Hi Virendra, It may be problem for rtp packet port forwarding if u can dial through DID number.
You need to open rtp port range in firewal. e.g. 10,000 to 20,000 port. please, write how can you dial call mobile or other devices. e.g. DID number, PRI number etc. -- Best Regards, Rajnikant Vanza Call : +91-9737456583 Software Engineer ------------------------------------------------------- Working On Linux,C/C++,Asterisk Technology Gandhinagar - Gujarat On Thu, Jun 9, 2011 at 12:13 AM, virendra bhati <[email protected]> wrote: > Hi List, > > 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. > > What is the problem in this case please help me.. > > -- > > > > ----- > Thanks and regards > > Virendra Bhati > +91-9172341457 > Asterisk Engineer > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
