Hi all,
Have a HandyTone 386 ATA set up with both analog ports going to a 2
line phone for testing. It seems to work ok most of the time, but two
problems remain.
If i don't use the ATA for "a while", when i make a test call to the
one of the lines i go right to voicemail since callweaver gets a
message that the line is busy as evidenced from the callweaver cli:
-- Called grandstream11
-- Got SIP response 486 "Busy" back from 192.168.1.227
-- SIP/grandstream11-2428 is busy
In this same state, if i pick up the handset on the analog phone on
the ATA i hear a click but i get no dial tone. Curious is that if i
put the handset down, then pick it up again i get a dial tone and if
i then hang up the hand set i can call the phone on the ATA and it
rings.
I have updated to the latest firmware and no changes. I have looked
on the net for this problem but can not find any references.
The ATA is on the LAN with callweaver. I don't think it is callweaver
since the ATA is giving the busy message back.
Any ideas?
Peace,
Dan
PS: Here is sip.conf extract for two lines on ATA:
; grandstream unit 1 line 1
[grandstream11]
type=friend
secret=grandstream11
context=from-internal-sip ; the internal context controls what
we can do
nat=no ; This phone is not natted
host=dynamic ; This device registers with us
defaultip=192.168.1.227 ; is fixed IP -- but not using host since
callweaver complained
; host=192.168.1.227 ; This device registers with us
canreinvite=yes ; allow RTP voice traffic to bypass OpenPBX
; dtmfmode=inband ; Choices are inband, rfc2833, or info (sjphone
needs inband) otherwise trial and errpr
dtmfmode=info ; either RFC2833 or INFO for the BudgeTone
[EMAIL PROTECTED] ; Activate the message waiting light if this
callerid="Dan" <104> ; my caller ID
disallow=all
allow=ulaw
; grandstream unit 1 line 2
[grandstream12]
type=friend
secret=grandstream12
context=from-internal-sip ; the internal context controls what
we can do
;qualify=yes ; Qualify peer is no more than 2000 ms away
nat=no ; This phone is not natted
host=dynamic ; This device registers with us
defaultip=192.168.1.227 ; is fixed IP -- but not using host since
callweaver complained
; host=192.168.1.227 ; This device registers with us
canreinvite=yes ; allow RTP voice traffic to bypass OpenPBX
; dtmfmode=inband ; Choices are inband, rfc2833, or info (sjphone
needs inband) otherwise trial and errpr
dtmfmode=info ; either RFC2833 or INFO for the BudgeTone
[EMAIL PROTECTED] ; Activate the message waiting light if this
callerid="Amy" <105> ; my caller ID
disallow=all
allow=ulaw
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