<div dir="rtl"><div class="gmail_extra"><div dir="ltr">Hi,</div><div dir="ltr"> </div><div dir="ltr"> Yes, you need two profiles - one for incoming and one for outgoing. I enclose here the relevant fragments from my old config:</div>
<div dir="ltr"> </div><div dir="ltr"><font face="courier new,monospace">voice service voip<br> qsig decode<br> dtmf-interworking rtp-nte</font> (RFC-2833)<br> <font face="courier new,monospace">srtp fallback<br> allow-connections sip to sip<br>
fax protocol t38 ls-redundancy 4 hs-redundancy 2 fallback cisco<br></font></div><div dir="ltr">Not sure this is relevant:</div><div dir="ltr"> </div><div dir="ltr"><font face="courier new,monospace">interface Serial0/0/0:15<br>
no ip address<br> encapsulation hdlc<br> isdn switch-type primary-qsig<br> isdn overlap-receiving<br> isdn not-end-to-end 64<br> isdn incoming-voice voice<br> isdn supp-service name calling profile Network-Extension operation-value-tag local<br>
isdn negotiate-bchan<br> isdn outgoing ie facility<br> isdn outgoing ie notify-indicator<br> isdn outgoing display-ie<br> isdn outgoing ie caller-number<br> isdn outgoing ie called-number<br> no cdp enable<br></font></div>
<div dir="ltr">This is for calls from PSTN to VoIP:</div><div dir="ltr"><br><font face="courier new,monospace">dial-peer voice 201 voip<br> destination-pattern 40.....<br> progress_ind setup enable 1<br> progress_ind progress enable 8<br>
session protocol sipv2<br> session target ipv4:<a href="http://10.1.1.1:5070">10.1.1.1:5070</a></font> Note the special port - it lands on a different profile<br> <font face="courier new,monospace">session transport udp<br>
dtmf-relay sip-notify</font> Use INFO/NOTIFY instead of RFC-2833<br> <font face="courier new,monospace">codec g711alaw<br> fax-relay sg3-to-g3<br> fax protocol t38 ls-redundancy 4 hs-redundancy 2 fallback cisco<br>
no vad<br></font><br>The VoIP to PSTN has nothing special:</div><div dir="ltr"><font face="courier new,monospace">dial-peer voice 1 pots<br> destination-pattern 8T<br> progress_ind setup enable 3<br> progress_ind progress enable 8<br>
no digit-strip<br> direct-inward-dial<br> port 0/0/0:15<br></font></div><div dir="ltr"> </div><div dir="ltr"> Regards, __Yehavi:</div><div dir="ltr"> </div><div dir="ltr" class="gmail_quote">2012/4/26 <span dir="ltr"><<a href="mailto:bernard.david.murphy@gmail.com" target="_blank">bernard.david.murphy@gmail.com</a>></span><br>
<blockquote style="margin:0px 0px 0px 0.8ex;padding-left:1ex;border-left-color:rgb(204,204,204);border-left-width:1px;border-left-style:solid" class="gmail_quote">To do that I'd have to have another sip profile for inbound call legs on a different port I suppose ?<br>
<div class="im HOEnZb"><br>
Sent from my BlackBerry® wireless device<br>
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-----Original Message-----<br>
From: Yehavi Bourvine <<a href="mailto:yehavi.bourvine@gmail.com">yehavi.bourvine@gmail.com</a>><br>
Sender: <a href="mailto:freeswitch-users-bounces@lists.freeswitch.org">freeswitch-users-bounces@lists.freeswitch.org</a><br>
</div><div class="HOEnZb"><div class="h5">Date: Thu, 26 Apr 2012 12:20:56<br>
To: FreeSWITCH Users Help<<a href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</a>><br>
Reply-To: FreeSWITCH Users Help <<a href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</a>><br>
Subject: Re: [Freeswitch-users] DTMF on BLeg not recognized by Cisco Gateway<br>
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