Two situations:<br><br>Shoretel Phone --> ShoreTel --> PRI --> Asterisk --> Softswitch --> VoIP/PSTN -- This situation does NOT work. User hears audio, but Asterisk does not appear to process DTMF. Note, this is ONLY on IVR applications where we are not getting 200/connect passed through even though we're hearing IVR audio (early media)
<br><br>Second situation:<br><br>IP Phone --> Asterisk --> Softswitch --> VoIP/PSTN -- Works as expected. You can dial the IVR and still send DTMF.<br><br>It seems like there are some early media problems when routing traffic through the PRI.
<br><br><br><br><div><span class="gmail_quote">On 5/19/06, <b class="gmail_sendername">Alexander Lopez</b> <<a href="mailto:Alex.Lopez@opsys.com">Alex.Lopez@opsys.com</a>> wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
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<p><font color="navy" face="Arial" size="2"><span style="font-size: 10pt; font-family: Arial; color: navy;">Is it Phone -> ShoreTel -> Asterisk
-> PSTN ???</span></font></p>
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<p><b><font face="Tahoma" size="2"><span style="font-size: 10pt; font-family: Tahoma; font-weight: bold;">From:</span></font></b><font face="Tahoma" size="2"><span style="font-size: 10pt; font-family: Tahoma;">
<a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">asterisk-users-bounces@lists.digium.com</a>
[mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">asterisk-users-bounces@lists.digium.com</a>] <b><span style="font-weight: bold;">On Behalf Of
</span></b>Anthony Cennami<br>
<b><span style="font-weight: bold;">Sent:</span></b> Friday, May 19, 2006 1:47 PM<br>
<b><span style="font-weight: bold;">To:</span></b> Doug<br>
<b><span style="font-weight: bold;">Cc:</span></b> Asterisk
Users Mailing List - Non-Commercial Discussion<br>
<b><span style="font-weight: bold;">Subject:</span></b> Re: [Asterisk-Users] PRI
dialing IVR with inband DTMF</span></font></p>
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<p style="margin-bottom: 12pt;"><font face="Times New Roman" size="3"><span style="font-size: 12pt;">Well the communication
between the Asterisk and Shoretel is ISDN PRI. If a station on the
Shoretel calls a regular number (company auto-attendant, cellphone voicemail,
etc) and that number "ANSWERS" then there do not appear to be any
problems with DTMF. <br>
<br>
When a station dials an IVR which does not "ANSWER" but does Early
Media, that stations DTMF is not being received by the PRI. My
understanding is that this should typically be handled over the D channel, but
in a number of test calls I discovered that all DTMF is being sent Inband from
Shoretel over the PRI. <br>
<br>
<br>
</span></font></p>
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<p><span><font face="Times New Roman" size="3"><span style="font-size: 12pt;">On 5/19/06, <b><span style="font-weight: bold;">Doug</span></b>
<<a href="mailto:Doug@natel.net" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">Doug@natel.net</a>> wrote:</span></font></span></p>
<p style="margin-bottom: 12pt;"><font face="Times New Roman" size="3"><span style="font-size: 12pt;">At 12:06 5/19/2006,
Anthony Cennami wrote:<br>
>I have a client who is using a Shoretel PBX. This PBX apparently<br>
>does not send DTMF information OOB, but instead sends this inband<br>
>via the B-channel.<br>
> <br>
>This is traversing an Asterisk box via a PRI. The user calls the<br>
>IVR (1-800-CALL-DHL), receives audio, but is not able to present<br>
>DTMF to engage the IVR. With some light research it appears that<br>
>the DSP is not activating until the call is answered.<br>
><br>
>DTMF on the SIP side is set to RFC2833 -- calls all work fine when<br>
>originating from a SIP phone connected to the same device.<br>
><br>
>Any suggestions on what needs to be done to pre-emptively enable DSP <br>
>and or early media on the PRI (outbound)??<br>
><br>
>Thanks,<br>
><br>
>Anthony<br>
><br>
>---SIP---->Asterisk----PRI---->Shoretel<br>
<br>
Hey Anthony,<br>
<br>
I don't know if this will help you but, we had <br>
a hard time getting touchtones (DTMF) to work until<br>
we set both ends to "INFO" (sometimes called "SIP Info")<br>
<br>
RFC2833, Inband, Auto, etc. did not work.</span></font></p>
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-- <br>
Anthony D Cennami</span></font></p>
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http://lists.digium.com/mailman/listinfo/asterisk-users</a><br><br><br></blockquote></div><br><br clear="all"><br>-- <br>Anthony D Cennami<br>