5060;branch=z9hG4bK1790cd99
Server: Linksys/SPA2100-3.3.6
Content-Length: 0