tested-clients
Table of Contents
Tested SIP trunk 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
- 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
Recommended generel SIP settings
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
SIP phones
- Snom: all models
- Mitel/Aastra: all models
- Yealink: all models
- Gigaset Dect: all models
Remarks
To present a specific phone number, enter it into Displayname in E.164 format.
ATA and gateways
- AudioCodes: all models supporting SIP
- Linksys/Cisco SPA: all models
- Grandstream HT8xx
Aastra PBX
- 2025
- 2065
- 3xx
- 4xx
Remarks
SIP/RTP does not have to go through the PBX, redirection of RTP, even to a different IP address is supported.
tested-clients.txt · Last modified: by Per Møller