<font color="#330000"><font face="comic sans ms,sans-serif">Hi,<br>Thank you all<br></font></font><font color="#330000"><font face="comic sans ms,sans-serif"><span id="result_box" class="" lang="en"><span class="hps"> But i</span> <span class="hps">thought</span> <span class="hps">to do this</span> <span class="hps">by hanging up</span> <span class="hps">the call</span> of<span class="hps"></span> <span class="hps">mobile</span> <span class="hps">after</span> <span class="hps">some</span> <span class="hps">time</span> <span class="hps">if the mobile phone does not respond<br>
Is opensips can do that? <br>thank you for your help<br></span></span></font></font><br><div class="gmail_quote">2012/10/9 Schneur Rosenberg <span dir="ltr"><<a href="mailto:rosenberg11219@gmail.com" target="_blank">rosenberg11219@gmail.com</a>></span><br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><p dir="ltr">That's true but from my experience it worked for me with a few carriers, of course every provider might treat it differently, even those that played a message did not actually pick up the call it was done with early media.</p>
<div class="HOEnZb"><div class="h5">
<div class="gmail_quote">On Oct 9, 2012 2:55 PM, "Adrian Georgescu" <<a href="mailto:ag@ag-projects.com" target="_blank">ag@ag-projects.com</a>> wrote:<br type="attribution"><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div style="word-wrap:break-word">You cannot control the behavior of the GSM operators. They may decide to play a prompt and practically answer the call without returning any code when the target device is out of coverage, voicemail box may kick in or not, roaming or other scenarios you are not even aware about. If instead of a response code, you end-up being connected or in early media, OpenSIPS cannot retarget the call in all situations. Thinks work random and fail for unknown reasons due to un-deterministic behavior of the GSM operators involved.<div>
<br></div><div>Adrian</div><div><div><div><div><div><br><div><div><div>On Oct 9, 2012, at 2:40 PM, Schneur Rosenberg wrote:</div><br><blockquote type="cite"><p dir="ltr">If you disable voicemail on the mobile phone through the carrier then you should get a busy signal from the carrier, if you also disable call waiting then you should also get busy a busy signal from the carrier, its not really required if you don't mind being disturbed by the beep , this might not always be a solution because you loose voicemail and call waiting on your phone.</p>
<p dir="ltr">Another solution might be to send the call through a Asterisk server and use the amd (answering machine detection) function.</p>
<div class="gmail_quote">On Oct 8, 2012 11:17 PM, "Adrian Georgescu" <<a href="mailto:ag@ag-projects.com" target="_blank">ag@ag-projects.com</a>> wrote:<br type="attribution"><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div style="word-wrap:break-word">There is no standard for communicating signaling information from a non-IP network like GSM to a SIP based network. What you try to achieve is close to impossible unless you own both networks and are able to set up some proprietary signaling mechanism between them.<div>
<br></div><div><div>Adrian<br><div> <br><div><div>On Oct 8, 2012, at 10:56 PM, Engineer voip wrote:</div><br><blockquote type="cite"><font face="comic sans ms,sans-serif">Hi,<br>I want to associate a mobile phone number to each desktop phone and want to ringing at the same time the desktop phone and mobile phone:<br>
** if the mobile phone is off service or already in communication (call falls on the responder) the opensips ignore this mobile and keep ringing the desktop phone.<br>
<br>i hope this is clear<br></font><br><div class="gmail_quote">2012/10/8 Duane Larson <span dir="ltr"><<a href="mailto:duane.larson@gmail.com" target="_blank">duane.larson@gmail.com</a>></span><br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
Can you explain in more detail what you are trying to do? <div><div><br><br><div class="gmail_quote">On Mon, Oct 8, 2012 at 8:32 AM, Engineer voip <span dir="ltr"><<a href="mailto:forvoip4@gmail.com" target="_blank">forvoip4@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><font color="#330000"><font face="comic sans ms,sans-serif">Hello,<br>I do that and its good but </font></font><span lang="en"><span style="font-family:comic sans ms,sans-serif"><span>if user A</span> <span>hangs up</span> <span>the call</span><span>, the user B</span> <span>is</span> <span>hanging</span> <span>too.</span><br>
<span>I want</span> <span>when user A</span> <span>hangs up or</span> responder<span></span>, <span>the user B</span> <span>can</span> </span><span><span style="font-family:comic sans ms,sans-serif">take the call!<br>
have you an idea to resolve that?<br></span><br></span></span><div class="gmail_quote">2012/10/8 Vlad Paiu <span dir="ltr"><<a href="mailto:vladpaiu@opensips.org" target="_blank">vladpaiu@opensips.org</a>></span><br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><u></u>
<div bgcolor="#ffffff" text="#000000">
Hello,<br>
<br>
This is the usual SIP parallel forking, so yes this can be done with
OpenSIPS 1.8. See the append_branch() function at [1] .<br>
<br>
[1] <a href="http://www.opensips.org/Resources/DocsCoreFcn18#toc106" target="_blank">http://www.opensips.org/Resources/DocsCoreFcn18#toc106</a><span style="padding-right:16px;width:16px;min-height:16px"></span><br>
<br>
Regards,<br>
<pre cols="72">Vlad Paiu
OpenSIPS Developer
<a href="http://www.opensips-solutions.com/" target="_blank">http://www.opensips-solutions.com</a><span style="padding-right:16px;width:16px;min-height:16px"></span> </pre>
<div><div>
<br>
On 10/08/2012 12:00 AM, Engineer voip wrote:
</div></div><blockquote type="cite"><div><div><span><span style="background-color:rgb(51,0,0)"><span style="background-color:rgb(255,255,255)"><font face="comic sans ms,sans-serif">Hi All,<br>
</font></span></span></span><span lang="en"><span>It</span><span></span><span></span><span></span><span></span><span></span><span></span><span></span><span></span><span></span><span> is</span> <span>possible to call a standard (fix) phone and mobile
phone at</span> the same time <span>with</span> <span>OpenSIPS</span></span> 1.8 ??<br clear="all">
<br>
-- <br>
<font color="#663300"><font face="comic sans ms,sans-serif"><br>
<span>Best Regards.</span><br>
</font></font><br>
<br>
</div></div><pre><fieldset></fieldset>
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<br></blockquote></div><br><br clear="all"><div><br></div>-- <br></div></div>--<br>*--*--*--*--*--*<br>Duane<br>*--*--*--*--*--*<br>--<br>
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