[Asterisk-Users] Grandstream BT102 Busy signal on hangup

[Steve Totaro]

hangup hangs up the channel, thats why you get a busy sound on your phone.
you have to phsically hang up the phone.


----- Original Message ----- 
From: "Doug Lytle" <support at drdos.info>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>
Sent: Friday, November 26, 2004 10:09 PM
Subject: [Asterisk-Users] Grandstream BT102 Busy signal on hangup


> Hey everybody,
>
> I've been playing around with Asterisk (Current CVS Stable dated: Asterisk
CVS-v1-0-11/23/04).
>
> I've purchased 2 GS BT102 SIP phones.  Both have been upgraded to firmware
1.0.5.18.  I've also have installed on my desktop and laptop, X-Lite.
>
> I've been using these to learn how to setup Asterisk.  I've got a Wildcat
X100P on order and will be here next week.
>
> My problem seems to be with the BT102s.  I can't seem to get them to
hangup when issued a Hangup().
>
> Asterisk's logs show it's has hung up, but I get a busy signal on the
phones themselves.
> This was also happening under firmware 1.0.5.16.  Configuration listed
below; all of them are called via the #include statement:
>
> <<<snip>>>
>
> my.sip.conf
>
> [5574]
> type = friend
> host = dynamic
> auth=md5
> username=5574
> reinvite=no
> canreinvite=no
> qualify=300
> nat=yes
> dtmfmode = info
> ;dtmfmode = also tried rfc2833, didn't make any difference.
> context = sip
> mailbox = 5574
> disallow=all
> allow=ulaw
> allow=alaw
> callerid = Doug Lytle <5574
>
> [5558]
> type = friend
> host = dynamic
> auth=md5
> username=5558
> qualify=300
> reinvite=no
> canreinvite=no
> nat=yes
> dtmfmode = rfc2833
> context = sip
> mailbox = 5558
> disallow=all
> allow=ulaw
> allow=alaw
> callerid = Brian Squires <5558>
>
> [5129]
> type = friend
> host = dynamic
> auth=md5
> username=5129
> reinvite=no
> canreinvite=no
> qualify=300
> nat=yes
> dtmfmode = rfc2833
> context = sip
> mailbox = 5129
> disallow=all
> allow=ulaw
> allow=alaw
> allow=ilbc
> callerid = Teone Taylor <5129>
>
>
> my.extensions.conf
>
> [sip]
>
> ; (Voicemail)
> exten => 5700,1,Wait,1
> exten => 5700,2,VoicemailMain
> exten => 5700,3,Hangup
>
> ; (Say Unix Time)
> exten => 13,1,SayUnixTime()
> exten => 13,2,Playback(beep)
> exten => 13,3,Hangup
>
> ; (Something to make you laugh)
> exten => 11,1,Playback(tt-monkeys)
> exten => 11,2,Hangup
>
> ; (MeetME Channels)
>
> exten => 5600,1,Meetme(1000)
> exten => 5600,2,Hangup
>
> ; (Doug Lytle)
>
> exten => 5574,1,Dial(SIP/5574,28,rt)
> exten => 5574,2,Voicemail(u5574)
> exten => 5574,103,Voicemail(b5574)
> exten => 5574,3,Hangup
>
> ; (Brian Squires)
>
> exten => 5558,1,Dial(SIP/5558,28,rt)
> exten => 5558,2,Voicemail(u5558)
> exten => 5558,103,Voicemail(b5558)
> exten => 5558,3,Hangup
>
> ; (Teone Taylor)
>
> exten => 5129,1,Dial(SIP/5129,28,rt)
> exten => 5129,2,Voicemail(u5129)
> exten => 5129,103,Voicemail(b5129)
> exten => 5129,3,Hangup
>
> my.iax.conf
>
> [5574]
> username=5574
> host=dynamic
> trunk=no
>
> [5574]
> type=user
> secret=12345
>
> [5558]
> username=5558
> host=dynamic
> trunk=no
>
> [5558]
> type=user
> secret=12345
>
> [5129]
> username=5129
> host=dynamic
> trunk=no
>
> [5129]
> type=user
> secret=12345
>
> my.voicemail.conf
>
> [sip]
> 5574 => 1242,Doug Lytle,dlytle at somedomain.com
> 5558 => 5567,Brian Squires,bsquires at somedomain.com
> 5129 => 1244,Teone Taylor,ttaylor at somedomain.com
>
> <<<snip>>>
>
> I see there are several people using the GS BT10x phones with no mention
of this problem.  I've Googled, searched the archives for year 2004 and
haven't found a solution.  Anybody have any suggestions?
>
> Doug Lytle
>
>
>
>
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