I have been working on implementing a Sipml5 phone for a clients website for the past week or so. I had a asterisk 1.8 system running on the clients end (current config w/ zoiper and hard phones, etc). Their new system will use a webphone (thats the plan). SO far putting Sipml5 together has proved more difficult than I initially thought.
The actual tutorial for it is completely useless, and I messed around with multiple asterisk installs until I finally got it working as far as registering, etc… But the Sound was only 1 way, different version is one way and the other way is static, etc…
I finally got the webrtc2sip server installed aswell but there is basically 0 documentation on how to configure it properly.
Today after reading on this forum that it worked with 11.5 I installed 11.5 (I had 11.5 before but it wasnt working) and using the configs I made over the past days it works calling from chrome to chrome or chrome to chromium. But I cannot call a regular SIP client and the asterisk server keeps crashing after every call ends, or in the call and there are no errors, not even in debug, it just says asterisk ending clean(0).
Any ideas on this? Any help would be greatly appreciated! Thanks in Advance!