[asterisk-users] $100 Bounty: Level3/Asterisk/Adtran T.38 Pass-through

[Christopher Harrington]

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 <EWieling at nyigc.com> wrote:

> It does not appear to make any difference.  Calls are still failing.
>
> -----Original Message-----
> From: Christopher Harrington [mailto:chris at 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 <EWieling at 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-bounces at lists.digium.com [mailto:
> asterisk-users-bounces at 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 <EWieling at nyigc.com>
> wrote:
>
>
>                 We have set directmedia=yes as well as directmedia=no.
>  There is no NAT involved.
>
>
>
>
>                 -----Original Message-----
>                 From: asterisk-users-bounces at lists.digium.com [mailto:
> asterisk-users-bounces at 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 <EWieling at 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.
>
>
>
>                         --
>
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>
>
>
>         --
>         -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
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