About me

caruizdiaz

caruizdiaz profile

I’m Carlos Ruiz Díaz, a telecommunications developer from Paraguay, now living in Mexico City.

If you want to get in touch with me, send me an email to carlos.ruizdiaz@gmail.com or call me to toky.co/caruizdiaz

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4 thoughts on “About me

  1. Estimadisimo Carlos Ruiz Diaz

    Que gusto saber que el genio de Dinstar SMS api es paraguayo! que alegria y que orgullo compatriota, espectacular toky!! que lastima estas en mexico! por avisa cuando estes por paraguay!. yo trabaje en una empresa en londres que se llamaba micromuse y ese equipo de gente ahora esta en http://www.moogsoft.com soft que estamos representando en latam, y parte del grupo http://www.redpoint.com . asesoranos con nuestro dinstar y sms api please (dinos como girarte dinero) y podemos ganar una verdadera fortuna vendiendo http://www.moogsoft.com a claro mexico aprovechando que estar por ahi. un placer

    manu morales

  2. Hello.
    My name is Rodion.
    I use WebRTC to receive calls in browser. I use JsSIP library on frontend (Chrome browser). And I use Asterisk + Kamailio on backend. In updated Chrome version 47 there were some WebRTC improvements and changes.
    Now when I make outgoing calls in my system everything works fine. But when I receive incoming calls there is no sound from caller and callee. Caller can’t hear the voice from the other side and the same is for callee. There is no error messages in browser console and on backend. There is no any warning messages even.
    In the previous versions of Chrome (46) everything works fine. I see the same sip logging messages in Chrome 45, 46 and 47. Everything looks the same. But I have that issue with no sound during incoming calls in browser while using Chrome 47.

    I spent more than 10 days trying to solve this issue… And I can’t solve it. May be you can help me to solve the issue? To advice something to force it back to work in Chrome 47?

    Thank you, Rodion

  3. Hello.
    My name is Rodion.
    I use WebRTC to receive calls in browser. I use JsSIP library on frontend (Chrome browser). And I use Asterisk + Kamailio on backend. In updated Chrome version 47 there were some WebRTC improvements and changes.
    Now when I make outgoing calls in our system everything works fine. But when I receive incoming calls there is no sound from caller and callee. Caller can’t hear the voice from the other side and the same is for callee. There is no error messages in browser console and on backend. There is no any warning messages even.
    In the previous versions of Chrome (46) everything works fine. I see the same sip logging messages in Chrome 45, 46 and 47. Everything looks the same. But I have that issue with no sound during incoming calls in browser while using Chrome 47.

    I spent more than 10 days trying to solve this issue… And I can’t solve it. May be you can help me to solve the issue? To advice something to force it back to work in Chrome 47?

    Thank you, Rodion

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