asterisk users - Nov 2004 - 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@somedomain.com
5558 => 5567,Brian Squires,bsquires@somedomain.com
5129 => 1244,Teone Taylor,ttaylor@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

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@drdos.info>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users@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@somedomain.com
> 5558 => 5567,Brian Squires,bsquires@somedomain.com
> 5129 => 1244,Teone Taylor,ttaylor@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
>
>
>
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>

> hangup hangs up the channel, thats why you get a busy sound on your phone.
> you have to phsically hang up the phone.
Which is basically happens when someone hangs up on you, no? ;)

Wilson Pickett wrote:
>>hangup hangs up the channel, thats why you get a busy sound on your
phone.
>>you have to phsically hang up the phone.
>>    
>>
>
>Which is basically happens when someone hangs up on you, no? ;)
>
>  
>
Not really, no. 

On an analog line, after 30 or so seconds, yes this is the norm.  On a 
digital line (Definity PBX) the line is dropped and the phone hangs up 
itself.  Figuring the later was a closer match to Asterisk,  I was 
guessing this behavior. 
I guess I can put a wait(5) or so to give the end user time to hangup 
before being hit with a busy signal.

Thanks!


Doug

> I guess I can put a wait(5) or so to give the end user time to hangup
> before being hit with a busy signal.
Or hang up faster!