I seem someone made whatsapp gateway by name Sip To Whatsapp, sip2wa,
how I can integrate those gateway into asterisk by using trunk, the demo is using some free sip server like antisip, sip.linphone.org etc …
Any example for using free sip account in those services as trunk ?
Free SIP services are unlikely to want to be used from PABXes, and may actively try to detect them. However, apart from that I would expect them to work the same way as single line paid services, so just take the settings they provide and match them to the example PJSIP trunk settings in the normal way.
My understanding is that Meta do not approve of the use of their protocol by third party code that has simply reverse engineered it. They could well change it without warning, or attempt to detect and block accounts doing so.
A number of things about the referenced page make me think that the author has a poor understanding of intellectual property.
https://wiki.asterisk.org/wiki/display/AST/res_pjsip+Configuration+Examples
based on the page that has presentation, the sip gw app just providing sound as input to microphone and likewise, its android framework job (which open source) that modified. so basically no reverse enginering of licensed application, its still using official apk (I might be wrong)
its same thing as using chan_dongle huawei, instead output sound to pc its redirected to sip rtp.
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