<--- SIP read from 172.25.51.1:10011 --->
INVITE sip:375@172.24.10.188;user=phone SIP/2.0
Supported: replaces,100rel
User-Agent: ABS GW v5.1.0
P-Asserted-Identity: "ISDN_T2" <sip:+521776599@mercury>
Content-Type: application/sdp
To: <sip:375@172.24.10.188;user=phone>
From: "ISDN_T2" <sip:+521776599@mercury>;tag=cc01ff37a60521d35da001f98edda0ac
Contact: sip:172.25.51.1
Call-ID: 82f9cd2d2e6c712d7528ef88802ba138@172.25.51.1
CSeq: 684819861 INVITE
Via: SIP/2.0/UDP 172.25.51.1;branch=z9hG4bK3e9645a733e59941286088d7a4945d8c
Max-Forwards: 70
Content-Length: 314

v=0
o=OXE 1279704517 1279704517 IN IP4 172.25.51.1
s=abs
c=IN IP4 172.25.51.4
t=0 0
m=audio 32712 RTP/AVP 8 0 4 97
a=sendrecv
a=rtpmap:8 PCMA/8000
a=ptime:20
a=maxptime:30
a=rtpmap:0 PCMU/8000
a=ptime:20
a=maxptime:30
a=rtpmap:4 G723/8000
a=ptime:30
a=maxptime:30
a=rtpmap:97 telephone-event/8000

<------------->
--- (13 headers 17 lines) ---
Sending to 172.25.51.1 : 5060 (no NAT)
Using INVITE request as basis request - 82f9cd2d2e6c712d7528ef88802ba138@172.25.51.1
Found peer '800'
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 4
Found RTP audio format 97
Peer audio RTP is at port 172.25.51.4:32712
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format G723 for ID 4
Found audio description format telephone-event for ID 97
Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0xd (g723|ulaw|alaw)/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 172.25.51.4:32712
Looking for 375 in sedoc (domain 172.24.10.188)
list_route: hop: <sip:172.25.51.1>

<--- Transmitting (no NAT) to 172.25.51.1:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.25.51.1;branch=z9hG4bK3e9645a733e59941286088d7a4945d8c;received=172.25.51.1
From: "ISDN_T2" <sip:+521776599@mercury>;tag=cc01ff37a60521d35da001f98edda0ac
To: <sip:375@172.24.10.188;user=phone>
Call-ID: 82f9cd2d2e6c712d7528ef88802ba138@172.25.51.1
CSeq: 684819861 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:375@172.24.10.188>
Content-Length: 0


<------------>
    -- Executing [375@sedoc:1] AGI("SIP/800-084250f8", "agi://127.0.0.1/mercury.agi") in new stack
    -- AGI Script agi://127.0.0.1/mercury.agi completed, returning 0
  == Auto fallthrough, channel 'SIP/800-084250f8' status is 'UNKNOWN'
Scheduling destruction of SIP dialog '82f9cd2d2e6c712d7528ef88802ba138@172.25.51.1' in 32000 ms (Method: INVITE)

<--- Reliably Transmitting (no NAT) to 172.25.51.1:5060 --->
SIP/2.0 603 Declined
Via: SIP/2.0/UDP 172.25.51.1;branch=z9hG4bK3e9645a733e59941286088d7a4945d8c;received=172.25.51.1
From: "ISDN_T2" <sip:+521776599@mercury>;tag=cc01ff37a60521d35da001f98edda0ac
To: <sip:375@172.24.10.188;user=phone>;tag=as3455cb36
Call-ID: 82f9cd2d2e6c712d7528ef88802ba138@172.25.51.1
CSeq: 684819861 INVITE
ser-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


<------------>
ccsedoc*CLI>
<--- SIP read from 172.25.51.1:10011 --->
ACK sip:375@172.24.10.188;user=phone SIP/2.0
Call-ID: 82f9cd2d2e6c712d7528ef88802ba138@172.25.51.1
From: "ISDN_T2" <sip:+521776599@mercury>;tag=cc01ff37a60521d35da001f98edda0ac
To: <sip:375@172.24.10.188;user=phone>;tag=as3455cb36
Via: SIP/2.0/UDP 172.25.51.1;branch=z9hG4bK3e9645a733e59941286088d7a4945d8c
CSeq: 684819861 ACK
Content-Length: 0


<------------->