[asterisk-users] FFA SendFax rejects T.38 reINVITE (488 Not acceptable here)
VoIP Question
voip.question at gmail.com
Tue Oct 19 14:17:25 CDT 2010
Fair enough Kevin :-) It's just that your documentation for this product is
so limited that without extensive search online and the assistance of
others, it would have been impossible for us to make any progress and we
haven't reached the ReceiveFax part yet ;)
Anyway, specifically, we installed Asterisk 1.6.2.11. As far as we
know/understand, the SendFax application is running.
This is the full log of the call, until it's rejected for the first time.
The remote switch resends the INVITEs a few more times, but it's all the
same, so I didn't include it:
sip*CLI> -- Attempting call on Local/12345678 at outgoing for s at outboundfax:1
(Retry 1)
sip*CLI> -- Executing [12345678 at outgoing:1]
Dial("Local/12345678 at outgoing-2c36;2", "SIP/12345678 at main,50,tTr") in new
stack
sip*CLI> == Using SIP RTP CoS mark 5
sip*CLI> == Using SIP VRTP CoS mark 6
sip*CLI> == Using UDPTL CoS mark 5
sip*CLI> Audio is at yyy.yyy.yyy.yyy port 10714
sip*CLI> Adding codec 0x100 (g729) to SDP
sip*CLI> Adding codec 0x2 (gsm) to SDP
sip*CLI> Adding non-codec 0x1 (telephone-event) to SDP
sip*CLI> Reliably Transmitting (no NAT) to xxx.xxx.xxx.xx8:5060:
INVITE sip:12345678 at xxx.xxx.xxx.xx8 SIP/2.0
Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=z9hG4bK5b6bc617;rport
Max-Forwards: 70
From: "Fax" <sip:98765432 at yyy.yyy.yyy.yyy>;tag=as28606a47
To: <sip:12345678 at xxx.xxx.xxx.xx8>
Contact: <sip:98765432 at yyy.yyy.yyy.yyy>
Call-ID: 2d965b0926e0134e0b211f882cbd2cc3 at yyy.yyy.yyy.yyy
CSeq: 102 INVITE
User-Agent: PBX
Date: Tue, 19 Oct 2010 16:41:16 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 312
v=0
o=root 697508180 697508180 IN IP4 yyy.yyy.yyy.yyy
s=Asterisk PBX 1.6.2.11
c=IN IP4 yyy.yyy.yyy.yyy
t=0 0
m=audio 10714 RTP/AVP 18 3 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
sip*CLI> -- Called 12345678 at main
sip*CLI>
<--- SIP read from UDP:xxx.xxx.xxx.xx8:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.0.3:5060;branch=z9hG4bK5b6bc617;rport=5060
From: "Fax" <sip:98765432 at 10.0.0.3:5060>;tag=as28606a47
To: <sip:12345678 at xxx.xxx.xxx.xx8:5060>;tag=gK028217ef
Call-ID: 2d965b0926e0134e0b211f882cbd2cc3 at yyy.yyy.yyy.yyy
CSeq: 102 INVITE
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
sip*CLI>
<--- SIP read from UDP:xxx.xxx.xxx.xx8:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.0.0.3:5060;branch=z9hG4bK5b6bc617;rport=5060
From: "Fax" <sip:98765432 at 10.0.0.3:5060>;tag=as28606a47
To: <sip:12345678 at xxx.xxx.xxx.xx8:5060>;tag=gK028217ef
Call-ID: 2d965b0926e0134e0b211f882cbd2cc3 at yyy.yyy.yyy.yyy
CSeq: 102 INVITE
Contact: <sip:12345678 at xxx.xxx.xxx.xx8:5060>
Allow: INVITE,ACK,CANCEL,BYE,INFO,NOTIFY,PRACK,UPDATE,MESSAGE,PUBLISH
Content-Length: 262
Content-Disposition: session; handling=required
Content-Type: application/sdp
v=0
o=Sonus_UAC 28160 32050 IN IP4 xxx.xxx.xxx.xx8
s=SIP Media Capabilities
c=IN IP4 xxx.xxx.xxx.xx7
t=0 0
m=audio 6256 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=maxptime:20
<------------->
--- (11 headers 12 lines) ---
Found RTP audio format 18
Found RTP audio format 101
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - 0x102 (gsm|g729), peer - audio=0x100 (g729)/video=0x0
(nothing)/text=0x0 (nothing), combined - 0x100 (g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port xxx.xxx.xxx.xx7:6256
sip*CLI> -- SIP/main-0000002a is making progress passing it to
Local/12345678 at outgoing-2c36;2
sip*CLI>
<--- SIP read from UDP:xxx.xxx.xxx.xx8:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.3:5060;branch=z9hG4bK5b6bc617;rport=5060
From: "Fax" <sip:98765432 at 10.0.0.3:5060>;tag=as28606a47
To: <sip:12345678 at xxx.xxx.xxx.xx8:5060>;tag=gK028217ef
Call-ID: 2d965b0926e0134e0b211f882cbd2cc3 at yyy.yyy.yyy.yyy
CSeq: 102 INVITE
Accept: application/sdp, application/isup, application/dtmf,
application/dtmf-relay, multipart/mixed
Contact: <sip:12345678 at xxx.xxx.xxx.xx8:5060>
Allow: INVITE,ACK,CANCEL,BYE,INFO,NOTIFY,PRACK,UPDATE,MESSAGE,PUBLISH
Require: timer
Supported: timer
Session-Expires: 1800;refresher=uac
Content-Length: 262
Content-Disposition: session; handling=required
Content-Type: application/sdp
v=0
o=Sonus_UAC 28160 32050 IN IP4 xxx.xxx.xxx.xx8
s=SIP Media Capabilities
c=IN IP4 xxx.xxx.xxx.xx7
t=0 0
m=audio 6256 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=maxptime:20
<------------->
--- (15 headers 12 lines) ---
list_route: hop: <sip:12345678 at xxx.xxx.xxx.xx8:5060>
set_destination: Parsing <sip:12345678 at xxx.xxx.xxx.xx8:5060> for
address/port to send to
set_destination: set destination to xxx.xxx.xxx.xx8, port 5060
Transmitting (no NAT) to xxx.xxx.xxx.xx8:5060:
ACK sip:12345678 at xxx.xxx.xxx.xx8:5060 SIP/2.0
Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=z9hG4bK7c7557f9;rport
Max-Forwards: 70
From: "Fax" <sip:98765432 at yyy.yyy.yyy.yyy>;tag=as28606a47
To: <sip:12345678 at xxx.xxx.xxx.xx8>;tag=gK028217ef
Contact: <sip:98765432 at yyy.yyy.yyy.yyy>
Call-ID: 2d965b0926e0134e0b211f882cbd2cc3 at yyy.yyy.yyy.yyy
CSeq: 102 ACK
User-Agent: PBX
Content-Length: 0
---
sip*CLI> -- SIP/main-0000002a answered Local/12345678 at outgoing-2c36;2
sip*CLI> > Channel Local/12345678 at outgoing-2c36;1 was answered.
-- Executing [s at outboundfax:1] Set("Local/12345678 at outgoing-2c36;1",
"FAXOPT(filename)=/tmp/1585867851.tif") in new stack
[Oct 19 18:41:17] WARNING[31185]: res_fax.c:2454 acf_faxopt_write: channel
'Local/12345678 at outgoing-2c36;1' set FAXOPT(filename) to
'/tmp/1585867851.tif' is unhandled!
-- Executing [s at outboundfax:2] Set("Local/12345678 at outgoing-2c36;1",
"FAXOPT(ecm)=yes") in new stack
sip*CLI> -- Executing [s at outboundfax:3]
Set("Local/12345678 at outgoing-2c36;1", "FAXOPT(headerinfo)=Smartel") in new
stack
-- Executing [s at outboundfax:4] Set("Local/12345678 at outgoing-2c36;1",
"FAXOPT(localstationid)=+972-72-278-0008") in new stack
-- Executing [s at outboundfax:5] Set("Local/12345678 at outgoing-2c36;1",
"FAXOPT(maxrate)=14400") in new stack
-- Executing [s at outboundfax:6] Set("Local/12345678 at outgoing-2c36;1",
"FAXOPT(minrate)=2400") in new stack
-- Executing [s at outboundfax:7] SendFAX("Local/12345678 at outgoing-2c36;1",
"/tmp/1585867851.tif,d") in new stack
-- Channel 'Local/12345678 at outgoing-2c36;1' sending FAX:
-- /tmp/1585867851.tif
sip*CLI> -- Channel 'Local/12345678 at outgoing-2c36;1' FAX session '6'
started
sip*CLI> -- FAX handle 0: [ 000.000152 ], STAT_EVT_STRT_TX st:
IDLE rt: IDLENSTX
-- FAX handle 0: [ 000.000214 ], STAT_EVT_TX_HW_RDY st: WT_TX_HW_RDY
rt: TRDYNHTY
-- FAX handle 0: [ 000.000240 ], P30EVN_SEND_STARTED
sip*CLI> == Spawn extension (outgoing, 12345678, 1) exited non-zero on
'Local/12345678 at outgoing-2c36;2'
sip*CLI> > Channel 'Local/12345678 at outgoing-2c36;1' fax session '6',
[ 000.075767 ], channel sent 2 frames (40 ms) of silence.
sip*CLI> > Channel 'Local/12345678 at outgoing-2c36;1' fax session '6',
[ 000.588736 ], stack sent 28 frames (560 ms) of energy.
sip*CLI> > Channel 'Local/12345678 at outgoing-2c36;1' fax session '6',
[ 003.566428 ], stack sent 149 frames (2980 ms) of silence.
sip*CLI> > Channel 'Local/12345678 at outgoing-2c36;1' fax session '6',
[ 004.089420 ], stack sent 26 frames (520 ms) of energy.
sip*CLI>
<--- SIP read from UDP:xxx.xxx.xxx.xx8:5060 --->
INVITE sip:98765432 at 10.0.0.3:5060 SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xx8:5060;branch=z9hG4bK02B020504a6f7a14f9a
From: <sip:12345678 at xxx.xxx.xxx.xx8:5060>;tag=gK028217ef
To: "Fax" <sip:98765432 at yyy.yyy.yyy.yyy>;tag=as28606a47
Call-ID: 2d965b0926e0134e0b211f882cbd2cc3 at yyy.yyy.yyy.yyy
CSeq: 8530 INVITE
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,INFO,NOTIFY,PRACK,UPDATE,MESSAGE,PUBLISH
Accept: application/sdp, application/isup, application/dtmf,
application/dtmf-relay, multipart/mixed
Contact: <sip:12345678 at xxx.xxx.xxx.xx8:5060>
Supported: timer
Session-Expires: 1800;refresher=uas
Min-SE: 90
Content-Length: 306
Content-Disposition: session; handling=required
Content-Type: application/sdp
v=0
o=Sonus_UAC 28160 32051 IN IP4 xxx.xxx.xxx.xx8
s=SIP Media Capabilities
c=IN IP4 xxx.xxx.xxx.xx7
t=0 0
m=image 6256 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:262
a=T38FaxMaxDatagram:176
a=T38FaxUdpEC:t38UDPRedundancy
a=sendrecv
<------------->
--- (16 headers 13 lines) ---
Sending to xxx.xxx.xxx.xx8 : 5060 (no NAT)
Got T.38 offer in SDP in dialog
2d965b0926e0134e0b211f882cbd2cc3 at yyy.yyy.yyy.yyy
Capabilities: us - 0x102 (gsm|g729), peer - audio=0x0 (nothing)/video=0x0
(nothing)/text=0x0 (nothing), combined - 0x0 (nothing)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0
(nothing), combined - 0x0 (nothing)
Got T.38 Re-invite without audio. Keeping RTP active during T.38 session.
<--- Transmitting (no NAT) to xxx.xxx.xxx.xx8:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
xxx.xxx.xxx.xx8:5060;branch=z9hG4bK02B020504a6f7a14f9a;received=xxx.xxx.xxx.xx8
From: <sip:12345678 at xxx.xxx.xxx.xx8:5060>;tag=gK028217ef
To: "Fax" <sip:98765432 at yyy.yyy.yyy.yyy>;tag=as28606a47
Call-ID: 2d965b0926e0134e0b211f882cbd2cc3 at yyy.yyy.yyy.yyy
CSeq: 8530 INVITE
Server: PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:98765432 at yyy.yyy.yyy.yyy>
Content-Length: 0
<------------>
sip*CLI>
<--- Reliably Transmitting (no NAT) to xxx.xxx.xxx.xx8:5060 --->
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP
xxx.xxx.xxx.xx8:5060;branch=z9hG4bK02B020504a6f7a14f9a;received=xxx.xxx.xxx.xx8
From: <sip:12345678 at xxx.xxx.xxx.xx8:5060>;tag=gK028217ef
To: "Fax" <sip:98765432 at yyy.yyy.yyy.yyy>;tag=as28606a47
Call-ID: 2d965b0926e0134e0b211f882cbd2cc3 at yyy.yyy.yyy.yyy
CSeq: 8530 INVITE
Server: PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
<------------>
sip*CLI>
<--- SIP read from UDP:xxx.xxx.xxx.xx8:5060 --->
ACK sip:98765432 at 10.0.0.3:5060 SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xx8:5060;branch=z9hG4bK02B020504a6f7a14f9a
From: <sip:12345678 at xxx.xxx.xxx.xx8:5060>;tag=gK028217ef
To: "Fax" <sip:98765432 at yyy.yyy.yyy.yyy>;tag=as28606a47
Call-ID: 2d965b0926e0134e0b211f882cbd2cc3 at yyy.yyy.yyy.yyy
CSeq: 8530 ACK
Max-Forwards: 70
Content-Length: 0
Thanks,
Michael
On Tue, Oct 19, 2010 at 8:56 PM, Kevin P. Fleming <kpfleming at digium.com>wrote:
> On 10/19/2010 12:01 PM, VoIP Question wrote:
> > Digium claims that their FFA is the best and most compatible solution
> > and they give one channel for free, but do not provide support for those
> > that do not buy more channels, but why buy more channels if the
> > free/test one doesn't work?
> >
> > I know they read (and sometimes respond) to this list, so I don't
> > understand why they don't clarify this issue.
>
> When you are asking for free help on a mailing list, patience is a
> virtue :-) You posted your question approximately four hours ago.
>
> --
> Kevin P. Fleming
> Digium, Inc. | Director of Software Technologies
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> skype: kpfleming | jabber: kfleming at digium.com
> Check us out at www.digium.com & www.asterisk.org
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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