[asterisk-users] linksys PAP2t and asterisk

Jose Flores Galicia flojose at gmail.com
Sat Feb 14 15:23:27 CST 2009


Man, as the CLI says:
SIP/us-092acb78 is ringing  (here it gives me a fake ring)

It's the channel SIP/us/something, which is generating ring signalling.



2009/2/14 wassim Darwish <wassim505 at hotmail.com>

>  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.com
> To: asterisk-users at lists.digium.com
> Date: Fri, 13 Feb 2009 20:08:20 +0000
> Subject: [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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-- 
Jose Flores Galicia
<<FloJoSe at gmail.com>>
BriefCode && Code Based Training
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