The VOIP dial-peers will always be chosen first due to preference. When the
phone is in SRST and we receive an incoming call we will STILL choose the
VOIP dial-peers even though the SUB/PUB are both down. We want the timer
(h225 tcp establish timer) to be a low value (1 second) otherwise there will
Fantastic Vik. That jogs my memory! Very much appreciated.
For the voice-class h323, are you referring to the "telephony-service
ccm-compatible" command? I just read up on that.
From: Vik Malhi [mailto:vma...@ipexpert.com]
Sent: May-10-10 2:42 PM
To: Ryan Schwab; OSL Group
Subject: Re: [OS
Take a look at this. #2 looks nice. You are VERY dependent on voice-class
h323 in your VOIP dial-peers.
HQ-RTR(config)#dial-peer hunt ?
<0-7> Dial-peer hunting choices, listed in hunting order within each
choice:
0 - Longest match in phone number, explicit preference, random selection.
1 -
Assuming that for the Unknown type you have the prefix occurring and that
you reset the GW
I¹m wondering if the Calling # type is being set to Subscriber when the ANI
is 7 digits. Try putting a prefix in the SUBSCRIBER type setting and reset
and see how you get on.
--
Vik Malhi CCIE #13890
I was looking over a the CME Admin Guide on SRST Fallback and I had a
question:
http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/admin/configuration/guid
e/cmesrst.html
It's mentioned that if you want to pre-build DN's, to place a lower
preference on the DN so it doesn't over-ride any exist
Absolutely agree.
You HAVE TO ENABLE FRTS for the policy-map (LLQ) to take effect. Running
FRTS on the physical interface is mandatory unless you are using class-based
shaping (nested policy-map).
--
Vik Malhi CCIE #13890
Instructor - IPexpert, Inc.
Mailto: vma...@ipexpert.com
Telephone: +1.810
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--
Message: 2
Date: Mon, 10 May 2010 06:47:10 -0700 (PDT)
From: akash patel
Subject: [OSL | CCIE_Voice] DTMF Relay issue for CME to Unity
This might be a stupid question or I might be tired but something aint
right.
I am sending unknown type local numbers (7 digits) from pstn to my h323 gw @
BR2. calling number on the phone shows up as 7digit and same shows up in
missed calls directory.
On my GW settings under CUCM, i have the appr
Can you post a copy of your config?
Matthew Berry
**Sent from my iPhone**
Skype/Twitter: ciscovoiceguru
Google Voice: +1 612 424 5044
On May 10, 2010, at 6:47 AM, akash patel wrote:
I have integrated CME and Unity Connection as per documentation. I
am running 7961-SIP phones on CME and t
I have integrated CME and Unity Connection as per documentation. I am running
7961-SIP phones on CME and they are registered.
dtmf-relay is set up tortp-nte under both ; voice register pool & SIP dial-peer
point to Unity connection.
When I press Message Button, I hear 'Sign-in' prompt, but wh
Hi everyone,
I'm having trouble getting the Gatekeeper lab working, Question 4.2
The config is as per the lab, but it's seems that the transcoder isn't being
invoked.
So from CCM -> G.729 -> HQ-RTR (outvia ipipgw) -> G.711 BR2
When I place a call and issue the "show gatek calls" I can see the G
If you are configuring FRF.12 you have to write frame-relay traffic shaping
under the physical interface to work properly.
Thanks and best regards,
EHAB SALEM
Cisco Instructor | Sigma IT - Egypt
From: Omotayo [mailto:adefilabi...@gmail.com]
Sent: Monday, May 10,
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