<p>Binan, avp db is usually used for temporary data storage e.g. To share some value between two call session etc. So while avp db can be used for this but then you need to make sure you never call avp db clear in your script.</p>
<p>Secondly you can use drouting to route any call, not just calls to gw. You need to create rule that call should not be routed to user B during 11h to 14h. So drouting lookup will fail. Then you check in alias db which will tell you user c is alias to user b and you can then route call to user c. This way you can create N x N drouting to alias rules.</p>
<p>Thank you.<br></p>
<p>On Oct 18, 2012 10:34 AM, "Binan AL Halabi" <<a href="mailto:binanalhalabi@yahoo.com">binanalhalabi@yahoo.com</a>> wrote:<br>
><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div style="font-size:10pt;font-family:times new roman,new york,times,serif"><div><span></span></div><div><span>Hi,</span></div>
<div style="font-style:normal;font-size:13.3333px;background-color:transparent;font-family:times new roman,new york,times,serif"><span>Add them as Attribute and Value(opensipsctl <span>avp</span> add ....... ): Add them as two separate AVP. <br>
</span></div><div style="font-style:normal;font-size:13.3333px;background-color:transparent;font-family:times new roman,new york,times,serif"><span>Then load them in your script as AVP (avp_db_load function).<br></span></div>
<div style="font-style:normal;font-size:13.3333px;background-color:transparent;font-family:times new roman,new york,times,serif"><br><span></span></div><div style="font-style:normal;font-size:13.3333px;background-color:transparent;font-family:times new roman,new york,times,serif">
<span>// Binan</span></div><div><br></div> <div style="font-family:times new roman,new york,times,serif;font-size:10pt"> <div style="font-family:times new roman,new york,times,serif;font-size:12pt"> <div dir="ltr"> <font face="Arial"> <hr size="1">
<b><span style="font-weight:bold">Från:</span></b> Engineer voip <<a href="mailto:forvoip4@gmail.com" target="_blank">forvoip4@gmail.com</a>><br> <b><span style="font-weight:bold">Till:</span></b> Binan AL Halabi <<a href="mailto:binanalhalabi@yahoo.com" target="_blank">binanalhalabi@yahoo.com</a>>; OpenSIPS users mailling list <<a href="mailto:users@lists.opensips.org" target="_blank">users@lists.opensips.org</a>> <br>
<b><span style="font-weight:bold">Skickat:</span></b> torsdag, 18 oktober 2012 10:21<br> <b><span style="font-weight:bold">Ämne:</span></b> Re: [OpenSIPS-Users] Transfering a call by opensips<br> </font> </div> <br>
<div>Hi,<br><br><span lang="en"><span>I don't have</span> <span>the fields in user_preferences </span> <span>table or i add them ?</span></span><br><br>
<div>2012/10/18 Binan AL Halabi <span dir="ltr"><<a rel="nofollow" href="mailto:binanalhalabi@yahoo.com" target="_blank">binanalhalabi@yahoo.com</a>></span><br><blockquote style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div><div style="font-size:10pt;font-family:times new roman,new york,times,serif">Hi,<br><br>Store in usr_preferences table the time_from and time_to<br>then read them in your script using avpops module <br>Take the hour of the call and compare it to the range you got it from database.<br>
<br><a href="http://www.opensips.org/html/docs/modules/1.7.x/avpops.html" target="_blank">http://www.opensips.org/html/docs/modules/1.7.x/avpops.html</a><br><br>// Binan <br><br><div><span></span></div><div style="font-family:times new roman,new york,times,serif;font-size:10pt">
<div style="font-family:times new roman,new york,times,serif;font-size:12pt"> <div dir="ltr"> <font face="Arial"> <hr size="1"> <b><span style="font-weight:bold">Från:</span></b> Engineer Voip <<a rel="nofollow" href="mailto:forvoip4@gmail.com" target="_blank">forvoip4@gmail.com</a>><br>
<b><span style="font-weight:bold">Till:</span></b> <a rel="nofollow" href="mailto:users@lists.opensips.org" target="_blank">users@lists.opensips.org</a> <br> <b><span style="font-weight:bold">Skickat:</span></b> torsdag, 18 oktober 2012 8:53<br>
<b><span style="font-weight:bold">Ämne:</span></b> [OpenSIPS-Users] Transfering a call by opensips<br> </font> </div><div><div> <br>
Hello all, <br>I want to transfert the call to user C when user A calls user B in interval of time for example: 11h-14h<br>I can do that by asterisk but i prefer to do it by opensips<br>It's possible to do that by opensips?<br>
<br>Cordialement. <br>Envoyé de mon iPhone<br></div></div><div>_______________________________________________<br>Users mailing list<br><a rel="nofollow" href="mailto:Users@lists.opensips.org" target="_blank">Users@lists.opensips.org</a><br>
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<br></blockquote></div><br><br clear="all"><br>-- <br><font color="#663300"><font face="comic sans ms,sans-serif"><br><span style>Best Regards.</span><br></font></font><br><br>
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<br></blockquote></p>