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

[Eric Wieling]

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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