<-- SIP read from 10.131.2.1:5060:
INVITE sip:*2@10.131.0.1;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.131.2.1;branch=z9hG4bKacDtiFcRG
Max-Forwards: 70
From: <sip:070001@10.131.0.1>;tag=1c1683226402
To: <sip:*2@10.131.0.1;user=phone>
Call-ID: 3123424816EbiX@10.131.2.1
CSeq: 1 INVITE
Contact: <sip:070001@10.131.2.1>
Supported: em,100rel,timer,replaces,path
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
User-Agent: Audiocodes-Sip-Gateway-MP-124 FXS/v.4.60A.008.006
Content-Type: application/sdp
Content-Length: 242

v=0
o=AudiocodesGW 601101 377877 IN IP4 10.131.2.1
s=Phone-Call
c=IN IP4 10.131.2.1
t=0 0
m=audio 6070 RTP/AVP 8 0 96
a=rtpmap:8 pcma/8000
a=rtpmap:0 pcmu/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
a=ptime:20
a=sendrecv
--- (13 headers 12 lines)---
Oct  4 19:13:37 DEBUG[11867]: chan_sip.c:3101 find_call: **** Creating new SIP dialog for incoming request 3123424816EbiX@10.131.2.1
Using INVITE request as basis request - 3123424816EbiX@10.131.2.1
Sending to 10.131.2.1 : 5060 (non-NAT)
Oct  4 19:13:37 DEBUG[11867]: chan_sip.c:6910 check_user_full: Setting NAT on RTP to 0
Reliably Transmitting (no NAT) to 10.131.2.1:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.131.2.1;branch=z9hG4bKacDtiFcRG;received=10.131.2.1
From: <sip:070001@10.131.0.1>;tag=1c1683226402
To: <sip:*2@10.131.0.1;user=phone>;tag=as3eb9a673
Call-ID: 3123424816EbiX@10.131.2.1
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:*2@10.131.0.1>
Proxy-Authenticate: Digest realm="asterisk", nonce="25ef9f25"
Content-Length: 0


---
Scheduling destruction of call '3123424816EbiX@10.131.2.1' in 15000 ms
Found user '070001'
localhost*CLI>
<-- SIP read from 10.131.2.1:5060:
ACK sip:*2@10.131.0.1;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.131.2.1;branch=z9hG4bKacDtiFcRG
Max-Forwards: 70
From: <sip:070001@10.131.0.1>;tag=1c1683226402
To: <sip:*2@10.131.0.1;user=phone>;tag=as3eb9a673
Call-ID: 3123424816EbiX@10.131.2.1
CSeq: 1 ACK
Contact: <sip:070001@10.131.2.1>
Supported: em,timer,replaces,path
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
User-Agent: Audiocodes-Sip-Gateway-MP-124 FXS/v.4.60A.008.006
Content-Length: 0

--- (12 headers 0 lines)---
Oct  4 19:13:37 DEBUG[11867]: chan_sip.c:3094 find_call: **** Found existing SIP dialog for incoming request 3123424816EbiX@10.131.2.1
Oct  4 19:13:37 DEBUG[11867]: chan_sip.c:1347 __sip_ack: Stopping retransmission on '3123424816EbiX@10.131.2.1' of Response 1: Match Found
localhost*CLI>
<-- SIP read from 10.131.2.1:5060:
INVITE sip:*2@10.131.0.1;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.131.2.1;branch=z9hG4bKacmDKrxwl
Max-Forwards: 70
From: <sip:070001@10.131.0.1>;tag=1c1683226402
To: <sip:*2@10.131.0.1;user=phone>
Call-ID: 3123424816EbiX@10.131.2.1
CSeq: 2 INVITE
Proxy-Authorization: Digest username="070001",realm="asterisk",nonce="25ef9f25" ",uri="sip:*2@10.131.0.1",algorithm=MD5,response="9ddb77341d596656ee77cf121ad99497"
Contact: <sip:070001@10.131.2.1>
Supported: em,100rel,timer,replaces,path
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
User-Agent: Audiocodes-Sip-Gateway-MP-124 FXS/v.4.60A.008.006
Content-Type: application/sdp
Content-Length: 242

v=0
o=AudiocodesGW 601101 377877 IN IP4 10.131.2.1
s=Phone-Call
c=IN IP4 10.131.2.1
t=0 0
m=audio 6070 RTP/AVP 8 0 96
a=rtpmap:8 pcma/8000
a=rtpmap:0 pcmu/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
a=ptime:20
a=sendrecv
--- (14 headers 12 lines)---
Oct  4 19:13:37 DEBUG[11867]: chan_sip.c:3094 find_call: **** Found existing SIP dialog for incoming request 3123424816EbiX@10.131.2.1
Using INVITE request as basis request - 3123424816EbiX@10.131.2.1
Sending to 10.131.2.1 : 5060 (non-NAT)
Oct  4 19:13:37 DEBUG[11867]: chan_sip.c:6910 check_user_full: Setting NAT on RTP to 0
Oct  4 19:13:37 NOTICE[11867]: chan_sip.c:6189 check_auth: Bad authentication from '<sip:*2@10.131.0.1;user=phone>' (got , expected 751c5cfa)
Reliably Transmitting (no NAT) to 10.131.2.1:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.131.2.1;branch=z9hG4bKacmDKrxwl;received=10.131.2.1
From: <sip:070001@10.131.0.1>;tag=1c1683226402
To: <sip:*2@10.131.0.1;user=phone>;tag=as3eb9a673
Call-ID: 3123424816EbiX@10.131.2.1
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:*2@10.131.0.1>
Proxy-Authenticate: Digest realm="asterisk", nonce="751c5cfa"
Content-Length: 0


---
Scheduling destruction of call '3123424816EbiX@10.131.2.1' in 15000 ms
Found user '070001'
localhost*CLI>
<-- SIP read from 10.131.2.1:5060:
ACK sip:*2@10.131.0.1;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.131.2.1;branch=z9hG4bKacmDKrxwl
Max-Forwards: 70
From: <sip:070001@10.131.0.1>;tag=1c1683226402
To: <sip:*2@10.131.0.1;user=phone>;tag=as3eb9a673
Call-ID: 3123424816EbiX@10.131.2.1
CSeq: 2 ACK
Contact: <sip:070001@10.131.2.1>
Supported: em,timer,replaces,path
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
User-Agent: Audiocodes-Sip-Gateway-MP-124 FXS/v.4.60A.008.006
Content-Length: 0

--- (12 headers 0 lines)---
Oct  4 19:13:37 DEBUG[11867]: chan_sip.c:3094 find_call: **** Found existing SIP dialog for incoming request 3123424816EbiX@10.131.2.1
Oct  4 19:13:37 DEBUG[11867]: chan_sip.c:1347 __sip_ack: Stopping retransmission on '3123424816EbiX@10.131.2.1' of Response 2: Match Found
Oct  4 19:13:52 DEBUG[11867]: chan_sip.c:1269 __sip_autodestruct: Auto destroying call '3123424816EbiX@10.131.2.1'
Destroying call '3123424816EbiX@10.131.2.1'
localhost*CLI>