The clients below has been tested with the TelecomX SIP trunk, and is confirmed to work.
Only important setting, is to ensure that the User-Agent string contains the word “asterisk”, case is not important.
minexpiry=3600 defaultexpiry=3600 maxexpiry=3600 qualifyfreq=45 t1min=100 timert1=1000 timerb=32000 rtptimeout=45 rtpholdtimeout=300 rtpkeepalive=50 nat=force_rport,comedia dtmfmode=rfc2833 registerattempts=0 registertimeout=40
register => <username>:<password>@sip.telecomx.dk
[telecomx1] type=peer host=sip.telecomx.dk canreinvite=no qualify=yes context=trunk defaultuser=<username> fromdomain=sip.telecomx.dk secret=<password> insecure=invite
To present a specific phone number, enter it into Display name in E.164 format.
To present a specific phone number, enter it into Displayname in E.164 format.
All models should work fine.
To present a specific phone number, enter it into Displayname in E.164 format.
All models should work fine.
To present a specific phone number, enter it into Display name in E.164 format.
SIP/RTP does not have to go through the PBX, redirection of RTP, even to a different IP address is supported.