<--- SIP read from 10.0.101.13:5061 --->
INVITE sip:602@80.251.131.142 SIP/2.0
Call-ID: call-F127CA6A-2AA2-2110-0C1A-4B@10.0.101.13
Contact: <sip:603@10.0.101.13:5061>
Content-Length: 270
Content-Type: application/sdp
CSeq: 1 INVITE
From: <sip:603@80.251.131.142>;tag=a00650d-36
Max-Forwards: 70
Session-GUID: 859321956-926496051-876150800-0
Supported: 100rel
To: <sip:602@80.251.131.142>
User-Agent: Quintum/1.0.0 SN/0030E1103326 SW/S107-04-00
Via: SIP/2.0/UDP 10.0.101.13:5061;branch=z9hG4bK-tenor-0a00-650d-0181

v=0
o=Quintum 411 411 IN IP4 10.0.101.13
s=VoipCall
c=IN IP4 10.0.101.13
t=0 0
m=audio 10310 RTP/AVP 8 18 101
c=IN IP4 10.0.101.13
a=rtpmap:8 pcma/8000/1
a=ptime:20
a=fmtp:18 annexb=yes
a=rtpmap:18 G729/8000/1
a=rtpmap:101 telephone-event/8000/1
a=sendrecv
<------------->
--- (13 headers 13 lines) ---
Sending to 10.0.101.13 : 5061 (no NAT)
Using INVITE request as basis request - call-F127CA6A-2AA2-2110-0C1A-4B@10.0.101.13
srvtm11*CLI>
<--- Reliably Transmitting (no NAT) to 10.0.101.13:5061 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.0.101.13:5061;branch=z9hG4bK-tenor-0a00-650d-0181;received=10.0.101.13
From: <sip:603@80.251.131.142>;tag=a00650d-36
To: <sip:602@80.251.131.142>;tag=as54dfc0fc
Call-ID: call-F127CA6A-2AA2-2110-0C1A-4B@10.0.101.13
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="389097e5"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'call-F127CA6A-2AA2-2110-0C1A-4B@10.0.101.13' in 32000 ms (Method: INVITE)
Found user '603'
srvtm11*CLI>
<--- SIP read from 10.0.101.13:5061 --->
ACK sip:602@80.251.131.142 SIP/2.0
Call-ID: call-F127CA6A-2AA2-2110-0C1A-4B@10.0.101.13
Contact: <sip:603@10.0.101.13:5061>
Content-Length: 0
CSeq: 1 ACK
From: <sip:603@80.251.131.142>;tag=a00650d-36
Max-Forwards: 70
Session-GUID: 859321956-926496051-876150800-0
Supported: 100rel
To: <sip:602@80.251.131.142>;tag=as54dfc0fc
User-Agent: Quintum/1.0.0 SN/0030E1103326 SW/S107-04-00
Via: SIP/2.0/UDP 10.0.101.13:5061;branch=z9hG4bK-tenor-0a00-650d-0181

<------------->
--- (12 headers 0 lines) ---
srvtm11*CLI>
<--- SIP read from 10.0.101.13:5061 --->
INVITE sip:602@80.251.131.142 SIP/2.0
Call-ID: call-F127CA6A-2AA2-2110-0C1A-4B@10.0.101.13
Contact: <sip:603@10.0.101.13:5061>
Content-Length: 270
Content-Type: application/sdp
CSeq: 2 INVITE
From: <sip:603@80.251.131.142>;tag=a00650d-36
Max-Forwards: 70
Proxy-Authorization: Digest realm="asterisk", nonce="389097e5",algorithm=MD5, username="603", uri="sip:602@80.251.131.142", response="5b8215bbf500215cd5926fe224fdae4c"
Session-GUID: 859321956-926496051-876150800-0
Supported: 100rel
To: <sip:602@80.251.131.142>
User-Agent: Quintum/1.0.0 SN/0030E1103326 SW/S107-04-00
Via: SIP/2.0/UDP 10.0.101.13:5061;branch=z9hG4bK-tenor-0a00-650d-0182

v=0
o=Quintum 411 411 IN IP4 10.0.101.13
s=VoipCall
c=IN IP4 10.0.101.13
t=0 0
m=audio 10310 RTP/AVP 8 18 101
c=IN IP4 10.0.101.13
a=rtpmap:8 pcma/8000/1
a=ptime:20
a=fmtp:18 annexb=yes
a=rtpmap:18 G729/8000/1
a=rtpmap:101 telephone-event/8000/1
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 10.0.101.13 : 5061 (no NAT)
Using INVITE request as basis request - call-F127CA6A-2AA2-2110-0C1A-4B@10.0.101.13
Found user '603'
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 10.0.101.13:10310
Found audio description format pcma for ID 8
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - 0x108 (alaw|g729), peer - audio=0x108 (alaw|g729)/video=0x0 (nothing), combined - 0x108 (alaw|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 10.0.101.13:10310
Looking for 602 in permit-national (domain 80.251.131.142)
list_route: hop: <sip:603@10.0.101.13:5061>
srvtm11*CLI>
<--- Transmitting (no NAT) to 10.0.101.13:5061 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.101.13:5061;branch=z9hG4bK-tenor-0a00-650d-0182;received=10.0.101.13
From: <sip:603@80.251.131.142>;tag=a00650d-36
To: <sip:602@80.251.131.142>
Call-ID: call-F127CA6A-2AA2-2110-0C1A-4B@10.0.101.13
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:602@80.251.131.142>
Content-Length: 0


<------------>
Audio is at 80.251.131.142 port 19334
Adding codec 0x8 (alaw) to SDP
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
srvtm11*CLI>
<--- Transmitting (no NAT) to 10.0.101.13:5061 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.0.101.13:5061;branch=z9hG4bK-tenor-0a00-650d-0182;received=10.0.101.13
From: <sip:603@80.251.131.142>;tag=a00650d-36
To: <sip:602@80.251.131.142>;tag=as67ad695e
Call-ID: call-F127CA6A-2AA2-2110-0C1A-4B@10.0.101.13
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:602@80.251.131.142>
Content-Type: application/sdp
Content-Length: 289

v=0
o=root 5974 5974 IN IP4 80.251.131.142
s=session
c=IN IP4 80.251.131.142
t=0 0
m=audio 19334 RTP/AVP 8 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
srvtm11*CLI>
<--- Transmitting (no NAT) to 10.0.101.13:5061 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.0.101.13:5061;branch=z9hG4bK-tenor-0a00-650d-0182;received=10.0.101.13
From: <sip:603@80.251.131.142>;tag=a00650d-36
To: <sip:602@80.251.131.142>;tag=as67ad695e
Call-ID: call-F127CA6A-2AA2-2110-0C1A-4B@10.0.101.13
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:602@80.251.131.142>
Content-Length: 0


<------------>
srvtm11*CLI>
[Jul 6 14:50:39] WARNING[10047]: chan_sip.c:2994 create_addr: No such host: 989
[Jul 6 14:50:39] WARNING[10047]: app_dial.c:1272 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown)
[Jul 6 14:50:40] NOTICE[10046]: rtp.c:823 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 10.0.101.13
Audio is at 80.251.131.142 port 19334
Adding codec 0x8 (alaw) to SDP
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
srvtm11*CLI>
<--- Reliably Transmitting (no NAT) to 10.0.101.13:5061 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.101.13:5061;branch=z9hG4bK-tenor-0a00-650d-0182;received=10.0.101.13
From: <sip:603@80.251.131.142>;tag=a00650d-36
To: <sip:602@80.251.131.142>;tag=as67ad695e
Call-ID: call-F127CA6A-2AA2-2110-0C1A-4B@10.0.101.13
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:602@80.251.131.142>
Content-Type: application/sdp
Content-Length: 289

v=0
o=root 5974 5975 IN IP4 80.251.131.142
s=session
c=IN IP4 80.251.131.142
t=0 0
m=audio 19334 RTP/AVP 8 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
srvtm11*CLI>
<--- SIP read from 10.0.101.13:5061 --->
ACK sip:602@80.251.131.142 SIP/2.0
Call-ID: call-F127CA6A-2AA2-2110-0C1A-4B@10.0.101.13
Contact: <sip:603@10.0.101.13:5061>
Content-Length: 0
CSeq: 2 ACK
From: <sip:603@80.251.131.142>;tag=a00650d-36
Max-Forwards: 70
Proxy-Authorization: Digest realm="asterisk", nonce="389097e5",algorithm=MD5, username="603", uri="sip:602@80.251.131.142", response="5b8215bbf500215cd5926fe224fdae4c"
Session-GUID: 859321956-926496051-876150800-0
Supported: 100rel
To: <sip:602@80.251.131.142>;tag=as67ad695e
User-Agent: Quintum/1.0.0 SN/0030E1103326 SW/S107-04-00
Via: SIP/2.0/UDP 10.0.101.13:5061;branch=z9hG4bK-tenor-0a00-650d-0183

<------------->
--- (13 headers 0 lines) ---
srvtm11*CLI>
set_destination: Parsing <sip:603@10.0.101.13:5061> for address/port to send to
set_destination: set destination to 10.0.101.13, port 5061
Reliably Transmitting (no NAT) to 10.0.101.13:5061:
INVITE sip:603@10.0.101.13:5061 SIP/2.0
Via: SIP/2.0/UDP 80.251.131.142:5060;branch=z9hG4bK39639742;rport
From: <sip:602@80.251.131.142>;tag=as67ad695e
To: <sip:603@80.251.131.142>;tag=a00650d-36
Contact: <sip:602@80.251.131.142>
Call-ID: call-F127CA6A-2AA2-2110-0C1A-4B@10.0.101.13
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
X-asterisk-info: SIP re-invite (T38 switchover)
Content-Type: application/sdp
Content-Length: 266

v=0
o=root 5974 5976 IN IP4 80.251.131.142
s=session
c=IN IP4 80.251.131.142
t=0 0
m=image 10720 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:400
a=T38FaxMaxDatagram:400
a=T38FaxUdpEC:t38UDPFEC

---
srvtm11*CLI>
<--- SIP read from 10.0.101.13:5061 --->
SIP/2.0 200 OK
Call-ID: call-F127CA6A-2AA2-2110-0C1A-4B@10.0.101.13
Contact: <sip:603@10.0.101.13:5061>
Content-Length: 253
Content-Type: application/sdp
CSeq: 102 INVITE
From: <sip:602@80.251.131.142>;tag=as67ad695e
To: <sip:603@80.251.131.142>;tag=a00650d-36
User-Agent: Quintum/1.0.0 SN/0030E1103326 SW/S107-04-00
Via: SIP/2.0/UDP 80.251.131.142:5060;branch=z9hG4bK39639742;rport

v=0
o=Quintum 412 5976 IN IP4 10.0.101.13
s=VoipCall
c=IN IP4 10.0.101.13
t=0 0
m=image 10310 udptl t38
c=IN IP4 10.0.101.13
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
<------------->
--- (10 headers 12 lines) ---
Got T.38 offer in SDP in dialog call-F127CA6A-2AA2-2110-0C1A-4B@10.0.101.13
Got T.38 Re-invite without audio. Keeping RTP active during T.38 session. Callid call-F127CA6A-2AA2-2110-0C1A-4B@10.0.101.13
Capabilities: us - 0x108 (alaw|g729), peer - audio=0x0 (nothing)/video=0x0 (nothing), combined - 0x0 (nothing)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
set_destination: Parsing <sip:603@10.0.101.13:5061> for address/port to send to
set_destination: set destination to 10.0.101.13, port 5061
Transmitting (no NAT) to 10.0.101.13:5061:
ACK sip:603@10.0.101.13:5061 SIP/2.0
Via: SIP/2.0/UDP 80.251.131.142:5060;branch=z9hG4bK6e0315f3;rport
From: <sip:602@80.251.131.142>;tag=as67ad695e
To: <sip:603@80.251.131.142>;tag=a00650d-36
Contact: <sip:602@80.251.131.142>
Call-ID: call-F127CA6A-2AA2-2110-0C1A-4B@10.0.101.13
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
set_destination: Parsing <sip:603@10.0.101.13:5061> for address/port to send to
set_destination: set destination to 10.0.101.13, port 5061
Audio is at 80.251.131.142 port 19334
Adding codec 0x8 (alaw) to SDP
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.0.101.13:5061:
INVITE sip:603@10.0.101.13:5061 SIP/2.0
Via: SIP/2.0/UDP 80.251.131.142:5060;branch=z9hG4bK749ec85c;rport
From: <sip:602@80.251.131.142>;tag=as67ad695e
To: <sip:603@80.251.131.142>;tag=a00650d-36
Contact: <sip:602@80.251.131.142>
Call-ID: call-F127CA6A-2AA2-2110-0C1A-4B@10.0.101.13
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 289

v=0
o=root 5974 5977 IN IP4 80.251.131.142
s=session
c=IN IP4 80.251.131.142
t=0 0
m=audio 19334 RTP/AVP 8 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
srvtm11*CLI>
<--- SIP read from 10.0.101.13:5061 --->
SIP/2.0 200 OK
Call-ID: call-F127CA6A-2AA2-2110-0C1A-4B@10.0.101.13
Contact: <sip:603@10.0.101.13:5061>
Content-Length: 221
Content-Type: application/sdp
CSeq: 103 INVITE
From: <sip:602@80.251.131.142>;tag=as67ad695e
To: <sip:603@80.251.131.142>;tag=a00650d-36
User-Agent: Quintum/1.0.0 SN/0030E1103326 SW/S107-04-00
Via: SIP/2.0/UDP 80.251.131.142:5060;branch=z9hG4bK749ec85c;rport

v=0
o=Quintum 413 5977 IN IP4 10.0.101.13
s=VoipCall
c=IN IP4 10.0.101.13
t=0 0
m=audio 10310 RTP/AVP 8 101
c=IN IP4 10.0.101.13
a=rtpmap:8 pcma/8000/1
a=ptime:20
a=rtpmap:101 telephone-event/8000/1
a=sendrecv
<------------->
--- (10 headers 11 lines) ---
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 10.0.101.13:10310
Found audio description format pcma for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - 0x108 (alaw|g729), 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 10.0.101.13:10310
set_destination: Parsing <sip:603@10.0.101.13:5061> for address/port to send to
set_destination: set destination to 10.0.101.13, port 5061
Transmitting (no NAT) to 10.0.101.13:5061:
ACK sip:603@10.0.101.13:5061 SIP/2.0
Via: SIP/2.0/UDP 80.251.131.142:5060;branch=z9hG4bK53f3c32c;rport
From: <sip:602@80.251.131.142>;tag=as67ad695e
To: <sip:603@80.251.131.142>;tag=a00650d-36
Contact: <sip:602@80.251.131.142>
Call-ID: call-F127CA6A-2AA2-2110-0C1A-4B@10.0.101.13
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
set_destination: Parsing <sip:603@10.0.101.13:5061> for address/port to send to
set_destination: set destination to 10.0.101.13, port 5061
Reliably Transmitting (no NAT) to 10.0.101.13:5061:
INVITE sip:603@10.0.101.13:5061 SIP/2.0
Via: SIP/2.0/UDP 80.251.131.142:5060;branch=z9hG4bK1892a450;rport
From: <sip:602@80.251.131.142>;tag=as67ad695e
To: <sip:603@80.251.131.142>;tag=a00650d-36
Contact: <sip:602@80.251.131.142>
Call-ID: call-F127CA6A-2AA2-2110-0C1A-4B@10.0.101.13
CSeq: 104 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
X-asterisk-info: SIP re-invite (T38 switchover)
Content-Type: application/sdp
Content-Length: 266

v=0
o=root 5974 5978 IN IP4 80.251.131.142
s=session
c=IN IP4 80.251.131.142
t=0 0
m=image 10720 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:400
a=T38FaxMaxDatagram:400
a=T38FaxUdpEC:t38UDPFEC

---
srvtm11*CLI>
<--- SIP read from 10.0.101.13:5061 --->
SIP/2.0 200 OK
Call-ID: call-F127CA6A-2AA2-2110-0C1A-4B@10.0.101.13
Contact: <sip:603@10.0.101.13:5061>
Content-Length: 253
Content-Type: application/sdp
CSeq: 104 INVITE
From: <sip:602@80.251.131.142>;tag=as67ad695e
To: <sip:603@80.251.131.142>;tag=a00650d-36
User-Agent: Quintum/1.0.0 SN/0030E1103326 SW/S107-04-00
Via: SIP/2.0/UDP 80.251.131.142:5060;branch=z9hG4bK1892a450;rport

v=0
o=Quintum 414 5978 IN IP4 10.0.101.13
s=VoipCall
c=IN IP4 10.0.101.13
t=0 0
m=image 10310 udptl t38
c=IN IP4 10.0.101.13
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
<------------->
--- (10 headers 12 lines) ---
Got T.38 offer in SDP in dialog call-F127CA6A-2AA2-2110-0C1A-4B@10.0.101.13
Got T.38 Re-invite without audio. Keeping RTP active during T.38 session. Callid call-F127CA6A-2AA2-2110-0C1A-4B@10.0.101.13
Capabilities: us - 0x108 (alaw|g729), peer - audio=0x0 (nothing)/video=0x0 (nothing), combined - 0x0 (nothing)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
set_destination: Parsing <sip:603@10.0.101.13:5061> for address/port to send to
set_destination: set destination to 10.0.101.13, port 5061
Transmitting (no NAT) to 10.0.101.13:5061:
ACK sip:603@10.0.101.13:5061 SIP/2.0
Via: SIP/2.0/UDP 80.251.131.142:5060;branch=z9hG4bK41382474;rport
From: <sip:602@80.251.131.142>;tag=as67ad695e
To: <sip:603@80.251.131.142>;tag=a00650d-36
Contact: <sip:602@80.251.131.142>
Call-ID: call-F127CA6A-2AA2-2110-0C1A-4B@10.0.101.13
CSeq: 104 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
set_destination: Parsing <sip:603@10.0.101.13:5061> for address/port to send to
set_destination: set destination to 10.0.101.13, port 5061
Audio is at 80.251.131.142 port 19334
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.0.101.13:5061:
INVITE sip:603@10.0.101.13:5061 SIP/2.0
Via: SIP/2.0/UDP 80.251.131.142:5060;branch=z9hG4bK77b691b3;rport
From: <sip:602@80.251.131.142>;tag=as67ad695e
To: <sip:603@80.251.131.142>;tag=a00650d-36
Contact: <sip:602@80.251.131.142>
Call-ID: call-F127CA6A-2AA2-2110-0C1A-4B@10.0.101.13
CSeq: 105 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 242

v=0
o=root 5974 5979 IN IP4 80.251.131.142
s=session
c=IN IP4 80.251.131.142
t=0 0
m=audio 19334 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

---
srvtm11*CLI>
<--- SIP read from 10.0.101.13:5061 --->
SIP/2.0 200 OK
Call-ID: call-F127CA6A-2AA2-2110-0C1A-4B@10.0.101.13
Contact: <sip:603@10.0.101.13:5061>
Content-Length: 221
Content-Type: application/sdp
CSeq: 105 INVITE
From: <sip:602@80.251.131.142>;tag=as67ad695e
To: <sip:603@80.251.131.142>;tag=a00650d-36
User-Agent: Quintum/1.0.0 SN/0030E1103326 SW/S107-04-00
Via: SIP/2.0/UDP 80.251.131.142:5060;branch=z9hG4bK77b691b3;rport

v=0
o=Quintum 415 5979 IN IP4 10.0.101.13
s=VoipCall
c=IN IP4 10.0.101.13
t=0 0
m=audio 10310 RTP/AVP 8 101
c=IN IP4 10.0.101.13
a=rtpmap:8 pcma/8000/1
a=ptime:20
a=rtpmap:101 telephone-event/8000/1
a=sendrecv
<------------->
--- (10 headers 11 lines) ---
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 10.0.101.13:10310
Found audio description format pcma for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - 0x108 (alaw|g729), 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 10.0.101.13:10310
set_destination: Parsing <sip:603@10.0.101.13:5061> for address/port to send to
set_destination: set destination to 10.0.101.13, port 5061
Transmitting (no NAT) to 10.0.101.13:5061:
ACK sip:603@10.0.101.13:5061 SIP/2.0
Via: SIP/2.0/UDP 80.251.131.142:5060;branch=z9hG4bK4aa397a9;rport
From: <sip:602@80.251.131.142>;tag=as67ad695e
To: <sip:603@80.251.131.142>;tag=a00650d-36
Contact: <sip:602@80.251.131.142>
Call-ID: call-F127CA6A-2AA2-2110-0C1A-4B@10.0.101.13
CSeq: 105 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
Really destroying SIP dialog 'call-F149C1EF-09A2-2110-0C1E-0@10.0.101.13' Method: REGISTER
srvtm11*CLI>
set_destination: Parsing <sip:603@10.0.101.13:5061> for address/port to send to
set_destination: set destination to 10.0.101.13, port 5061
Reliably Transmitting (no NAT) to 10.0.101.13:5061:
INVITE sip:603@10.0.101.13:5061 SIP/2.0
Via: SIP/2.0/UDP 80.251.131.142:5060;branch=z9hG4bK637c1242;rport
From: <sip:602@80.251.131.142>;tag=as67ad695e
To: <sip:603@80.251.131.142>;tag=a00650d-36
Contact: <sip:602@80.251.131.142>
Call-ID: call-F127CA6A-2AA2-2110-0C1A-4B@10.0.101.13
CSeq: 106 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
X-asterisk-info: SIP re-invite (T38 switchover)
Content-Type: application/sdp
Content-Length: 266

v=0
o=root 5974 5980 IN IP4 80.251.131.142
s=session
c=IN IP4 80.251.131.142
t=0 0
m=image 10720 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:400
a=T38FaxMaxDatagram:400
a=T38FaxUdpEC:t38UDPFEC

---
srvtm11*CLI>
<--- SIP read from 10.0.101.13:5061 --->
SIP/2.0 200 OK
Call-ID: call-F127CA6A-2AA2-2110-0C1A-4B@10.0.101.13
Contact: <sip:603@10.0.101.13:5061>
Content-Length: 253
Content-Type: application/sdp
CSeq: 106 INVITE
From: <sip:602@80.251.131.142>;tag=as67ad695e
To: <sip:603@80.251.131.142>;tag=a00650d-36
User-Agent: Quintum/1.0.0 SN/0030E1103326 SW/S107-04-00
Via: SIP/2.0/UDP 80.251.131.142:5060;branch=z9hG4bK637c1242;rport

v=0
o=Quintum 416 5980 IN IP4 10.0.101.13
s=VoipCall
c=IN IP4 10.0.101.13
t=0 0
m=image 10310 udptl t38
c=IN IP4 10.0.101.13
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
<------------->
--- (10 headers 12 lines) ---
Got T.38 offer in SDP in dialog call-F127CA6A-2AA2-2110-0C1A-4B@10.0.101.13
Got T.38 Re-invite without audio. Keeping RTP active during T.38 session. Callid call-F127CA6A-2AA2-2110-0C1A-4B@10.0.101.13
Capabilities: us - 0x108 (alaw|g729), peer - audio=0x0 (nothing)/video=0x0 (nothing), combined - 0x0 (nothing)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
set_destination: Parsing <sip:603@10.0.101.13:5061> for address/port to send to
set_destination: set destination to 10.0.101.13, port 5061
Transmitting (no NAT) to 10.0.101.13:5061:
ACK sip:603@10.0.101.13:5061 SIP/2.0
Via: SIP/2.0/UDP 80.251.131.142:5060;branch=z9hG4bK035bc320;rport
From: <sip:602@80.251.131.142>;tag=as67ad695e
To: <sip:603@80.251.131.142>;tag=a00650d-36
Contact: <sip:602@80.251.131.142>
Call-ID: call-F127CA6A-2AA2-2110-0C1A-4B@10.0.101.13
CSeq: 106 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
set_destination: Parsing <sip:603@10.0.101.13:5061> for address/port to send to
set_destination: set destination to 10.0.101.13, port 5061
Audio is at 80.251.131.142 port 19334
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.0.101.13:5061:
INVITE sip:603@10.0.101.13:5061 SIP/2.0
Via: SIP/2.0/UDP 80.251.131.142:5060;branch=z9hG4bK3a8992a7;rport
From: <sip:602@80.251.131.142>;tag=as67ad695e
To: <sip:603@80.251.131.142>;tag=a00650d-36
Contact: <sip:602@80.251.131.142>
Call-ID: call-F127CA6A-2AA2-2110-0C1A-4B@10.0.101.13
CSeq: 107 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 242

v=0
o=root 5974 5981 IN IP4 80.251.131.142
s=session
c=IN IP4 80.251.131.142
t=0 0
m=audio 19334 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

---
srvtm11*CLI>
<--- SIP read from 10.0.101.13:5061 --->
SIP/2.0 200 OK
Call-ID: call-F127CA6A-2AA2-2110-0C1A-4B@10.0.101.13
Contact: <sip:603@10.0.101.13:5061>
Content-Length: 221
Content-Type: application/sdp
CSeq: 107 INVITE
From: <sip:602@80.251.131.142>;tag=as67ad695e
To: <sip:603@80.251.131.142>;tag=a00650d-36
User-Agent: Quintum/1.0.0 SN/0030E1103326 SW/S107-04-00
Via: SIP/2.0/UDP 80.251.131.142:5060;branch=z9hG4bK3a8992a7;rport

v=0
o=Quintum 417 5981 IN IP4 10.0.101.13
s=VoipCall
c=IN IP4 10.0.101.13
t=0 0
m=audio 10310 RTP/AVP 8 101
c=IN IP4 10.0.101.13
a=rtpmap:8 pcma/8000/1
a=ptime:20
a=rtpmap:101 telephone-event/8000/1
a=sendrecv
<------------->
--- (10 headers 11 lines) ---
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 10.0.101.13:10310
Found audio description format pcma for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - 0x108 (alaw|g729), 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 10.0.101.13:10310
set_destination: Parsing <sip:603@10.0.101.13:5061> for address/port to send to
set_destination: set destination to 10.0.101.13, port 5061
Transmitting (no NAT) to 10.0.101.13:5061:
ACK sip:603@10.0.101.13:5061 SIP/2.0
Via: SIP/2.0/UDP 80.251.131.142:5060;branch=z9hG4bK35a4ffa0;rport
From: <sip:602@80.251.131.142>;tag=as67ad695e
To: <sip:603@80.251.131.142>;tag=a00650d-36
Contact: <sip:602@80.251.131.142>
Call-ID: call-F127CA6A-2AA2-2110-0C1A-4B@10.0.101.13
CSeq: 107 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
set_destination: Parsing <sip:603@10.0.101.13:5061> for address/port to send to
set_destination: set destination to 10.0.101.13, port 5061
Reliably Transmitting (no NAT) to 10.0.101.13:5061:
INVITE sip:603@10.0.101.13:5061 SIP/2.0
Via: SIP/2.0/UDP 80.251.131.142:5060;branch=z9hG4bK6fc12158;rport
From: <sip:602@80.251.131.142>;tag=as67ad695e
To: <sip:603@80.251.131.142>;tag=a00650d-36
Contact: <sip:602@80.251.131.142>
Call-ID: call-F127CA6A-2AA2-2110-0C1A-4B@10.0.101.13
CSeq: 108 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
X-asterisk-info: SIP re-invite (T38 switchover)
Content-Type: application/sdp
Content-Length: 266

v=0
o=root 5974 5982 IN IP4 80.251.131.142
s=session
c=IN IP4 80.251.131.142
t=0 0
m=image 10720 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:400
a=T38FaxMaxDatagram:400
a=T38FaxUdpEC:t38UDPFEC

---
srvtm11*CLI>
<--- SIP read from 10.0.101.13:5061 --->
SIP/2.0 200 OK
Call-ID: call-F127CA6A-2AA2-2110-0C1A-4B@10.0.101.13
Contact: <sip:603@10.0.101.13:5061>
Content-Length: 253
Content-Type: application/sdp
CSeq: 108 INVITE
From: <sip:602@80.251.131.142>;tag=as67ad695e
To: <sip:603@80.251.131.142>;tag=a00650d-36
User-Agent: Quintum/1.0.0 SN/0030E1103326 SW/S107-04-00
Via: SIP/2.0/UDP 80.251.131.142:5060;branch=z9hG4bK6fc12158;rport

v=0
o=Quintum 418 5982 IN IP4 10.0.101.13
s=VoipCall
c=IN IP4 10.0.101.13
t=0 0
m=image 10310 udptl t38
c=IN IP4 10.0.101.13
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
<------------->
--- (10 headers 12 lines) ---
Got T.38 offer in SDP in dialog call-F127CA6A-2AA2-2110-0C1A-4B@10.0.101.13
Got T.38 Re-invite without audio. Keeping RTP active during T.38 session. Callid call-F127CA6A-2AA2-2110-0C1A-4B@10.0.101.13
Capabilities: us - 0x108 (alaw|g729), peer - audio=0x0 (nothing)/video=0x0 (nothing), combined - 0x0 (nothing)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
set_destination: Parsing <sip:603@10.0.101.13:5061> for address/port to send to
set_destination: set destination to 10.0.101.13, port 5061
Transmitting (no NAT) to 10.0.101.13:5061:
ACK sip:603@10.0.101.13:5061 SIP/2.0
Via: SIP/2.0/UDP 80.251.131.142:5060;branch=z9hG4bK2462385e;rport
From: <sip:602@80.251.131.142>;tag=as67ad695e
To: <sip:603@80.251.131.142>;tag=a00650d-36
Contact: <sip:602@80.251.131.142>
Call-ID: call-F127CA6A-2AA2-2110-0C1A-4B@10.0.101.13
CSeq: 108 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
set_destination: Parsing <sip:603@10.0.101.13:5061> for address/port to send to
set_destination: set destination to 10.0.101.13, port 5061
Audio is at 80.251.131.142 port 19334
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.0.101.13:5061:
INVITE sip:603@10.0.101.13:5061 SIP/2.0
Via: SIP/2.0/UDP 80.251.131.142:5060;branch=z9hG4bK3ff0cab0;rport
From: <sip:602@80.251.131.142>;tag=as67ad695e
To: <sip:603@80.251.131.142>;tag=a00650d-36
Contact: <sip:602@80.251.131.142>
Call-ID: call-F127CA6A-2AA2-2110-0C1A-4B@10.0.101.13
CSeq: 109 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 242

v=0
o=root 5974 5983 IN IP4 80.251.131.142
s=session
c=IN IP4 80.251.131.142
t=0 0
m=audio 19334 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

---
srvtm11*CLI>
<--- SIP read from 10.0.101.13:5061 --->
SIP/2.0 200 OK
Call-ID: call-F127CA6A-2AA2-2110-0C1A-4B@10.0.101.13
Contact: <sip:603@10.0.101.13:5061>
Content-Length: 221
Content-Type: application/sdp
CSeq: 109 INVITE
From: <sip:602@80.251.131.142>;tag=as67ad695e
To: <sip:603@80.251.131.142>;tag=a00650d-36
User-Agent: Quintum/1.0.0 SN/0030E1103326 SW/S107-04-00
Via: SIP/2.0/UDP 80.251.131.142:5060;branch=z9hG4bK3ff0cab0;rport

v=0
o=Quintum 419 5983 IN IP4 10.0.101.13
s=VoipCall
c=IN IP4 10.0.101.13
t=0 0
m=audio 10310 RTP/AVP 8 101
c=IN IP4 10.0.101.13
a=rtpmap:8 pcma/8000/1
a=ptime:20
a=rtpmap:101 telephone-event/8000/1
a=sendrecv
<------------->
--- (10 headers 11 lines) ---
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 10.0.101.13:10310
Found audio description format pcma for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - 0x108 (alaw|g729), 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 10.0.101.13:10310
set_destination: Parsing <sip:603@10.0.101.13:5061> for address/port to send to
set_destination: set destination to 10.0.101.13, port 5061
Transmitting (no NAT) to 10.0.101.13:5061:
ACK sip:603@10.0.101.13:5061 SIP/2.0
Via: SIP/2.0/UDP 80.251.131.142:5060;branch=z9hG4bK523ff881;rport
From: <sip:602@80.251.131.142>;tag=as67ad695e
To: <sip:603@80.251.131.142>;tag=a00650d-36
Contact: <sip:602@80.251.131.142>
Call-ID: call-F127CA6A-2AA2-2110-0C1A-4B@10.0.101.13
CSeq: 109 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
srvtm11*CLI>
set_destination: Parsing <sip:603@10.0.101.13:5061> for address/port to send to
set_destination: set destination to 10.0.101.13, port 5061
Reliably Transmitting (no NAT) to 10.0.101.13:5061:
INVITE sip:603@10.0.101.13:5061 SIP/2.0
Via: SIP/2.0/UDP 80.251.131.142:5060;branch=z9hG4bK4d2258e7;rport
From: <sip:602@80.251.131.142>;tag=as67ad695e
To: <sip:603@80.251.131.142>;tag=a00650d-36
Contact: <sip:602@80.251.131.142>
Call-ID: call-F127CA6A-2AA2-2110-0C1A-4B@10.0.101.13
CSeq: 110 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
X-asterisk-info: SIP re-invite (T38 switchover)
Content-Type: application/sdp
Content-Length: 266

v=0
o=root 5974 5984 IN IP4 80.251.131.142
s=session
c=IN IP4 80.251.131.142
t=0 0
m=image 10720 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:400
a=T38FaxMaxDatagram:400
a=T38FaxUdpEC:t38UDPFEC

---
srvtm11*CLI>
<--- SIP read from 10.0.101.13:5061 --->
SIP/2.0 200 OK
Call-ID: call-F127CA6A-2AA2-2110-0C1A-4B@10.0.101.13
Contact: <sip:603@10.0.101.13:5061>
Content-Length: 253
Content-Type: application/sdp
CSeq: 110 INVITE
From: <sip:602@80.251.131.142>;tag=as67ad695e
To: <sip:603@80.251.131.142>;tag=a00650d-36
User-Agent: Quintum/1.0.0 SN/0030E1103326 SW/S107-04-00
Via: SIP/2.0/UDP 80.251.131.142:5060;branch=z9hG4bK4d2258e7;rport

v=0
o=Quintum 420 5984 IN IP4 10.0.101.13
s=VoipCall
c=IN IP4 10.0.101.13
t=0 0
m=image 10310 udptl t38
c=IN IP4 10.0.101.13
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
<------------->
--- (10 headers 12 lines) ---
Got T.38 offer in SDP in dialog call-F127CA6A-2AA2-2110-0C1A-4B@10.0.101.13
Got T.38 Re-invite without audio. Keeping RTP active during T.38 session. Callid call-F127CA6A-2AA2-2110-0C1A-4B@10.0.101.13
Capabilities: us - 0x108 (alaw|g729), peer - audio=0x0 (nothing)/video=0x0 (nothing), combined - 0x0 (nothing)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
set_destination: Parsing <sip:603@10.0.101.13:5061> for address/port to send to
set_destination: set destination to 10.0.101.13, port 5061
Transmitting (no NAT) to 10.0.101.13:5061:
ACK sip:603@10.0.101.13:5061 SIP/2.0
Via: SIP/2.0/UDP 80.251.131.142:5060;branch=z9hG4bK6c98ecb5;rport
From: <sip:602@80.251.131.142>;tag=as67ad695e
To: <sip:603@80.251.131.142>;tag=a00650d-36
Contact: <sip:602@80.251.131.142>
Call-ID: call-F127CA6A-2AA2-2110-0C1A-4B@10.0.101.13
CSeq: 110 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
set_destination: Parsing <sip:603@10.0.101.13:5061> for address/port to send to
set_destination: set destination to 10.0.101.13, port 5061
Audio is at 80.251.131.142 port 19334
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.0.101.13:5061:
INVITE sip:603@10.0.101.13:5061 SIP/2.0
Via: SIP/2.0/UDP 80.251.131.142:5060;branch=z9hG4bK14d3aee7;rport
From: <sip:602@80.251.131.142>;tag=as67ad695e
To: <sip:603@80.251.131.142>;tag=a00650d-36
Contact: <sip:602@80.251.131.142>
Call-ID: call-F127CA6A-2AA2-2110-0C1A-4B@10.0.101.13
CSeq: 111 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 242

v=0
o=root 5974 5985 IN IP4 80.251.131.142
s=session
c=IN IP4 80.251.131.142
t=0 0
m=audio 19334 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

---
srvtm11*CLI>
<--- SIP read from 10.0.101.13:5061 --->
SIP/2.0 200 OK
Call-ID: call-F127CA6A-2AA2-2110-0C1A-4B@10.0.101.13
Contact: <sip:603@10.0.101.13:5061>
Content-Length: 221
Content-Type: application/sdp
CSeq: 111 INVITE
From: <sip:602@80.251.131.142>;tag=as67ad695e
To: <sip:603@80.251.131.142>;tag=a00650d-36
User-Agent: Quintum/1.0.0 SN/0030E1103326 SW/S107-04-00
Via: SIP/2.0/UDP 80.251.131.142:5060;branch=z9hG4bK14d3aee7;rport

v=0
o=Quintum 421 5985 IN IP4 10.0.101.13
s=VoipCall
c=IN IP4 10.0.101.13
t=0 0
m=audio 10310 RTP/AVP 8 101
c=IN IP4 10.0.101.13
a=rtpmap:8 pcma/8000/1
a=ptime:20
a=rtpmap:101 telephone-event/8000/1
a=sendrecv
<------------->
--- (10 headers 11 lines) ---
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 10.0.101.13:10310
Found audio description format pcma for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - 0x108 (alaw|g729), 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 10.0.101.13:10310
set_destination: Parsing <sip:603@10.0.101.13:5061> for address/port to send to
set_destination: set destination to 10.0.101.13, port 5061
Transmitting (no NAT) to 10.0.101.13:5061:
ACK sip:603@10.0.101.13:5061 SIP/2.0
Via: SIP/2.0/UDP 80.251.131.142:5060;branch=z9hG4bK150def41;rport
From: <sip:602@80.251.131.142>;tag=as67ad695e
To: <sip:603@80.251.131.142>;tag=a00650d-36
Contact: <sip:602@80.251.131.142>
Call-ID: call-F127CA6A-2AA2-2110-0C1A-4B@10.0.101.13
CSeq: 111 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
set_destination: Parsing <sip:603@10.0.101.13:5061> for address/port to send to
set_destination: set destination to 10.0.101.13, port 5061
Reliably Transmitting (no NAT) to 10.0.101.13:5061:
INVITE sip:603@10.0.101.13:5061 SIP/2.0
Via: SIP/2.0/UDP 80.251.131.142:5060;branch=z9hG4bK1918d53a;rport
From: <sip:602@80.251.131.142>;tag=as67ad695e
To: <sip:603@80.251.131.142>;tag=a00650d-36
Contact: <sip:602@80.251.131.142>
Call-ID: call-F127CA6A-2AA2-2110-0C1A-4B@10.0.101.13
CSeq: 112 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
X-asterisk-info: SIP re-invite (T38 switchover)
Content-Type: application/sdp
Content-Length: 266

v=0
o=root 5974 5986 IN IP4 80.251.131.142
s=session
c=IN IP4 80.251.131.142
t=0 0
m=image 10720 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:400
a=T38FaxMaxDatagram:400
a=T38FaxUdpEC:t38UDPFEC

---
srvtm11*CLI>
<--- SIP read from 10.0.101.13:5061 --->
SIP/2.0 200 OK
Call-ID: call-F127CA6A-2AA2-2110-0C1A-4B@10.0.101.13
Contact: <sip:603@10.0.101.13:5061>
Content-Length: 253
Content-Type: application/sdp
CSeq: 112 INVITE
From: <sip:602@80.251.131.142>;tag=as67ad695e
To: <sip:603@80.251.131.142>;tag=a00650d-36
User-Agent: Quintum/1.0.0 SN/0030E1103326 SW/S107-04-00
Via: SIP/2.0/UDP 80.251.131.142:5060;branch=z9hG4bK1918d53a;rport

v=0
o=Quintum 422 5986 IN IP4 10.0.101.13
s=VoipCall
c=IN IP4 10.0.101.13
t=0 0
m=image 10310 udptl t38
c=IN IP4 10.0.101.13
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
<------------->
--- (10 headers 12 lines) ---
Got T.38 offer in SDP in dialog call-F127CA6A-2AA2-2110-0C1A-4B@10.0.101.13
Got T.38 Re-invite without audio. Keeping RTP active during T.38 session. Callid call-F127CA6A-2AA2-2110-0C1A-4B@10.0.101.13
Capabilities: us - 0x108 (alaw|g729), peer - audio=0x0 (nothing)/video=0x0 (nothing), combined - 0x0 (nothing)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
set_destination: Parsing <sip:603@10.0.101.13:5061> for address/port to send to
set_destination: set destination to 10.0.101.13, port 5061
Transmitting (no NAT) to 10.0.101.13:5061:
ACK sip:603@10.0.101.13:5061 SIP/2.0
Via: SIP/2.0/UDP 80.251.131.142:5060;branch=z9hG4bK02672c4c;rport
From: <sip:602@80.251.131.142>;tag=as67ad695e
To: <sip:603@80.251.131.142>;tag=a00650d-36
Contact: <sip:602@80.251.131.142>
Call-ID: call-F127CA6A-2AA2-2110-0C1A-4B@10.0.101.13
CSeq: 112 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
set_destination: Parsing <sip:603@10.0.101.13:5061> for address/port to send to
set_destination: set destination to 10.0.101.13, port 5061
Audio is at 80.251.131.142 port 19334
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.0.101.13:5061:
INVITE sip:603@10.0.101.13:5061 SIP/2.0
Via: SIP/2.0/UDP 80.251.131.142:5060;branch=z9hG4bK29e077d3;rport
From: <sip:602@80.251.131.142>;tag=as67ad695e
To: <sip:603@80.251.131.142>;tag=a00650d-36
Contact: <sip:602@80.251.131.142>
Call-ID: call-F127CA6A-2AA2-2110-0C1A-4B@10.0.101.13
CSeq: 113 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 242

v=0
o=root 5974 5987 IN IP4 80.251.131.142
s=session
c=IN IP4 80.251.131.142
t=0 0
m=audio 19334 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

---
srvtm11*CLI>
<--- SIP read from 10.0.101.13:5061 --->
SIP/2.0 200 OK
Call-ID: call-F127CA6A-2AA2-2110-0C1A-4B@10.0.101.13
Contact: <sip:603@10.0.101.13:5061>
Content-Length: 221
Content-Type: application/sdp
CSeq: 113 INVITE
From: <sip:602@80.251.131.142>;tag=as67ad695e
To: <sip:603@80.251.131.142>;tag=a00650d-36
User-Agent: Quintum/1.0.0 SN/0030E1103326 SW/S107-04-00
Via: SIP/2.0/UDP 80.251.131.142:5060;branch=z9hG4bK29e077d3;rport

v=0
o=Quintum 423 5987 IN IP4 10.0.101.13
s=VoipCall
c=IN IP4 10.0.101.13
t=0 0
m=audio 10310 RTP/AVP 8 101
c=IN IP4 10.0.101.13
a=rtpmap:8 pcma/8000/1
a=ptime:20
a=rtpmap:101 telephone-event/8000/1
a=sendrecv
<------------->
--- (10 headers 11 lines) ---
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 10.0.101.13:10310
Found audio description format pcma for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - 0x108 (alaw|g729), 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 10.0.101.13:10310
set_destination: Parsing <sip:603@10.0.101.13:5061> for address/port to send to
set_destination: set destination to 10.0.101.13, port 5061
Transmitting (no NAT) to 10.0.101.13:5061:
ACK sip:603@10.0.101.13:5061 SIP/2.0
Via: SIP/2.0/UDP 80.251.131.142:5060;branch=z9hG4bK44d96674;rport
From: <sip:602@80.251.131.142>;tag=as67ad695e
To: <sip:603@80.251.131.142>;tag=a00650d-36
Contact: <sip:602@80.251.131.142>
Call-ID: call-F127CA6A-2AA2-2110-0C1A-4B@10.0.101.13
CSeq: 113 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0