True, but it should bypass Asterisk when possible for SIP streams and may solve your problem.
On Thu, Dec 27, 2012 at 12:16 PM, Eric Wieling <[email protected]> wrote: > We have directrtpsetup=no because the comments in the sample config > indicates it does not work in all situations. > > -----Original Message----- > From: [email protected] [mailto: > [email protected]] On Behalf Of Christopher > Harrington > Sent: Thursday, December 27, 2012 1:13 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] $100 Bounty: Level3/Asterisk/Adtran T.38 > Pass-through > > directrtpsetup=yes in sip.conf? > > > > On Thu, Dec 27, 2012 at 12:09 PM, Eric Wieling <[email protected]> wrote: > > > We have set directmedia=yes as well as directmedia=no. There is > no NAT involved. > > > > > -----Original Message----- > From: [email protected] [mailto: > [email protected]] On Behalf Of Leandro Dardini > Sent: Thursday, December 27, 2012 1:08 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] $100 Bounty: Level3/Asterisk/Adtran > T.38 Pass-through > > Have you configured the canreinvite=yes in sip peer? > > I am currently off work for two days, but a 100% fail means a > configuration problem for sure. > > > Leandro > > > 2012/12/27 Eric Wieling <[email protected]> > > > We are offering $100 (paid via paypal or check) to the > first person who assists us in successfully sending and receiving faxes in > the setup described below. Offer expires Dec 31. We are a direct customer > of Level 3, there is no other carrier involved. > > What we want to work: > > Level 3 T.38 TN <-> MSX/Nextone SBC <-> Asterisk > 1.8.18.1 <-> Adtran NetVanta w/POTS and T.38 support. > > When we replace Asterisk with Kamailio faxes work fine. > When we put Asterisk there instead, then faxes fail nearly 100% of the > time. > > I see the switch to T.38 in the Adtran debug logs. We > can originate and terminate T.38 calls in Asterisk using SendFax/ReceiveFax > using T.38 so I'm assuming I have my udptl.conf and sip.conf settings > correct. > > > > -- > > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by > http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar > every Thurs: > http://www.asterisk.org/hello > > 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-- > New to Asterisk? Join us for a live introductory webinar every > Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > -- > -Chris Harrington > > ACSDi Office: 763.559.5800 > Mobile Phone: 612.326.4248 > > -- -Chris Harrington ACSDi Office: 763.559.5800 Mobile Phone: 612.326.4248
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
