Hi All,
Here is my SRST config
i am seeing issue not related to Brage.
Issue
BR1 and Br2 phone is having shared line in cucm
both are registered to SRST and shared line are appeared on phones
PSTN can make a call to Primary line ie 3001 and 3002
but call is simply disconnected when
Hi Adam,
I found this in CCM service settings for some of H323 parameters :
Ensure that calls over QSIG trunks do not utilize + because it is not
supported by QSIG. This parameter has no effect on H.323 outbound calls
because H.323 unconditionally strips the + sign when routing outbound calls.
Many Thanks Randoll for your great assistance, Have a nice day :)
On Wed, Mar 23, 2011 at 9:09 PM, Randall Saborio ill2...@gmail.com wrote:
Hi Wael,
This is what I got from some notes on how I usually do it:
1.
Define router as DNS Server
ip dns server
1.
Define the
Hi Shingei,
Have you set the MVA number under Media Resources in CUCM and is this number in
a PT that can be seen by the RDP CSS and possibly also incoming CSS of the MVA
GW?
You also need to be able to route to this number from the MVA GW with
appropriate dialpeers, if the MVA DN is in the
it's fine on SIP, it's only h323 that can't handle the + sign.
and it's not a dial-peer that strips it, when call manager sends the h225
setup, + is already NOT present there...
Regards,
On Thu, Mar 24, 2011 at 2:08 AM, Adam Thompson phoe...@fatturtle.comwrote:
The + gets stripped by the voip
Hi Roger,
I think you have mis-read the question. The problem we are having is the
following:
PT-INTERNAL --- extensions ,500X, 100X
PT-XLATE-MVA - [15]XXX - translate calling to +44.. - APPLIED
CSS-INTERNAL -- contains PT-INTERNAL
CSS-MVA contains PT-XLATE-MVA
CSS-MVA is a CSS applied to
Hi George,
What I ment was that you need to see the MVA DN set under Media Resources in
the RDP CSS. But I never said that you need to put that DN in PT-INTERNAL, put
that in PT-XLATE-MVA and try it out.
Sincerely
Roger Källberg
CCIE #26199 (Voice)
Consultant
Cygate AB
Eric Perssons väg 21,
Hi Roger,
I don't see that as a requirement nowhere... I will try it out though next
time I do it.
Anyway, just to let you know, that while internal calls didn't work, I had
an RP to 90014158884343 on PT-XLATE-MVA and that did work!!! so I'm not sure
I need MVA DN's PT in CSS-MVA.
From what I
Hi Guys,
Glad to see you all here at this time while now is time for dinnerfor
asia,are u guys sleepless?
I fire up my home lab after received the suggestion from Roger. Testing out
now...
20mins later...
Hi Roger,
Your magic works! As you mentioned, i changed the mva# partition from
pt-phones
Hi Shingei,
Great news then! I'll try it out,
Cheers,
On Thu, Mar 24, 2011 at 10:54 AM, ShinGei Yong shingei.y...@gmail.comwrote:
Hi Guys,
Glad to see you all here at this time while now is time for dinnerfor
asia,are u guys sleepless?
I fire up my home lab after received the suggestion
This has been a pretty long discussion about the normal behaviour of
H323. The '+' expression is recognised by MGCP and SIP GWs, not H323.
If you want an H323 GW to send the full E164 number starting with '+'
you need to apply a voice translation rule. For example you can apply
the rule on the
one small correction. It would be translate calling since we are talking
about callerid.
On Thu, Mar 24, 2011 at 8:20 AM, Bill Lake whl...@gmail.com wrote:
This has been a pretty long discussion about the normal behaviour of
H323. The '+' expression is recognised by MGCP and SIP GWs, not
Hi everyone,
I would like to get your input about the subject above.
Actually I have a Softswitch and one Proxy... I would like to know
which parameters I have to configured to get the link up and making
calls going thru between two h323 peers.
Any tip will be welcomed.
Thanks in advance,
Does this have to go on the voice port or can it go on the dial peer too?
-Original Message-
From: adam compton com...@gmail.com
Sender: ccie_voice-boun...@onlinestudylist.com
Date: Thu, 24 Mar 2011 08:24:41
To: Bill Lakewhl...@gmail.com
Cc: ccie_voice@onlinestudylist.com; Adam
Dial peers
On Thu, Mar 24, 2011 at 10:03 AM, matt.san...@myself.com wrote:
Hi everyone,
I would like to get your input about the subject above.
Actually I have a Softswitch and one Proxy... I would like to know which
parameters I have to configured to get the link up and making calls going
Hi experts,
I had configured MVA using hairpin method.
Everything worked and calls went out, but when I picked it up, there were no
audio!
Has anyone seen this problem before?
Where should I start to troubleshoot?
FYI, both of dial-peers voip are using no vad and G711ulaw. I can see calls
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