I'm developing a new application to test the good configuration of my voip server (asterisk). The principle is to use sipp tool as a sip client in order to simulate several scénarios (simple call, call transfer, etc). I want to test also the voicemail application configured and enabled in my voip server i.e : calling the voicemail by a sip client to perform many possible actions for example consult voicemails messages, erase them, listen to archived messages, etc. this interaction between sip client an voicemail is based on dtmf codes assigned for every menu on the voicemail. I'am asking if someone has an idea or an example about how to test/troubleshoot voicemail using sipp client : -xml scénario to call voicemail and navigate between menus -how to send correct dtmf digit (1-9 or #) as rtp packets -how to interpret correctly the voicemail menu invitation (how to convert rtp packets representing the menu option as an audio file or something similar)
I would really appreciate any expert or voip developer help.
Thanks in advance.
