<div dir="ltr"><div><div><div><div><div><div>Hello Deepika,<br><br></div>you must check your calls are up, and if they're down you call back again.<br><br></div>FreeSWITCH has provisions for calling out from the conference.<br><br></div>Obviously GSM lines, especially in rural areas, are prone to drop.<br><br></div>You can use SIP trunks like now (and you can check different prociders), or you can buy GSM-SIP adapters (alibaba is your friend).<br><br></div>Good luck with your project, and let us know how it goes.<br><br></div>-giovanni<br><div class="gmail_extra"><br><div class="gmail_quote">On 17 April 2017 at 09:26, Deepika Yadav <span dir="ltr"><<a href="mailto:deepikay@iiitd.ac.in" target="_blank">deepikay@iiitd.ac.in</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">Hi,<div><br></div><div>I involved in a research project where we have developed a mobile based training system. Our server is Freeswitch based. The server on receiving request from an android app establishs conference call amongst 21 participants over the cellular network by calling on their mobile numbers. To achieve this, a VOIP-GSm gateway service is used offered by Doorvaani. The participants locations are rural areas of India. We need to maintain the conference call for atleast 1 hour as there is some kind of training program.</div><div><br></div><div>We have tested our training platform by conducting 10 such conference calls. However, one persistent problem that we are facing is frequent call drops. In between of an on-going conference, some of the participants faces call drops, to which we have adopted a workaround of redialing them. </div><div><br></div><div>One of the potential cause is the poor signal state in some areas, but there are many people who have decent signal strength and still face call drops.</div><div><br></div><div>We are not sure how can we investigate the matter as neither the VOIP-GSM logs are available for us nor can we setup conference call with more than 5-6 participants using existing service providers.</div><div><br></div><div>Is this a problem from Freeswitch, can I debug? or should we setup our own VOIP-GSM gateway as currently we are using web service from a company we can not physically access the logs. I would be happy to get suggestions on how to deal with this.</div><div><br></div><div><div class="m_7849077176341595142m_5643757448331817742gmail_signature" data-smartmail="gmail_signature">Regards<br>Deepika<br><a href="https://www.iiitd.edu.in/~deepikay/" target="_blank">https://www.iiitd.edu.in/~deep<wbr>ikay/</a></div>
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<a href="http://www.freeswitch.org" rel="noreferrer" target="_blank">http://www.freeswitch.org</a><br></blockquote></div><br><br clear="all"><br>-- <br><div class="m_7849077176341595142gmail_signature" data-smartmail="gmail_signature"><div dir="ltr"><div><span><span><br>Sincerely,<br></span><span><font color="#888888"><br>
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