Re: [OpenSIPS-Users] SIP to WebRTC via OpenSIPS mid-registrar fails: forced proto 6 not matching sips uri

2021-02-12 Thread juancarlosg6
Hi Mark, is working for us more or less, because is strange situtation, if the extension WSS received a incoming call working good but if a put the call in on-hold inmediatly hangout, but if the same extension do the call to a external number or a SIP extension or another Webrtc extension work

Re: [OpenSIPS-Users] SIP to WebRTC via OpenSIPS mid-registrar fails: forced proto 6 not matching sips uri

2020-12-23 Thread juancarlosg6
Hi, iam having the same situation, can you copy the opensips.cfg as reference, iam having issue with incoming calls to webrtc extensions, my opensips version is 3.1 and is working with mid_registar, i can do calls to SIP extensions, SIP Trunk, but not working when incoming calls to extension

Re: [OpenSIPS-Users] SIP to WebRTC via OpenSIPS mid-registrar fails: forced proto 6 not matching sips uri

2020-12-23 Thread juancarlosg6
Hi Mark, can you share the configuration? i need to do something similar but without success, thank you. Using your example: REGISTER: SIP softphone (LinPhone) -> OpenSIPS Mid-registrar -> Asterisk = success REGISTER: WebRTC webphone (Mizutech) -> OpenSIPS Mid-registrar -> Asterisk