
<--- SIP read from UDP:127.0.0.1:5061 --->
INVITE sip:K0001@127.0.0.1:5060 SIP/2.0
Via: SIP/2.0/UDP 127.0.1.1:5061;branch=z9hG4bK-19236-1-0
From: sipp <sip:sipp@127.0.1.1:5061>;tag=19236SIPpTag001
To: sut <sip:K0001@127.0.0.1:5060>
Call-ID: 1-19236@127.0.1.1
CSeq: 1 INVITE
Contact: sip:sipp@127.0.1.1:5061
Max-Forwards: 70
Subject: Performance Test
Content-Type: application/sdp
Content-Length:   127

v=0
o=user1 53655765 2353687637 IN IP4 127.0.1.1
s=-
c=IN IP4 127.0.1.1
t=0 0
m=audio 6000 RTP/AVP 0
a=rtpmap:0 PCMU/8000
<------------->
--- (11 headers 7 lines) ---
  == Using SIP RTP CoS mark 5
  == Using UDPTL CoS mark 5
Sending to 127.0.1.1 : 5061 (no NAT)
Using INVITE request as basis request - 1-19236@127.0.1.1
No matching peer for 'sipp' from '127.0.0.1:5061'
Found RTP audio format 0
Found audio description format PCMU for ID 0
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 127.0.1.1:6000
Looking for K0001 in default (domain 127.0.0.1)
list_route: hop: <sip:sipp@127.0.1.1:5061>

<--- Transmitting (no NAT) to 127.0.1.1:5061 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 127.0.1.1:5061;branch=z9hG4bK-19236-1-0;received=127.0.0.1
From: sipp <sip:sipp@127.0.1.1:5061>;tag=19236SIPpTag001
To: sut <sip:K0001@127.0.0.1:5060>
Call-ID: 1-19236@127.0.1.1
CSeq: 1 INVITE
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:K0001@127.0.0.1>
Content-Length: 0


<------------>
    -- Executing [K0001@default:1] NoOp("SIP/127.0.1.1:5061-00000066", "Ein Fax kommt an auf der ") in new stack
    -- Executing [K0001@default:2] Set("SIP/127.0.1.1:5061-00000066", "fax_id=1293707767102") in new stack
    -- Executing [K0001@default:3] Set("SIP/127.0.1.1:5061-00000066", "ORIG-TO-DDI=") in new stack
    -- Executing [K0001@default:4] Set("SIP/127.0.1.1:5061-00000066", "customer=K0001") in new stack
    -- Executing [K0001@default:5] Set("SIP/127.0.1.1:5061-00000066", "efax=") in new stack
    -- Executing [K0001@default:6] Set("SIP/127.0.1.1:5061-00000066", "pcap=20101230/12/20101230-121607-sipp-K0001-*") in new stack
    -- Executing [K0001@default:7] Set("SIP/127.0.1.1:5061-00000066", "LOCALSTATIONID=") in new stack
    -- Executing [K0001@default:8] ReceiveFAX("SIP/127.0.1.1:5061-00000066", "/tmp/fax-1293707767102.tif,f") in new stack
Audio is at 127.0.0.1 port 16648
Adding codec 0x4 (ulaw) to SDP

<--- Reliably Transmitting (no NAT) to 127.0.1.1:5061 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 127.0.1.1:5061;branch=z9hG4bK-19236-1-0;received=127.0.0.1
From: sipp <sip:sipp@127.0.1.1:5061>;tag=19236SIPpTag001
To: sut <sip:K0001@127.0.0.1:5060>;tag=as4174a974
Call-ID: 1-19236@127.0.1.1
CSeq: 1 INVITE
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:K0001@127.0.0.1>
Content-Type: application/sdp
Content-Length: 200

v=0
o=root 226736178 226736178 IN IP4 127.0.0.1
s=Asterisk PBX 1.6.2.13
c=IN IP4 127.0.0.1
t=0 0
m=audio 16648 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:127.0.0.1:5061 --->
ACK sip:K0001@127.0.0.1:5060 SIP/2.0
Via: SIP/2.0/UDP 127.0.1.1:5061;branch=z9hG4bK-19236-1-5
From: sipp <sip:sipp@127.0.1.1:5061>;tag=19236SIPpTag001
To: sut <sip:K0001@127.0.0.1:5060>;tag=as4174a974
Call-ID: 1-19236@127.0.1.1
CSeq: 1 ACK
Contact: sip:sipp@127.0.1.1:5061
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
set_destination: Parsing <sip:sipp@127.0.1.1:5061> for address/port to send to
set_destination: set destination to 127.0.1.1, port 5061
Reliably Transmitting (no NAT) to 127.0.1.1:5061:
INVITE sip:sipp@127.0.1.1:5061 SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK3c294cc8;rport
Max-Forwards: 70
From: sut <sip:K0001@127.0.0.1:5060>;tag=as4174a974
To: sipp <sip:sipp@127.0.1.1:5061>;tag=19236SIPpTag001
Contact: <sip:K0001@127.0.0.1>
Call-ID: 1-19236@127.0.1.1
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.13
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: 289

v=0
o=root 226736178 226736179 IN IP4 127.0.0.1
s=Asterisk PBX 1.6.2.13
c=IN IP4 127.0.0.1
t=0 0
m=image 4278 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxFillBitRemoval
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxDatagram:1400
a=T38FaxUdpEC:t38UDPRedundancy

---

<--- SIP read from UDP:127.0.0.1:5061 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK3c294cc8;rport
From: sut <sip:K0001@127.0.0.1:5060>;tag=as4174a974
To: sipp <sip:sipp@127.0.1.1:5061>;tag=19236SIPpTag001;tag=19236SIPpTag011
Call-ID: 1-19236@127.0.1.1
CSeq: 102 INVITE
Contact: <sip:127.0.1.1:5061;transport=UDP>
Content-Length: 0



testfax01*CLI> core show channels
Channel              Location             State   Application(Data)
SIP/127.0.1.1:5061-0 K0001@default:8      Up      ReceiveFAX(/tmp/fax-1293707767
1 active channel
1 active call
100 calls processed
testfax01*CLI>
testfax01*CLI>
[Dec 30 12:21:14] WARNING[19237]: app_fax.c:507 transmit_audio: It looks like we hung. Aborting.
[Dec 30 12:21:14] WARNING[19237]: app_fax.c:223 phase_e_handler: Error transmitting fax. result=49: The call dropped prematurely.
[Dec 30 12:21:14] WARNING[19237]: app_fax.c:817 transmit: Transmission error
  == Spawn extension (default, K0001, 8) exited non-zero on 'SIP/127.0.1.1:5061-00000066'
    -- Executing [h@default:1] NoOp("SIP/127.0.1.1:5061-00000066", "###       FAXSTATUS: FAILED") in new stack
    -- Executing [h@default:2] NoOp("SIP/127.0.1.1:5061-00000066", "###        FAXERROR: The call dropped prematurely") in new stack
    -- Executing [h@default:3] NoOp("SIP/127.0.1.1:5061-00000066", "###        FAXPAGES: ") in new stack
    -- Executing [h@default:4] NoOp("SIP/127.0.1.1:5061-00000066", "###      FAXBITRATE: ") in new stack
    -- Executing [h@default:5] NoOp("SIP/127.0.1.1:5061-00000066", "###   FAXRESOLUTION: ") in new stack
    -- Executing [h@default:6] NoOp("SIP/127.0.1.1:5061-00000066", "### REMOTESTATIONID: ") in new stack
    -- Executing [h@default:7] Set("SIP/127.0.1.1:5061-00000066", "remote_id=unbekannt") in new stack
testfax01*CLI>

 -------> SIP Channel is still activ forever! <-------
testfax01*CLI> sip show channels
Peer             User/ANR         Call ID          Format           Hold     Last Message    Expiry
127.0.1.1        sipp             1-19236@127.0.1  0x0 (nothing)    No       Tx: INVITE
1 active SIP dialog