The fax is originated from a fax machine connected to an ata which supports t38.
On Wed, Sep 24, 2008 at 11:54 PM, C F <[EMAIL PROTECTED]> wrote: > On Wed, Sep 24, 2008 at 5:43 AM, Rizwan Hisham <[EMAIL PROTECTED]> > wrote: > > Hi all, > > Sorry to interrupt. I need some help regarding fax passthru mode. > > > > We are trying to configure fax passthru mode in asterisk using sip. For > out > > of network calls/fax we use trunk configuration. i am using asterisk > 1.4.2. > > The user has to use fax machine connected to their ata and dial the > callee > > number, the call is originated just like a regular voice call. have not > > defined any special context for sending faxes. Have enabled t38 and > > canreinvite in peer/user and trunk configuration. But the fax is not > going > > thru. Our service provider does support fax passthru. Following is the > trunk > > and user/peer configuration: > > They support passthru, and the originating send fax is what? PSTN? or > VoIP ATA with t38 support? > There has to one that does the t38, if the point where it gets > converted to VoIP does not support t38 then passthru will not help > you. > > > > > TRUNK CONF > > [TRUNK-OUT] > > type=peer > > host=XXX > > port=5060 > > context=default > > country=us > > dtmfmode=rfc2833 > > restrictcid=no > > canreinvite=yes > > insecure=no > > disallow=all > > allow=ulaw > > allow=alaw > > allow=g729 > > allow=gsm > > promiscredir=yes > > t38_udptl=yes > > > > USER/PEER > > > > [abc] > > username=abc > > type=friend > > secret=123 > > qualify=25000 > > nat=yes > > mailbox=12129339037 > > insecure=port,invite > > incominglimit=2 > > outgoinglimit=2 > > intl_trunk=TRUNK-OUT > > local_trunk=TRUNK-OUT > > host=dynamic > > dtmfmode=inband > > context=uscan > > canreinvite=yes > > callerid="Rizwan Qureshi" <12222222222> > > accountcode=1:0:abc > > amaflags=default > > disallow=all > > allow=ulaw > > allow=alaw > > allow=gsm > > t38_udptl=yes > > > > > > Any solutions? > > > > On Tue, Sep 23, 2008 at 10:21 PM, Andrew Joakimsen <[EMAIL PROTECTED]> > > wrote: > >> > >> On Tue, Sep 23, 2008 at 10:02 AM, Steve Totaro > >> <[EMAIL PROTECTED]> wrote: > >> > ATAs work OK I guess, just make sure to use a loss less codec such as > >> > ULAW. > >> > >> Since the OP stated he is using E1 lines then he should probably be > >> using alaw instead. > >> > >> _______________________________________________ > >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > >> > >> AstriCon 2008 - September 22 - 25 Phoenix, Arizona > >> Register Now: http://www.astricon.net > >> > >> asterisk-users mailing list > >> To UNSUBSCRIBE or update options visit: > >> http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > > -- > > Best Regards > > Rizwan Hisham > > > > > > _______________________________________________ > > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > > > AstriCon 2008 - September 22 - 25 Phoenix, Arizona > > Register Now: http://www.astricon.net > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2008 - September 22 - 25 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- Best Regards Rizwan Hisham
_______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
