I'm trying to fix a problem with the GrandStream Budgetone 102. I've been reading the
source code, mailing lists and other resources. Here's the scenario and the approach
I have been pursuing. I'm having some problems with the AGI calls and I hope someone
can give me some clarification.
PSTN <---> T1,PRI * <---> Grandstream BT 102 (12)
<---> TDM400 (1) Fax machine
End user requires that an inbound call ring 4 of the BT 102 phones so that any
available employee may answer. Generally, this works very well and the only problem I
have is when another call comes in. In this case the Grandstream BT 102 rings very
loudly in the ear piece and obliterates the conversation. The external party notices
that the audio is cut off while the BT 102 is ringing.
The solution I am trying to develop to work around the problem is to use an AGI script
to check each of the extensions in the group of phones that are supposed to be rung.
I want to remove extensions that are presently in a call and not ring them with the
new call. Also, I want to perform this test every 10 seconds so that I may include
any extension back into the group of phones to ring if their previous call has ended.
I plan to implement a variable and increment it in a loop using gotoif to provide
three or four 10 second trials for the dial. If no one answers, I'll send it to voice
mail.
I have tried using ChanIsAvail and Channel status to see if I can detect when the Sip
phone is busy. In the case of the ChanIsAvail, it doesn't matter if the phone is busy
or not, it will still return the channel as available. Maybe the definition of
channel is available does not have anything to do if it is in a call or not. In the
case of the Channel Status, it always returns 201 on the Sip channel. Actually, I'm
using the Asterisk Perl Modules by James Golovich so the
$AGI->channel_status('Zap/1-1') returns 4 and the $AGI->channel_status('Sip/2400')
returns -1.
I think that a major problem with the Channel Status is that the Sip channel is not
being correctly provided. Since it seems to work with the Zap channel. The code is
walking the channels to do a strcmp for an exact match. I'm lost to find out what the
Sip channel designator should look like.
It appears that the ChanIsAvail would be the correct call to make for this purpose.
Last, I can't get the options to work with the $AGI->exed('Dial', $newvar , '30,t');
command. It seems to ignore the options, so, I can't tell the dial command how long
to ring and to allow the called extension to transfer.
TIA, Walker
Here are some snips from my conf files and agi script:
myagi.agi
...
for $i ( split /,/,$ARGV[0] ) {
if ( $AGI->exec('ChanIsAvail', $i) == 0 ) {
if ( $count++ > 0 ) { $newvar .= "\&" };
$newvar .= $i;
$result = $AGI->channel_status($i); # always returns -1
print STDERR "$result\n";
}
}
$result = $AGI->channel_status('Zap/1-1'); # test this with Zap
print STDERR "$result\n"; # always returns 4
$AGI->exec('Dial', $newvar , '30,t'); # this is supposed to dial the
# extensions that are "not busy"
...
sip.conf
...
[2400] ; Grandstream Phone
context=intern
type=friend
insecure=yes
host=dynamic
permit=192.168.254.0/255.255.255.0
mailbox=2400
dtmfmode=inband
canreinvite=no
nat=no
...
extensions.conf
...
PHONE2=SIP/2400
PHONE3=SIP/2410
RECEPTION=${PHONE2},${PHONE3}
...
exten => 2200,1,AGI(myagi.agi,${RECEPTION})
...
console (asterisk -vvvvvvvc)
...
-- Executing AGI("Zap/1-1", "myagi.agi|SIP/2400,SIP/2410") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/myagi.agi
arg1 = SIP/2400,SIP/2410
Channel Status:
-- AGI Script Executing Application: (ChanIsAvail) Options: (SIP/2400)
-1
-- AGI Script Executing Application: (ChanIsAvail) Options: (SIP/2410)
-1
4
-- AGI Script Executing Application: (Dial) Options: (SIP/2400&SIP/2410)
-- Called 2400
-- Called 2410
-- SIP/2400-3320 is ringing
-- SIP/2410-a8ca is ringing
...
--
******** DataCrest, Inc. -- Technically Superior ******************
Walker Haddock http://www.datacrest.com
DataCrest, Inc. e-mail: [EMAIL PROTECTED]
1634A Montgomery Hwy. phone: 1-888-941-3282, 1-205-335-8589
Birmingham, AL 35216 fax: 1-205-823-7838
***********************************************************************
--
******** DataCrest, Inc. -- Technically Superior ******************
Walker Haddock http://www.datacrest.com
DataCrest, Inc. e-mail: [EMAIL PROTECTED]
1634A Montgomery Hwy. phone: 1-888-941-3282, 1-205-335-8589
Birmingham, AL 35216 fax: 1-205-823-7838
***********************************************************************
_______________________________________________
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
