Table of Contents

Tested clients

The clients below has been tested with the TelecomX SIP trunk, and is confirmed to work.

Asterisk

Only important setting, is to ensure that the User-Agent string contains the word “asterisk”, case is not important.

Versions

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

Sample registration

register => <username>:<password>@sip.telecomx.dk

Sample SIP peer

[telecomx1]
  type=peer
  host=sip.telecomx.dk
  canreinvite=no
  qualify=yes
  context=trunk
  defaultuser=<username>
  fromdomain=sip.telecomx.dk
  secret=<password>
  insecure=invite

Gigaset dect

Models

Remarks

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


SNOM

Models

Remarks

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


Audiocodes

Models

All models should work fine.

Remarks

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


Linksys ATA

Models

All models should work fine.

Remarks

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


Aastra PBX

Models

Remarks

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