I thought that this command " param aa-hunt3 3210 " meant that it specifies
option number 3 for hunt group 3210.
Am I wrong?
I have the BACD configured but no matter what I configure the mentioned
command as, the queue still recognizes 1 for the huntgroup and says 3 is
invalid.
application
serv
Calls to BR1 physical phone 2001 or 2002 fail but
calls to 2004 ipblue phone with same
location/aargrp/css-aar etc.
here is the debug -
fialed call to BR1 2001
P29-BR1-RTR(config)#
Feb 21 01:59:00.032: ISDN Se0/0/0:23 Q931: RX <- SETUP
pd = 8 callref = 0x0015
Bearer Capability i = 0x
when you use the "service session" can you get an active call and enter the
command "sh call act voice brief" and find out what codecs and what
protocols are being used. We'll have a better idea of what is going on when
you post the answer to this question.
The service session command invokes the
Hi Vik:
Thanks for your answer, to make the things clear, if I have different codecs
one of the legs shoud be H323, wright?, however if I use G711 in both legs,
then I can use sip-to-sip, wright?.
The other thing is, with the configuration I sent you adding the command
"service session" it
you cannot sip - sip. One leg must be H323. You must g711 on the inbound and
outbound sip call legs to make this work.
Vik Malhi - CCIE #13890, CCSI #31584
Sr Technical Instructor - IPexpert, Inc.
A Cisco Learning Partner - We Accept Learning Credits!
Telephone: +1.810.326.1444
Fax: +1.810
Thanks, i'll try that tomorrow.
JD
From: [EMAIL PROTECTED]: [EMAIL PROTECTED]: Wed, 20 Feb 2008 13:55:01 -0800CC:
[EMAIL PROTECTED]: Re: [OSL | CCIE_Voice] VM in SRST mode
I think your translation pattern should be changed- you want to set the fwding
# to 2003 no the calling #. so change
Hi Vik: Yes it is just for the calls coming from the WAN, What I am seeing is
the CME sends a SIP 302 message to the IPIPGW, and the IPIPGW shoud send a
re-invite to the number of the CUE, however the IPIPGW sends an ACK and I heard
a busy tone, attached are the configurations of the IPIPGW and
I think your translation pattern should be changed- you want to set the
fwding # to 2003 no the calling #. so change the called # mask- ccm then
tries to ring 2003 and since this is not registered it fwds to VM using 2003
as the fwding #
current:
translation pattern: 1773
Calling Party Tran
Does the call-forward work from the PSTN?
If it is specifically the calls from over the WAN that do not fwd then I go
along with DevilDoc and Jason- transcoder or incoming dial-peer not matching
correctly.
If it doesn't work for g711u end to end, then that potentially rules out the
xcoder as be
Sounds like your dial-peers are not engaging.
Tried a reboot yet?
On Wed, Feb 20, 2008 at 3:37 PM, Jose Linero Welcker <
[EMAIL PROTECTED]> wrote:
> Hi Jason:
>
> Thanks for your answer, yes I have a Txcodec configured, and I tried
> configuring the cinming dial-peer with G711ulaw and still rece
Hi Jason:
Thanks for your answer, yes I have a Txcodec configured, and I tried
configuring the cinming dial-peer with G711ulaw and still receive the SIP
message:
Feb 20 21:37:13.366: //-1//SIP/Msg/ccsipDisplayMsg:Sent: SIP/2.0
302 Moved TemporarilyVia: SIP/2.0/UDP 162.1.103.1:5
You might have a codec issue here.
Do you have transcoder configured on the cme router.
If you do, try to configure the sip incoming dial-peer with "codec g711ulaw"
On Wed, Feb 20, 2008 at 3:29 PM, Jose Linero Welcker <
[EMAIL PROTECTED]> wrote:
> Hi JD:
>
> Actually I have it:
>
> dial-peer vo
Hi JD:
Actually I have it:
dial-peer voice 4 voip session protocol sipv2 incoming called-number 3...
dtmf-relay rtp-nte
Any ideas?
Regards,
Jose
From: [EMAIL PROTECTED]: [EMAIL PROTECTED]; [EMAIL PROTECTED]: Wed, 20 Feb 2008
13:22:47 -0800Subject: Re: [OSL | CCIE_Voice] CUE FNA FB Pr
You are missing an incoming dialpeer for your g729 voip connection on your cme
router. Either create a dedicated incoming dialpeer for your g729 voip
connections or you can create a general dialpeer like the one below.
dial-peer voice 100 voip
incoming called-number .
JD
From: [EMAIL PROTE
Hi all:
I have this configuration:
CM --- H323 Trunk --- IPIPGW --- SIP Trunk --- CME
The H323 is using G711 and the the SIP Trunk is using G729, the calls between
the phones in CCM and CME are working without problems, however when I am going
to forward the call to CUE due CFNA or CFB I r
Hi Vik,
For the workaround of the SRST and RDNIS IE issue, i created an alias command
for each DN under the call-manager-fallback. In addition, i also created a CTI
route point to catch all incoming numbers from the alias commands and use a
custom voicemail profile to map the route point DN
I would also bring awareness to the other solution - mapping each DN to a
unique Call forward DN via the alias command and translating on the
CallManager. Please see my post on Feb 18th.
In the real world, if you have a PRI then I guess the best solution would be
to put pressure on your provider t
you can try clearing the telnet session.
service-m servie-en 0/0 sess clear
-Senthil
On Feb 20, 2008 8:19 AM, Olson, Pete <[EMAIL PROTECTED]> wrote:
> I attended Vic's vLecture on CUE. One of the recommendations was to
> reset the CUE to factory defaults. I tried this in a lab session. I
> coul
I attended Vic's vLecture on CUE. One of the recommendations was to
reset the CUE to factory defaults. I tried this in a lab session. I
could no longer talk to the CUE afterward. I would get a connection
refused message. Any suggestions on how to establish communication if I
run into this again? An
In addition to what Vik memtioned, also make sure you have a specify
dial-peer that matches the VM number exactly (no 9) and does a
'forward-digits all'
Mark Snow
Sr Technical Instructor
IPexpert, Inc.
Sent from my iPhone
On Feb 19, 2008, at 11:41 PM, anil batra <[EMAIL PROTECTED]> wrote:
Try entering this command in the cmr router (not into CUE):
no aaa new-model
Also please send your router and cue config.
Mark Snow
Sr Technical Instructor
IPexpert, Inc.
Sent from my iPhone
On Feb 19, 2008, at 10:18 PM, DSCP46EF <[EMAIL PROTECTED]> wrote:
Hi Guys,
Can someone point me in
I believe vm-integration is required in any case where RDNIS doesn't work.
-matt
ccievoice1 wrote:
Hi Vik,
Can I assume during "*normal*" srst failover, vm-integration is not
required. However, due to IOS bug for a certain IOS version,
vm-integration is required. Am I right to say that?
Th
Hi Vik,
Can I assume during "*normal*" srst failover, vm-integration is not
required. However, due to IOS bug for a certain IOS version, vm-integration
is required. Am I right to say that?
Thanks.
On Wed, Feb 20, 2008 at 1:19 PM, anil batra <[EMAIL PROTECTED]> wrote:
> yes that's correct..1599
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