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 [88017736288...@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: wassim...@hotmail.comto: asterisk-us...@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.
Windows Live™: E-mail. Chat. Share. Get more ways to connect. Check it out.
_________________________________________________________________
Windows Live™: E-mail. Chat. Share. Get more ways to connect.
http://windowslive.com/explore?ocid=TXT_TAGLM_WL_t2_allup_explore_022009
_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
