[Asterisk-Users] Bug Help or Suggestion - Grandstream
GXP2000(firmware 1.0.2.8) - BLF, Hints, call-limit
Richard Cheung
asterisk-list at usinfolab.com
Tue Mar 14 22:18:41 MST 2006
firmware 1.0.2.8 is a beta version, the latest beta version is 1.0.2.13.
You may want to try the latest beta.
http://www.voip-info.org/tiki-index.php?page=GXP-2000
I am using stable firmware 1.0.1.12, and it has no major problem.
-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com]On Behalf Of Corporate
IT Solutions - Michael Dunne
Sent: Tuesday, March 14, 2006 8:30 PM
To: asterisk-users at lists.digium.com
Subject: [Asterisk-Users] Bug Help or Suggestion - Grandstream
GXP2000(firmware 1.0.2.8) - BLF, Hints, call-limit
Version - Asterisk SVN-trunk-r12793M (1.2.4)
I have 4 Grandstream GXP 2000 phones configured. However at the moment,
I have had to disable BLF, Hints, and Call Limiting due to an extremely
annoying bug which seems to make the phones channels "lock" in busy
after a call has been hungup.
If I do a show hints after say extension 200 has hung up I get the
following
-= Registered Asterisk Dial Plan Hints =-
203 : SIP/203 State:Idle
Watchers 4
202 : SIP/202 State:Idle
Watchers 4
201 : SIP/201 State:Idle
Watchers 4
200 : SIP/200 State:Busy
Watchers 4
----------------
- 5 hints registered
However extension 200 has just hung up and is not in fact busy as it
claims.
I am not sure where to even begin troubleshooting this one. I don't know
if its an Asterisk hint bug, however when debugging the sip (sip debug
peer 200) I do see the hangup notification sent to Asterisk.
If there is a "best practice" configuration for the Grandstreams on
Asterisk then I would love to see how its done to minimise errors and
bugs.
The phones themselves are nice enough (although on the cheap side) its
just there appears to be one too many bugs to make it a bullet proof
solution. Especially considering I could buy four Grandstreams for the
price of one Cisco/Snom it's a high seller for obvious reasons.
Thanks for any assistance offered.
Extensions.conf, sip.conf available on request if this can be tracked
back to an asterisk config issue.
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