[asterisk-users] linksys PAP2t and asterisk

[wassim Darwish]

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:
-- Executing [88017736288155 at direct:1] Dial("SIP/490115-092bacc8", "SIP/us/88017736288155") in new stack    -- Called us/88017736288155    -- Call on SIP/us-092acb78 left from hold    -- SIP/us-092acb78 is making progress passing it to SIP/490115-092bacc8    -- SIP/us-092acb78 is ringing  (here it gives me a fake ring)
 
how can i disable this ringing . 



From: wassim505 at hotmail.comTo: asterisk-users at lists.digium.comDate: Fri, 13 Feb 2009 20:08:20 +0000Subject: [asterisk-users] linksys PAP2t and asterisk

Hi all: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. any suggestions please. 



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