On some analogs systems caller id is sent after first ring, so removing "<span class="Apple-style-span" style="font-family: arial, sans-serif; font-size: 13px; background-color: rgb(255, 255, 255); ">callerid=asreceived" may help</span><div>
<div><font class="Apple-style-span" face="arial, sans-serif"><br></font></div><div><font class="Apple-style-span" face="arial, sans-serif"><br></font></div><div>Nasir Iqbal<br><br>ICT Innovations<br><a href="http://www.ictinnovations.com/" target="_blank">http://www.ictinnovations.com/</a><br>
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<br><br><div class="gmail_quote">On Tue, Oct 4, 2011 at 4:38 AM, neo haux <span dir="ltr"><<a href="mailto:neo.haux@gmx.com">neo.haux@gmx.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
<div>Hi</div><div><br></div><div>I am testing a degium TDP400P (2fxo+2fxs) on my asterisk</div><div><br></div><div>I configured incoming calls from pstn to ring my SIP phone (extension : 100)</div><div><br></div><div>cat extensions.conf</div>
<div>...</div><div>[from-pstn]</div><div>exten => s,1,Dial(SIP/100,10)</div><div> same => n,VoiceMail(100,u)</div><div><br></div><div><br></div><div><br></div><div><br></div><div>root@PC-debian:/etc/asterisk# cat dahdi-channels.conf</div>
<div>...</div><div>...</div><div>...</div><div>;;; line="1 WCTDM/0/0 FXSKS (EC: MG2 - INACTIVE)"</div><div>signalling=fxs_ks</div><div>callerid=asreceived</div><div>group=0</div><div>context=from-pstn</div><div>
channel => 1</div><div>callerid=</div><div>group=</div><div>context=default</div><div>...</div><div>...</div><div>...</div><div><br></div><div>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.</div>
<div><br></div><div>I did those modifications in the file /etc/asterisk/chan_dahdi.conf without improuvement ( After restarting Asterisk)</div><div><br></div><div>[channels]</div><div>cidstart=ring</div><div>immediate=yes</div>
<div>faxdetect=no</div><div>usecallerid=no</div><div><br></div><div><br></div><div><br></div><div><br></div><div>Here is the debug from Asterisk console</div><div><br></div><div>*CLI> -- Starting simple switch on 'DAHDI/1-1'</div>
<div> -- Executing [s@from-pstn:1] Dial("DAHDI/1-1", "SIP/100,10") in new stack</div><div> == Using SIP RTP CoS mark 5</div><div> -- Called SIP/100</div><div> -- SIP/100-00000001 is ringing</div>
<div> == Spawn extension (from-pstn, s, 1) exited non-zero on 'DAHDI/1-1'</div><div> -- Hanging up on 'DAHDI/1-1'</div><div> -- Hungup 'DAHDI/1-1'</div>
<br>--<br>
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