Enable dtmf on the logger.conf and see if you get some # or ** or whatever key you have configured at features.conf for transfer, maybe you could see something into the logs. I get something similar with some Linksys PAP2.<br>
<br>Adria Vidal<br><br><div class="gmail_quote">On Tue, Apr 1, 2008 at 10:15 PM, Tim Nelson <<a href="mailto:tnelson@rockbochs.com">tnelson@rockbochs.com</a>> wrote:<br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
Hello! I'm having a bit of an issue with one of my installations that I cannot figure out. For some reason, when two people are in a call (both local to the * box, same subnet, pure SIP), the call will randomly be placed on hold and provide MOH to the other party. We're using Polycom IP430 handsets almost exclusively for this installation. Can anyone think of a reason why a call would randomly go on hold?<br>
<br>
Tim Nelson<br>
Systems/Network Support<br>
Rockbochs Inc.<br>
<br>
<br>
_______________________________________________<br>
-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>
<br>
asterisk-users mailing list<br>
To UNSUBSCRIBE or update options visit:<br>
<a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br>
</blockquote></div><br><br clear="all"><br>-- <br>--<br>AdriĆ Vidal<br><a href="mailto:adriavidal@gmail.com">adriavidal@gmail.com</a>