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<body class='hmmessage'><div dir='ltr'>Thanks<font style="" face="'times new roman', serif"> M.Salman</font><br>When I am configured the endpoint to do the mixing of media it works, I will look for documentation as to how integrating Kamailio with Asterisk. So, Asterisk more feature rich? anybody know if there is a mailing list for Siremis( a web interface)? I setup SIREMIS with Apache server and the siremis installation complete from step 1 to step 4 and after i got the congratulation message , it just go back to step 1.<br>Thanks<br>Abdulmalik <br><br><div><hr id="stopSpelling">Date: Fri, 7 Mar 2014 21:13:19 +0500<br>From: msalman212@gmail.com<br>To: sr-users@lists.sip-router.org<br>Subject: Re: [SR-Users] (no subject)<br><br><div dir="ltr"><div><div><br></div>Hi Malik.<br></div> Kamailio is a SIP proxy it does not cater media, you can look for Asterisk, FreeSWITCH along with Kamailio to do the needful.<br></div><div class="ecxgmail_extra">
<br><br><div class="ecxgmail_quote">On Fri, Mar 7, 2014 at 9:03 PM, malik sherif <span dir="ltr"><<a href="mailto:asherif74@hotmail.com" target="_blank">asherif74@hotmail.com</a>></span> wrote:<br><blockquote class="ecxgmail_quote" style="border-left:1px #ccc solid;padding-left:1ex;">
<div><div dir="ltr">Hello,<br>I am using kamailio 4.1.1, I am wondering if kamailio supports call forwarding , three-way call, and call transfer natively? Does it expect the endpoints to do the RTP mixing?<br>Thanks<br>Absul<br>
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<br></blockquote></div><br><br clear="all"><br>-- <br><div dir="ltr"><font face="'times new roman', serif">Regards</font><div><font face="'times new roman', serif"><br></font><div><pre><font face="'times new roman', serif">M. Salman Zafar<br>
VoIP Professional<br></font><br></pre></div></div></div>
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