<div>Hi Virendra,<br></div><div><br></div><div>It may be problem for rtp packet port forwarding if u can dial through DID number. </div><div><br></div><div>You need to open rtp port range in firewal. e.g. 10,000 to 20,000 port.</div>
<div><br></div><div>please, write how can you dial call mobile or other devices. e.g. DID number, PRI number etc.</div><div><br></div><div><br>-- <br>Best Regards,<br><br>Rajnikant Vanza<br>Call : +91-9737456583<br>Software Engineer<br>
-------------------------------------------------------<br>Working On Linux,C/C++,Asterisk Technology<br>Gandhinagar - Gujarat</div><br><div class="gmail_quote">On Thu, Jun 9, 2011 at 12:13 AM, virendra bhati <span dir="ltr"><<a href="mailto:virbhati@gmail.com">virbhati@gmail.com</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 List,<br><br>When we make calls into asterisk with the help of our mobile, landline number, Cisco 79XX series then we didn't able to here any IVR which is playing into asterisk server. But when we dial from SIP softphone then all is working fine and we are able to here the IVR sound files.<br>
<br>What is the problem in this case please help me..<br clear="all"><br>-- <br><div dir="ltr"><br><br><br>-----<br>Thanks and regards<br><br> Virendra Bhati<br>+91-9172341457<br>Asterisk Engineer<br></div><br>
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