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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