| View previous topic :: View next topic |
| Author |
Message |
GameGamer43
Co-op Board Member
Joined: 06 May 2007
Posts: 16
|
 Posted: Fri Apr 04, 2008 1:46 pm Post subject: Configuring HT503 for use with an Asterisk Server |
 |
|
As the rather new Broadcom HT503 device has been out for some time now, and as with all Grandstream products documentation is rare I have listed the settings below to get the HT503 working with your asterisk server. This will enable you to field calls from your PSTN line of make outgoing calls on the PSTN line when your Internet of VoIP provider is down. Should you have any questions, please feel free to ask me in this forum or by private message.
-----------------------
SIP Trunk Settings
-----------------------
Outgoing Settings
| Code: | Trunk Name: userid
Peer Details:
dtmfmode=rfc2833
host=dynamic
secret=password for userid
type=friend
username=userid |
Incoming Settings
| Code: | USER Context: from-userid
USER Details: |
------------------
HT503 Settings
------------------
Status (of what your device should say after completion of this setup)
| Code: | Port Hook Registration DND Forward Busy Forward Delayed Forward
FXO Idle Registered No |
Basic Settings
| Code: | | Unconditional Call Forward to VOIP: extension@voipserver:5060 |
FXO Port
| Code: | Account Active: Yes
SIP Server: ipaddress of your asterisk server
SIP User ID: userid
Authenticate ID: userid
Authenticate Password: password for userid
Caller ID Minimum RX Level (dB): -18
Caller ID Transport Type: Relay via SIP From
Number of Rings: 4
Stage Dialing: 1 |
Caller ID Minimum RX Level (dB) may need some tweaking based on your Landline provider. The settings above are for a Verizon Landline in the NY area. |
|
| Back to top |
|
 |
farhansabir
Joined: 25 Apr 2008
Posts: 1
Location: Ontario
|
| Posted: Fri Apr 25, 2008 10:53 am Post subject: |
|
|
Yours is the only posting I could find on HT503 with config settings. However, I have not been able to connect the FXO to asterisk as trunk. I sent a PM too.
Can u tell me if the ATA connects to asterisk or Asterisk connects to the ATA?
While configuring the trunk, I initially tried using the ATA IP, but that was also useless. Then used dynamic as you mentioned. Also, when its functioning will i be able to see the status in 'sip show peers' ?
I really don’t know am i doing wrong in configuring this ATA. If you can please be kind enough to repeat the steps with any exclusive user ID/pwd to be defined in ATA/Asterisk. It may be helpful. I can also provide you remote access through vnc.
Thanks
Farhan Sabir
farhansabir@yahoo.com
farhan@cigear.com
_________________
--
Farhan Sabir
System Analyst
CIGear.com |
|
| Back to top |
|
|
|
FAQ
Search
Memberlist
Usergroups
Register
Profile
Log in to check your private messages
Log in
