Archives : November-2014
2014-11-12 2:45 GMT-02:00 Luis Eduardo Cortes : I solved my issue by changing type=friend with type=peer in [555555555] section, afterwards, googling Ive found this article that explain me why: http://forums.digium.com/viewtopic.php?t=79338#p161214 ..
tengo la siguiente pagina pero no se como seguir despues del punto 22http://highsecurity.blogspot.com/2012/12/webrtc-and-asterisk-11-using-sipml5.ht..
Change my Dynastar E1 gateway to Cisco with E1 module, but cant make easiest dialplan. All my routing i made on asterisk, so i need that cisco all calls from E1 send via sip to Asterisk and all calls came from Asterisk by sip send to E1. From E1 to Aster..
Asterisk users and developers, The last few weeks we had several crashes on live asterisks running versions12.2.0rc1 / 12.6.1 with PJPROJECT versions 2.1.0 / 2.2.1. We opened a ticket – ASTERISK-24471.After investigating the issue I can say that ..