>
> On Jul 5, 2021, at 8:11 AM, Abbas Wali wrote:
>
>
> could it be the phone firmware, i am using 12-5-1sr2-2
>
> On Fri, 2 Jul 2021 at 19:28, Myron Young wrote:
>
>> I have it working with an 8841 and we have both the SIP dial rule as well
>> as the blank
ds the xlate.
>
> Did you do both, or just one and then the other?
>
> On Fri, Jul 2, 2021 at 5:49 AM Abbas Wali wrote:
>
>> having issues with config. PLAR for a SIP 8851 phone. tried the blank
>> Part didnt work, tried the SIP dial rules with button or b
t;
> 2) Use AXL to create the end users instead of BAT
>
> Otherwise, if you don't hit the issue, then you can just BAT them in like
> normal.
>
> On Thu, Feb 18, 2016 at 12:37 PM, abbas Wali <abba...@gmail.com> wrote:
>
>> hi all,
>>
>> need to add aro
; *From:* cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] *On Behalf
> Of *abbas Wali
> *Sent:* Thursday, February 18, 2016 10:38 AM
> *To:* cisco-voip@puck.nether.net
> *Subject:* [cisco-voip] cucm9 insert Local enduser via bat AD sync enable
>
>
>
> hi all,
>
>
&
s and region settings or transcoders which will
> allow devices to talk to it at G711ulaw.
>
> On Tue, Feb 16, 2016 at 5:51 PM, abbas wali <abba...@gmail.com> wrote:
>
>> Yes indeed the system is set for G729 in the SParameters.
>>
>>
>>
>>
Yes indeed the system is set for G729 in the SParameters.
But if changed all the existing prompts in 729 will not play?
From: James Buchanan [mailto:james.buchan...@gmail.com]
Sent: 16 February 2016 14:29
To: abbas wali <abba...@gmail.com>
Cc: cisco-voip@puck.nether.net
Subje
> one of these in Audacity.
>
> --
> Andreas Sikkema
> ___
> cisco-voip mailing list
> cisco-voip@puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
--
*Abbas Wali*
___
cisco-voip mailing
ou can do it in Audacity under Other Formats when you export the file.
> However, I've never seen an option for G729.
>
> On Tue, Feb 16, 2016 at 10:17 AM, abbas Wali <abba...@gmail.com> wrote:
>
>> yes seen them but again they save it in g711's. also in the new audacity
>>
nified-call-centre-express-prompts-uccx-using-audacity/
>
>
>
>
>
> *From:* cisco-voip [mailto:cisco-voip-boun...@puck.nether.net
> <cisco-voip-boun...@puck.nether.net>] *On Behalf Of *abbas Wali
> *Sent:* Tuesday, February 16, 2016 7:09 AM
> *To:* James Buchanan <james.bu
Hi guys,
Just need a quick help here.
Every prompt I record ( via UnityC or Audacity etc ) upload and can only
hear gibberish.
But when I load an already saved file in G729 - it plays okay.
I have checked an my regions for phone dpool and application trigger are in
the same
Hi all,
I am a newish bee in the UCCX world.
currently need to write a script which will
- ring agent phone
- if not picked then put agent not ready mode.
( cant change the cluster wide as other need that )
- call queues and plays inqueue mesg
- after certain time it goes to VM etc.
any help
[mailto:ryanh...@outlook.com]
Sent: 24 November 2015 15:32
To: abbas wali <abba...@gmail.com>; cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] Jabber phone mode outbound calls issue
I assume this is only with the Jabber clients and not IP phones as well?
The annunciation message
[mailto:ryanh...@outlook.com]
Sent: 24 November 2015 18:30
To: abbas wali <abba...@gmail.com>; cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] Jabber phone mode outbound calls issue
Also and if it isn't too late, before you pull the SDL trace on the test
call, can you verify that the
Hi all,
Jabber phone only mode (10.5.2) is unable to make any outbound calls
including any internal calls even to reach the voicemail.
Inbound calls are working.
This is happening in CUCM 9.1
When dial anything , I get the "your call cannt be completed as dialled
please consult."
ee...@chemeketa.edu>
> (503)589-7776 <cseee...@chemeketa.edu>
>
>
>
>
> “Make space in your life for the things that matter, for family and
> friends, love and generosity, fun and joy. Without this, you will burn out
> in mid-career and wonder where your life went.”
>
This doesn’t make any sense ..
Created a new account with the same parameters for that agent/user and it does
login.
That’s absurd
From: abbas wali [mailto:abba...@gmail.com]
Sent: 15 September 2015 19:23
To: 'Ryan Huff' <ryanh...@outlook.com>; ealeather...@gmail.com
Cc:
com <mailto:ealeather...@gmail.com> >
Sent: Tuesday, September 15, 2015 01:36 PM
To: abbas wali <abba...@gmail.com <mailto:abba...@gmail.com> >
Subject: Re: [cisco-voip] UCCX 9 EM agents cant login
CC: Ryan Huff <ryanh...@outlook.com <mailto:ryanh...@outlook.com> >,Cis
Sorry that will mean !!
From: Ryan Huff [mailto:ryanh...@outlook.com]
Sent: 15 September 2015 15:18
To: abbas wali <abba...@gmail.com>; cisco-voip@puck.nether.net
Subject: RE: [cisco-voip] UCCX 9 EM agents cant login
Have is the subsystem in partial service?
_
From
Hi all,
Urgent issue here.
Ext Mob. Enabled agents cant login. Getting "Login failed due to a
configuration error with your phone and JTAPI or UCM. Contact your admin.."
The users profile is in the controlled list for RM application user.
The phone they are loggin in - is used by
figured the outcome this provides might meet the requirements of the user.
>
>
>
> Dan
>
>
>
>
>
> *From:* cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] *On Behalf
> Of *abbas Wali
> *Sent:* Wednesday, September 09, 2015 10:57 AM
> *To:* cisco-voip@puck
how to NOT play the
Sorry "user" is not avalible, record your message at the tone...
just want to play a personal recording OR a recorded name and take a message
this is for a system call handler
and have tried the untick [ Play the "Record Your Message at the Tone"
Prompt]
but it doesn't work.
Hi all,
basic question
how to send email notifications (O365 is setup already) to another
subscriber who already has got their own VM box.
so e.g.
VM1: 12345
VM2: 98765
both are setup with mailbox. VM1 when receives a VM, needs to send
notification via email to VM2 email box with the
hi all,
bit unclear about having a primary and secondary hubs for a single spoke in
ILS.
with us - there are ( or will be ) SMEs acting as Hubs for their region.
have seen a leaf can point to redundant SME in case of a failure
but if you run ILS on top of that - then similarly we can have a
Hi folks,
CUCM 9 and H323 gateways with ISDN channels to PSTN. We want to redirect to
VM when all the channels are full.
Can CUCM do it or Dpeer?
Thanks
___
cisco-voip mailing list
cisco-voip@puck.nether.net
if the random value is 0-2 or 3 and
use a redirect step to send the call to 2 different places based on which
random number comes up.
Brian
On Fri, May 22, 2015 at 11:26 AM, abbas wali abba...@gmail.com wrote:
Thanks Phil,
Its just calls coming into a particular number – calling party doesn’t
Hi all,
Is there a way to distribute ¼ of calls to one number/ddi and rest to a
different set of numbers.
CUCM 9 HP cant do that for me.
Anything in UCCX 9!!
TIA.
___
cisco-voip mailing list
cisco-voip@puck.nether.net
[mailto:philip.wale...@polycom.com]
Sent: 22 May 2015 16:14
To: abbas wali; cisco-voip@puck.nether.net
Subject: RE: [cisco-voip] 1/4 calls to DDI while rest to ext.
It would help if we had a little more understanding of what you are trying
to accomplish.
Are you trying to distribute calls ¼ and ¾ over
]
Sent: 22 May 2015 16:34
To: abbas wali
Cc: Walenta, Philip; cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] 1/4 calls to DDI while rest to ext.
use case might help us to understand...
But things that come to might are using UCCx for sending calls where you want
them to go. you
Thanks mate.
From: Anthony Holloway [mailto:avholloway+cisco-v...@gmail.com]
Sent: 15 April 2015 21:46
To: abbas Wali
Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] PER CALL BANDWIDTH CONSUMPTION OVER ETHERNET+802.1Q
I read somewhere that a phone could generate up to 2.5x call
To: Abbas Wali; 'Anthony Holloway'
Cc: cisco-voip@puck.nether.net
Subject: RE: [cisco-voip] PER CALL BANDWIDTH CONSUMPTION OVER ETHERNET+802.1Q
With BiB it is 3x your codec.
G.711 example:
1. 80k down far-end audio (remote party-current user)
2. 80k up current user audio
boot camps, and
medianet is not a part of the blueprint.
On Wed, Apr 15, 2015 at 4:22 AM abbas Wali abba...@gmail.com wrote:
hi all,
Vik Malhi posted that for a successful g711 call
HQSW(config-pmap-c)#police 90500 8000 exc drop ! 0 packet loss
now, as per Ciso medianet 4
The VoIP
more than a single call's worth of RTP to ingress the switch port, in
which case your 128kbps would not be enough and you would have issues with
things such as network recording or silent monitoring.
On Wed, Apr 15, 2015 at 9:19 AM abbas Wali abba...@gmail.com wrote:
medianet is
http
.
many thanks
--
*Abbas Wali*
___
cisco-voip mailing list
cisco-voip@puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip
On Oct 6, 2014, at 9:05 AM, Bala Singaram mmailb...@gmail.com wrote:
Yes you are right.
On Mon, Oct 6, 2014 at 3:57 AM, abbas Wali abba...@gmail.com wrote:
thanks Bala,
so you have to upload/install that to all the nodes/subs and no reboot
required. I have also just seen a cop file
:
Hi Abbas,
Install the pack in PUB first then other nodes [ SUB ], no need to reboot
the server, since the device pack will be active version itself.
Regards,
Bala
On Mon, Oct 6, 2014 at 3:31 AM, abbas Wali abba...@gmail.com wrote:
hi all,
added new Device Pack for 7800 SIP phones
Gary,
and upload/install the DPack on all of them right.
how about just going for the COP file option?
On 6 October 2014 12:31, Gary Parker g.j.par...@lboro.ac.uk wrote:
On 6 Oct 2014, at 11:57, abbas Wali abba...@gmail.com wrote:
thanks Bala,
so you have to upload/install that to all
12 Cadillac Drive Suite 130, Brentwood, TN 37027
D: 615.866.5729 | C: 931.797.2326 | F: 615.866.5781 |
jbucha...@presidio.com
PRESIDIO
Practical thinking for a connected world.
Follow Us: www.twitter.com/presidio
From: abbas Wali [abba
ǀ BLOCK ǀ
Cisco Global EMEAR Partner of the Year 2013
*From:* abbas Wali [mailto:abba...@gmail.com]
*Sent:* 11 February 2014 11:01
*To:* Matthew Collins
*Cc:* Buchanan, James; cisco-voip@puck.nether.net
*Subject:* Re: [cisco-voip] dial into ddi and get a secondary dial tone
to call
38 matches
Mail list logo
