[asterisk-users] FFA SendFax rejects T.38 reINVITE (488 Not acceptable here)

Tue Oct 19 10:48:50 CDT 2010

We don't have an ATA and fax machine.

The whole point (as I specified in the header and initial message) is the
attempt to use "Fax for Asterisk" to send the message.

As I showed in the logs, the remote carrier sends a proper T.38 reINVITE,
but our Asterisk doesn't accept, despite the fact that this provider is
defined in sip.conf with both canreinvite and t38pt_udptl enabled, so the
only question is (as far as we understand) is why in this scenario, the T.38
is rejected.

Here are the logs (sip debug is open) again, since we get the reINVITE:

<--- SIP read from UDP:xxx.xxx.xxx.xx8:5060 --->
INVITE sip:1234567 at 10.0.0.3:5060 SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xx8:5060;branch=z9hG4bK0dB004c7e5e3c3e60a8
From: <sip:98765432 at xxx.xxx.xxx.xx8:5060><sip:98765432 at xxx.xxx.xxx.xx8:5060>;tag=gK0d817deb

To: "Fax" <sip:1234567 at yyy.yyy.yyy.yyy>
<sip:1234567 at yyy.yyy.yyy.yyy>;tag=as0ddeacb5

Call-ID: 74ca1e4e3e86a1b873428773477e201f at yyy.yyy.yyy.yyy
CSeq: 1785 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:98765432 at xxx.xxx.xxx.xx8:5060><sip:98765432 at xxx.xxx.xxx.xx8:5060>
Supported: timer
Session-Expires: 1800;refresher=uas
Min-SE: 90
Content-Length:  303
Content-Disposition: session; handling=required
Content-Type: application/sdp

v=0
o=Sonus_UAC 218 7126 IN IP4 xxx.xxx.xxx.xx8
s=SIP Media Capabilities
c=IN IP4 xxx.xxx.xxx.xx7
t=0 0
m=image 6202 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
74ca1e4e3e86a1b873428773477e201f 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=z9hG4bK0dB004c7e5e3c3e60a8;received=xxx.xxx.xxx.xx8

From: <sip:98765432 at xxx.xxx.xxx.xx8:5060><sip:98765432 at xxx.xxx.xxx.xx8:5060>;tag=gK0d817deb

To: "Fax" <sip:1234567 at yyy.yyy.yyy.yyy>
<sip:1234567 at yyy.yyy.yyy.yyy>;tag=as0ddeacb5

Call-ID: 74ca1e4e3e86a1b873428773477e201f at yyy.yyy.yyy.yyy
CSeq: 1785 INVITE
Server: Smartel-PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:1234567 at yyy.yyy.yyy.yyy> <sip:1234567 at yyy.yyy.yyy.yyy>
Content-Length: 0

<------------>

<--- 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=z9hG4bK0dB004c7e5e3c3e60a8;received=xxx.xxx.xxx.xx8

From: <sip:98765432 at xxx.xxx.xxx.xx8:5060><sip:98765432 at xxx.xxx.xxx.xx8:5060>;tag=gK0d817deb

To: "Fax" <sip:1234567 at yyy.yyy.yyy.yyy>
<sip:1234567 at yyy.yyy.yyy.yyy>;tag=as0ddeacb5

Call-ID: 74ca1e4e3e86a1b873428773477e201f at yyy.yyy.yyy.yyy
CSeq: 1785 INVITE
Server: Smartel-PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

Thanks.

Michael

On Tue, Oct 19, 2010 at 5:40 PM, David Backeberg <dbackeberg at gmail.com>wrote:

> On Tue, Oct 19, 2010 at 11:21 AM, VoIP Question <voip.question at gmail.com>
> wrote:
> > It's set to yes for this peer.
> >
> > also t38pt_udptl is set to yes.
> >
> > :(
>
> You don't say anything about what you're trying to send / receive against.
>
> Here's how you should troubleshoot:
>
> * start with a 'real fax machine' if you have one, on an analog line
> if you have one. If you can't receive / send with that against your
> target, blame your target.
> * move to audio-pass through fax on asterisk. No T.38. If that works.
> * add in T.38
>
> You will learn things in that process and be able to tell at what
> layer your troubles are happening.
>
> It could be coincidental that things give up during the reinvite. It
> could actually be giving up for noise on the line, packet drops, etc.
>
> At the very least, start recording the call. You'll at least be able
> to hear up to the re-invite.
>
> Definitely record the audio passthrough attempt and listen back to it.
>
> --
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