Atcom AG188N
From VoIP.ms Wiki
| [draft revision] | [draft revision] |
(→SIP Configuration Section:) |
|||
| Line 36: | Line 36: | ||
**'''Enable Via rport:''' Enabled | **'''Enable Via rport:''' Enabled | ||
**'''Local SIP Port:''' 5060 | **'''Local SIP Port:''' 5060 | ||
| - | **'''Register Expire Time:''' | + | **'''Register Expire Time:''' 180 |
**'''RFC Protocol Edition:''' RFC3261 | **'''RFC Protocol Edition:''' RFC3261 | ||
**'''Server Type:''' common | **'''Server Type:''' common | ||
| Line 49: | Line 49: | ||
[[File:Atcom.JPG|Atcom AG188N Configuration Screen]] | [[File:Atcom.JPG|Atcom AG188N Configuration Screen]] | ||
| - | |||
| - | |||
==Known Issues with this Device:== | ==Known Issues with this Device:== | ||
Revision as of 20:42, 17 December 2013
Contents |
Documentation:
Accessing Your Device´s Configuration Page:
Get the IP Address to access your AG-188N from the handset:
First please pick up the handset and dial #*111#
Then you will hear the AG-188N’s IP address, please write it down.
Enter the AG-188N’s IP address into a web browser and press enter, then you can access AG-188N’s web manage interface.
Remember the account login is admin/admin for administrator and guest/guest for user.
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: 180
- 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.
