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

[Nasir Iqbal]

On some analogs systems caller id is sent after first ring, so
removing "callerid=asreceived"
may help


Nasir Iqbal

ICT Innovations
http://www.ictinnovations.com/



On Tue, Oct 4, 2011 at 4:38 AM, neo haux <neo.haux at gmx.com> 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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