[asterisk-users] Delay before ringing from PSTN`s call

John Novack jnovack at stromberg-carlson.org
Tue Oct 4 13:49:55 CDT 2011


You need to make sure your (DAHDI or ZAPTEL ) is set up properly for your country's CLID protocol
In the US CLID is sent between the first and second rings, and with a proper configuration Asterisk waits a ring before processing the call
Other parts of the world use different methods and protocols
You will need to dig into that first.

John Novack


neo haux wrote:
> Hi
>
> I am testing a degium TDP400P (2fxo+2fxs) on my asterisk
>
> I configured incoming calls from pstn to ring my SIP phone (extension : 100)
>
> cat  extensions.conf
> ...
> [from-pstn]
> exten => s,1,Dial(SIP/100,10)
>  same => n,VoiceMail(100,u)
>
>
>
>
> root at PC-debian:/etc/asterisk# cat dahdi-channels.conf
> ...
> ...
> ...
> ;;; line="1 WCTDM/0/0 FXSKS  (EC: MG2 - INACTIVE)"
> signalling=fxs_ks
> callerid=asreceived
> group=0
> context=from-pstn
> channel => 1
> callerid=
> group=
> context=default
> ...
> ...
> ...
>
> What I don`t understand is why the SIPphone rings after 3 secondes after Astererisk detects the incoming call. Moreover, after hanging off the external caller the SIPphone continue to ring for 3 seconds.
>
> I did those modifications in the file  /etc/asterisk/chan_dahdi.conf without improuvement ( After restarting Asterisk)
>
> [channels]
> cidstart=ring
> immediate=yes
> faxdetect=no
> usecallerid=no
>
>
>
>
> Here is the debug from Asterisk console
>
> *CLI>     -- Starting simple switch on 'DAHDI/1-1'
>     -- Executing [s at from-pstn:1] Dial("DAHDI/1-1", "SIP/100,10") in new stack
>   == Using SIP RTP CoS mark 5
>     -- Called SIP/100
>     -- SIP/100-00000001 is ringing
>   == Spawn extension (from-pstn, s, 1) exited non-zero on 'DAHDI/1-1'
>     -- Hanging up on 'DAHDI/1-1'
>     -- Hungup 'DAHDI/1-1'
>
>
> --
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-- 

Dog is my Co-pilot

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