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<p>I did turn on debug and made inbound call and see that gateway is sending 3 invite but not getting any response. </p>
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<p>The other question is, has anyone configured any SIP FAX as 3rd party sip device in CUCM? Is that even capable.</p>
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<p>thanks</p>
<p>Hamu <br>
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<div id="divRplyFwdMsg" dir="ltr"><font color="#000000" face="Calibri, sans-serif" style="font-size: 11pt;"><b>From:</b> Sreekanth <sknth.n@gmail.com><br>
<b>Sent:</b> Wednesday, November 1, 2017 3:01 AM<br>
<b>To:</b> Hamu Ebiso<br>
<b>Cc:</b> cisco-voip voyp list<br>
<b>Subject:</b> Re: [cisco-voip] Need help with SIP FAX Configuration.</font>
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<div>Check if the call is being answered first. The first step of a fax call is the voice call getting established, and then getting escalated to a fax call.
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So usual troubleshooting for a voice call to see why the call is not answered.<br>
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The next step is to check what protocol of fax you are using and check if that is configured correctly end-to-end.<br>
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<div class="gmail_quote">On 1 November 2017 at 02:11, Hamu Ebiso <span dir="ltr">
<<a href="mailto:hebiso2010@hotmail.com" target="_blank">hebiso2010@hotmail.com</a>></span> wrote:<br>
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<p>Hi Everyone,</p>
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<p>Has any body experiencing configuring <span>RICOH MP C5503 for SIP FAX on Either CUCM as 3th party SIP or on voice gateway?</span></p>
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<p>11.0 CUCM</p>
<p><span>SIP FAX: RICOH MP C5503 </span></p>
<p>Voice router 4431.</p>
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<p>I have configured Route Pattern in CUCM and pointed the router group and then to the voice gateway.</p>
<p>I configured Dial peer in voice router but when I dial the FAX is not picking up. When I try to dial out, nothing is happening as well.</p>
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<p>Dial Peer as shown below:</p>
<p></p>
<div>dial-peer voice 500 voip<br>
description Ricoh 072<br>
destination-pattern 10 digit #<br>
session protocol sipv2<br>
session target ipv4:x.x.x.x<br>
voice-class codec 1<br>
dtmf-relay rtp-nte<br>
no vad<br>
!<br>
dial-peer voice 501 voip<br>
description Ricoh 073<br>
destination-pattern 10 digit #<br>
session protocol sipv2<br>
session target ipv4:x.x.x.x<br>
voice-class codec 1<br>
dtmf-relay rtp-nte</div>
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<div>I'm just wondering if I'm missing any configuration.<br>
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<p><span></span></p>
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<p>if you can point to the any document that I can follow to make sure everything is setup correctly, that would be appreciated.</p>
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