<html><head></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; ">A couple of us have asked this guy to post relevant configs so we can help him but so far nothing has been posted.<div><br></div><div><br><div><div>On Nov 12, 2010, at 8:00 AM, Ryan Ratliff wrote:</div><br class="Apple-interchange-newline"><blockquote type="cite"><div style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; ">I see lots of discussion around config and call flow but have you actually done any troubleshooting? One way audio most of the time comes down to the simple fact that one party is not receiving RTP from the other. For these one-way audio calls you need to determine what IP addresses are involved. Next verify this in the signaling via the SDPs in the SIP messages. You can then use show commands on the router to confirm where it thinks it should be sending and receiving RTP to/from and if in fact packet counters are incrementing.<div><br></div><div>If no MTP is being used for the call, try forcing it to use one and see if that fixes the issue.</div><div>If you are using media flow-through (default) does changing it to flow-around fix the issue? </div><div><br><div>
<span class="Apple-style-span" style="border-collapse: separate; font-family: Helvetica; font-size: medium; font-style: normal; font-variant: normal; font-weight: normal; letter-spacing: normal; line-height: normal; orphans: 2; text-indent: 0px; text-transform: none; white-space: normal; widows: 2; word-spacing: 0px; -webkit-border-horizontal-spacing: 0px; -webkit-border-vertical-spacing: 0px; -webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: auto; -webkit-text-stroke-width: 0px; "><div>-Ryan</div></span>
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<br><div><div>On Nov 12, 2010, at 8:09 AM, Bill Riley wrote:</div><br class="Apple-interchange-newline"><span class="Apple-style-span" style="border-collapse: separate; font-family: 'Lucida Grande'; font-style: normal; font-variant: normal; font-weight: normal; letter-spacing: normal; line-height: normal; orphans: 2; text-indent: 0px; text-transform: none; white-space: normal; widows: 2; word-spacing: 0px; -webkit-border-horizontal-spacing: 0px; -webkit-border-vertical-spacing: 0px; -webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: auto; -webkit-text-stroke-width: 0px; font-size: medium; "><div lang="EN-US" link="blue" vlink="purple"><div class="WordSection1" style="page: WordSection1; "><div style="margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 11pt; font-family: Calibri, sans-serif; "><span style="color: rgb(31, 73, 125); ">I shouldn’t need an MTP for this connection. All SIP traffic is sourced from one interface. I do have SCCP traffic sourced from a different interface but it is only used for a conference bridge, not MTP.<o:p></o:p></span></div><div style="margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 11pt; font-family: Calibri, sans-serif; "><span style="color: rgb(31, 73, 125); "><o:p> </o:p></span></div><div><div style="border-right-style: none; border-bottom-style: none; border-left-style: none; border-width: initial; border-color: initial; border-top-style: solid; border-top-color: rgb(181, 196, 223); border-top-width: 1pt; padding-top: 3pt; padding-right: 0in; padding-bottom: 0in; padding-left: 0in; "><div style="margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 11pt; font-family: Calibri, sans-serif; "><b><span style="font-size: 10pt; font-family: Tahoma, sans-serif; ">From:</span></b><span style="font-size: 10pt; font-family: Tahoma, sans-serif; "><span class="Apple-converted-space"> </span>Cheng, Karen [mailto:Karen.Cheng@racq.com.au]<span class="Apple-converted-space"> </span><br><b>Sent:</b><span class="Apple-converted-space"> </span>Thursday, November 11, 2010 9:14 PM<br><b>To:</b><span class="Apple-converted-space"> </span>'Bill Riley'<br><b>Subject:</b><span class="Apple-converted-space"> </span>RE: [cisco-voip] SIP trunk one way audio<o:p></o:p></span></div></div></div><div style="margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 11pt; font-family: Calibri, sans-serif; "><o:p> </o:p></div><div style="margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 11pt; font-family: Calibri, sans-serif; "><span lang="EN-AU" style="color: rgb(31, 73, 125); ">Not sure if you have checked already but is your SIP trunk using one interface and your MTP/SCCP interface a different interface?<o:p></o:p></span></div><div style="margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 11pt; font-family: Calibri, sans-serif; "><span lang="EN-AU" style="color: rgb(31, 73, 125); "><o:p> </o:p></span></div><div style="margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 11pt; font-family: Calibri, sans-serif; "><span lang="EN-AU" style="color: rgb(31, 73, 125); ">I had one-way audio/no audio problems ages back due to this because our integrator had configured the SIP trunk to point to int gi0/0’s IP and then configured the SCCP interface to loopback0.<o:p></o:p></span></div><div style="margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 11pt; font-family: Calibri, sans-serif; "><span lang="EN-AU" style="color: rgb(31, 73, 125); "><o:p> </o:p></span></div><div style="margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 11pt; font-family: Calibri, sans-serif; "><span lang="EN-AU" style="color: rgb(31, 73, 125); ">Regards<o:p></o:p></span></div><div style="margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 11pt; font-family: Calibri, sans-serif; "><span lang="EN-AU" style="color: rgb(31, 73, 125); "><o:p> </o:p></span></div><div><div><div style="margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 11pt; font-family: Calibri, sans-serif; "><b><i><span lang="EN-AU" style="font-size: 10pt; font-family: Arial, sans-serif; color: rgb(0, 52, 120); ">Karen Cheng</span></i></b><span lang="EN-AU" style="font-size: 12pt; font-family: 'Times New Roman', serif; color: rgb(31, 73, 125); "><o:p></o:p></span></div></div><div><div style="margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 11pt; font-family: Calibri, sans-serif; "><i><span lang="EN-AU" style="font-size: 10pt; font-family: Arial, sans-serif; color: rgb(0, 52, 120); ">Voice Network Engineer</span></i><span lang="EN-AU" style="font-size: 12pt; font-family: 'Times New Roman', serif; color: rgb(31, 73, 125); "><o:p></o:p></span></div></div><div><div style="margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 11pt; font-family: Calibri, sans-serif; "><span lang="EN-AU" style="font-size: 12pt; font-family: 'Times New Roman', serif; color: rgb(31, 73, 125); "> <o:p></o:p></span></div></div><div><div><div style="margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 11pt; font-family: Calibri, sans-serif; "><span lang="EN-AU" style="font-size: 12pt; font-family: 'Times New Roman', serif; color: rgb(31, 73, 125); "> <o:p></o:p></span></div></div></div></div><div style="margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 11pt; font-family: Calibri, sans-serif; "><span lang="EN-AU" style="font-size: 12pt; font-family: 'Times New Roman', serif; color: rgb(31, 73, 125); "> </span><span lang="EN-AU" style="color: rgb(31, 73, 125); "><o:p></o:p></span></div><div><div style="border-right-style: none; border-bottom-style: none; border-left-style: none; border-width: initial; border-color: initial; border-top-style: solid; border-top-color: rgb(181, 196, 223); border-top-width: 1pt; padding-top: 3pt; padding-right: 0in; padding-bottom: 0in; padding-left: 0in; "><div style="margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 11pt; font-family: Calibri, sans-serif; "><b><span style="font-size: 10pt; font-family: Tahoma, sans-serif; ">From:</span></b><span style="font-size: 10pt; font-family: Tahoma, sans-serif; "><span class="Apple-converted-space"> </span><a href="mailto:cisco-voip-bounces@puck.nether.net" style="color: blue; text-decoration: underline; ">cisco-voip-bounces@puck.nether.net</a><span class="Apple-converted-space"> </span>[mailto:cisco-voip-bounces@puck.nether.net]<span class="Apple-converted-space"> </span><b>On Behalf Of<span class="Apple-converted-space"> </span></b>Bill Riley<br><b>Sent:</b><span class="Apple-converted-space"> </span>Friday, 12 November 2010 2:15 AM<br><b>To:</b><span class="Apple-converted-space"> </span><a href="mailto:cisco-voip@puck.nether.net" style="color: blue; text-decoration: underline; ">cisco-voip@puck.nether.net</a><br><b>Subject:</b><span class="Apple-converted-space"> </span>[cisco-voip] SIP trunk one way audio<o:p></o:p></span></div></div></div><div style="margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 11pt; font-family: Calibri, sans-serif; "><span lang="EN-AU"><o:p> </o:p></span></div><div style="margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 11pt; font-family: Calibri, sans-serif; ">I have a new SIP trunk terminating in a 2921 CUBE bundle. When I call in from the Trunk directly to an IP phone it works correctly. If I call from the trunk to IP phone and the IP phone transfers the call without waiting for the remote party to answer I get one way audio. From reading this looks like and MTP issue but I have an MTP set in the MRGL for the trunk.<o:p></o:p></div><div style="margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 11pt; font-family: Calibri, sans-serif; "><span lang="EN-AU" style="font-size: 12pt; font-family: 'Times New Roman', serif; "><br></span><span lang="EN-AU" style="font-size: 7.5pt; font-family: Verdana, sans-serif; color: green; ">75% of inspected vehicle have a defect*. Call 13 1905 and book an RACQ Vehicle Inspection today. *Based on the RACQ Vehicle Defect Report January 2008<br><br></span><span lang="EN-AU" style="font-size: 12pt; font-family: 'Times New Roman', serif; "><br></span><span lang="EN-AU" style="font-size: 10pt; font-family: Arial, sans-serif; color: gray; ">Please Note: If you are not the intended recipient, please delete this email as its use is prohibited. RACQ does not warrant or represent that this email is free from viruses or defects. If you do not wish to receive any further commercial electronic messages from RACQ please e-mail<span class="Apple-converted-space"> </span><a href="mailto:unsubscribe@racq.com.au" style="color: blue; text-decoration: underline; ">unsubscribe@racq.com.au</a><span class="Apple-converted-space"> </span>or contact RACQ on 13 19 05.</span><span lang="EN-AU" style="font-size: 12pt; font-family: 'Times New Roman', serif; "><o:p></o:p></span></div></div>_______________________________________________<br>cisco-voip mailing list<br><a href="mailto:cisco-voip@puck.nether.net" style="color: blue; text-decoration: underline; ">cisco-voip@puck.nether.net</a><br><a href="https://puck.nether.net/mailman/listinfo/cisco-voip" style="color: blue; text-decoration: underline; ">https://puck.nether.net/mailman/listinfo/cisco-voip</a><br></div></span></div><br></div></div>_______________________________________________<br>cisco-voip mailing list<br><a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a><br>https://puck.nether.net/mailman/listinfo/cisco-voip<br></blockquote></div><br></div></body></html>