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this post is attached to the prevoius post, this is what i have on CLI when i call from Linksys pap2t to asterisk and then asterisk bridge the call to a sip provider:<BR>
-- Executing [88017736288155@direct:1] Dial("SIP/490115-092bacc8", "SIP/us/88017736288155") in new stack<BR> -- Called us/88017736288155<BR> -- Call on SIP/us-092acb78 left from hold<BR> -- SIP/us-092acb78 is making progress passing it to SIP/490115-092bacc8<BR> -- <FONT color=#ff0000>SIP/us-092acb78 is ringing (here it gives me a fake ring)</FONT><BR>
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<FONT color=#ff0000><FONT color=#0c0c0c>how can i disable this ringing .</FONT> <BR></FONT><BR><BR><BR>
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From: wassim505@hotmail.com<BR>To: asterisk-users@lists.digium.com<BR>Date: Fri, 13 Feb 2009 20:08:20 +0000<BR>Subject: [asterisk-users] linksys PAP2t and asterisk<BR><BR>
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Hi all:<BR>when i make a call from linksys pap2t to an asterisk server a fake ring is heard some times ,but when sending calls between 2 asterisk servers through sip no fake ring is heard but real one. <BR>any suggestions please.<BR> <BR><BR><BR><BR>
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