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<title>Re: [asterisk-users] Asterisk Avaya SIP Trunking One Way Audio</title>
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<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>First, I’m pretty sure avaya peer needs to friend. Try adding
the below to sip.conf and do a reload.<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>[general]<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>externip = the.wan.ext.ip<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>localnet = 192.168.1.0/255.255.255.0<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'> If that doesn’t work, add nat=yes to avaya peer=friend<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Skyler<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
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<p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span></b><span
style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'> asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] <b>On Behalf Of </b>Lyle Giese<br>
<b>Sent:</b> Thursday, April 07, 2011 6:39 PM<br>
<b>To:</b> asterisk-users@lists.digium.com<br>
<b>Subject:</b> Re: [asterisk-users] Asterisk Avaya SIP Trunking One Way Audio<o:p></o:p></span></p>
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<p class=MsoNormal><o:p> </o:p></p>
<p><span style='font-size:10.0pt'>On 04/07/11 03:00, Shariq Khan wrote:<br>
> I am facing one way audio problem in sip trunking between asterisk and<br>
> avaya.<br>
><br>
>
+-------------+ +----+<br>
>
| avaya sip |-------| P1 |<br>
>
+-------------+ +----+<br>
>
|<br>
>
|<br>
>
|<br>
>
+-------------+<br>
>
| Asterisk
|
WAN<br>
> -------------------------------------------------<br>
>
|
|
LAN<br>
>
+-------------+<br>
>
|<br>
>
/<br>
> +----+ /<br>
> | P2 |--+<br>
> +----+<br>
><br>
> When P1 dial P2, P2 hears voice clear but P1 could not hear any voice.<br>
><br>
> My sip.conf is<br>
><br>
> [avaya]<br>
> type=peer<br>
> fromdomain=xx.xx.xx.xx<br>
> host=xx.xx.xx.xx<br>
> disallow=all<br>
> allow=ulaw<br>
> dtmfmode=rfc2833<br>
> canreinvite=yes<br>
><br>
><br>
> --<br>
> Regards,<br>
> Shariq Khan<br>
> 0333-3501125<br>
><br>
><br>
><br>
> --<br>
> _____________________________________________________________________<br>
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<br>
Turn off reinvite on all extensions and SIP trunks involved and try again.<br>
<br>
Lyle Giese<br>
LCR Computer Services, Inc.<br>
<br>
--<br>
_____________________________________________________________________<br>
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<o:p></o:p></p>
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<p class=avgcert>No virus found in this message.<br>
Checked by AVG - <a href="http://www.avg.com">www.avg.com</a><br>
Version: 10.0.1204 / Virus Database: 1498/3523 - Release Date: 03/22/11<br>
Internal Virus Database is out of date.<o:p></o:p></p>
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