Yes, disabling reinvites solved the problem :)<div><br></div><div>Thanks.</div><div><br></div><div>--<br><div><div>Regards,</div><div>Shariq Khan</div><div>0333-3501125</div><br>
<br><br><div class="gmail_quote">On Fri, Apr 8, 2011 at 6:38 AM, Lyle Giese <span dir="ltr"><<a href="mailto:lyle@lcrcomputer.net">lyle@lcrcomputer.net</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
<div><div></div><div class="h5">On 04/07/11 03:00, Shariq Khan wrote:<br>
</div></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div></div><div class="h5">
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></div></div>
--<br>
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</blockquote>
<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>
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</blockquote></div><br></div></div>