Hi,
I was wondering if someone could shed some light on the issue im having. Currently having working perfectly asterisk with my Trunk Sip VOIP provider (UNE based on colombia south America) Currently trying to use zoiper using SIP for the users to call the office, the issue is that they cant hear each other. after hours of researching found out in the sip.nat config i would put this
after reloaded it starts to working the zoiper calls both sides can hear BUT after few mins users started to say that the calls weren't coming in to the Trunk Sip of my VOIP provider so i had to remove those lines on the sip nat config any idea how to troubleshoot this i know its a nat thing but dont know where to look.
@tjreid thanks for the reply, as for zoiper the IP is facing to the Phone server which is the 20x.xx.xx.xx and the extensions are facing to that trunk, the issue is that lets say ext 405 tries to call 105 and 405 is using a data plan outside of the company they cant hear each other. So i went into the SIP nat config of elastix added the config above and then BAM it starts working flawless, but then what happens now the SIP trunk the calls get dropped so i know it has to be something with the NAT somewhere