Atcom AG188N
From VoIP.ms Wiki
Contents |
SIP Configuration Section:
- Fill the following fields according to your account:
- Register Server Addr: atlanta.voip.ms (one of our multiple servers)
- Register Server Port: 5060
- Register Username: 100000 (your VoIP.ms username)
- Register Password: ******** (the account password)
- Detect Interval Time: 60
- DTMF Mode: DTMF_RFC2833
- Enable Via rport: Enabled
- Local SIP Port: 5060
- Register Expire Time: 60
- RFC Protocol Edition: RFC3261
- Server Type: common
- SIP Default Protocol: Enabled
- Click on the "Apply" button at the bottom of the form.
Configuration Screen
Known Issues with this Device:
Problems when dialing *97 for Voicemail:
With one phone hooked to the ‘phone’ jack, you can dial a star code to access the PSTN line instead of the SIP/IAX configured on the ATA. So, one analog phone enables you to use either SIP/IAX or the PSTN. However, people using the ATA with Asterisk have a problem when they want to dial Asterisk * codes like *97 for voicemail (as the * takes you to PSTN). The solution is to delete the “lifeline *T” option from “Dial-Peer” menu.
