We are using t38pt_udptl=yes,redundancy,maxdatagram=400 Without the maxdatagram we get errors in the CLI. We also tried using FEC instead of redundancy, but no difference.
-----Original Message----- From: Christopher Harrington [mailto:ch...@acsdi.com] Sent: Thursday, December 27, 2012 2:23 PM To: Eric Wieling Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] $100 Bounty: Level3/Asterisk/Adtran T.38 Pass-through Last thing to check, just for sanity's sake: t38pt_udptl=yes in sip.conf? It defaults to off. On Thu, Dec 27, 2012 at 12:32 PM, Eric Wieling <ewiel...@nyigc.com> wrote: It does not appear to make any difference. Calls are still failing. -----Original Message----- From: Christopher Harrington [mailto:ch...@acsdi.com] Sent: Thursday, December 27, 2012 1:20 PM To: Eric Wieling Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] $100 Bounty: Level3/Asterisk/Adtran T.38 Pass-through 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 <ewiel...@nyigc.com> wrote: We have directrtpsetup=no because the comments in the sample config indicates it does not work in all situations. -----Original Message----- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] 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 <ewiel...@nyigc.com> wrote: We have set directmedia=yes as well as directmedia=no. There is no NAT involved. -----Original Message----- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] 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 <ewiel...@nyigc.com> 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 -- -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
