[asterisk-users] T.38 SIP Issues

[Ricardo Carvalho]

I made some tests with FAX in Asterisk 1.4 using T.38 between two ATAs
connected to legacy FAX machines, and realized that only SIP can make
passthrough in the server while RTP go direct between endpoints. Is it
possible for RTP data stream also to make passthrough in Asterisk?

Thanks,
Ricardo Carvalho.




On Fri, Mar 14, 2008 at 1:13 PM, Steve Underwood <steveu at coppice.org> wrote:

> Mindaugas Kezys wrote:
> > Hello,
> >
> > Higher speeds then 9600kbps are not permited by patents.
> >
> Would you care to name one that prevents 14,400?
> > Regards,
> > Mindaugas Kezys
> > http://www.kolmisoft.com
> > MOR PRO - Advanced Billing Solution for Asterisk PBX
> >
> >
> > -----Original Message-----
> > From: asterisk-users-bounces at lists.digium.com
> > [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Andreas
> van
> > dem Helge
> > Sent: Friday, March 14, 2008 3:28 AM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: Re: [asterisk-users] T.38 SIP Issues
> >
> > Has someone submitted a bugreport regarding enabling > 9600kbps fax? I
> > always wonder why it would never negociate 14400kbps... when it did
> > work a single page on fine resolution would take 4 minutes.
> >
> > Thank you very much for that link. I knew there had to be more
> > possible configurations for T.38. I will give it a try... but I think
> > I can get away without patching chan_sip.c, no? that just seems to
> > enable higher bitrates.
> >
> > And Linksys SPA2102 is one of the exact devices I have in my lab.
> >
> > On Thu, Mar 13, 2008 at 1:52 PM, Mindaugas Kezys <mkezys at gmail.com>
> wrote:
> >
> >> Hello,
> >>
> >>  This can help:
> >>
> > http://80.86.84.71/kolmiwiki/index.php/Send_Receive_Fax-T38
> >
> >>  Regards,
> >>  Mindaugas Kezys
> >>  http://www.kolmisoft.com
> >>  MOR PRO - Advanced Billing for Asterisk PBX
> >>
> >>
> >>
> >>
> >>  -----Original Message-----
> >>  From: asterisk-users-bounces at lists.digium.com
> >>  [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Andreas
> van
> >>  dem Helge
> >>  Sent: Thursday, March 13, 2008 5:16 AM
> >>  To: Asterisk Users Mailing List - Non-Commercial Discussion
> >>  Subject: [asterisk-users] T.38 SIP Issues
> >>
> >>  Is there any trick to getting T.38 fax to work with SIP? I had it
> >>  working and one day with no changes *poof* it stopped working and
> >>  hasn't worked for months. The only common factor is Asterisk 1.4.x
> >>  (always try to use the latest version) and NAT.
> >>
> >>  I've tried:
> >>
> >>  -Linksys ATA
> >>  -Grandstream ATA
> >>  -Audicodes ATA
> >>
> >>  All do the same thing. Call connects, hear the first 2sec of fax tone
> >>  and then just silence, but the call usually stays open.
> >>
> >>  I've tried two T.38-capable providers.
> >>
> >>  I've tried two different routers:
> >>  -Linksys WRT54GS running DD-WRT (Linux)
> >>  -Dell Optiplex 170L running PFSense (BSD)
> >>
> >>  Different Linux distros on the servers:
> >>  -SuSE 64bit
> >>  -RHEL 32bit
> >>  -SuSE 32bit
> >>
> >>  Is there any magic to get this to work? As far as I can tell the only
> >>  possible config option is "t38pt_udptl = yes" which I have set under
> >>  [general] & the peer.
> >>
> Steve
>
>
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