Tested SIP trunk clients

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.

• 1.6.x
• 1.8.x
• 10.x
• 11.x
• 12.x
• 13.x
• 16.x
• 17.x
• 18.x
• 20.x
• 21.x
• 22.x

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

• Snom: all models
• Mitel/Aastra: all models
• Yealink: all models
• Gigaset Dect: all models

To present a specific phone number, enter it into Displayname in E.164 format.

• AudioCodes: all models supporting SIP
• Linksys/Cisco SPA: all models
• Grandstream HT8xx

• 2025
• 2065
• 3xx
• 4xx

SIP/RTP does not have to go through the PBX, redirection of RTP, even to a different IP address is supported.