<html><head></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; color: rgb(0, 0, 0); font-size: 14px; font-family: Calibri, sans-serif; "><div>Do you have debugs from the CUBE? Are MTPs involved in the call flow?</div><div><br></div><span id="OLK_SRC_BODY_SECTION"><div style="font-family:Calibri; font-size:11pt; text-align:left; color:black; BORDER-BOTTOM: medium none; BORDER-LEFT: medium none; PADDING-BOTTOM: 0in; PADDING-LEFT: 0in; PADDING-RIGHT: 0in; BORDER-TOP: #b5c4df 1pt solid; BORDER-RIGHT: medium none; PADDING-TOP: 3pt"><span style="font-weight:bold">From: </span> Lelio Fulgenzi <<a href="mailto:lelio@uoguelph.ca">lelio@uoguelph.ca</a>><br><span style="font-weight:bold">Date: </span> Monday 30 July 2012 10:22 PM<br><span style="font-weight:bold">To: </span> Robert Kulagowski <<a href="mailto:rkulagow@gmail.com">rkulagow@gmail.com</a>><br><span style="font-weight:bold">Cc: </span> Cisco VOIP <<a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a>><br><span style="font-weight:bold">Subject: </span> Re: [cisco-voip] SIP trunk fax troubleshooting<br></div><div><br></div><div><style type="text/css">p { margin: 0; }</style><div><div style="font-family: Verdana; font-size: 10pt; color: #000000">I recall being at a Live session whereby they said just that, if you're still doing FAXing without some sort of FAX server, keep using a PRI for that. By "they", I mean David Hanes, Gonzalo Salgueiro, the writers of the <a href="http://www.ciscopress.com/bookstore/product.asp?isbn=1587052695">Modem/FAX book</a> who were doing the session. Or at least I'm pretty sure about that anyways.<span><br><br>It's one of the (many) reasons I've steered away from SIP for now.<br><br><span name="x"></span>---<br>Lelio Fulgenzi, B.A.<br>Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1<br>(519) 824-4120 x56354 (519) 767-1060 FAX (ANNU)<br>^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^<br>Cooking with unix is easy. You just sed it and forget it. <br> - LFJ (with apologies to Mr. Popeil)<br><span name="x"></span><br></span><br><hr id="zwchr"><b>From: </b>"Robert Kulagowski" <<a href="mailto:rkulagow@gmail.com">rkulagow@gmail.com</a>><br><b>To: </b>"Cisco VOIP" <<a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a>><br><b>Sent: </b>Monday, July 30, 2012 12:43:18 PM<br><b>Subject: </b>[cisco-voip] SIP trunk fax troubleshooting<br><br>Outbound faxes aren't working.<br><br>Fax machine is connected to a VGC -> CUCM -> SIP trunk -> CUBE -> bandwidth.com<br><br>Cisco IOS Software, C2900 Software (C2900-UNIVERSALK9-M), Version<br>15.1(4)M2, RELEASE SOFTWARE (fc1)<br>Technical Support: <a href="http://www.cisco.com/techsupport">http://www.cisco.com/techsupport</a><br>Copyright (c) 1986-2011 by Cisco Systems, Inc.<br>Compiled Mon 26-Sep-11 17:37 by prod_rel_team<br><br>ROM: System Bootstrap, Version 15.0(1r)M12, RELEASE SOFTWARE (fc1)<br><br>voice service voip<br> ip address trusted list<br> ipv4 216.82.224.202<br> ipv4 216.82.225.202<br> ipv4 10.255.23.7<br> ipv4 10.255.1.21<br> ipv4 10.255.29.20<br> address-hiding<br> mode border-element<br> allow-connections sip to sip<br> no supplementary-service sip moved-temporarily<br> no supplementary-service sip refer<br> fax protocol pass-through g711ulaw<br> sip<br> header-passing<br> error-passthru<br> no update-callerid<br> early-offer forced<br> midcall-signaling passthru<br> privacy-policy passthru<br> g729 annexb-all<br><br>dial-peer voice 10000 voip<br> description Outbound SIP to Bandwidth.com<br> translation-profile outgoing SIP-OUT<br> destination-pattern 8T<br> session protocol sipv2<br> session target ipv4:216.82.224.202<br> voice-class sip dtmf-relay force rtp-nte<br> dtmf-relay rtp-nte<br> codec g711ulaw<br> fax-relay ecm disable<br> ip qos dscp cs5 media<br> ip qos dscp cs4 signaling<br> clid restrict<br> no vad<br><br>The CUBE book makes it seem like it should "just work".<br><br>I've also tried:<br><br>dial-peer voice 10000 voip<br> description Outbound SIP to Bandwidth.com<br> translation-profile outgoing SIP-OUT<br> preference 1<br> destination-pattern 8T<br> session protocol sipv2<br> session target ipv4:216.82.224.202<br> voice-class sip dtmf-relay force rtp-nte<br> dtmf-relay rtp-nte<br> codec g711ulaw<br> fax-relay ecm disable<br> fax rate 9600<br> fax protocol pass-through g711ulaw<br> ip qos dscp cs5 media<br> ip qos dscp cs4 signaling<br> clid restrict<br> no vad<br><br>No joy there either.<br><br>Seems like fax and modem are perpetual issues with SIP trunking from<br>what I can see in the archives? The easiest would be to just continue<br>to send fax to PRI / analogs, but where's the fun in that?<br>_______________________________________________<br>cisco-voip mailing list<br><a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a><br><a href="https://puck.nether.net/mailman/listinfo/cisco-voip">https://puck.nether.net/mailman/listinfo/cisco-voip</a><br></div></div></div>_______________________________________________
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