Displaying 2 results from an estimated 2 matches for "to_pstn".
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2015 Jan 19
2
SEMI-OFFTOPIC openvox
Hi list, I write on the list looking for help, buy a openvox gw gsm
for four channels and I'm a little disappointed with the support
openvox, for some reason , The call doesn?t get trough
support tells me it was my asterisk server, but does not really work
me and my internal calls are working perfectly, I tested with another
sangoma FXO gateway and works perfectly.
the problem is that
2015 Jan 21
0
asterisk-users Digest, Vol 126, Issue 18 mtr
...com>
> Content-Type: text/plain; charset=UTF-8
>
> Hi, when I make an outgoing call sends me a busy here, and no one is making call
>
> Contact: <sip:984783842 at 50.X.X.X:5060>
> Content-Length: 0
>
>
> <------------>
> -- Executing [984783842 at to_pstn:1] Dial("SIP/101-0000004e",
> "SIP/5001/84783842@,40,rRT") in new stack
> == Using SIP VIDEO TOS bits 136
> == Using SIP VIDEO CoS mark 6
> == Using SIP RTP TOS bits 184
> == Using SIP RTP CoS mark 5
> Audio is at 13780
> Video is at 50.X.X.X:18488
>...