Hello all,
I am new to Asterisk@home and am having a strange issue with both my new
Grandstream HT-286 & BT-101. The issue is as follows:
Example is with BT-101 (HT-286 shows same behavior)
1) Device registers to Asterisk
2) I can place a call via the BT-101 out my Zap or SIP provider
3) Conversation takes place (yay!)
4) I hang up BT-101
5) BT-101 will no longer dial out until:
a) I give asterisk a "restart now"
b) I place a call from another extension in my home TO the BT-101,
answer BT-101, hang up BT-101, and all is well for another single
outbound call
Nothing is logged via "sip debug peer 7213" when the phone will not
dial.
After I reset it above, everything looks/works fine.
This behavior also occurs with internal extension to extension calls
between the BT-101 (7213) and HT-286 (7214).
To me, it sounds like I have something incorrect with my extension
configuration (teardown?) for both the BT-101 and HT-286, however, I also
have 2 X-Lite softphones, with identical extension configurations as the
Grandstream devices, and both of the softphones work flawlessly, and have
for several weeks now.
Here are my configs.
First X-Lite softphone:
[7211]
username=7211
type=friend
record_out=Adhoc
record_in=Adhoc
qualify=no
port=5060
nat=never
mailbox=7211@device
host=dynamic
dtmfmode=rfc2833
context=from-internal
canreinvite=no
callerid=device <7211>
BT-101 (firmware 1.0.8.16):
[7213]
username=7213
type=friend
record_out=Adhoc
record_in=Adhoc
qualify=no
port=5060
nat=never
mailbox=7213@device
host=dynamic
dtmfmode=rfc2833
context=from-internal
canreinvite=no
callerid=device <7213>
Please let me know if you need any more of my configurations; any and all
help would be appreciated.
Thank you!