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<p class="MsoNormal"><span lang="EN-US" style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D">Thanks you fpr yours reply, Let me check my config.<o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-US" style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D"><o:p> </o:p></span></p>
<p class="MsoNormal"><span lang="EN-US" style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D">Best Regards.<o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-US" style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D"><o:p> </o:p></span></p>
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<p class="MsoNormal"><b><span lang="ES" style="font-size:10.0pt;font-family:"Tahoma","sans-serif"">De:</span></b><span lang="ES" style="font-size:10.0pt;font-family:"Tahoma","sans-serif""> Mark Holloway [mailto:mh@markholloway.com]
<br>
<b>Enviado el:</b> viernes, 21 de octubre de 2011 10:26 a.m.<br>
<b>Para:</b> shary shary<br>
<b>CC:</b> Santiago Figueroa; cisco-voip-bounces@puck.nether.net; cisco-voip@puck.nether.net<br>
<b>Asunto:</b> Re: [cisco-voip] SIP signal<o:p></o:p></span></p>
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<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal">The 180 Ringing should definitely trigger the ring back to the PSTN. You may want to debug ISDN as well and see what's happening. The SIP part is correct. <o:p></o:p></p>
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<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal">On Oct 18, 2011, at 3:34 AM, shary shary wrote:<o:p></o:p></p>
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<br>
<o:p></o:p></p>
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<p class="MsoNormal"><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif"">can ou share your config, did u use rel100xx in SIP bind voice service ?<br>
<br>
</span><span style="font-size:10.0pt;font-family:"Lucida Handwriting";color:blue">Regards,</span><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif""><br>
</span><span style="font-size:10.0pt;font-family:"Lucida Handwriting";color:blue">Sheeraz Saeed lodhi.</span><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif""><br>
<br>
<o:p></o:p></span></p>
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<p class="MsoNormal"><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif"">> From:<span class="apple-converted-space"> </span><a href="mailto:sfigueroa@mnet.com.mx">sfigueroa@mnet.com.mx</a><br>
> To:<span class="apple-converted-space"> </span><a href="mailto:cisco-voip-bounces@puck.nether.net">cisco-voip-bounces@puck.nether.net</a>;<span class="apple-converted-space"> </span><a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a><br>
> Date: Mon, 17 Oct 2011 23:39:52 -0500<br>
> Subject: [cisco-voip] SIP signal<br>
><span class="apple-converted-space"> </span><br>
> Hello experts<br>
><span class="apple-converted-space"> </span><br>
><span class="apple-converted-space"> </span><br>
> I have a question with SIP to do work ringback, I have this step of signal SIP but do not if this is the instruction to PSTN hear ringback? My problem is that PSTN do not hear ringback.<br>
</span><span lang="EN-US" style="font-size:10.0pt;font-family:"Tahoma","sans-serif"">><span class="apple-converted-space"> </span><br>
><span class="apple-converted-space"> </span><br>
><span class="apple-converted-space"> </span><br>
> SIP/2.0 180 Ringing<br>
> Via: SIP/2.0/UDP 207.249.152.70:5060;branch=z9hG4bK*002e-00005424-07c1,SIP/2.0/UDP 10.15.15.7:5060;branch=z9hG4bK45F12EA<br>
> From: <</span><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif""><a href="sip:8181113405@10.15.15.7"><span lang="EN-US">sip:8181113405@10.15.15.7</span></a></span><span lang="EN-US" style="font-size:10.0pt;font-family:"Tahoma","sans-serif"">>;tag=9D58538-D02<br>
> To: <</span><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif""><a href="sip:335518400@207.249.152.75"><span lang="EN-US">sip:335518400@207.249.152.75</span></a></span><span lang="EN-US" style="font-size:10.0pt;font-family:"Tahoma","sans-serif"">>;tag=7B104564-2709<br>
> Date: Mon, 17 Oct 2011 22:50:25 GMT<br>
> Call-ID: </span><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif""><a href="mailto:C839BDD3-F84611E0-89E8ACDE-1882742B@10.15.15.37%3cmailto:C839BDD3-F84611E0-89E8ACDE-1882742B@10.15.15.37"><span lang="EN-US">C839BDD3-F84611E0-89E8ACDE-1882742B@10.15.15.37<mailto:C839BDD3-F84611E0-89E8ACDE-1882742B@10.15.15.37</span></a></span><span lang="EN-US" style="font-size:10.0pt;font-family:"Tahoma","sans-serif"">><br>
> Timestamp: 1318890771<br>
> CSeq: 101 INVITE<br>
> Require: 100rel<br>
> RSeq: 4915<br>
> Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER<br>
> Allow-Events: telephone-event<br>
> Remote-Party-ID: "B Ivr Q" <</span><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif""><a href="sip:8451@10.14.208.19"><span lang="EN-US">sip:8451@10.14.208.19</span></a></span><span lang="EN-US" style="font-size:10.0pt;font-family:"Tahoma","sans-serif"">>;party=called;screen=no;privacy=off<br>
> Contact: <</span><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif""><a href="sip:3325518400@10.14.208.19:5060"><span lang="EN-US">sip:3325518400@10.14.208.19:5060</span></a></span><span lang="EN-US" style="font-size:10.0pt;font-family:"Tahoma","sans-serif"">><br>
> Record-Route: <</span><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif""><a href="sip:Gk-int@207.249.152.75:5060;lr"><span lang="EN-US">sip:Gk-int@207.249.152.75:5060;lr</span></a></span><span lang="EN-US" style="font-size:10.0pt;font-family:"Tahoma","sans-serif"">><br>
> Server: Cisco-SIPGateway/IOS-12.x<br>
> Content-Length: 0<br>
><span class="apple-converted-space"> </span><br>
> can anyone help me?<br>
</span><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif"">><span class="apple-converted-space"> </span><br>
> Thanks.<br>
><span class="apple-converted-space"> </span><br>
><span class="apple-converted-space"> </span><br>
><span class="apple-converted-space"> </span><br>
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