Can't call out from BR1 phone in SIP SRST but I can call into it. When I
dial from SIP phone in SRST the moment I press dial softkey I hangs up and
if I press speaker phone and dial the first digit it hangs up.
Here si the config for SIP SRST pls...
voice service voip
allow-connections h323 to
HQ>>BR2>>>CUE getting fast busy
CUE is inte to CCM, it works when the region between HQ and BR2 is setup as
G711. I do have transcodes reg and in MRGL of HQ and BR2. Hq mrgl has tran
config on hq router and br2 mrgl has tran config on br2 router. CUE CTI RP
and Ports are in BR2 DP. Any clue pls?
I had my license, and by mistake I unregister the license, so VTGO phone works
in demo mode now. I couldn't find the license file, is there a way to contact
VTGO or by any other means to find that S/N so I can register phones again.
Thanks in advance,
Hi Sean,
Your incoming pots dial peer and voip outgoing dial peers (and perhaps your
voice port) don't have any called number translation, so from your h323 gw
config in ucm set significant digits to 4, also make sure the h.323
interface in your br2 gw has the h323-gateway voip interface and
h323-
Hi,
I'd say yes if the negotiated codec between the gw and ucm is g.729r8
(default codec for the dial peer), so make sure the gw is using that codec
when talking to ucm or at least it's in the list for the voice class codec
used by the dial peer,
Thanks,
On Wed, Feb 10, 2010 at 2:07 PM, vccie201
Is anyone having this problem? I can access all other website and even can
ping all servers and routers of my rack.
Thanks___
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Wierd H323 gateway problem PSTN inbound call to BR2 phones does not ring
phones.. calls come in and i verify using debug isdn q931 but phone does not
ring, if i use csim start from gateway to call phone, it rings, so that
eliminates partition and inbound CSS issue. phone can successfully make
o
