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

[neo haux]

Hi,

I commented the option callerid in the file dahdi-channels.conf without
success, My SIP phone still ring after 4-5 secondes :-(

; Span 1: WCTDM/0 "Wildcard TDM410P" (MASTER)
;;; line="1 WCTDM/0/0 FXSKS  (EC: MG2 - INACTIVE)"
signalling=fxs_ks
;callerid=asreceived

I am living in Canada, so I guess I should use USA signaling ? If so in
which file ?



Message: 7
Date: Tue, 04 Oct 2011 14:49:55 -0400
From: John Novack <jnovack at stromberg-carlson.org>
Subject: Re: [asterisk-users] Delay before ringing from PSTN`s call
To: Asterisk Users Mailing List - Non-Commercial Discussion
       <asterisk-users at lists.digium.com>
Cc: neo haux <neo.haux at gmx.com>
Message-ID: <4E8B5553.2030505 at stromberg-carlson.org>
Content-Type: text/plain; charset="iso-8859-1"; Format="flowed"

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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