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 <[email protected]> Subject: Re: [asterisk-users] Delay before ringing from PSTN`s call To: Asterisk Users Mailing List - Non-Commercial Discussion <[email protected]> Cc: neo haux <[email protected]> Message-ID: <[email protected]> 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@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@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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