asterisk users - Jul 2010 - Reinvite to alaw after T.38 reception

I'm having issues with T.38 on Asterisk 1.6.2.8. A few lines are received OK
and then I get only garbage. I'm using ReceiveFAX() provided by app_fax to
receive the faxes.

After talking to the engineers on the telco, they said Asterisk is sending a
REINVITE to alaw after the T.38 reception is complete, and that could be the
cause of the problems.

I personally am not totally convinced of that, but they asked me if it's
possible to make Asterisk not reinvite to alaw after a T.38 fax reception. Is
that possible at all?

Here's the relevant sip.conf and extensions.conf portions:

[voxip]
username=5421047000
nat=yes
type=peer
secret=supersecret
port=5060
canreinvite=no
insecure=port,invite
host=10.150.65.16
fromuser=5421047000
fromdomain=10.153.66.146
dtmfmode=rfc2833
context=entrada-e1
disallow=all
allow=alaw
qualify=no
t38pt_udptl=yes


[macro-recebefax]
exten => s,1,Set(DB(fax/count)=$[${DB(fax/count)} + 1])
exten => s,n,Set(FAXCOUNT=${DB(fax/count)})
exten => s,n,Set(FAXFILE=fax-${DB(fax/count)}-rx)
exten => s,n,Set(LOCALSTATIONID=5421047008)
exten => s,n,ReceiveFAX(/var/spool/asterisk/fax/${FAXFILE}.tif)




Vin?cius Fontes
Gerente de Seguran?a da Informa??o
Canall Tecnologia em Comunica??es
Passo Fundo - RS - Brasil
+55 54 2104-7000

Information Security Manager
Canall Tecnologia em Comunica??es
Passo Fundo - RS - Brazil
+55 54 2104-7000

----- "Kyle Kienapfel" <doctor.whom at gmail.com> escreveu:
> On Mon, Jul 5, 2010 at 4:20 AM, --[ UxBoD ]-- <uxbod at splatnix.net>
> wrote:
> >
> > ----- Original Message -----
> >> Hi,
> >>
> >> We have tried upgrading from 1.6.1.14 to 1.6.2.9 and have found
> that
> >> we are unable to URI dial our clients. We run a multi-tenant
> server
> >> and have set sip.conf to forward calls to a public context based
> on
> >> incoming domain name. This was all working before but not it is
> >> complaining of a loop back as the source and target server are the
> >> same.
> >>
> >> Any ideas on how to overcome this problem as we dial our clients
> based
> >> on their email address.
> >
> > Grabbing a SIP debug I see:
> >
> > <--- Transmitting (no NAT) to 10.172.120.5:5060 --->
> > SIP/2.0 100 Trying
> > Via: SIP/2.0/UDP
>
10.172.120.5:5060;branch=z9hG4bK-dadp6piblhin;received=10.172.120.5;rport=5060
> > From: "User A" <sip:usera at
172.30.14.8>;tag=c3zqlidz1u
> > To: <sip:userb at seconddomain.com>
> > Call-ID: 66b3314cc6d1-jxu0nhluv4zt
> > CSeq: 2 INVITE
> > Server: secret
> > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
> INFO
> > Supported: replaces, timer
> > Require: timer
> > Session-Expires: 1800;refresher=uas
> > Contact: <sip:userb at 172.30.14.8>
> > Content-Length: 0
> >
> > And am guessing that as the source from IP matches the Contact:
> address Asterisk sees that as a loop ?
> 
> I don't know these things, but you should probably post more of a SIP
> trace. Maybe turn on full sip debug to a file for long enough to see
> what the SIP conversation looks like that asterisk 1.6.2.9 is having
> with itself.
> 
> -- 
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