We dont have any Digium cards, we just have a GrandStream FXS 8-port device with 2 analog phones and one Grand stream FXO 8-port device with one POTS line and both are connected to the netgear switch....very rarely the analog phones are used and its very rare that calls are made through POTS using FXO.............we get this DTMF problem with the SIP phones(when called out or when we receive a call randomly)......I will try to capture the dtmf from the asterisk console with higher verbosity mode and also set the relaxeddtmf parameter.....<br>
<br>Thanks<br>Sandesh<br><br><div class="gmail_quote">On Thu, Jul 22, 2010 at 3:48 AM, Benny Amorsen <span dir="ltr"><<a href="mailto:benny%2Busenet@amorsen.dk">benny+usenet@amorsen.dk</a>></span> wrote:<br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
I would appreciate it if you didn't top-post.<br>
<div class="im"><br>
das sandesh <<a href="mailto:sandesh440@gmail.com">sandesh440@gmail.com</a>> writes:<br>
<br>
</div><div class="im">> Hi Benny...<br>
><br>
> DTMF tones are heard at the SIP phones side and not the other<br>
> party.......'server side' means from the Asterisk side.....from the<br>
> wireshark captures it appeards that the dtmf digits were sent from the<br>
> asterisk server ip to the phone ip randomly through Cisco(just passes the<br>
> SIP packt) inbetween the conversation.......<br>
<br>
</div>How do you interface with the PSTN? A Digium card?<br>
<br>
Either way you may want relaxdtmf=no in dahdi.conf if you don't have<br>
that already.<br>
<br>
You can see the DTMF happening on the Asterisk console if you set<br>
verbosity high enough.<br>
<font color="#888888"><br>
<br>
/Benny<br>
<br>
</font></blockquote></div><br>