Very true. Yeah I have verified the packet it was. It was the 6608
keepalive messages. Anyway to remark them on the application layer (aka in
CM). I know I can create an access-list to remark them, but I wanted to see
if there was alternatives.
thanks,
Ryan Trauernicht
On Sun, Jan 11, 2009 at
Hi,
it is a home lab, I can restrict name and number with configuring CME to to
that but when it comes to ccm, it wouldn't kick in.
So I think problem lies in ccm part.
Jeremy
On Mon, Jan 12, 2009 at 6:08 AM, Christian Hennrich <
christian.hennr...@intact-is.com> wrote:
> Hi,
>
> if you have
Have you considered using a Sniffer like Wireshark to verify your markings
are being gotten across?
I also use access-lists and match on DSCP tos bits.
You can also re-write the markings at the port level if you feel the need
and think something is wrong on the server side.
Again Sniffer to the
This is a known bug. of the show policy-map int command. Is it strictly
cosmic. What version of code do you have? 12.2.25see?
Thanks,
Ryan Trauernicht
On Sun, Jan 11, 2009 at 10:49 PM, jeremy co wrote:
> Hi,
>
> Interestingly no traffic is matched by class map, anybody face this be4?
>
> IP
Hi,
Interestingly no traffic is matched by class map, anybody face this be4?
IP phones attached to fa0/1-2
mls qos map policed-dscp 0 24 26 to 10
mls qos
!
class-map match-all VOICE
match any
!
!
policy-map PHONE
class VOICE
set ip dscp 46
!
interface FastEthernet0/1
switchport acces
Hi Ryan,
I am using PVDM2. If I remove the IPIPGW (removed the outvia command on
the gk), all the calls will work fine. Only when IPIPGW is invoke and call
from SiteB will fail. The only way I can get the calls to work is by
putting the transcoder in the 711-only DP. But by doing that, the f
I can proudly say I am a knuckle head sometimes. Sometimes you overlook the
smallest more common commands.
Debug voice ccapi inout
Jan 6 23:19:03.543: //29/00C0BA8E0200/CCAPI/ccCallDisconnect:
Cause Value=16, Tag=0x0, Call Entry(Previous Disconnect Cause=0,
Disconnect Cause=0)
That got m
I created a separate class-map to match on AF31 just to see if I have
anything that is still matching on AF31 since there really shouldnt be.
I have changed the IPVMSA service param to 0x60 and i change the CTI service
param's as well to CS3. If i set a policy map output on the serial
interface it
What is the work around for IPCC express when trying to add in a JTAPI
controlled group with the default DP?
It just sits there and does nothing.
Only way I found around it is to set the JTAPI integration only to the
publisher and configure everything... then add in the sub later.
Thanks,
Ryan Tr
that is how I always test it.
Just for testing to make sure it isnt a CSS issue...
Take a Translation Pattern and set it to 1999 with the CSS of the IPMA RP
and have it translate to the Manager DN.
Then dial 1999 and see if the manager's phone rings. That is pretty much
the same as the CTI Route
I configured IPMA again last night and same issue. The CSS are all ok, I have
double triple check with IPExpert solutions.
I have internal/external, forward no answer internal/external, forward no
coverage internal/external, forward on failure internal/external to the manager
DN with CSS manager
>From the CME router, can you collect the following debugs and send it as a
txt attachment?
debug voip ccapi inout
debug ras
debug h225 asn1
debug h245 asn1
Configure 'service sequence-numbers' as well.
On Sun, Jan 11, 2009 at 2:53 PM, Ryan Trauernicht
wrote:
> The TCS was unchecked already. M
Yeah I can prob do that... just looking to see what the flexibility is.
thanks,
Ryan Trauernicht
On Sun, Jan 11, 2009 at 4:18 PM, wafers44 wrote:
> AFAIK, the GK will load balance (random select) b/w the endpoints
> registered to the zone that you are trying to hopoff to. I don't believe
> there
AFAIK, the GK will load balance (random select) b/w the endpoints registered
to the zone that you are trying to hopoff to. I don't believe there's a way
to set a preference among the endpoints registered to the hopoff zone.
If you don't want CM to register w/ a TP, can you configure a TP on the GK
What type of hardware MTP are you using. You need to remember that some
older MTP hardware only does Xcoding and not MTP. I believe PVDM1's do not
do MTP... only Transcoding (even though it is called MTP).
6608 port can do hardward MTP.
thanks,
Ryan Trauernicht
On Sat, Jan 10, 2009 at 7:09 PM,
Erwin, You see below that you may have Channels 1-3 on the PSTN
configured... but the 6608 is still going bottom up be default and trying
channel 23...
Bearer Capability i = 0x8090A2
Standard = CCITT
Transfer Capability = Speech
Transfer Mode = Circuit
Transfer Rate = 64 kbit/s
Channel I
I agree with Ryan H. Only way that would work is if you checked the VM box.
Verify again that your IPMA CTI Route Point has the CSS for Managers and
not Everyone.
Call-forward no coverage is the one you need would need to be set to
managers CSS.
Thanks,
Ryan Trauernicht
2009/1/10 Ryan Hicks
>
That options seems like it would save alot of time. I have always just
pulled the MOH file from CM and used sound recorder to corp it to the delay
length. Then I play that prompt in a menu statement with option 0.
So the prompt is actually playing the MOH file.
No need for hold/menu/unhold
On
Prob a stupid question is there any way to set a preference command for
the hopoff command on a gatekeeper?
so I dont want CM to register with a tech-prefix but I want to use the
hopoff. I don't see a priority command like do you with a normal zone
prefix command.
thanks,
Ryan Trauernicht
The TCS was unchecked already. MTP required was unchecked. So no good.
Any other ideas?
Thanks,
Ryan Trauernicht
On Fri, Jan 9, 2009 at 9:14 PM, Vik Malhi wrote:
> Wait for H245 TCS on the trunk page within CCMAdmin should be unchecked.
> Let us know how you get on with that.
> --
> Vik Malh
I saw a very similar problem a while ago: when trying to call CUE from CCM via
SIP trunk, IOS cannot convert DTMF between RTP-NTE and SIP-notify
http://www.cisco.com/en/US/docs/ios/voice/cube/configuration/guide/vb-gw-sipsip.html#wp1342177
:"rtp-nte" to "sip-notify" seems to be unsupported.
I f
Hi all,
If you configure a SIP trunk to a router, and a FXS phone on that router: is
it normal the DTMF digits are passed along in the voice bearer channel once
you pick up the FXS phone (only in that direction: so you don't hear them if
you call an UCM phone with the FXS phone -using the same trun
You only need to refresh the application.
kapil atrish schrieb:
Nope. I've done it manier times, you don't need to restart anything.
*/Mike O /* wrote:
When you change to a different script in IPCC do you need to restart
any services?
Thanks,
Mike
__
Hi,
if you have clid and calling name restricted, then it is normally send
to the ISDN and one flag, I do not know which one exactly, in the Q.931
Message tells, that the Name needs to get stripped on the last ISDN
Switch, before delivery.
One Question does this issue occurs on proctorlabs o
Hi,
possibly someonee from Europe can asssist ;-)
All Western Europe Countries except United Kingdom, Iceland and Portugal
is UTC +1 during Winter +60 Minutes. During Summer it is +2, so +120
Minutes.
It depends on when you are looking and if you have configured Daylight
Saving Time.
PST
Have you got any calling number transformations configured on the
route list? If there is anything there then that would override the
route pattern setting.
Vik Malhi - CCIE#13890
Senior Technical Instructor - IPexpert Inc
Telephone: +1.810.326.1444
Fax: +1.810.454.0130
Mailto: vma...@ipex
Nope. I've done it manier times, you don't need to restart anything.
Mike O wrote: When you change to a different script in
IPCC do you need to restart any services?
Thanks,
Mike
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