[asterisk-users] Disconnect supervision tone detection working for india

[asteriskguru asteriskguru]

Hi ,

Thanks danny nicholas.  Finally we get the things done with following.
If i specify busypatten=500,500  then asterisk does not recognize hang up
signal. After removing it only all are working fine.

I choosed  2nd option as per your suggestions.


working chan-dahdi.conf:
====================

signalling = fxs_ks
busycount = 3
busydetect = yes
callprogress = yes

progzone=in
usecallerid=yes
cidstart=ring
callerid=asreceived
group=0
context=from-pstn
channel => 1


Caller id Detection :
I have PSTN line with caller id display. After the 2nd ring,  caller id
display phone shows caller id.  Can you guide me on right to get it in
asterisk.

I am getting following error when line is ringing ,

*CLI>     -- Starting simple switch on 'DAHDI/1-1'
[Jul 31 15:26:24] ERROR[5216]: callerid.c:564 callerid_feed: No start bit
found in fsk data.
[Jul 31 15:26:24] WARNING[5216]: chan_dahdi.c:7016 ss_thread: Failed to
decode CallerID on channel 'DAHDI/1-1'
[Jul 31 15:26:24] WARNING[5216]: chan_dahdi.c:7121 ss_thread: CallerID
returned with error on channel 'DAHDI/1-1'
    -- Executing [s at from-pstn:1] AGI("DAHDI/1-1", "agi://localhost") in new
stack
    -- AGI Script Executing Application: (BACKGROUND) Options:
(/home/guest/adhearsion/songs_play/voices/welcome)
    -- <DAHDI/1-1> Playing
'/home/guest/adhearsion/songs_play/voices/welcome' (language 'en')


hope guidance from you,
Ashik



On Sat, Jul 31, 2010 at 1:50 PM, asteriskguru asteriskguru <
beaasteriskguru at gmail.com> wrote:

> hi,
>
> Although I changed those parameters. Asterisk does not detect "hangup
> signal"
>
>
> signalling = fxs_ks
> busydetect = yes
> busycount = 3
> busypattern = 500,500
> answeronpolarityswitch = no
> hanguponpolarityswitch = yes
>
> callprogress=no
> progzone=in
> usecallerid=yes
> cidstart=ring
> callerid=asreceived
> group=0
> context=from-pstn
> channel => 1
>
> by hearing attached tone anything can be done ?
>
> hope guidance from you,
> ashik
>
>
> On Fri, Jul 30, 2010 at 7:20 PM, Danny Nicholas <danny at debsinc.com> wrote:
>
>>  Your best bets are going to be
>>
>> #1 hanguponpolarityswitch=yes
>>
>> Or
>>
>> #2 callprogress=yes
>>
>>
>>
>> I’d hang my hat on #1 personally
>>
>> --
>> _____________________________________________________________________
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>               http://www.asterisk.org/hello
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100731/b3ca4bf1/attachment.htm