<div dir="ltr">Last thing to check, just for sanity's sake:<div><br></div><div style>t38pt_udptl=yes in sip.conf? It defaults to off.</div><div style><br></div><div style><br></div></div><div class="gmail_extra"><br><br>
<div class="gmail_quote">On Thu, Dec 27, 2012 at 12:32 PM, Eric Wieling <span dir="ltr"><<a href="mailto:EWieling@nyigc.com" target="_blank">EWieling@nyigc.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
It does not appear to make any difference. Calls are still failing.<br>
<div class="HOEnZb"><div class="h5"><br>
-----Original Message-----<br>
From: Christopher Harrington [mailto:<a href="mailto:chris@acsdi.com">chris@acsdi.com</a>]<br>
Sent: Thursday, December 27, 2012 1:20 PM<br>
To: Eric Wieling<br>
Cc: Asterisk Users Mailing List - Non-Commercial Discussion<br>
Subject: Re: [asterisk-users] $100 Bounty: Level3/Asterisk/Adtran T.38 Pass-through<br>
<br>
True, but it should bypass Asterisk when possible for SIP streams and may solve your problem.<br>
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On Thu, Dec 27, 2012 at 12:16 PM, Eric Wieling <<a href="mailto:EWieling@nyigc.com">EWieling@nyigc.com</a>> wrote:<br>
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<br>
We have directrtpsetup=no because the comments in the sample config indicates it does not work in all situations.<br>
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<br>
-----Original Message-----<br>
From: <a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a> [mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a>] On Behalf Of Christopher Harrington<br>
Sent: Thursday, December 27, 2012 1:13 PM<br>
To: Asterisk Users Mailing List - Non-Commercial Discussion<br>
Subject: Re: [asterisk-users] $100 Bounty: Level3/Asterisk/Adtran T.38 Pass-through<br>
<br>
directrtpsetup=yes in sip.conf?<br>
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<br>
On Thu, Dec 27, 2012 at 12:09 PM, Eric Wieling <<a href="mailto:EWieling@nyigc.com">EWieling@nyigc.com</a>> wrote:<br>
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<br>
We have set directmedia=yes as well as directmedia=no. There is no NAT involved.<br>
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<br>
<br>
-----Original Message-----<br>
From: <a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a> [mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a>] On Behalf Of Leandro Dardini<br>
Sent: Thursday, December 27, 2012 1:08 PM<br>
To: Asterisk Users Mailing List - Non-Commercial Discussion<br>
Subject: Re: [asterisk-users] $100 Bounty: Level3/Asterisk/Adtran T.38 Pass-through<br>
<br>
Have you configured the canreinvite=yes in sip peer?<br>
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I am currently off work for two days, but a 100% fail means a configuration problem for sure.<br>
<br>
<br>
Leandro<br>
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<br>
2012/12/27 Eric Wieling <<a href="mailto:EWieling@nyigc.com">EWieling@nyigc.com</a>><br>
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<br>
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.<br>
<br>
What we want to work:<br>
<br>
Level 3 T.38 TN <-> MSX/Nextone SBC <-> Asterisk 1.8.18.1 <-> Adtran NetVanta w/POTS and T.38 support.<br>
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When we replace Asterisk with Kamailio faxes work fine. When we put Asterisk there instead, then faxes fail nearly 100% of the time.<br>
<br>
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.<br>
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-Chris Harrington<br>
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ACSDi Office: <a href="tel:763.559.5800" value="+17635595800">763.559.5800</a><br>
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-Chris Harrington<br>
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ACSDi Office: <a href="tel:763.559.5800" value="+17635595800">763.559.5800</a><br>
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</div></div></blockquote></div><br><br clear="all"><div><br></div>-- <br>-Chris Harrington<br><div>ACSDi Office: 763.559.5800</div><div><div>Mobile Phone: 612.326.4248</div></div><div><br></div>
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