Hello all, I am new to VOIP and linphone. We are planning to develop a voip add on board to be interfaced with our host system, based on linux.
Our design of VoIP add on board goes this way... It has a telephone connected to it. The VoIP add on board has an application and hardware for SLIC and dsp codecs such as G.723.1, G.711, G.729 which does the voice compression and RTP packetization. So we are planning to use the console version of linphone on our host system. We want to make use of the SIP functionality available with linphone. As per the reading from the linphone it is found that it expects a SIP URL in the form of [EMAIL PROTECTED] We have only the provision to input the enduser SIP Phone no. using the telephone. My doubts are, 1. Is it possible and sufficient to use the telephone keypad alone for calling (dialing) the SIP based end user. 2. Also is it possible to give the same dtmf digits as input to SIP application in the linphone to get it connected to the other end user? We would appreciate your help in this regard. Please advice. Thanks & Regards, Vidhya Sankari _______________________________________________ Linphone-users mailing list [email protected] http://lists.nongnu.org/mailman/listinfo/linphone-users
