<html><head><meta http-equiv="Content-Type" content="text/html; charset=utf-8"></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; line-break: after-white-space;" class=""><div class="">Hey Ayub,</div><div class=""><br class=""></div><div class="">I would take it one step at a time. Kamailio is a toolkit with a bunch of modules. I find it useful to first figure out what features you need in a proxy. This includes understanding what are you trying to protect your backend media servers from, problems you are currently having and/or how are you trying to scale your backend servers. </div><div class=""><br class=""></div><div class="">Assuming you are using Asterisk or FreeSwitch as your backend media servers, here are some questions that might help derive the features you want:</div><div class=""><br class=""></div><div class="">- Are the backend servers being overwhelmed by REGISTER requests?</div><div class="">- Are you providing the proxy for phone end points, PBX’s or both</div><div class="">- What’s your High Availability Requirements? Are the backend media services replicated so it doesn’t mater what server gets the original INVITE requests?</div><div class="">- Are you handing registrations for 1 SIP domain or multiple SIP domains?</div><div class="">- Do you want to maintain CDR’s at the proxy or leave it to the backend media servers</div><div class=""><br class=""></div><div class=""><b class="">Mack Hendricks / Head of Support / dOpenSource</b></div><div class=""><div dir="auto" style="word-wrap: break-word; -webkit-nbsp-mode: space; line-break: after-white-space;" class=""><div style="color: rgb(0, 0, 0); font-family: Helvetica; font-size: 12px; font-style: normal; font-variant-caps: normal; font-weight: normal; letter-spacing: normal; text-align: start; text-indent: 0px; text-transform: none; white-space: normal; word-spacing: 0px; -webkit-text-stroke-width: 0px;">web: <a href="http://dopensource.com" class="">http://dopensource.com</a></div><div style="color: rgb(0, 0, 0); font-family: Helvetica; font-size: 12px; font-style: normal; font-variant-caps: normal; font-weight: normal; letter-spacing: normal; text-align: start; text-indent: 0px; text-transform: none; white-space: normal; word-spacing: 0px; -webkit-text-stroke-width: 0px;">support: +888-907-2085</div><div style="color: rgb(0, 0, 0); font-family: Helvetica; font-size: 12px; font-style: normal; font-variant-caps: normal; font-weight: normal; letter-spacing: normal; text-align: start; text-indent: 0px; text-transform: none; white-space: normal; word-spacing: 0px; -webkit-text-stroke-width: 0px;"><a href="http://dsiprouter.org" class="">dSIPRouter</a> - GUI focused on implementing Kamailio to provide SIP Trunking and PBX Hosting Services</div></div>
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<div><br class=""></div><div><br class=""><blockquote type="cite" class=""><div class="">On Mar 26, 2018, at 1:26 AM, Mohammed Ayub <<a href="mailto:payubpashap@gmail.com" class="">payubpashap@gmail.com</a>> wrote:</div><br class="Apple-interchange-newline"><div class=""><div dir="ltr" class="">HI,<div class=""><br class=""></div><div class="">Myself Ayub from Phonology, Bangalore, India.</div><div class=""><br class=""></div><div class="">We are trying to set up an SIP proxy server using Kamailio with Siremis as web interface.</div><div class=""><br class=""></div><div class="">Have installed and configured kamailio and siremis but need help in config if sip domain , users and etc...</div><div class=""><br class=""></div><div class="">Please revert.</div><div class=""><br class=""></div><div class="">Regards</div><div class="">Ayub</div></div>
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