Shirni,
I am aware of the default codec on the dial-peer. In my
scenario, I have an inbound dial-peer that only accepts g729 and a SIP endpoint
that only accepts G711ulaw. Transcoder has already been configured and its
operation has been verified with a test call from SIP endpo
By default voip dial-peer sends g729. To use G711 you have to define
codec in outgoing dial-peer.
sample config :
dial-peer v 1 voip
desttination-pattern 1...$
codec g711ulaw
session-target ipv4:X.X.X.X
session-protocol sipv2
Call coming from CUCM to CUCME will be g711 unless you have a incom
Bill,
I initially suspected that to be the issue. But my config checks out
fine. Also, note that the same config over SIP trunk works just fine. I only
face that problem when I send it over RAS. Here is my config:
May 4 03:52:44.577: //-1//DPM/dpMatchPeersCore:
Callin
It seems that you have a dial-peer with voice-class codec in it, this
is where it will get "all" the codecs from. I believe that George
indicated this also.
You should look at your CME config if it is the one sending all codecs.
Try removing
voice-class codec x from your dial-peer to prevent it
George,
Call flow from CME to CUCM is working fine. I am having issues
dialing from CUCM to CME over RAS. Same issue is experienced with or with
FastStart on H225 trunk. Incoming Voip call leg matches a dial-peer that only
has g729r8. Yet, when CME sends an invite to SIP endpoin
Hi Emin,
This Invite from what I understand is on outgoing call leg from CME to CUCM...
And it takes codecs from dial-peer... So check what you have on the dial-peer,
I'm sure you have voice class defined with g729 and g711...
Regards,
Sent from my iPad
On 3 May 2011, at 19:28, Emin Guliyev
Kobel,
Thanks for the reply. One more thing. Same setup over SIP trunk
works just fine. Over RAS, CUCME to CUCM dialing works fine but CUCM to CUCME
does not invoke a transcoder.
Thanks,
Emin
From: Miron Kobelski [mailto:findko...@gmail.com]
Sent: Tuesday, May 03, 2011 5:26 PM
Hi Emin,
I've noticed this also. My bet would be that voice register global
configuration affects incoming dial-peers created internally by CUCME. Even
if IP phone supports all codecs and DTMF transports, incoming dial-peer
filters them so only allowed items can be negotiated. It makes sense,
beca
Yes sir I replied back with correction , but max incoming calls on a dn
are controlled by huntstop channel.
On 5/3/2011 11:18 AM, Naoufal Kerboute wrote:
The max calls per button for outgoing and incoming calls
*From:*ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinest
Guys,
Quick question: Why is it that even though I explicitly define
codec g711ulaw under "voice register pool". When the "INVITE" is sent out SDP
always includes all the codecs and DTMF types. It is also the same for DTMF
relay. Even though I put in "rte-nte" only, it offers "
The max calls per button for outgoing and incoming calls
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Shrini
Sent: Tuesday, May 03, 2011 9:33 PM
To: Roig Borrell, Francesc Xavier
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CC
Hi guys,
I need an advice from CCIE's who already passed the exam.
In the call routing section, if the question doesn't ask for the calling number
type and plan, do I have to set the calling number type (Sub, Nat ...) vs the
called number or no need?
Also if the question doesn't ask for calling
Hi,
Where the spoken name script save the wav files?
Also do you have any idea how to use play the name of the agent before routing
the call to it?
Thank a lot
Naoufal
From: George Goglidze [mailto:gogli...@gmail.com]
Sent: Tuesday, May 03, 2011 1:38 PM
To: Naoufal Kerboute
Cc: ccie_voice@onlin
Although not going to happen often, I have actually had an issue with the
Proctor Lab equipment providing the PRI having issues and the help desk had to
do things on their end. Just keep that in mind when you are doing simple
things like getting a pri up.
From: adam compton
Sent: Monday, May
Actually max calls per button is for outgoing calls.
Below one I configured a while ago, will try later today.
Thanks
Shrini
On 5/3/2011 10:02 AM, Shrini wrote:
Hi Roig,
Try adding max-calls-per-button 5
Thanks
Shrini
On 5/3/2011 9:23 AM, Roig Borrell, Francesc Xavier wrote:
Hi Shrini,
S
Hi Roig,
Try adding max-calls-per-button 5
Thanks
Shrini
On 5/3/2011 9:23 AM, Roig Borrell, Francesc Xavier wrote:
Hi Shrini,
Sorry for not answering you before. I had to stop my studies for a
while due to work :-(
Now I have tested it and here are my conclusions
Ephone-dn 11 octo-line
Hi Shrini,
Sorry for not answering you before. I had to stop my studies for a while due to
work :-(
Now I have tested it and here are my conclusions
Ephone-dn 11 octo-line
Number 4023
Huntstop channel 5
ephone 1
busy-trigger-per-button 4
button 1:1 2:11
!
ephone 2
busy-trigger-per-bu
Hi,
To use the spoken name in another script, you need two steps, at least:
1) "Get User", where you can provide username, or agent extension to get the
user info.
2) "Get User Info", here you can retrieve Spoken Name into a prompt
variable.
I hope this helps,
Regards,
On Mon, May 2, 2011 at 8:
Hi Team,
I am facing problem with IP IVR Server; That is Prompts are not being playing,
when someone call to our direct PSTN number, he gets normal beep ringing.
Previouslly it was working fine, that is when someone call to this PSTN number
he listen IVR prompts, but now he is only getting norm
Looks like you want switch to get an IP with route.
Answer is data vlan.
Create dhcp pool for DATA vlan on router
on Switch
# int vlan
# ip address dhcp
On 5/2/2011 9:05 PM, Cristobal Priego wrote:
hello all
i know this is a stupid question but i'd like to know your opinion on this
if you
Did anyone tried Create CSQ Prompt ?
I tried this link with no success
https://supportforums.cisco.com/thread/265999
On 5/2/2011 2:56 PM, Miron Kobelski wrote:
Hi,
in the end user configuration on CUCM, there is a parameter ("Name
dialing" or similar, can't check this right now) which allows
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