<--- SIP read from UDP://10.150.65.16:5060 --->
INVITE sip:5421047008@10.153.66.146:5060;transport=UDP;user=phone SIP/2.0
f: <sip:5433142499@10.150.65.16:5060;user=phone>;tag=c4-45026-4a3cc2-235d1e0b-4a3cc2
t: <sip:5421047008@10.153.66.146:5060;user=phone>
i: a733f4f01041960a13c44a3cc221fd57e0cfa105d5bfd9f58-0322-5177
CSeq: 1 INVITE
User-agent: CS2000_NGSS/9.0
P-Asserted-Identity: <sip:5433142499@10.150.65.16;user=phone>
Allow: ACK,BYE,CANCEL,INVITE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY,PRACK
v: SIP/2.0/UDP PAE1CS2K:5060;maddr=10.150.65.16;branch=z9hG4bK-4a3cc2-21fd5906-37158b38
Max-Forwards: 70
m: <sip:10.150.65.16:5060;transport=UDP>
k: 100rel
c: application/sdp
l: 416

v=0
o=PVG 1265365362840 1265365362840 IN IP4 10.152.0.164
s=-
p=+1 6135555555
c=IN IP4 10.152.0.164
t=0 0
m=audio 49760 RTP/AVP 18 8 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=fmtp:18 annexb=no
m=image 57952 udptl t38
a=T38FaxVersion:0
a=T38FaxMaxBuffer:1100
a=T38FaxMaxDatagram:612
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxUdpEC:t38UDPRedundancy

<------------->
[Feb  5 08:24:25] --- (14 headers 18 lines) ---
[Feb  5 08:24:25]   == Using SIP RTP TOS bits 184
[Feb  5 08:24:25]   == Using SIP RTP CoS mark 5
[Feb  5 08:24:25]   == Using SIP VRTP TOS bits 136
[Feb  5 08:24:25]   == Using SIP VRTP CoS mark 6
[Feb  5 08:24:25]   == Using UDPTL TOS bits 184
[Feb  5 08:24:25]   == Using UDPTL CoS mark 5
[Feb  5 08:24:25] Sending to 10.150.65.16 : 5060 (NAT)
[Feb  5 08:24:25] Using INVITE request as basis request - a733f4f01041960a13c44a3cc221fd57e0cfa105d5bfd9f58-0322-5177
[Feb  5 08:24:25] Found peer 'voxip' for '5433142499' from 10.150.65.16:5060
[Feb  5 08:24:25] Found RTP audio format 18
[Feb  5 08:24:25] Found RTP audio format 8
[Feb  5 08:24:25] Found RTP audio format 101
[Feb  5 08:24:25] Found audio description format telephone-event for ID 101
[Feb  5 08:24:25] Got T.38 offer in SDP in dialog a733f4f01041960a13c44a3cc221fd57e0cfa105d5bfd9f58-0322-5177
[Feb  5 08:24:25] Capabilities: us - 0x108 (alaw|g729), peer - audio=0x108 (alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x108 (alaw|g729)
[Feb  5 08:24:25] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Feb  5 08:24:25] Peer audio RTP is at port 10.152.0.164:49760
[Feb  5 08:24:25] Looking for 5421047008 in entrada-e1 (domain 10.153.66.146)
[Feb  5 08:24:25] list_route: hop: <sip:10.150.65.16:5060;transport=UDP>
[Feb  5 08:24:25] 
<--- Transmitting (no NAT) to 10.150.65.16:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP PAE1CS2K:5060;maddr=10.150.65.16;branch=z9hG4bK-4a3cc2-21fd5906-37158b38;received=10.150.65.16
From: <sip:5433142499@10.150.65.16:5060;user=phone>;tag=c4-45026-4a3cc2-235d1e0b-4a3cc2
To: <sip:5421047008@10.153.66.146:5060;user=phone>
Call-ID: a733f4f01041960a13c44a3cc221fd57e0cfa105d5bfd9f58-0322-5177
CSeq: 1 INVITE
Server: Asterisk PBX 1.6.1.14
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:5421047008@10.153.66.146>
Content-Length: 0


<------------>
[Feb  5 08:24:25]     -- Executing [5421047008@entrada-e1:1] Goto("SIP/voxip-0000006c", "interno,7008,1") in new stack
[Feb  5 08:24:25]     -- Goto (interno,7008,1)
[Feb  5 08:24:25]     -- Executing [7008@interno:1] Goto("SIP/voxip-0000006c", "macro-recebefax,s,1") in new stack
[Feb  5 08:24:25]     -- Goto (macro-recebefax,s,1)
[Feb  5 08:24:25]     -- Executing [s@macro-recebefax:1] Set("SIP/voxip-0000006c", "DB(fax/count)=18") in new stack
[Feb  5 08:24:25]     -- Executing [s@macro-recebefax:2] Set("SIP/voxip-0000006c", "FAXCOUNT=18") in new stack
[Feb  5 08:24:25]     -- Executing [s@macro-recebefax:3] Set("SIP/voxip-0000006c", "FAXFILE=fax-18-rx") in new stack
[Feb  5 08:24:25]     -- Executing [s@macro-recebefax:4] ReceiveFAX("SIP/voxip-0000006c", "/var/spool/asterisk/fax/fax-18-rx.tif,d") in new stack
[Feb  5 08:24:25] Audio is at 10.153.66.146 port 13624
[Feb  5 08:24:25] Adding codec 0x100 (g729) to SDP
[Feb  5 08:24:25] Adding codec 0x8 (alaw) to SDP
[Feb  5 08:24:25] Adding non-codec 0x1 (telephone-event) to SDP
[Feb  5 08:24:25] 
<--- Reliably Transmitting (no NAT) to 10.150.65.16:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP PAE1CS2K:5060;maddr=10.150.65.16;branch=z9hG4bK-4a3cc2-21fd5906-37158b38;received=10.150.65.16
From: <sip:5433142499@10.150.65.16:5060;user=phone>;tag=c4-45026-4a3cc2-235d1e0b-4a3cc2
To: <sip:5421047008@10.153.66.146:5060;user=phone>;tag=as20749f8b
Call-ID: a733f4f01041960a13c44a3cc221fd57e0cfa105d5bfd9f58-0322-5177
CSeq: 1 INVITE
Server: Asterisk PBX 1.6.1.14
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:5421047008@10.153.66.146>
Content-Type: application/sdp
Content-Length: 311

v=0
o=root 124129144 124129144 IN IP4 10.153.66.146
s=Asterisk PBX 1.6.1.14
c=IN IP4 10.153.66.146
t=0 0
m=audio 13624 RTP/AVP 18 8 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
m=image 0 udptl t38

<------------>
[Feb  5 08:24:25] 
<--- SIP read from UDP://10.150.65.16:5060 --->
ACK sip:5421047008@10.153.66.146 SIP/2.0
f: <sip:5433142499@10.150.65.16:5060;user=phone>;tag=c4-45026-4a3cc2-235d1e0b-4a3cc2
To: <sip:5421047008@10.153.66.146:5060;user=phone>;tag=as20749f8b
i: a733f4f01041960a13c44a3cc221fd57e0cfa105d5bfd9f58-0322-5177
CSeq: 1 ACK
User-agent: CS2000_NGSS/9.0
Max-Forwards: 70
Allow: ACK,BYE,CANCEL,INVITE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY,PRACK
v: SIP/2.0/UDP PAE1CS2K:5060;maddr=10.150.65.16;branch=z9hG4bK-4a3cc2-21fd592f-2227dbf0
m: <sip:10.150.65.16:5060;transport=UDP>
k: 100rel
l: 0


<------------->
[Feb  5 08:24:25] --- (12 headers 0 lines) ---
[Feb  5 08:24:25] set_destination: Parsing <sip:10.150.65.16:5060;transport=UDP> for address/port to send to
[Feb  5 08:24:25] set_destination: set destination to 10.150.65.16, port 5060
[Feb  5 08:24:25] Reliably Transmitting (no NAT) to 10.150.65.16:5060:
INVITE sip:10.150.65.16:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.153.66.146:5060;branch=z9hG4bK2cf6c44b;rport
Max-Forwards: 70
From: <sip:5421047008@10.153.66.146:5060;user=phone>;tag=as20749f8b
To: <sip:5433142499@10.150.65.16:5060;user=phone>;tag=c4-45026-4a3cc2-235d1e0b-4a3cc2
Contact: <sip:5421047008@10.153.66.146>
Call-ID: a733f4f01041960a13c44a3cc221fd57e0cfa105d5bfd9f58-0322-5177
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.1.14
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 346

v=0
o=root 124129144 124129145 IN IP4 10.153.66.146
s=Asterisk PBX 1.6.1.14
c=IN IP4 10.153.66.146
t=0 0
m=image 4915 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxFillBitRemoval
a=T38FaxTranscodingMMR
a=T38FaxTranscodingJBIG
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxDatagram:1400
a=T38FaxUdpEC:t38UDPRedundancy

---
[Feb  5 08:24:25] 
<--- SIP read from UDP://10.150.65.16:5060 --->
SIP/2.0 100 Trying
From: <sip:5421047008@10.153.66.146:5060;user=phone>;tag=as20749f8b
t: <sip:5433142499@10.150.65.16:5060;user=phone>;tag=c4-45026-4a3cc2-235d1e0b-4a3cc2
Call-ID: a733f4f01041960a13c44a3cc221fd57e0cfa105d5bfd9f58-0322-5177
CSeq: 102 INVITE
Server: CS2000_NGSS/9.0
k: 100rel
Allow: ACK,BYE,CANCEL,INVITE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY,PRACK
Via: SIP/2.0/UDP 10.153.66.146:5060;rport=5060;branch=z9hG4bK2cf6c44b
ontact: <sip:10.150.65.16:5060;transport=UDP>
Content-Length: 0


<------------->
[Feb  5 08:24:25] --- (11 headers 0 lines) ---
[Feb  5 08:24:25] 
<--- SIP read from UDP://10.150.65.16:5060 --->
SIP/2.0 200 OK
From: <sip:5421047008@10.153.66.146:5060;user=phone>;tag=as20749f8b
t: <sip:5433142499@10.150.65.16:5060;user=phone>;tag=c4-45026-4a3cc2-235d1e0b-4a3cc2
Call-ID: a733f4f01041960a13c44a3cc221fd57e0cfa105d5bfd9f58-0322-5177
CSeq: 102 INVITE
Server: CS2000_NGSS/9.0
k: 100rel
Allow: ACK,BYE,CANCEL,INVITE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY,PRACK
Via: SIP/2.0/UDP 10.153.66.146:5060;rport=5060;branch=z9hG4bK2cf6c44b
Contact: <sip:10.150.65.16:5060;transport=UDP>
c: application/sdp
Content-Length: 299

v=0
o=PVG 1265365363260 1265365363260 IN IP4 10.152.0.164
s=-
p=+1 6135555555
c=IN IP4 10.152.0.164
t=0 0
m=image 57952 udptl t38
a=T38FaxVersion:0
a=T38FaxMaxBuffer:1100
a=T38FaxMaxDatagram:612
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxUdpEC:t38UDPRedundancy

<------------->
[Feb  5 08:24:25] --- (12 headers 13 lines) ---
[Feb  5 08:24:25] Got T.38 offer in SDP in dialog a733f4f01041960a13c44a3cc221fd57e0cfa105d5bfd9f58-0322-5177
[Feb  5 08:24:25] Capabilities: us - 0x108 (alaw|g729), peer - audio=0x0 (nothing)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x0 (nothing)
[Feb  5 08:24:25] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
[Feb  5 08:24:25] Got T.38 Re-invite without audio. Keeping RTP active during T.38 session.
[Feb  5 08:24:25] set_destination: Parsing <sip:10.150.65.16:5060;transport=UDP> for address/port to send to
[Feb  5 08:24:25] set_destination: set destination to 10.150.65.16, port 5060
[Feb  5 08:24:25] Transmitting (no NAT) to 10.150.65.16:5060:
ACK sip:10.150.65.16:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.153.66.146:5060;branch=z9hG4bK2b68c474;rport
Max-Forwards: 70
From: <sip:5421047008@10.153.66.146:5060;user=phone>;tag=as20749f8b
To: <sip:5433142499@10.150.65.16:5060;user=phone>;tag=c4-45026-4a3cc2-235d1e0b-4a3cc2
Contact: <sip:5421047008@10.153.66.146>
Call-ID: a733f4f01041960a13c44a3cc221fd57e0cfa105d5bfd9f58-0322-5177
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.1.14
Content-Length: 0