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
calleridgroupcontext=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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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' > > -- > _____________________________________________________________________ > -- 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/20111004/d7d12869/attachment.htm>
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' > > > -- > _____________________________________________________________________ > -- 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-- Dog is my Co-pilot -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20111004/dadfa7ba/attachment.htm>
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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