<p dir="ltr">Hi Nabeel,</p>
<p dir="ltr">Point 1 I cant imagine how those lines possibly relate to no media error in asterisk, I guess it depends on your config setup.</p>
<p dir="ltr">The logical answer to your point 2 would be Asterisk realtime. However this is not going to be as staraight forward as making asterisk use subscriber table for voicemail since thats not how asterisk realtime works.</p>
<p dir="ltr">You can tell asterisk which table to point to when finding a vmail box but that table needs to have voicemail related columns. <br>
There could be more complicated ways to instruct asterisk to do what you want can be, for example, ARI /AGI/custom dialplan and IVR. However, all of those should be discussed in Asterisk mailing list.</p>
<p dir="ltr">The OpenSIPS integration guide did serve it's purpose in showing you the path, agreed that it should be updated but I believe OpenSIPS team is doing much bigger things. Your efforts here might be very helpful to alot of people.</p>
<p dir="ltr">Regards,<br>
Sammy</p>
<div class="gmail_quot<blockquote class=" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">The last error message has been solved by removing the following lines from opensips.cfg:<div><br></div><div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-style:solid;border-left-color:rgb(204,204,204);padding-left:1ex"> if (!db_does_uri_exist()) {<br> send_reply("420","Bad Extension");<br> exit;<br> }<br></blockquote><div> </div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-style:solid;border-left-color:rgb(204,204,204);padding-left:1ex"> t_newtran();<br> t_reply("480", "Temporarily Unavailable");</blockquote></div><div><br></div><div><br></div><div>Now voicemail seems to be working, but only if manually adding users to the voicemail.conf file. So the following questions remain:</div><div><br></div><div>1. Adding "limit = 1" to sipusers mysql view resolves a database error. Is this the correct way to fix the error for voicemail integration?</div><div><br></div><div>2. How can voicemail.conf file be configured to use variable substitution for all users in the OpenSIPS subscriber table?</div><div><br></div><div>Nabeel</div></div><div class="gmail_extra"><br><div class="gmail_quote">On 2 July 2016 at 13:41, Nabeel <span dir="ltr"><<a href="mailto:nabeelshikder@gmail.com" target="_blank">nabeelshikder@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><p dir="ltr">In the latest version of Asterisk, there is a new file voicemail.conf which must be configured correctly for voicemail, but the tutorial does not mention this file at all. Please let me know how to configure this file for integration with OpenSIPS. </p><span><font color="#888888">
<p dir="ltr">Nabeel</p>
</font></span></blockquote></div><br></div>
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