<--- SIP read from UDP:127.0.0.1:5061 --->
jaK
<------------->

<--- SIP read from UDP:192.168.1.10:5061 --->
OPTIONS sip:615@192.168.1.10 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5061;rport;branch=z9hG4bK176322717
From: <sip:Minitel@192.168.0.1>;tag=1182982813
To: <sip:615@192.168.1.10>
Call-ID: 1646215648
CSeq: 20 OPTIONS
Accept: application/sdp
Max-Forwards: 70
User-Agent: Linphone/3.3.2 (eXosip2/3.3.0)
Expires: 120
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Looking for 3615 in default (domain 192.168.1.10)

<--- Transmitting (NAT) to 192.168.1.10:5061 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.1.10:5061;branch=z9hG4bK176322717;received=192.168.1.10;rport=5061
From: <sip:Minitel@192.168.0.1>;tag=1182982813
To: <sip:615@192.168.1.10>;tag=as313cc129
Call-ID: 1646215648
CSeq: 20 OPTIONS
Server: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Accept: application/sdp
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '1646215648' in 32000 ms (Method: OPTIONS)

<--- SIP read from UDP:192.168.1.10:5061 --->
INVITE sip:615@192.168.1.10 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5061;rport;branch=z9hG4bK1910835768
From: <sip:Minitel@192.168.0.1>;tag=1170425729
To: <sip:615@192.168.1.10>
Call-ID: 1434997741
CSeq: 20 INVITE
Contact: <sip:Minitel@192.168.1.10:5061>
Content-Type: application/sdp
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Max-Forwards: 70
User-Agent: Linphone/3.3.2 (eXosip2/3.3.0)
Subject: Phone call
Content-Length: 204

v=0
o=Minitel 123456 654321 IN IP4 192.168.1.10
s=A conversation
c=IN IP4 192.168.1.10
t=0 0
m=audio 7078 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-11
<------------->
--- (13 headers 9 lines) ---
Sending to 192.168.1.10:5061 (NAT)
Using INVITE request as basis request - 1434997741
Found peer 'Minitel' for 'Minitel' from 192.168.1.10:5061

<--- Reliably Transmitting (NAT) to 192.168.1.10:5061 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.10:5061;branch=z9hG4bK1910835768;received=192.168.1.10;rport=5061
From: <sip:Minitel@192.168.0.1>;tag=1170425729
To: <sip:615@192.168.1.10>;tag=as446755d0
Call-ID: 1434997741
CSeq: 20 INVITE
Server: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="72c2ce29"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '1434997741' in 32000 ms (Method: INVITE)

<--- SIP read from UDP:192.168.1.10:5061 --->
ACK sip:615@192.168.1.10 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5061;rport;branch=z9hG4bK1910835768
From: <sip:Minitel@192.168.0.1>;tag=1170425729
To: <sip:3615@192.168.1.10>;tag=as446755d0
Call-ID: 1434997741
CSeq: 20 ACK
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---

<--- SIP read from UDP:192.168.1.10:5061 --->
INVITE sip:615@192.168.1.10 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5061;rport;branch=z9hG4bK1422638500
From: <sip:Minitel@192.168.0.1>;tag=1170425729
To: <sip:3615@192.168.1.10>
Call-ID: 1434997741
CSeq: 21 INVITE
Contact: <sip:Minitel@192.168.1.10:5061>
Authorization: Digest username="Minitel", realm="asterisk", nonce="72c2ce29", uri="sip:3615@192.168.1.10", response="ee01bcb8c8ede83d564573fea64f254f", algorithm=MD5
Content-Type: application/sdp
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Max-Forwards: 70
User-Agent: Linphone/3.3.2 (eXosip2/3.3.0)
Subject: Phone call
Content-Length: 204

v=0
o=Minitel 123456 654321 IN IP4 192.168.1.10
s=A conversation
c=IN IP4 192.168.1.10
t=0 0
m=audio 7078 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-11
<------------->
--- (14 headers 9 lines) ---
Sending to 192.168.1.10:5061 (NAT)
Using INVITE request as basis request - 1434997741
Found peer 'Minitel' for 'Minitel' from 192.168.1.10:5061
  == Using SIP RTP CoS mark 5
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=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 192.168.1.10:7078
Looking for 615 in minitel_service (domain 192.168.1.10)
list_route: hop: <sip:Minitel@192.168.1.10:5061>

<--- Transmitting (NAT) to 192.168.1.10:5061 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.10:5061;branch=z9hG4bK1422638500;received=192.168.1.10;rport=5061
From: <sip:Minitel@192.168.0.1>;tag=1170425729
To: <sip:615@192.168.1.10>
Call-ID: 1434997741
CSeq: 21 INVITE
Server: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:615@192.168.1.10:5060>
Content-Length: 0


<------------>
    -- Executing [3615@minitel_service:1] Answer("SIP/Minitel-00000007", "") in new stack
Audio is at 15386
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to 192.168.1.10:5061 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.10:5061;branch=z9hG4bK1422638500;received=192.168.1.10;rport=5061
From: <sip:Minitel@192.168.0.1>;tag=1170425729
To: <sip:615@192.168.1.10>;tag=as591831fc
Call-ID: 1434997741
CSeq: 21 INVITE
Server: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:3615@192.168.1.10:5060>
Content-Type: application/sdp
Content-Length: 249

v=0
o=root 414451446 414451446 IN IP4 192.168.1.10
s=Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
c=IN IP4 192.168.1.10
t=0 0
m=audio 15386 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:30
a=sendrecv

<------------>
Retransmitting #1 (NAT) to 192.168.1.10:5061:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.10:5061;branch=z9hG4bK1422638500;received=192.168.1.10;rport=5061
From: <sip:Minitel@192.168.0.1>;tag=1170425729
To: <sip615@192.168.1.10>;tag=as591831fc
Call-ID: 1434997741
CSeq: 21 INVITE
Server: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:3615@192.168.1.10:5060>
Content-Type: application/sdp
Content-Length: 249

v=0
o=root 414451446 414451446 IN IP4 192.168.1.10
s=Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
c=IN IP4 192.168.1.10
t=0 0
m=audio 15386 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:30
a=sendrecv

---
    -- Executing [615@minitel_service:2] Monitor("SIP/Minitel-00000007", "wav,,m") in new stack
    -- Executing [615@minitel_service:3] Mymodule("SIP/Minitel-00000007", "") in new stack
[Apr 28 19:29:47] WARNING[5671]: app_dummy23.c:187 dummy23_exec: V23: slen total 29400

<--- SIP read from UDP:192.168.1.10:5061 --->
ACK sip:615@192.168.1.10:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5061;rport;branch=z9hG4bK1090991479
From: <sip:Minitel@192.168.0.1>;tag=1170425729
To: <sip:3615@192.168.1.10>;tag=as591831fc
Call-ID: 1434997741
CSeq: 21 ACK
Contact: <sip:Minitel@192.168.1.10:5061>
Max-Forwards: 70
User-Agent: Linphone/3.3.2 (eXosip2/3.3.0)
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---

<--- SIP read from UDP:192.168.1.10:5061 --->
BYE sip:615@192.168.1.10:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5061;rport;branch=z9hG4bK353204485
From: <sip:Minitel@192.168.0.1>;tag=1170425729
To: <sip:3615@192.168.1.10>;tag=as591831fc
Call-ID: 1434997741
CSeq: 22 BYE
Contact: <sip:Minitel@192.168.1.10:5061>
Max-Forwards: 70
User-Agent: Linphone/3.3.2 (eXosip2/3.3.0)
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Sending to 192.168.1.10:5061 (NAT)
Scheduling destruction of SIP dialog '1434997741' in 32000 ms (Method: BYE)

<--- Transmitting (NAT) to 192.168.1.10:5061 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.10:5061;branch=z9hG4bK353204485;received=192.168.1.10;rport=5061
From: <sip:Minitel@192.168.0.1>;tag=1170425729
To: <sip:615@192.168.1.10>;tag=as591831fc
Call-ID: 1434997741
CSeq: 22 BYE
Server: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0