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Summary:ASTERISK-17261: No Re-Invite message from Asterisk to the peer when trying for FAX pass-through
Reporter:pkolusu (pkolusu)Labels:
Date Opened:2011-01-18 04:17:16.000-0600Date Closed:2011-06-07 14:04:47
Priority:MinorRegression?No
Status:Closed/CompleteComponents:PBX/General
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:
Description:I have been trying to test fax pass-through with Asterisk.
The scenario is like this.
Two fax machines are connected to the ports that are configured with Asterisk server.

When i am trying to send fax from one port to another port, the receiving port is detecting the fax tone, after initial call set up, and sending the re-invite to change the codec to G711(PCMA), it is receving the response from the Asterisk (Trying and 200 OK), but there is no re-invite message from the Asterisk server to the peer(in this case fax initiator).

i am expecting the re-invite message from the Asterisk to the peer to change the codec to G711(PCMA), but it is not happening. what is the problem i did not understand.

I have tried with canreinvite=yes and no and nat=yes and nerver. but the behaviour is same. no re-invite message from the Asterisk to the peer.


Could you please help me to resolve this issue.

****** ADDITIONAL INFORMATION ******

I am using pjsua-1.7 SIP software for my ATA.
both fax machines are in same network
Comments:By: Matthew Nicholson (mnicholson) 2011-01-18 14:29:12.000-0600

Is the receiving side sending a T.38 reinvite?  Please post a 'sip debug' trace of this problem in action.  I am not sure if there is a bug here, what you have described thus far seems like normal behavior.
By: pkolusu (pkolusu) 2011-01-18 21:31:47.000-0600

I am sorry, i did not understand your question "Is the receiving side sending a T.38 reinvite?". u mean FAX receiving side?. if so, it is sending re-invite message after completion of initial call set up and also it is receiving response (200 OK) from Asterisk.

But the problem is, there is no re-invite message from the asterisk to other side (this is FAX initiator, which is configured to different codec during initial call setup, unless it receives re-invite message from the Asterisk it does not know what codec it has to use for FAX).

And i am not using T.38 for FAX. I am trying FAX-passthrough with G711.

I am explaining signaling message flow of FAX-Passthrough, please correct me if it is not correct.

In FAX-Passthrough,
Initial call set up is normal as any voice call,
After initial call setup completion, the FAX receiving party detects the fax and sends re-invite message with required information(codec, etc) to the other party(FAX initiator). when the other side (FAX initiator) receives the re-invite, it(FAX initiator) configures for the specified codec and sends 200 OK message to the FAX receiver. The FAX receiver configures with negotiated codec after it receives 200 OK from FAX initiator.

In my case, FAX initiator not receiving any re-invite message from Astersik, which i am expecting as per my understanding of FAX-PASSTHROUGH signaling flow.

I have tested same code with Brekeke SIP server, it is working fine, the signaling flow is as i explained above.but with Asterisk, there is not re-invite message from Asterisk to the FAX initiator.


Following are the traces.
--------------------------

Fax Initiating side SIP Traces
================================
=========================================================


<CC-MODULE> [ch 0] off-hook
<SIPUA> [ch 0] sip account reg status - 200 ()
<SIPUA> ****** LED Status sent to Led Controller : 4 2 17 ******
<CC-MODULE> [ch 0] Playing Dail Tone - STATE_OUTGOING
<CC-MODULE> [ch 0] start timer (9 sec)
<CC-MODULE> [ch 0] DTMF/PULSE digit  3
<CC-MODULE> [ch 0] DTMF/PULSE digit  3
<CC-MODULE> [ch 0] start timer (10 sec)
<CC-MODULE> [ch 0] DTMF/PULSE digit  3
<CC-MODULE> [ch 0] Digit timer expired
<CC-MODULE> [ch 0] calling to the number - 333
20:41:22.498   pjsua_call.c  Making call with acc #0 to sip:333@192.168.7.28
20:41:22.498  pjsua_media.c  Media index 0 selected for call 1
20:41:22.503   pjsua_core.c  TX 914 bytes Request msg INVITE/cseq=3872 (tdta0x1016cae0) to UDP 192.168.7.28:5060:
INVITE sip:333@192.168.7.28 SIP/2.0
Via: SIP/2.0/UDP 192.168.8.54:5060;rport;branch=z9hG4bKPjadsT83YtGksroRsGWl.PEMLjV4.8PzyP
Max-Forwards: 70
From: sip:222@192.168.7.28;tag=ReYStR8O5z7kCkGv3JyE1X9fLeK-RYZl
To: sip:333@192.168.7.28
Contact: <sip:222@192.168.8.54:5060>
Call-ID: e.VhXnDqscoRBixjz63.q.usCq22.Wxt
CSeq: 3872 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: PJSUA v1.7/powerpc-unknown-none
Content-Type: application/sdp
Content-Length:   301

v=0
o=- 3159117682 3159117682 IN IP4 192.168.8.54
s=pjmedia
c=IN IP4 192.168.8.54
t=0 0
a=X-nat:0
m=audio 4002 RTP/AVP 18 8 0 101
a=rtcp:4003 IN IP4 192.168.8.54
a=rtpmap:18 G729/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

--end msg--
20:41:22.504    pjsua_app.c  Call 1 state changed to CALLING
<SIPUA> ************ [ch 0] call id : -1   ----- [ch 1] call id : -1 **************
<SIPUA> [ch 0] call_id - 1 **********************
<SIPUA> ************ [ch 0] call id : 1   ----- [ch 1] call id : -1 **************
 20:41:22.511   pjsua_core.c  RX 537 bytes Response msg 401/INVITE/cseq=3872 (rdata0x10165a0c) from UDP 192.168.7.28:5060:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.8.54:5060;rport;branch=z9hG4bKPjadsT83YtGksroRsGWl.PEMLjV4.8PzyP;received=192.168.8.54
From: sip:222@192.168.7.28;tag=ReYStR8O5z7kCkGv3JyE1X9fLeK-RYZl
To: sip:333@192.168.7.28;tag=as53cb249e
Call-ID: e.VhXnDqscoRBixjz63.q.usCq22.Wxt
CSeq: 3872 INVITE
Server: Asterisk PBX 1.6.2.11
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1361dec3"
Content-Length: 0


--end msg--
20:41:22.512   pjsua_core.c  TX 330 bytes Request msg ACK/cseq=3872 (tdta0x101979f0) to UDP 192.168.7.28:5060:
ACK sip:333@192.168.7.28 SIP/2.0
Via: SIP/2.0/UDP 192.168.8.54:5060;rport;branch=z9hG4bKPjadsT83YtGksroRsGWl.PEMLjV4.8PzyP
Max-Forwards: 70
From: sip:222@192.168.7.28;tag=ReYStR8O5z7kCkGv3JyE1X9fLeK-RYZl
To: sip:333@192.168.7.28;tag=as53cb249e
Call-ID: e.VhXnDqscoRBixjz63.q.usCq22.Wxt
CSeq: 3872 ACK
Content-Length:  0


--end msg--
20:41:22.515   pjsua_core.c  TX 1076 bytes Request msg INVITE/cseq=3873 (tdta0x1016cae0) to UDP 192.168.7.28:5060:
INVITE sip:333@192.168.7.28 SIP/2.0
Via: SIP/2.0/UDP 192.168.8.54:5060;rport;branch=z9hG4bKPjyKw3f8j6vpI7BC7sznaJOy0NYL2JrDE4
Max-Forwards: 70
From: sip:222@192.168.7.28;tag=ReYStR8O5z7kCkGv3JyE1X9fLeK-RYZl
To: sip:333@192.168.7.28
Contact: <sip:222@192.168.8.54:5060>
Call-ID: e.VhXnDqscoRBixjz63.q.usCq22.Wxt
CSeq: 3873 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: PJSUA v1.7/powerpc-unknown-none
Authorization: Digest username="222", realm="asterisk", nonce="1361dec3", uri="sip:333@192.168.7.28", response="8c3546e50396098d2c06436dc3d8e9a0", algorith
m=MD5
Content-Type: application/sdp
Content-Length:   301

v=0
o=- 3159117682 3159117682 IN IP4 192.168.8.54
s=pjmedia
c=IN IP4 192.168.8.54
t=0 0
a=X-nat:0
m=audio 4002 RTP/AVP 18 8 0 101
a=rtcp:4003 IN IP4 192.168.8.54
a=rtpmap:18 G729/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

--end msg--
20:41:22.516    pjsua_app.c  Call 1 state changed to CALLING
20:41:22.524   pjsua_core.c  RX 526 bytes Response msg 100/INVITE/cseq=3873 (rdata0x10165a0c) from UDP 192.168.7.28:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.8.54:5060;rport;branch=z9hG4bKPjyKw3f8j6vpI7BC7sznaJOy0NYL2JrDE4;received=192.168.8.54
From: sip:222@192.168.7.28;tag=ReYStR8O5z7kCkGv3JyE1X9fLeK-RYZl
To: sip:333@192.168.7.28
Call-ID: e.VhXnDqscoRBixjz63.q.usCq22.Wxt
CSeq: 3873 INVITE
Server: Asterisk PBX 1.6.2.11
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: 1800;refresher=uas
Contact: <sip:333@192.168.7.28>
Content-Length: 0


--end msg--
20:41:23.192   pjsua_core.c  RX 542 bytes Response msg 180/INVITE/cseq=3873 (rdata0x10165a0c) from UDP 192.168.7.28:5060:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.8.54:5060;rport;branch=z9hG4bKPjyKw3f8j6vpI7BC7sznaJOy0NYL2JrDE4;received=192.168.8.54
From: sip:222@192.168.7.28;tag=ReYStR8O5z7kCkGv3JyE1X9fLeK-RYZl
To: sip:333@192.168.7.28;tag=as6d3023d4
Call-ID: e.VhXnDqscoRBixjz63.q.usCq22.Wxt
CSeq: 3873 INVITE
Server: Asterisk PBX 1.6.2.11
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: 1800;refresher=uas
Contact: <sip:333@192.168.7.28>
Content-Length: 0


--end msg--
<SIPUA> ringback_start function is called ......
<SIPUA> ************ [ch 0] call id : 1   ----- [ch 1] call id : -1 **************
<CC-MODULE> [ch 0] START RINGBACK TONE
20:41:23.195    pjsua_app.c  Call 1 state changed to EARLY (180 Ringing)
20:41:23.196   pjsua_core.c  RX 542 bytes Response msg 180/INVITE/cseq=3873 (rdata0x10165a0c) from UDP 192.168.7.28:5060:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.8.54:5060;rport;branch=z9hG4bKPjyKw3f8j6vpI7BC7sznaJOy0NYL2JrDE4;received=192.168.8.54
From: sip:222@192.168.7.28;tag=ReYStR8O5z7kCkGv3JyE1X9fLeK-RYZl
To: sip:333@192.168.7.28;tag=as6d3023d4
Call-ID: e.VhXnDqscoRBixjz63.q.usCq22.Wxt
CSeq: 3873 INVITE
Server: Asterisk PBX 1.6.2.11
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: 1800;refresher=uas
Contact: <sip:333@192.168.7.28>
Content-Length: 0


--end msg--
20:41:23.197    pjsua_app.c  Call 1 state changed to EARLY (180 Ringing)
20:41:25.603   pjsua_core.c  RX 878 bytes Response msg 200/INVITE/cseq=3873 (rdata0x10165a0c) from UDP 192.168.7.28:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.8.54:5060;rport;branch=z9hG4bKPjyKw3f8j6vpI7BC7sznaJOy0NYL2JrDE4;received=192.168.8.54
From: sip:222@192.168.7.28;tag=ReYStR8O5z7kCkGv3JyE1X9fLeK-RYZl
To: sip:333@192.168.7.28;tag=as6d3023d4
Call-ID: e.VhXnDqscoRBixjz63.q.usCq22.Wxt
CSeq: 3873 INVITE
Server: Asterisk PBX 1.6.2.11
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: 1800;refresher=uas
Contact: <sip:333@192.168.7.28>
Content-Type: application/sdp
Content-Length: 308

v=0
o=root 1052619648 1052619648 IN IP4 192.168.7.28
s=Asterisk PBX 1.6.2.11
c=IN IP4 192.168.7.28
t=0 0
m=audio 14064 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

--end msg--
20:41:25.604    pjsua_app.c  Call 1 state changed to CONNECTING
20:41:25.606 strm0x1019c3dc  VAD temporarily disabled
20:41:25.607 strm0x1019c3dc  Encoder stream started
20:41:25.607 strm0x1019c3dc  Decoder stream started
<SIPUA> ************ [ch 0] call id : 1   ----- [ch 1] call id : -1 **************
 20:41:25.608  pjsua_media.c  Media updates, stream #0: PCMU (sendrecv)
********* before ring_stop *********************
************* Stopping Ring Back Tone *********************
<SIPUA> ************ [ch 0] call id : 1   ----- [ch 1] call id : -1 **************
<CC-MODULE> [ch 0] Stopping Dial tone
<CC-MODULE> [ch 0] Stopping ringback tone
<CC-MODULE> [ch 0] STOP TONE
********* In connect_sound check *********************
<SIPUA> ************ [ch 0] call id : 1   ----- [ch 1] call id : -1 **************
 20:41:25.610  pjsua_media.c  Opening vinetic ch0 ..
************ port_map[0] : 2 ******************
20:41:25.611   conference.c  Port 2 (sip:333@192.168.7.28) transmitting to port 0 (channel0)
20:41:25.611   conference.c  Port 0 (channel0) transmitting to port 2 (sip:333@192.168.7.28)
codec_info, codec_name: PCMU     ptime: 20----------------------+++++
selected codec PCMU
<CC-MODULE> IFX_PKT_RTP_CFG_SET , [SSRC - 0x38a55aa2] [Seq No : 0x5db0] ret: 0
<CC-MODULE> received ptime from the SDP message - 20
<CC-MODULE> CODEC ULAW (PCMU) configured
<SIPUA> ************ [ch 0] call id : 1   ----- [ch 1] call id : -1 **************
<SIPUA> ************* cmdid : 2 , ch:0, rtp_port_min: 4000 rtp_port_max: 4010 remote_rtp_port:14064
<SIPUA> [ch 0] enabling start codec
<CC-MODULE> [ch 0] Codec encoding Started
<CC-MODULE> [ch 0] Codec decoding Started
********* After calling conf_connect for both directions *********************
20:41:25.621    pjsua_app.c  Media for call 1 is active
20:41:25.623   pjsua_core.c  TX 330 bytes Request msg ACK/cseq=3873 (tdta0x101a11b0) to UDP 192.168.7.28:5060:
ACK sip:333@192.168.7.28 SIP/2.0
Via: SIP/2.0/UDP 192.168.8.54:5060;rport;branch=z9hG4bKPjPQ7.gAZVHYnmzcPIACz-3xKbwsKkKGh4
Max-Forwards: 70
From: sip:222@192.168.7.28;tag=ReYStR8O5z7kCkGv3JyE1X9fLeK-RYZl
To: sip:333@192.168.7.28;tag=as6d3023d4
Call-ID: e.VhXnDqscoRBixjz63.q.usCq22.Wxt
CSeq: 3873 ACK
Content-Length:  0


--end msg--
20:41:25.623   pjsua_call.c  Got answer with multiple codecs, start updating media session to use only one codec..
20:41:25.624    pjsua_app.c  <SIPUA> Call 1 state is CONFIRMED
20:41:25.624    pjsua_app.c  Call 1 state changed to CONFIRMED
20:41:25.826   pjsua_core.c  TX 848 bytes Request msg INVITE/cseq=3874 (tdta0x101a4b20) to UDP 192.168.7.28:5060:
INVITE sip:333@192.168.7.28 SIP/2.0
Via: SIP/2.0/UDP 192.168.8.54:5060;rport;branch=z9hG4bKPjWlyE3-OBjzmLkIYx-KqQ8BVoaD3Y9l4Z
Max-Forwards: 70
From: sip:222@192.168.7.28;tag=ReYStR8O5z7kCkGv3JyE1X9fLeK-RYZl
To: sip:333@192.168.7.28;tag=as6d3023d4
Contact: <sip:222@192.168.8.54:5060>
Call-ID: e.VhXnDqscoRBixjz63.q.usCq22.Wxt
CSeq: 3874 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800;refresher=uas
Min-SE: 90
Content-Type: application/sdp
Content-Length:   251

v=0
o=- 3159117682 3159117683 IN IP4 192.168.8.54
s=pjmedia
c=IN IP4 192.168.8.54
t=0 0
a=X-nat:0
m=audio 4002 RTP/AVP 0 101
a=rtcp:4003 IN IP4 192.168.8.54
a=rtpmap:0 PCMU/8000
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

--end msg--
20:41:25.833   pjsua_core.c  RX 541 bytes Response msg 100/INVITE/cseq=3874 (rdata0x10165a0c) from UDP 192.168.7.28:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.8.54:5060;rport;branch=z9hG4bKPjWlyE3-OBjzmLkIYx-KqQ8BVoaD3Y9l4Z;received=192.168.8.54
From: sip:222@192.168.7.28;tag=ReYStR8O5z7kCkGv3JyE1X9fLeK-RYZl
To: sip:333@192.168.7.28;tag=as6d3023d4
Call-ID: e.VhXnDqscoRBixjz63.q.usCq22.Wxt
CSeq: 3874 INVITE
Server: Asterisk PBX 1.6.2.11
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: 1800;refresher=uas
Contact: <sip:333@192.168.7.28>
Content-Length: 0


--end msg--
20:41:25.835   pjsua_core.c  RX 807 bytes Response msg 200/INVITE/cseq=3874 (rdata0x10165a0c) from UDP 192.168.7.28:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.8.54:5060;rport;branch=z9hG4bKPjWlyE3-OBjzmLkIYx-KqQ8BVoaD3Y9l4Z;received=192.168.8.54
From: sip:222@192.168.7.28;tag=ReYStR8O5z7kCkGv3JyE1X9fLeK-RYZl
To: sip:333@192.168.7.28;tag=as6d3023d4
Call-ID: e.VhXnDqscoRBixjz63.q.usCq22.Wxt
CSeq: 3874 INVITE
Server: Asterisk PBX 1.6.2.11
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: 1800;refresher=uas
Contact: <sip:333@192.168.7.28>
Content-Type: application/sdp
Content-Length: 237

v=0
o=root 1052619648 1052619649 IN IP4 192.168.7.28
s=Asterisk PBX 1.6.2.11
c=IN IP4 192.168.7.28
t=0 0
m=audio 14064 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

--end msg--
<SIPUA> ************ [ch 0] call id : 1   ----- [ch 1] call id : -1 **************
<SIPUA> closing ch0
20:41:26.504  pjsua_media.c  Closing null sound device 0..
<SIPUA> [ch 0] Calling Stop Codecs
<CC-MODULE> [ch 0] Codec decoding Stopped
<CC-MODULE> [ch 0] Codec encoding Stopped
<SIPUA> ************** In On_Stream_destroyed - 1 ****************
20:41:26.522  pjsua_media.c  Media session for call 1 is destroyed
20:41:26.523 strm0x1019c3dc  VAD temporarily disabled
20:41:26.524 strm0x1019c3dc  Encoder stream started
20:41:26.524 strm0x1019c3dc  Decoder stream started
<SIPUA> ************ [ch 0] call id : 1   ----- [ch 1] call id : -1 **************
 20:41:26.524  pjsua_media.c  Media updates, stream #0: PCMU (sendrecv)
********* before ring_stop *********************
********* In connect_sound check *********************
<SIPUA> ************ [ch 0] call id : 1   ----- [ch 1] call id : -1 **************
 20:41:26.525  pjsua_media.c  Opening vinetic ch0 ..
************ port_map[0] : 2 ******************
20:41:26.532   conference.c  Port 2 (sip:333@192.168.7.28) transmitting to port 0 (channel0)
20:41:26.532   conference.c  Port 0 (channel0) transmitting to port 2 (sip:333@192.168.7.28)
codec_info, codec_name: PCMU     ptime: 20----------------------+++++
selected codec PCMU
<CC-MODULE> IFX_PKT_RTP_CFG_SET , [SSRC - 0x6175452b] [Seq No : 0x339] ret: 0
<CC-MODULE> received ptime from the SDP message - 20
<CC-MODULE> CODEC ULAW (PCMU) configured
<SIPUA> ************ [ch 0] call id : 1   ----- [ch 1] call id : -1 **************
<SIPUA> ************* cmdid : 2 , ch:0, rtp_port_min: 4000 rtp_port_max: 4010 remote_rtp_port:14064
<SIPUA> [ch 0] enabling start codec
<CC-MODULE> [ch 0] Codec encoding Started
<CC-MODULE> [ch 0] Codec decoding Started
********* After calling conf_connect for both directions *********************
20:41:26.540    pjsua_app.c  Media for call 1 is active
20:41:26.542   pjsua_core.c  TX 330 bytes Request msg ACK/cseq=3874 (tdta0x1019ec78) to UDP 192.168.7.28:5060:
ACK sip:333@192.168.7.28 SIP/2.0
Via: SIP/2.0/UDP 192.168.8.54:5060;rport;branch=z9hG4bKPjEK8lVegRWfTKgGprQ0wAgr9ncxm.S-JH
Max-Forwards: 70
From: sip:222@192.168.7.28;tag=ReYStR8O5z7kCkGv3JyE1X9fLeK-RYZl
To: sip:333@192.168.7.28;tag=as6d3023d4
Call-ID: e.VhXnDqscoRBixjz63.q.usCq22.Wxt
CSeq: 3874 ACK
Content-Length:  0


--end msg--
20:41:26.543   pjsua_core.c  RX 807 bytes Response msg 200/INVITE/cseq=3874 (rdata0x10165a0c) from UDP 192.168.7.28:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.8.54:5060;rport;branch=z9hG4bKPjWlyE3-OBjzmLkIYx-KqQ8BVoaD3Y9l4Z;received=192.168.8.54
From: sip:222@192.168.7.28;tag=ReYStR8O5z7kCkGv3JyE1X9fLeK-RYZl
To: sip:333@192.168.7.28;tag=as6d3023d4
Call-ID: e.VhXnDqscoRBixjz63.q.usCq22.Wxt
CSeq: 3874 INVITE
Server: Asterisk PBX 1.6.2.11
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: 1800;refresher=uas
Contact: <sip:333@192.168.7.28>
Content-Type: application/sdp
Content-Length: 237

v=0
o=root 1052619648 1052619649 IN IP4 192.168.7.28
s=Asterisk PBX 1.6.2.11
c=IN IP4 192.168.7.28
t=0 0
m=audio 14064 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

--end msg--
20:41:27.210   pjsua_core.c  TX 330 bytes Request msg ACK/cseq=3874 (tdta0x1019ec78) to UDP 192.168.7.28:5060:
ACK sip:333@192.168.7.28 SIP/2.0
Via: SIP/2.0/UDP 192.168.8.54:5060;rport;branch=z9hG4bKPjEK8lVegRWfTKgGprQ0wAgr9ncxm.S-JH
Max-Forwards: 70
From: sip:222@192.168.7.28;tag=ReYStR8O5z7kCkGv3JyE1X9fLeK-RYZl
To: sip:333@192.168.7.28;tag=as6d3023d4
Call-ID: e.VhXnDqscoRBixjz63.q.usCq22.Wxt
CSeq: 3874 ACK
Content-Length:  0


--end msg--
20:41:27.211   pjsua_core.c  RX 807 bytes Response msg 200/INVITE/cseq=3874 (rdata0x10165a0c) from UDP 192.168.7.28:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.8.54:5060;rport;branch=z9hG4bKPjWlyE3-OBjzmLkIYx-KqQ8BVoaD3Y9l4Z;received=192.168.8.54
From: sip:222@192.168.7.28;tag=ReYStR8O5z7kCkGv3JyE1X9fLeK-RYZl
To: sip:333@192.168.7.28;tag=as6d3023d4
Call-ID: e.VhXnDqscoRBixjz63.q.usCq22.Wxt
CSeq: 3874 INVITE
Server: Asterisk PBX 1.6.2.11
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: 1800;refresher=uas
Contact: <sip:333@192.168.7.28>
Content-Type: application/sdp
Content-Length: 237

v=0
o=root 1052619648 1052619649 IN IP4 192.168.7.28
s=Asterisk PBX 1.6.2.11
c=IN IP4 192.168.7.28
t=0 0
m=audio 14064 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

--end msg--
20:41:27.215   pjsua_core.c  TX 330 bytes Request msg ACK/cseq=3874 (tdta0x1019ec78) to UDP 192.168.7.28:5060:
ACK sip:333@192.168.7.28 SIP/2.0
Via: SIP/2.0/UDP 192.168.8.54:5060;rport;branch=z9hG4bKPjEK8lVegRWfTKgGprQ0wAgr9ncxm.S-JH
Max-Forwards: 70
From: sip:222@192.168.7.28;tag=ReYStR8O5z7kCkGv3JyE1X9fLeK-RYZl
To: sip:333@192.168.7.28;tag=as6d3023d4
Call-ID: e.VhXnDqscoRBixjz63.q.usCq22.Wxt
CSeq: 3874 ACK
Content-Length:  0


--end msg--
<rtcp_build_rtcp> Called IFX_TAPI_PKT_RTCP_STATISTICS_GET, ch : 0, ret : 0 , ssrc : 0x6175452b , rtp_ts : 0x89f2, psent : 0xa4, osend : 0x5888, rssrc : 193
5680436, frat : 0,lost : 0x0, last seq : 0x1879, jitter : 0x0

20:41:31.611   pjsua_core.c  RX 519 bytes Request msg OPTIONS/cseq=102 (rdata0x10165a0c) from UDP 192.168.7.28:5060:
OPTIONS sip:222@192.168.8.54:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.7.28:5060;branch=z9hG4bK6fae7962
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.7.28>;tag=as7b0643d1
To: <sip:222@192.168.8.54:5060>
Contact: <sip:Unknown@192.168.7.28>
Call-ID: 5c7cf61a159731820d1262a630b130b2@192.168.7.28
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.11
Date: Wed, 19 Jan 2011 03:00:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


--end msg--
20:41:31.616   pjsua_core.c  TX 1037 bytes Response msg 200/OPTIONS/cseq=102 (tdta0x101a3018) to UDP 192.168.7.28:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.7.28:5060;received=192.168.7.28;branch=z9hG4bK6fae7962
Call-ID: 5c7cf61a159731820d1262a630b130b2@192.168.7.28
From: "Unknown" <sip:Unknown@192.168.7.28>;tag=as7b0643d1
To: <sip:222@192.168.8.54>;tag=z9hG4bK6fae7962
CSeq: 102 OPTIONS
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Accept: application/sdp, application/pidf+xml, application/xpidf+xml, application/simple-message-summary, message/sipfrag;version=2.0, application/im-iscom
posing+xml, text/plain
Supported: replaces, 100rel, timer, norefersub
Allow-Events: presence, message-summary, refer
User-Agent: PJSUA v1.7/powerpc-unknown-none
Content-Type: application/sdp
Content-Length:   290

v=0
o=- 3159117691 3159117691 IN IP4 192.168.8.54
s=pjmedia
c=IN IP4 192.168.8.54
t=0 0
m=audio 4000 RTP/AVP 18 8 0 101
a=rtcp:4001 IN IP4 192.168.8.54
a=rtpmap:18 G729/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

--end msg--
20:41:33.202   pjsua_core.c  RX 519 bytes Request msg OPTIONS/cseq=102 (rdata0x10165a0c) from UDP 192.168.7.28:5060:
OPTIONS sip:111@192.168.8.54:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.7.28:5060;branch=z9hG4bK778c64db
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.7.28>;tag=as6d5c21f3
To: <sip:111@192.168.8.54:5060>
Contact: <sip:Unknown@192.168.7.28>
Call-ID: 5cb9aab344674b331bbb70293886efc0@192.168.7.28
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.11
Date: Wed, 19 Jan 2011 03:00:06 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


--end msg--
20:41:33.204   pjsua_core.c  TX 1037 bytes Response msg 200/OPTIONS/cseq=102 (tdta0x101a3018) to UDP 192.168.7.28:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.7.28:5060;received=192.168.7.28;branch=z9hG4bK778c64db
Call-ID: 5cb9aab344674b331bbb70293886efc0@192.168.7.28
From: "Unknown" <sip:Unknown@192.168.7.28>;tag=as6d5c21f3
To: <sip:111@192.168.8.54>;tag=z9hG4bK778c64db
CSeq: 102 OPTIONS
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Accept: application/sdp, application/pidf+xml, application/xpidf+xml, application/simple-message-summary, message/sipfrag;version=2.0, application/im-iscom
posing+xml, text/plain
Supported: replaces, 100rel, timer, norefersub
Allow-Events: presence, message-summary, refer
User-Agent: PJSUA v1.7/powerpc-unknown-none
Content-Type: application/sdp
Content-Length:   290

v=0
o=- 3159117693 3159117693 IN IP4 192.168.8.54
s=pjmedia
c=IN IP4 192.168.8.54
t=0 0
m=audio 4000 RTP/AVP 18 8 0 101
a=rtcp:4001 IN IP4 192.168.8.54
a=rtpmap:18 G729/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

--end msg--
<rtcp_build_rtcp> Called IFX_TAPI_PKT_RTCP_STATISTICS_GET, ch : 0, ret : 0 , ssrc : 0x6175452b , rtp_ts : 0x13b69, psent : 0x12f, osend : 0x99df, rssrc : 1
285703938, frat : 0,lost : 0x0, last seq : 0x18fd, jitter : 0x4

<rtcp_build_rtcp> Called IFX_TAPI_PKT_RTCP_STATISTICS_GET, ch : 0, ret : 0 , ssrc : 0x6175452b , rtp_ts : 0x1ce56, psent : 0x215, osend : 0x1290a, rssrc :
1285703938, frat : 0,lost : 0x0, last seq : 0x19df, jitter : 0x9

<rtcp_build_rtcp> Called IFX_TAPI_PKT_RTCP_STATISTICS_GET, ch : 0, ret : 0 , ssrc : 0x6175452b , rtp_ts : 0x26129, psent : 0x2fb, osend : 0x1b7a0, rssrc :
1285703938, frat : 0,lost : 0x0, last seq : 0x1a6b, jitter : 0x3

<rtcp_build_rtcp> Called IFX_TAPI_PKT_RTCP_STATISTICS_GET, ch : 0, ret : 0 , ssrc : 0x6175452b , rtp_ts : 0x2f545, psent : 0x3e4, osend : 0x24816, rssrc :
1285703938, frat : 0,lost : 0x0, last seq : 0x1b0e, jitter : 0x2

<rtcp_build_rtcp> Called IFX_TAPI_PKT_RTCP_STATISTICS_GET, ch : 0, ret : 0 , ssrc : 0x6175452b , rtp_ts : 0x38829, psent : 0x4cf, osend : 0x2daf6, rssrc :
1285703938, frat : 0,lost : 0x0, last seq : 0x1b77, jitter : 0x4

<rtcp_build_rtcp> Called IFX_TAPI_PKT_RTCP_STATISTICS_GET, ch : 0, ret : 0 , ssrc : 0x6175452b , rtp_ts : 0x4200b, psent : 0x5be, osend : 0x36fc1, rssrc :
1285703938, frat : 0,lost : 0x0, last seq : 0x1bd6, jitter : 0x2

<rtcp_build_rtcp> Called IFX_TAPI_PKT_RTCP_STATISTICS_GET, ch : 0, ret : 0 , ssrc : 0x6175452b , rtp_ts : 0x4b2ea, psent : 0x6a1, osend : 0x3fb4d, rssrc :
1285703938, frat : 0,lost : 0x0, last seq : 0x1c60, jitter : 0x1

<rtcp_build_rtcp> Called IFX_TAPI_PKT_RTCP_STATISTICS_GET, ch : 0, ret : 0 , ssrc : 0x6175452b , rtp_ts : 0x547a9, psent : 0x786, osend : 0x488ae, rssrc :
1285703938, frat : 0,lost : 0x0, last seq : 0x1ceb, jitter : 0x2

<rtcp_build_rtcp> Called IFX_TAPI_PKT_RTCP_STATISTICS_GET, ch : 0, ret : 0 , ssrc : 0x6175452b , rtp_ts : 0x5da96, psent : 0x86b, osend : 0x516a4, rssrc :
1285703938, frat : 0,lost : 0x0, last seq : 0x1d78, jitter : 0xa

<rtcp_build_rtcp> Called IFX_TAPI_PKT_RTCP_STATISTICS_GET, ch : 0, ret : 0 , ssrc : 0x6175452b , rtp_ts : 0x66d6e, psent : 0x8eb, osend : 0x55658, rssrc :
1285703938, frat : 0,lost : 0x0, last seq : 0x1d98, jitter : 0x3

<rtcp_build_rtcp> Called IFX_TAPI_PKT_RTCP_STATISTICS_GET, ch : 0, ret : 0 , ssrc : 0x6175452b , rtp_ts : 0x72b65, psent : 0x9b0, osend : 0x5c1c1, rssrc :
1285703938, frat : 0,lost : 0x0, last seq : 0x1dfa, jitter : 0x4

20:42:26.577   pjsua_core.c  TX 515 bytes Request msg REGISTER/cseq=37166 (tdta0x101a3018) to UDP 192.168.7.28:5060:
REGISTER sip:192.168.7.28 SIP/2.0
Via: SIP/2.0/UDP 192.168.8.54:5060;rport;branch=z9hG4bKPjQT8d9ivMUoeQJwV8VCL.UvIBip96UHRZ
Max-Forwards: 70
From: <sip:222@192.168.7.28>;tag=upr6rEkxRk74YwmRWnmvw-A2UUNrsNw7
To: <sip:222@192.168.7.28>
Call-ID: Z03XMZX5B.cYwuoimasH8h3GYlw9lV7b
CSeq: 37166 REGISTER
User-Agent: PJSUA v1.7/powerpc-unknown-none
Contact: <sip:222@192.168.8.54:5060>
Expires: 300
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Content-Length:  0


--end msg--
20:42:26.584   pjsua_core.c  RX 544 bytes Response msg 401/REGISTER/cseq=37166 (rdata0x10165a0c) from UDP 192.168.7.28:5060:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.8.54:5060;rport;branch=z9hG4bKPjQT8d9ivMUoeQJwV8VCL.UvIBip96UHRZ;received=192.168.8.54
From: <sip:222@192.168.7.28>;tag=upr6rEkxRk74YwmRWnmvw-A2UUNrsNw7
To: <sip:222@192.168.7.28>;tag=as61f171b7
Call-ID: Z03XMZX5B.cYwuoimasH8h3GYlw9lV7b
CSeq: 37166 REGISTER
Server: Asterisk PBX 1.6.2.11
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2683cdce"
Content-Length: 0


--end msg--
20:42:26.591   pjsua_core.c  TX 673 bytes Request msg REGISTER/cseq=37167 (tdta0x101a3018) to UDP 192.168.7.28:5060:
REGISTER sip:192.168.7.28 SIP/2.0
Via: SIP/2.0/UDP 192.168.8.54:5060;rport;branch=z9hG4bKPjhd4jTT7A4TECs9Z8Qfq9Zv0LHCuP.UuP
Max-Forwards: 70
From: <sip:222@192.168.7.28>;tag=upr6rEkxRk74YwmRWnmvw-A2UUNrsNw7
To: <sip:222@192.168.7.28>
Call-ID: Z03XMZX5B.cYwuoimasH8h3GYlw9lV7b
CSeq: 37167 REGISTER
User-Agent: PJSUA v1.7/powerpc-unknown-none
Contact: <sip:222@192.168.8.54:5060>
Expires: 300
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Authorization: Digest username="222", realm="asterisk", nonce="2683cdce", uri="sip:192.168.7.28", response="6ff4eee9ecc45b9b6cac977a1a2f1d4d", algorithm=MD
5
Content-Length:  0


--end msg--
20:42:26.602   pjsua_core.c  RX 519 bytes Request msg OPTIONS/cseq=102 (rdata0x10165a0c) from UDP 192.168.7.28:5060:
OPTIONS sip:222@192.168.8.54:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.7.28:5060;branch=z9hG4bK2241f0b2
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.7.28>;tag=as4780fa89
To: <sip:222@192.168.8.54:5060>
Contact: <sip:Unknown@192.168.7.28>
Call-ID: 052829ab77f6ee6805ed2caa1496c6bf@192.168.7.28
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.11
Date: Wed, 19 Jan 2011 03:01:00 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


--end msg--
20:42:26.604   pjsua_core.c  TX 1037 bytes Response msg 200/OPTIONS/cseq=102 (tdta0x101a4b20) to UDP 192.168.7.28:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.7.28:5060;received=192.168.7.28;branch=z9hG4bK2241f0b2
Call-ID: 052829ab77f6ee6805ed2caa1496c6bf@192.168.7.28
From: "Unknown" <sip:Unknown@192.168.7.28>;tag=as4780fa89
To: <sip:222@192.168.8.54>;tag=z9hG4bK2241f0b2
CSeq: 102 OPTIONS
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Accept: application/sdp, application/pidf+xml, application/xpidf+xml, application/simple-message-summary, message/sipfrag;version=2.0, application/im-iscom
posing+xml, text/plain
Supported: replaces, 100rel, timer, norefersub
Allow-Events: presence, message-summary, refer
User-Agent: PJSUA v1.7/powerpc-unknown-none
Content-Type: application/sdp
Content-Length:   290

v=0
o=- 3159117746 3159117746 IN IP4 192.168.8.54
s=pjmedia
c=IN IP4 192.168.8.54
t=0 0
m=audio 4000 RTP/AVP 18 8 0 101
a=rtcp:4001 IN IP4 192.168.8.54
a=rtpmap:18 G729/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

--end msg--
20:42:26.608   pjsua_core.c  RX 559 bytes Response msg 200/REGISTER/cseq=37167 (rdata0x10165a0c) from UDP 192.168.7.28:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.8.54:5060;rport;branch=z9hG4bKPjhd4jTT7A4TECs9Z8Qfq9Zv0LHCuP.UuP;received=192.168.8.54
From: <sip:222@192.168.7.28>;tag=upr6rEkxRk74YwmRWnmvw-A2UUNrsNw7
To: <sip:222@192.168.7.28>;tag=as61f171b7
Call-ID: Z03XMZX5B.cYwuoimasH8h3GYlw9lV7b
CSeq: 37167 REGISTER
Server: Asterisk PBX 1.6.2.11
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Expires: 300
Contact: <sip:222@192.168.8.54:5060>;expires=300
Date: Wed, 19 Jan 2011 03:01:00 GMT
Content-Length: 0


--end msg--
20:42:27.276    pjsua_acc.c  sip:222@192.168.7.28: registration success, status=200 (OK), will re-register in 300 seconds
20:42:27.277    pjsua_acc.c  Keep-alive timer started for acc 0, destination:192.168.7.28:5060, interval:15s
<ewl_cc_sip> <send_led> [0] received registered & port is in use
<SIPUA> ****** LED Status sent to Led Controller : 4 2 17 ******
<SIPUA> [acc_id 0] has_registration - 1 , info.status - 200
20:42:28.166   pjsua_core.c  TX 515 bytes Request msg REGISTER/cseq=46971 (tdta0x101a4b20) to UDP 192.168.7.28:5060:
REGISTER sip:192.168.7.28 SIP/2.0
Via: SIP/2.0/UDP 192.168.8.54:5060;rport;branch=z9hG4bKPjdGjcYnuxftzL7Y-2TMZy8nmBVLVajF6a
Max-Forwards: 70
From: <sip:111@192.168.7.28>;tag=U6GcdnM9FAIgk7BiOdRRM3-gB5qUHXiv
To: <sip:111@192.168.7.28>
Call-ID: AFODxsFwtFPoEoySgREm0yj1dO-pXqBO
CSeq: 46971 REGISTER
User-Agent: PJSUA v1.7/powerpc-unknown-none
Contact: <sip:111@192.168.8.54:5060>
Expires: 300
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Content-Length:  0


--end msg--
20:42:28.173   pjsua_core.c  RX 544 bytes Response msg 401/REGISTER/cseq=46971 (rdata0x10165a0c) from UDP 192.168.7.28:5060:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.8.54:5060;rport;branch=z9hG4bKPjdGjcYnuxftzL7Y-2TMZy8nmBVLVajF6a;received=192.168.8.54
From: <sip:111@192.168.7.28>;tag=U6GcdnM9FAIgk7BiOdRRM3-gB5qUHXiv
To: <sip:111@192.168.7.28>;tag=as0c6822e7
Call-ID: AFODxsFwtFPoEoySgREm0yj1dO-pXqBO
CSeq: 46971 REGISTER
Server: Asterisk PBX 1.6.2.11
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="759841fa"
Content-Length: 0


--end msg--
20:42:28.175   pjsua_core.c  TX 673 bytes Request msg REGISTER/cseq=46972 (tdta0x101a4b20) to UDP 192.168.7.28:5060:
REGISTER sip:192.168.7.28 SIP/2.0
Via: SIP/2.0/UDP 192.168.8.54:5060;rport;branch=z9hG4bKPjD-GTp78wXbYE6BxtIzM4Ev4HXOosvbpl
Max-Forwards: 70
From: <sip:111@192.168.7.28>;tag=U6GcdnM9FAIgk7BiOdRRM3-gB5qUHXiv
To: <sip:111@192.168.7.28>
Call-ID: AFODxsFwtFPoEoySgREm0yj1dO-pXqBO
CSeq: 46972 REGISTER
User-Agent: PJSUA v1.7/powerpc-unknown-none
Contact: <sip:111@192.168.8.54:5060>
Expires: 300
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Authorization: Digest username="111", realm="asterisk", nonce="759841fa", uri="sip:192.168.7.28", response="fcf029b2bffe9e9332cd937545a6c483", algorithm=MD
5
Content-Length:  0


--end msg--
20:42:28.186   pjsua_core.c  RX 519 bytes Request msg OPTIONS/cseq=102 (rdata0x10165a0c) from UDP 192.168.7.28:5060:
OPTIONS sip:111@192.168.8.54:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.7.28:5060;branch=z9hG4bK38ec8cd6
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.7.28>;tag=as266e1de6
To: <sip:111@192.168.8.54:5060>
Contact: <sip:Unknown@192.168.7.28>
Call-ID: 3dd4aae942b7099d1baf82717fdfa743@192.168.7.28
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.11
Date: Wed, 19 Jan 2011 03:01:01 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


--end msg--
20:42:28.191   pjsua_core.c  TX 1037 bytes Response msg 200/OPTIONS/cseq=102 (tdta0x101aaf48) to UDP 192.168.7.28:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.7.28:5060;received=192.168.7.28;branch=z9hG4bK38ec8cd6
Call-ID: 3dd4aae942b7099d1baf82717fdfa743@192.168.7.28
From: "Unknown" <sip:Unknown@192.168.7.28>;tag=as266e1de6
To: <sip:111@192.168.8.54>;tag=z9hG4bK38ec8cd6
CSeq: 102 OPTIONS
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Accept: application/sdp, application/pidf+xml, application/xpidf+xml, application/simple-message-summary, message/sipfrag;version=2.0, application/im-iscom
posing+xml, text/plain
Supported: replaces, 100rel, timer, norefersub
Allow-Events: presence, message-summary, refer
User-Agent: PJSUA v1.7/powerpc-unknown-none
Content-Type: application/sdp
Content-Length:   290

v=0
o=- 3159117748 3159117748 IN IP4 192.168.8.54
s=pjmedia
c=IN IP4 192.168.8.54
t=0 0
m=audio 4000 RTP/AVP 18 8 0 101
a=rtcp:4001 IN IP4 192.168.8.54
a=rtpmap:18 G729/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

--end msg--
20:42:28.192   pjsua_core.c  RX 559 bytes Response msg 200/REGISTER/cseq=46972 (rdata0x10165a0c) from UDP 192.168.7.28:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.8.54:5060;rport;branch=z9hG4bKPjD-GTp78wXbYE6BxtIzM4Ev4HXOosvbpl;received=192.168.8.54
From: <sip:111@192.168.7.28>;tag=U6GcdnM9FAIgk7BiOdRRM3-gB5qUHXiv
To: <sip:111@192.168.7.28>;tag=as0c6822e7
Call-ID: AFODxsFwtFPoEoySgREm0yj1dO-pXqBO
CSeq: 46972 REGISTER
Server: Asterisk PBX 1.6.2.11
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Expires: 300
Contact: <sip:111@192.168.8.54:5060>;expires=300
Date: Wed, 19 Jan 2011 03:01:01 GMT
Content-Length: 0


--end msg--
20:42:28.860    pjsua_acc.c  sip:111@192.168.7.28: registration success, status=200 (OK), will re-register in 300 seconds
20:42:28.864    pjsua_acc.c  Keep-alive timer started for acc 1, destination:192.168.7.28:5060, interval:15s
<SIPUA> ****** LED Status sent to Led Controller : 4 1 16 ******
<SIPUA> [acc_id 1] has_registration - 1 , info.status - 200
<rtcp_build_rtcp> Called IFX_TAPI_PKT_RTCP_STATISTICS_GET, ch : 0, ret : 0 , ssrc : 0x6175452b , rtp_ts : 0x7c665, psent : 0xaa1, osend : 0x6560d, rssrc :
1285703938, frat : 0,lost : 0x0, last seq : 0x1e88, jitter : 0x6

<rtcp_build_rtcp> Called IFX_TAPI_PKT_RTCP_STATISTICS_GET, ch : 0, ret : 0 , ssrc : 0x6175452b , rtp_ts : 0x862a6, psent : 0xb34, osend : 0x6a2cb, rssrc :
1285703938, frat : 0,lost : 0x0, last seq : 0x1eff, jitter : 0x3

<rtcp_build_rtcp> Called IFX_TAPI_PKT_RTCP_STATISTICS_GET, ch : 0, ret : 0 , ssrc : 0x6175452b , rtp_ts : 0x90fca, psent : 0xbd0, osend : 0x6f240, rssrc :
1285703938, frat : 0,lost : 0x0, last seq : 0x1f07, jitter : 0x3

<rtcp_build_rtcp> Called IFX_TAPI_PKT_RTCP_STATISTICS_GET, ch : 0, ret : 0 , ssrc : 0x6175452b , rtp_ts : 0x9a2a7, psent : 0xcaf, osend : 0x77b4c, rssrc :
1285703938, frat : 0,lost : 0x0, last seq : 0x1f82, jitter : 0x2

<rtcp_build_rtcp> Called IFX_TAPI_PKT_RTCP_STATISTICS_GET, ch : 0, ret : 0 , ssrc : 0x6175452b , rtp_ts : 0xa60a9, psent : 0xdb5, osend : 0x81810, rssrc :
1285703938, frat : 0,lost : 0x0, last seq : 0x1ff6, jitter : 0x2

20:42:54.723   pjsua_core.c  RX 409 bytes Request msg BYE/cseq=102 (rdata0x10165a0c) from UDP 192.168.7.28:5060:
BYE sip:222@192.168.8.54:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.7.28:5060;branch=z9hG4bK0a121638
Max-Forwards: 70
From: sip:333@192.168.7.28;tag=as6d3023d4
To: sip:222@192.168.7.28;tag=ReYStR8O5z7kCkGv3JyE1X9fLeK-RYZl
Call-ID: e.VhXnDqscoRBixjz63.q.usCq22.Wxt
CSeq: 102 BYE
User-Agent: Asterisk PBX 1.6.2.11
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


--end msg--
20:42:54.725   pjsua_core.c  TX 287 bytes Response msg 200/BYE/cseq=102 (tdta0x101a3018) to UDP 192.168.7.28:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.7.28:5060;received=192.168.7.28;branch=z9hG4bK0a121638
Call-ID: e.VhXnDqscoRBixjz63.q.usCq22.Wxt
From: <sip:333@192.168.7.28>;tag=as6d3023d4
To: <sip:222@192.168.7.28>;tag=ReYStR8O5z7kCkGv3JyE1X9fLeK-RYZl
CSeq: 102 BYE
Content-Length:  0


--end msg--
<SIPUA> ************ [ch 0] call id : 1   ----- [ch 1] call id : -1 **************
<CC-MODULE> [ch 0] START BUSY TONE
<SIPUA> ************ [ch 0] call id : 1   ----- [ch 1] call id : -1 **************
<SIPUA> ************* cmdid : 3 , ch:0, rtp_port_min: 4000 rtp_port_max: 4010 remote_rtp_port:269758956
 20:42:54.730    pjsua_app.c  Call 1 is DISCONNECTED [reason=200 (Normal call clearing)]
<PJSUA> ************Active Calls List ************************
<PJSUA> *******************************************************
<SIPUA> ************ [ch 0] call id : 1   ----- [ch 1] call id : -1 **************
<SIPUA> closing ch0
20:42:54.730  pjsua_media.c  Closing null sound device 0..
<SIPUA> [ch 0] Calling Stop Codecs
<CC-MODULE> [ch 0] Codec decoding Stopped
<CC-MODULE> [ch 0] Codec encoding Stopped
<SIPUA> ************** In On_Stream_destroyed - 1 ****************
20:42:54.745  pjsua_media.c  Media session for call 1 is destroyed
EWL:on_reset_call +++++++++++++++++++ channel 0 callid:-1, channel 1 callid:-1




FAX Receiving side SIP Traces
=============================
==========================================

--end msg--
03:19:37.142   pjsua_core.c  RX 801 bytes Request msg INVITE/cseq=102 (rdata0x10165a0c) from UDP 192.168.7.28:5060:
INVITE sip:333@192.168.8.51:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.7.28:5060;branch=z9hG4bK56701eaa
Max-Forwards: 70
From: "voip2" <sip:222@192.168.7.28>;tag=as606f84c2
To: <sip:333@192.168.8.51:5060>
Contact: <sip:222@192.168.7.28>
Call-ID: 35474d8e747875700681800c796f5778@192.168.7.28
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.11
Date: Wed, 19 Jan 2011 02:59:56 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 261

v=0
o=root 1683219920 1683219920 IN IP4 192.168.7.28
s=Asterisk PBX 1.6.2.11
c=IN IP4 192.168.7.28
t=0 0
m=audio 15384 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

--end msg--
EWL:on_reset_call +++++++++++++++++++ channel 0 callid:-1, channel 1 callid:-1
03:19:37.143  pjsua_media.c  Media index 0 selected for call 1
03:19:37.147   pjsua_core.c  TX 278 bytes Response msg 100/INVITE/cseq=102 (tdta0x10194170) to UDP 192.168.7.28:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.7.28:5060;received=192.168.7.28;branch=z9hG4bK56701eaa
Call-ID: 35474d8e747875700681800c796f5778@192.168.7.28
From: "voip2" <sip:222@192.168.7.28>;tag=as606f84c2
To: <sip:333@192.168.8.51>
CSeq: 102 INVITE
Content-Length:  0


--end msg--
<CC-MODULE> caller id : 222
<CC-MODULE> callerid in rincid : 222
<CC-MODULE> [ch 1] START RINGING
03:19:37.152   pjsua_core.c  TX 446 bytes Response msg 180/INVITE/cseq=102 (tdta0x10194170) to UDP 192.168.7.28:5060:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.7.28:5060;received=192.168.7.28;branch=z9hG4bK56701eaa
Call-ID: 35474d8e747875700681800c796f5778@192.168.7.28
From: "voip2" <sip:222@192.168.7.28>;tag=as606f84c2
To: <sip:333@192.168.8.51>;tag=lG1gaxqt6zYg6X4Q5q1ZtjtE2yzui2l4
CSeq: 102 INVITE
Contact: <sip:333@192.168.8.51:5060>
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Content-Length:  0


--end msg--
03:19:37.153    pjsua_app.c  Call 1 state changed to EARLY (180 Ringing)
<CC-MODULE> [ch 1] off-hook
<CC-MODULE> Sending CMD_CC_OFF_HOOK to SIPUA for the channel - 1
<SIPUA> ****** LED Status sent to Led Controller : 4 1 17 ******
03:19:39.992 strm0x10197a34  VAD temporarily disabled
03:19:39.993 strm0x10197a34  Encoder stream started
03:19:39.993 strm0x10197a34  Decoder stream started
<SIPUA> ************ [ch 0] call id : -1   ----- [ch 1] call id : 1 **************
 03:19:39.994  pjsua_media.c  Media updates, stream #0: PCMU (sendrecv)
********* before ring_stop *********************
<SIPUA> ************ [ch 0] call id : -1   ----- [ch 1] call id : 1 **************
<CC-MODULE> [ch 1] STOP RINGING
<SIPUA> ************ [ch 0] call id : -1   ----- [ch 1] call id : 1 **************
********* In connect_sound check *********************
<SIPUA> ************ [ch 0] call id : -1   ----- [ch 1] call id : 1 **************
 03:19:39.995  pjsua_media.c  Opening vinetic ch1 ..
************ port_map[1] : 2 ******************
03:19:39.996   conference.c  Port 2 (sip:222@192.168.7.28) transmitting to port 1 (channel1)
03:19:39.996   conference.c  Port 1 (channel1) transmitting to port 2 (sip:222@192.168.7.28)
codec_info, codec_name: PCMU     ptime: 20----------------------+++++
selected codec PCMU
<CC-MODULE> IFX_PKT_RTP_CFG_SET , [SSRC - 0x10d856cf] [Seq No : 0xd4a] ret: 0
<CC-MODULE> received ptime from the SDP message - 20
<CC-MODULE> CODEC ULAW (PCMU) configured
<SIPUA> ************ [ch 0] call id : -1   ----- [ch 1] call id : 1 **************
<SIPUA> ************* cmdid : 2 , ch:1, rtp_port_min: 16384 rtp_port_max: 16394 remote_rtp_port:15384
<SIPUA> [ch 1] enabling start codec
<CC-MODULE> [ch 1] Codec encoding Started
<CC-MODULE> [ch 1] Codec decoding Started
********* After calling conf_connect for both directions *********************
03:19:40.002    pjsua_app.c  Media for call 1 is active
03:19:40.003   pjsua_core.c  TX 776 bytes Response msg 200/INVITE/cseq=102 (tdta0x10194170) to UDP 192.168.7.28:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.7.28:5060;received=192.168.7.28;branch=z9hG4bK56701eaa
Call-ID: 35474d8e747875700681800c796f5778@192.168.7.28
From: "voip2" <sip:222@192.168.7.28>;tag=as606f84c2
To: <sip:333@192.168.8.51>;tag=lG1gaxqt6zYg6X4Q5q1ZtjtE2yzui2l4
CSeq: 102 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Contact: <sip:333@192.168.8.51:5060>
Supported: replaces, 100rel, timer, norefersub
Content-Type: application/sdp
Content-Length:   253

v=0
o=- 3504395977 3504395978 IN IP4 192.168.8.51
s=pjmedia
c=IN IP4 192.168.8.51
t=0 0
a=X-nat:0
m=audio 16386 RTP/AVP 0 101
a=rtcp:16387 IN IP4 192.168.8.51
a=rtpmap:0 PCMU/8000
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

--end msg--
03:19:40.004    pjsua_app.c  Call 1 state changed to CONNECTING
03:19:40.011   pjsua_core.c  RX 399 bytes Request msg ACK/cseq=102 (rdata0x10165a0c) from UDP 192.168.7.28:5060:
ACK sip:333@192.168.8.51:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.7.28:5060;branch=z9hG4bK51351b06
Max-Forwards: 70
From: "voip2" <sip:222@192.168.7.28>;tag=as606f84c2
To: <sip:333@192.168.8.51:5060>;tag=lG1gaxqt6zYg6X4Q5q1ZtjtE2yzui2l4
Contact: <sip:222@192.168.7.28>
Call-ID: 35474d8e747875700681800c796f5778@192.168.7.28
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.2.11
Content-Length: 0


--end msg--
03:19:40.011    pjsua_app.c  <SIPUA> Call 1 state is CONFIRMED
03:19:40.012    pjsua_app.c  Call 1 state changed to CONFIRMED
<rtcp_build_rtcp> Called IFX_TAPI_PKT_RTCP_STATISTICS_GET, ch : 1, ret : 0 , ssrc : 0x10d856cf , rtp_ts : 0x1e9a, psent : 0x15, osend : 0x90d, rssrc : 682606238, frat : 0,lost : 0xffffffff, last seq : 0x89c1, jitter : 0x1ce

<TAPI_SIGNAL> fTX
<TAPI_SIGNAL> [ch 1] Received IFX_TAPI_EVENT_FAXMODEM_CED
<TAPI_SIGNAL> fTX
<TAPI_SIGNAL> [ch 1] Received IFX_TAPI_EVENT_FAXMODEM_CED
<TAPI_SIGNAL> fRX
<TAPI_SIGNAL> [ch 1] Received IFX_TAPI_EVENT_FAXMODEM_CED
<TAPI_SIGNAL> fRX
<TAPI_SIGNAL> [ch 1] Received IFX_TAPI_EVENT_FAXMODEM_CEDEND
<TAPI_SIGNAL> fRX
<TAPI_SIGNAL> [ch 1] Received IFX_TAPI_EVENT_FAXMODEM_CED
<TAPI_SIGNAL> fRX
<TAPI_SIGNAL> [ch 1] Received IFX_TAPI_EVENT_FAXMODEM_CEDEND
<TAPI_SIGNAL> fRX
<TAPI_SIGNAL> [ch 1] Received IFX_TAPI_EVENT_FAXMODEM_CNGFAX
<TAPI-SIGNALS><ClearChannel> NO CODEC Selected
<TAPI_SIGNALS> before disabling WLEC  -- nNBNEwindow : 8 , nNBFEwindow : 8 , nWBNEwindow : 8
<SIPUA> [ch 1] fax detected with call_id____+_+_+_++__+_+_+_+_++_+_+_+_+_++_++++++++++++++++ : 1
03:19:44.606   pjsua_call.c  EWL:fmt_count codec for FAX, -_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_2

local seq-no:18609  remote seq-no:102 _____________________________________
03:19:44.611   pjsua_core.c  TX 926 bytes Request msg INVITE/cseq=18609 (tdta0x10194a90) to UDP 192.168.7.28:5060:
INVITE sip:222@192.168.7.28 SIP/2.0
Via: SIP/2.0/UDP 192.168.8.51:5060;rport;branch=z9hG4bKPjaMUVSBPeAJJZZKPJLVHXQvKf7hEcpuFC
Max-Forwards: 70
From: <sip:333@192.168.8.51>;tag=lG1gaxqt6zYg6X4Q5q1ZtjtE2yzui2l4
To: "voip2" <sip:222@192.168.7.28>;tag=as606f84c2
Contact: <sip:333@192.168.8.51:5060>
Call-ID: 35474d8e747875700681800c796f5778@192.168.7.28
CSeq: 18609 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: PJSUA v1.7/powerpc-unknown-none
Content-Type: application/sdp
Content-Length:   272

v=0
o=- 3504395977 3504395979 IN IP4 192.168.8.51
s=pjmedia
c=IN IP4 192.168.8.51
t=0 0
a=X-nat:0
m=audio 16386 RTP/AVP 8 101
a=rtcp:16387 IN IP4 192.168.8.51
a=rtpmap:8 PCMA/8000
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=silenceSupp:off

--end msg--
<CC-MODULE> Sent CMD_FAX_TONE_DETECTED to SIPUA
03:19:44.620   pjsua_core.c  RX 514 bytes Response msg 100/INVITE/cseq=18609 (rdata0x10165a0c) from UDP 192.168.7.28:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.8.51:5060;rport;branch=z9hG4bKPjaMUVSBPeAJJZZKPJLVHXQvKf7hEcpuFC;received=192.168.8.51
From: <sip:333@192.168.8.51>;tag=lG1gaxqt6zYg6X4Q5q1ZtjtE2yzui2l4
To: "voip2" <sip:222@192.168.7.28>;tag=as606f84c2
Call-ID: 35474d8e747875700681800c796f5778@192.168.7.28
CSeq: 18609 INVITE
Server: Asterisk PBX 1.6.2.11
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:222@192.168.7.28>
Content-Length: 0


--end msg--
03:19:44.622   pjsua_core.c  RX 780 bytes Response msg 200/INVITE/cseq=18609 (rdata0x10165a0c) from UDP 192.168.7.28:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.8.51:5060;rport;branch=z9hG4bKPjaMUVSBPeAJJZZKPJLVHXQvKf7hEcpuFC;received=192.168.8.51
From: <sip:333@192.168.8.51>;tag=lG1gaxqt6zYg6X4Q5q1ZtjtE2yzui2l4
To: "voip2" <sip:222@192.168.7.28>;tag=as606f84c2
Call-ID: 35474d8e747875700681800c796f5778@192.168.7.28
CSeq: 18609 INVITE
Server: Asterisk PBX 1.6.2.11
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:222@192.168.7.28>
Content-Type: application/sdp
Content-Length: 237

v=0
o=root 1683219920 1683219921 IN IP4 192.168.7.28
s=Asterisk PBX 1.6.2.11
c=IN IP4 192.168.7.28
t=0 0
m=audio 15384 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

--end msg--
<SIPUA> ************ [ch 0] call id : -1   ----- [ch 1] call id : 1 **************
<SIPUA> closing ch1
03:19:44.624  pjsua_media.c  Closing null sound device 1..
<SIPUA> [ch 1] Calling Stop Codecs
<CC-MODULE> [ch 1] Codec decoding Stopped
<CC-MODULE> [ch 1] Codec encoding Stopped
<SIPUA> ************** In On_Stream_destroyed - 1 ****************
03:19:44.638  pjsua_media.c  Media session for call 1 is destroyed
03:19:44.639 strm0x101978c4  VAD temporarily disabled
<TAPI_SIGNAL> fTX
<TAPI_SIGNAL> [ch 1] Received IFX_TAPI_EVENT_FAXMODEM_CEDEND
<TAPI_SIGNAL> fTX
<TAPI_SIGNAL> [ch 1] Received IFX_TAPI_EVENT_FAXMODEM_HOLDEND
03:19:44.642 strm0x101978c4  Encoder stream started
03:19:44.642 strm0x101978c4  Decoder stream started
<SIPUA> ************ [ch 0] call id : -1   ----- [ch 1] call id : 1 **************
FAX Restore channel Setting default JB size [ADAPT_VOICE] ret 0 JJJJJJJJJJJJJJJJJJJBBBBBBBBBBBBBBB

<TAPI_SIGNAL> fRX
<TAPI_SIGNAL> [ch 1] Received IFX_TAPI_EVENT_FAXMODEM_HOLDEND
FAX Restore channel Setting default JB size [ADAPT_VOICE] ret 0 JJJJJJJJJJJJJJJJJJJBBBBBBBBBBBBBBB

03:19:44.648  pjsua_media.c  Media updates, stream #0: PCMA (sendrecv)
********* before ring_stop *********************
********* In connect_sound check *********************
<SIPUA> ************ [ch 0] call id : -1   ----- [ch 1] call id : 1 **************
 03:19:45.317  pjsua_media.c  Opening vinetic ch1 ..
************ port_map[1] : 2 ******************
03:19:45.318   conference.c  Port 2 (sip:222@192.168.7.28) transmitting to port 1 (channel1)
03:19:45.319   conference.c  Port 1 (channel1) transmitting to port 2 (sip:222@192.168.7.28)
codec_info, codec_name: PCMA     ptime: 20----------------------+++++
selected codec PCMA
<CC-MODULE> IFX_PKT_RTP_CFG_SET , [SSRC - 0x3c37c363] [Seq No : 0x1f10] ret: 0
<CC-MODULE> received ptime from the SDP message - 20
<CC-MODULE>  CODEC ALAW (PCMA) configured
<SIPUA> ************ [ch 0] call id : -1   ----- [ch 1] call id : 1 **************
<SIPUA> ************* cmdid : 2 , ch:1, rtp_port_min: 16384 rtp_port_max: 16394 remote_rtp_port:15384
<SIPUA> [ch 1] enabling start codec
<CC-MODULE> [ch 1] Codec encoding Started
<CC-MODULE> [ch 1] Codec decoding Started
********* After calling conf_connect for both directions *********************
03:19:45.324    pjsua_app.c  Media for call 1 is active
03:19:45.326   pjsua_core.c  TX 356 bytes Request msg ACK/cseq=18609 (tdta0x10199f58) to UDP 192.168.7.28:5060:
ACK sip:222@192.168.7.28 SIP/2.0
Via: SIP/2.0/UDP 192.168.8.51:5060;rport;branch=z9hG4bKPjsHKbWNIlGI3wMcp-1UGdOTI7sTjvT01m
Max-Forwards: 70
From: <sip:333@192.168.8.51>;tag=lG1gaxqt6zYg6X4Q5q1ZtjtE2yzui2l4
To: "voip2" <sip:222@192.168.7.28>;tag=as606f84c2
Call-ID: 35474d8e747875700681800c796f5778@192.168.7.28
CSeq: 18609 ACK
Content-Length:  0


--end msg--
03:19:45.327   pjsua_core.c  RX 780 bytes Response msg 200/INVITE/cseq=18609 (rdata0x10165a0c) from UDP 192.168.7.28:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.8.51:5060;rport;branch=z9hG4bKPjaMUVSBPeAJJZZKPJLVHXQvKf7hEcpuFC;received=192.168.8.51
From: <sip:333@192.168.8.51>;tag=lG1gaxqt6zYg6X4Q5q1ZtjtE2yzui2l4
To: "voip2" <sip:222@192.168.7.28>;tag=as606f84c2
Call-ID: 35474d8e747875700681800c796f5778@192.168.7.28
CSeq: 18609 INVITE
Server: Asterisk PBX 1.6.2.11
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:222@192.168.7.28>
Content-Type: application/sdp
Content-Length: 237

v=0
o=root 1683219920 1683219921 IN IP4 192.168.7.28
s=Asterisk PBX 1.6.2.11
c=IN IP4 192.168.7.28
t=0 0
m=audio 15384 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

--end msg--
03:19:45.328   pjsua_core.c  TX 356 bytes Request msg ACK/cseq=18609 (tdta0x10199f58) to UDP 192.168.7.28:5060:
ACK sip:222@192.168.7.28 SIP/2.0
Via: SIP/2.0/UDP 192.168.8.51:5060;rport;branch=z9hG4bKPjsHKbWNIlGI3wMcp-1UGdOTI7sTjvT01m
Max-Forwards: 70
From: <sip:333@192.168.8.51>;tag=lG1gaxqt6zYg6X4Q5q1ZtjtE2yzui2l4
To: "voip2" <sip:222@192.168.7.28>;tag=as606f84c2
Call-ID: 35474d8e747875700681800c796f5778@192.168.7.28
CSeq: 18609 ACK
Content-Length:  0


--end msg--
03:19:45.329   pjsua_core.c  RX 780 bytes Response msg 200/INVITE/cseq=18609 (rdata0x10165a0c) from UDP 192.168.7.28:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.8.51:5060;rport;branch=z9hG4bKPjaMUVSBPeAJJZZKPJLVHXQvKf7hEcpuFC;received=192.168.8.51
From: <sip:333@192.168.8.51>;tag=lG1gaxqt6zYg6X4Q5q1ZtjtE2yzui2l4
To: "voip2" <sip:222@192.168.7.28>;tag=as606f84c2
Call-ID: 35474d8e747875700681800c796f5778@192.168.7.28
CSeq: 18609 INVITE
Server: Asterisk PBX 1.6.2.11
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:222@192.168.7.28>
Content-Type: application/sdp
Content-Length: 237

v=0
o=root 1683219920 1683219921 IN IP4 192.168.7.28
s=Asterisk PBX 1.6.2.11
c=IN IP4 192.168.7.28
t=0 0
m=audio 15384 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

--end msg--
<TAPI_SIGNAL> fTX
<TAPI_SIGNAL> [ch 1] Received IFX_TAPI_EVENT_FAXMODEM_DIS
03:19:45.998   pjsua_core.c  TX 356 bytes Request msg ACK/cseq=18609 (tdta0x10199f58) to UDP 192.168.7.28:5060:
ACK sip:222@192.168.7.28 SIP/2.0
Via: SIP/2.0/UDP 192.168.8.51:5060;rport;branch=z9hG4bKPjsHKbWNIlGI3wMcp-1UGdOTI7sTjvT01m
Max-Forwards: 70
From: <sip:333@192.168.8.51>;tag=lG1gaxqt6zYg6X4Q5q1ZtjtE2yzui2l4
To: "voip2" <sip:222@192.168.7.28>;tag=as606f84c2
Call-ID: 35474d8e747875700681800c796f5778@192.168.7.28
CSeq: 18609 ACK
Content-Length:  0


--end msg--
<rtcp_build_rtcp> Called IFX_TAPI_PKT_RTCP_STATISTICS_GET, ch : 1, ret : 0 , ssrc : 0x3c37c363 , rtp_ts : 0xb3a1, psent : 0xb1, osend : 0x55e8, rssrc : 943776983, frat : 0,lost : 0x0, last seq : 0x8ab4, jitter : 0x3

<rtcp_build_rtcp> Called IFX_TAPI_PKT_RTCP_STATISTICS_GET, ch : 1, ret : 0 , ssrc : 0x3c37c363 , rtp_ts : 0x1512b, psent : 0x1a8, osend : 0xed5f, rssrc : 943776983, frat : 0,lost : 0x0, last seq : 0x8bab, jitter : 0x5

<rtcp_build_rtcp> Called IFX_TAPI_PKT_RTCP_STATISTICS_GET, ch : 1, ret : 0 , ssrc : 0x3c37c363 , rtp_ts : 0x1eeaa, psent : 0x26a, osend : 0x151ab, rssrc : 943776983, frat : 0,lost : 0x0, last seq : 0x8ca0, jitter : 0x5

<rtcp_build_rtcp> Called IFX_TAPI_PKT_RTCP_STATISTICS_GET, ch : 1, ret : 0 , ssrc : 0x3c37c363 , rtp_ts : 0x28c24, psent : 0x32e, osend : 0x1b1fa, rssrc : 943776983, frat : 0,lost : 0x0, last seq : 0x8d96, jitter : 0x6

<rtcp_build_rtcp> Called IFX_TAPI_PKT_RTCP_STATISTICS_GET, ch : 1, ret : 0 , ssrc : 0x3c37c363 , rtp_ts : 0x3299d, psent : 0x3d2, osend : 0x1f877, rssrc : 943776983, frat : 1,lost : 0x1, last seq : 0x8e92, jitter : 0x8

<rtcp_build_rtcp> Called IFX_TAPI_PKT_RTCP_STATISTICS_GET, ch : 1, ret : 0 , ssrc : 0x3c37c363 , rtp_ts : 0x3c9aa, psent : 0x456, osend : 0x22a5f, rssrc : 943776983, frat : 0,lost : 0x1, last seq : 0x8f87, jitter : 0x9

<rtcp_build_rtcp> Called IFX_TAPI_PKT_RTCP_STATISTICS_GET, ch : 1, ret : 0 , ssrc : 0x3c37c363 , rtp_ts : 0x4671f, psent : 0x507, osend : 0x2881e, rssrc : 943776983, frat : 0,lost : 0x1, last seq : 0x907d, jitter : 0x3

03:20:25.402   pjsua_core.c  RX 519 bytes Request msg OPTIONS/cseq=102 (rdata0x10165a0c) from UDP 192.168.7.28:5060:
OPTIONS sip:333@192.168.8.51:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.7.28:5060;branch=z9hG4bK54ac690c
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.7.28>;tag=as40e2682e
To: <sip:333@192.168.8.51:5060>
Contact: <sip:Unknown@192.168.7.28>
Call-ID: 10f32a851643cf1c6c757010391f8ec7@192.168.7.28
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.11
Date: Wed, 19 Jan 2011 03:00:44 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


--end msg--
03:20:25.403   pjsua_core.c  TX 1039 bytes Response msg 200/OPTIONS/cseq=102 (tdta0x10194a90) to UDP 192.168.7.28:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.7.28:5060;received=192.168.7.28;branch=z9hG4bK54ac690c
Call-ID: 10f32a851643cf1c6c757010391f8ec7@192.168.7.28
From: "Unknown" <sip:Unknown@192.168.7.28>;tag=as40e2682e
To: <sip:333@192.168.8.51>;tag=z9hG4bK54ac690c
CSeq: 102 OPTIONS
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Accept: application/sdp, application/pidf+xml, application/xpidf+xml, application/simple-message-summary, message/sipfrag;version=2.0, application/im-iscomposing+xml, text/plain
Supported: replaces, 100rel, timer, norefersub
Allow-Events: presence, message-summary, refer
User-Agent: PJSUA v1.7/powerpc-unknown-none
Content-Type: application/sdp
Content-Length:   292

v=0
o=- 3504396025 3504396025 IN IP4 192.168.8.51
s=pjmedia
c=IN IP4 192.168.8.51
t=0 0
m=audio 16384 RTP/AVP 18 8 0 101
a=rtcp:16385 IN IP4 192.168.8.51
a=rtpmap:18 G729/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

--end msg--
<rtcp_build_rtcp> Called IFX_TAPI_PKT_RTCP_STATISTICS_GET, ch : 1, ret : 0 , ssrc : 0x3c37c363 , rtp_ts : 0x509a8, psent : 0x5b5, osend : 0x2e492, rssrc : 943776983, frat : 0,lost : 0x1, last seq : 0x916d, jitter : 0xb

<rtcp_build_rtcp> Called IFX_TAPI_PKT_RTCP_STATISTICS_GET, ch : 1, ret : 0 , ssrc : 0x3c37c363 , rtp_ts : 0x5a71c, psent : 0x649, osend : 0x331f0, rssrc : 943776983, frat : 0,lost : 0x1, last seq : 0x91e9, jitter : 0x7

<rtcp_build_rtcp> Called IFX_TAPI_PKT_RTCP_STATISTICS_GET, ch : 1, ret : 0 , ssrc : 0x3c37c363 , rtp_ts : 0x67054, psent : 0x6e9, osend : 0x37e13, rssrc : 943776983, frat : 0,lost : 0x1, last seq : 0x92a3, jitter : 0x3

<rtcp_build_rtcp> Called IFX_TAPI_PKT_RTCP_STATISTICS_GET, ch : 1, ret : 0 , ssrc : 0x3c37c363 , rtp_ts : 0x70de0, psent : 0x796, osend : 0x3d9e7, rssrc : 943776983, frat : 0,lost : 0x1, last seq : 0x9395, jitter : 0xb

<rtcp_build_rtcp> Called IFX_TAPI_PKT_RTCP_STATISTICS_GET, ch : 1, ret : 0 , ssrc : 0x3c37c363 , rtp_ts : 0x7d854, psent : 0x837, osend : 0x42615, rssrc : 943776983, frat : 0,lost : 0x1, last seq : 0x9452, jitter : 0x4

<rtcp_build_rtcp> Called IFX_TAPI_PKT_RTCP_STATISTICS_GET, ch : 1, ret : 0 , ssrc : 0x3c37c363 , rtp_ts : 0x875d6, psent : 0x873, osend : 0x42d51, rssrc : 943776983, frat : 0,lost : 0x1, last seq : 0x94b6, jitter : 0x3

<rtcp_build_rtcp> Called IFX_TAPI_PKT_RTCP_STATISTICS_GET, ch : 1, ret : 0 , ssrc : 0x3c37c363 , rtp_ts : 0x91361, psent : 0x908, osend : 0x47990, rssrc : 943776983, frat : 0,lost : 0x1, last seq : 0x95a8, jitter : 0x5

<rtcp_build_rtcp> Called IFX_TAPI_PKT_RTCP_STATISTICS_GET, ch : 1, ret : 0 , ssrc : 0x3c37c363 , rtp_ts : 0x9b127, psent : 0x9a9, osend : 0x4c5be, rssrc : 943776983, frat : 0,lost : 0x1, last seq : 0x9698, jitter : 0x7

<CC-MODULE> [ch 1] on-hook
<CC-MODULE> Currently No Dail/Busy/Ringback tone enabled
<SIPUA> [ch 1] call hangup with call_id : 1
03:21:09.123   pjsua_core.c  TX 401 bytes Request msg BYE/cseq=18610 (tdta0x10194a90) to UDP 192.168.7.28:5060:
BYE sip:222@192.168.7.28 SIP/2.0
Via: SIP/2.0/UDP 192.168.8.51:5060;rport;branch=z9hG4bKPjuCX5QD5DewIjk4HHuPHKyptrCuzH.QsQ
Max-Forwards: 70
From: <sip:333@192.168.8.51>;tag=lG1gaxqt6zYg6X4Q5q1ZtjtE2yzui2l4
To: "voip2" <sip:222@192.168.7.28>;tag=as606f84c2
Call-ID: 35474d8e747875700681800c796f5778@192.168.7.28
CSeq: 18610 BYE
User-Agent: PJSUA v1.7/powerpc-unknown-none
Content-Length:  0


--end msg--
<SIPUA> [ch 1] call_id - 1 **********************
<CC-MODULE> Sending CMD_CC_ON_HOOK to SIPUA for the channel - 1
<SIPUA> [ch 1] sip account reg status - 200 ()
<SIPUA> ****** LED Status sent to Led Controller : 4 1 16 ******
03:21:09.130   pjsua_core.c  RX 474 bytes Response msg 200/BYE/cseq=18610 (rdata0x10165a0c) from UDP 192.168.7.28:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.8.51:5060;rport;branch=z9hG4bKPjuCX5QD5DewIjk4HHuPHKyptrCuzH.QsQ;received=192.168.8.51
From: <sip:333@192.168.8.51>;tag=lG1gaxqt6zYg6X4Q5q1ZtjtE2yzui2l4
To: "voip2" <sip:222@192.168.7.28>;tag=as606f84c2
Call-ID: 35474d8e747875700681800c796f5778@192.168.7.28
CSeq: 18610 BYE
Server: Asterisk PBX 1.6.2.11
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


--end msg--
<SIPUA> ************ [ch 0] call id : -1   ----- [ch 1] call id : 1 **************
<SIPUA> ************ [ch 0] call id : -1   ----- [ch 1] call id : 1 **************
<SIPUA> ************* cmdid : 3 , ch:1, rtp_port_min: 16384 rtp_port_max: 16394 remote_rtp_port:269758956
 03:21:09.132    pjsua_app.c  Call 1 is DISCONNECTED [reason=200 (Normal call clearing)]
<PJSUA> ************Active Calls List ************************
<PJSUA> *******************************************************
<SIPUA> ************ [ch 0] call id : -1   ----- [ch 1] call id : 1 **************
<SIPUA> closing ch1
03:21:09.132  pjsua_media.c  Closing null sound device 1..
<SIPUA> [ch 1] Calling Stop Codecs
<CC-MODULE> [ch 1] Codec decoding Stopped
<CC-MODULE> [ch 1] Codec encoding Stopped
<SIPUA> ************** In On_Stream_destroyed - 1 ****************
03:21:09.155  pjsua_media.c  Media session for call 1 is destroyed
EWL:on_reset_call +++++++++++++++++++ channel 0 callid:-1, channel 1 callid:-1
By: Matthew Nicholson (mnicholson) 2011-01-19 10:08:46.000-0600

This is not the way fax passthrough works in asterisk.  If you would like to do passthrough in this manner, set up the channel for ulaw or alaw from the start, or use T.38 passthrough.  T.38 passthrough is the only fax passthrough configuration that we officially support.