Hi Lisa,
There is a company based in India that I have used in the past that I
believe have a product that meets your requirements.
http://www.parsec-tech.com/ProjectDocs/MARS%20Panic%20Button.pdf
I haven’t used this specific product before but used another (silent
monitoring) and was very
Hi Jonatan,
We are doing it for one of our customers I found the below data sheet
really useful especially in regards to what features are currently
available to SRST (based on your IOS version)
http://www.cisco.com/c/en/us/products/collateral/unified-communications/unified-survivable-remote-site
Just ran into this one
*Topology:*
8845 > CUCM 8.6 > H323 GW > T1 > PSTN
Desc of problem: Video capable handsets cannot make outbound calls where as
non video phones work fine. Calls are passed to h323 GW OK
*Troubleshooting steps:*
Devices have same Class of service/device pool/within su
tually did want to use video for onnet calls or through
> trunks but do support video payloads.
>
> Thanks,
>
> Ryan
>
> On Mar 31, 2017, at 3:26 PM, Ryan Huff wrote:
>
> conf t
>
> (config)# voice-port0/0/0:23
>
> (config-voiceport)# bearer-cap speech
>
> ex
7 at 4:04 PM, Nick Britt
> wrote:
>
>> Funnily enough the clients requests were,
>>
>> - get outbound calls working
>> - disable video calls
>>
>> if only I had disabled calls first.
>>
>> Cheers Ryan.
>>
>> On Fri, Mar 31, 2017 at 1
Hi Anthony,
Sorry to grave dig but just wondered if you ever got an answer for this?
I am about to go down this rabbit hole myself as a customer wants a better
explanation (documentation) as to how this is supposed to be configured
and how this should behave.
Cheers
Nick
On Wed, Jun 15, 2016
Hi Joe,
Can I ask which version of CUCM you are using? there are some known issues
with DTMF in 10.5.2
On Mon, Aug 28, 2017 at 10:40 AM, Brian Meade wrote:
> CUCM will try to negotiate inband (RFC2833) and out of band.
>
> I don't believe you can edit length of tone or on/off times for RFC2833.
Buy cheap, Buy twice.
If you are committed to getting the cert, go with Vik (in person if
possible), it will save you a huge amount of time/money in the long run.
On Thu, Apr 4, 2019 at 5:49 AM Benjamin Turner
wrote:
> Also,
>
>
>
> Kevin Wallace is soon to release the Collab V2 video and workb
Hi Ryan, I am no QOS expert either but I think you will need to match the
protocol RTP Audio, to pickup any packets that haven't been marked with EF,
although I think the standard phone template in CUCM does this.
class-map match-all VOICE
match protocol rtp audio
On Sat, May 2, 2015 at 11:55 P
class-map match-any CALL_SIGNALLING
match protocol skinny
match protocol h323
match protocol sip
!
class-map match-any VOICE
match access-group name VOICE_TRAFFIC
match protocol rtp audio
On Mon, May 4, 2015 at 9:13 AM, Nick Britt wrote:
> Hi Ryan, I am no QOS expert either but I th
I have used redbox voice recording, previously and was very impressed with
it. The end user interface is great, which is good for us (less training)
On Fri, Jun 19, 2015 at 4:07 AM, Scott Voll wrote:
> We have had telrex callrex now enghouse QMS for about 2+ years. System
> works with forked au
Seconded brilliant product. Pity it can't fix the responsiveness of the
8945/9951's
On Wed, Sep 30, 2015 at 12:44 AM, Terry Oakley
wrote:
> For those of you looking at UnifiedFX I will add my experience. First it
> is limited experience as we have only been using the product for 6 months.
> So
to clarify the product works fine with 9971's and 8945's the phones are
just slow to respond. like in real life.
On Tue, Oct 6, 2015 at 7:32 AM, Nick Britt wrote:
> Seconded brilliant product. Pity it can't fix the responsiveness of the
> 8945/9951's
>
>
Sorry to Grave dig here, trying to configure CPA on UCCX on 10.6(1.1.1)
with a CISCO2921/K9 on Version 15.4(3)M4, RELEASE SOFTWARE (fc1).
It appears CSCui62525 was resolved on our version of UCCX but we tried
removing RTP-NTE from all dialpeers with no luck (no update messages being
sent from
Didn't try VAD maybe should have though! Resolution below, in short we
changed the trans-coder to universal (not sure this helped the issue)
forced TCP in both the SIP gateway within UCCX and on the incoming
dial-peers then also removed RTP-NTE DTMF on the dial-peers.
Here is a summary:
*Proble
ever do g711 out to the PSTN then? And your ITSP was onboard
> with no relay?
>
> Was the TCP vs UDP requirement documented somewhere and you just missed it
> originally, or was that a shot in the dark to try TCP?
>
> Thanks again for sharing your journey.
>
> On Thu, Oct
Hi There!
Our customer is using the outbound progressive CUBE dialer using Finesse as
the front end and UCCX 10.6.
When using the Preview dialer the wrap-uptimers seem to take affect but
when using the progressive dialer no wrap-up times are received the next
call the agent goes straight to "not
AU/>
>
> Think before you print.
>
> This email may contain confidential and privileged material for the sole
> use of the intended recipient. Any review, use, distribution or disclosure
> by others is strictly prohibited. If you are not the intended recipient (or
> autho
+1 plus for unified fx saved my bacon numerous times ;-)
On Friday, 30 October 2015, Charles Goldsmith wrote:
> Variphy (variphy.com), and it comes with a lot of other very handy tools.
>
> On Thu, Oct 29, 2015 at 4:51 PM, Scott Voll > wrote:
>
>> So what vendor do you use for Remote phone cont
e Abhiram!
On Fri, Oct 23, 2015 at 5:47 PM, Nick Britt
wrote:
> Thanks for the explanation Abhiram very well detailed. I havent had much
> experience with progressive diallers the client maintains the position that
> this is possible with other vendors progressive diallers but I cannot see
Hey Guys,
Sorry to grave dig, doing an upgrade to 10.6.1 su1 from 9.0.2 su1 (via
9.0.2 su3) is there anyway to obtain CAD 10.6.1.1057 in advance? Need to
publish this Citrix.
On Fri, Oct 2, 2015 at 10:01 PM, Jason Aarons (AM) <
jason.aar...@dimensiondata.com> wrote:
> It would be really nice i
wrong time and wrong hemisphere
But admin's I for one would like to see more of this.
On Tue, Apr 26, 2016 at 6:37 AM, Horton, Jamin
wrote:
> Hello everyone,
>
>
>
> I am currently seeking a UC Collaboration Deployment Engineer. We are a
> large Cisco VAR and the opportunity is based in the De
Hi Ed,
Only ever used the non Video Dallas Delta's which seem to work quite well.
Although this appears to do what you are asking but havent used them
personally.
http://www.2n.cz/en/products/intercom-systems/ip-intercoms/helios-ip-vario/tech-specs/#product-content
On Tue, May 17, 2016 at 11:53
Sorry does the 81 get removed or its not added?
I am assuming the call is blindly transferred and then not answered?
Are you using the remote number field under the device?
I will be honest I haven't tested transferred calls while using the remote
number calling transformation approach but I hav
Hey There!
I have a requirement to force an FXO line into the Busy/offhook state from
the CLI.
The provider has a round robin hunt group across a few FXO's but in
previous testing we have realised that when the port was in a "shut" state
the Telco still sent calls to the disabled ports.
Only whe
*Got it.*
*!voice-port 0/0/0busyout forced!*
*who needs google when you have "sh run all | sec" *
On Wed, Jun 1, 2016 at 2:34 PM, Nick Britt wrote:
> Hey There!
>
> I have a requirement to force an FXO line into the Busy/offhook state from
> the CLI.
>
> Th
Hi David,
Can I ask Which version of IOS you are using?
Also could you post your incoming dial peer configuration or are you just
using the default DP 0?
Ive experienced a similar issue before (luckily I didn't configure this
particular deployment)
Before IOS 15 (I believe) direct in ward dial
ated with the line plugged into your router, you'd get dial tone and
> from there you could dial an number the dial plan allowed.
>
>
>
> Sent from my iPhone
>
> On Sep 11, 2016, at 11:49 AM, Nick Britt > wrote:
>
> Hi David,
>
> Can I ask Which version of
Hey Guys/Gals,
I seem to be having a problem with an install of Cisco IP Communicator
(8.6.3.0) not picking up the auxiliary Voice VLAN configuration.
In short we have the option 150 details listed on the voice vlans and not
the data and I keep receiving a TFTP server not found error. But wh
rian
>
>
> On Tue, Apr 22, 2014 at 12:53 PM, Nick Britt wrote:
>
>>
>> Hey Guys/Gals,
>>
>>
>>
>> I seem to be having a problem with an install of Cisco IP Communicator
>> (8.6.3.0) not picking up the auxiliary Voice VLAN configuration.
>&g
Morning/Afternoon,
We have a need for calls to ring a shared line (unanswered) for up to 15
minutes.
CUCM 9.1.2 SIP enviroment 8945/9951 endpoints.
We are currently are seeing disconnected on internal (SIP to SIP calls)
after 3 minutes and 2:30minutes for external calls (E1 > MD110 > MGCP >
CUCM
nnect the call if it's coming from CUCM.
> Same thing for your MGCP scenario.
>
> Brian
>
>
>
> On Thu, Jun 12, 2014 at 5:14 AM, Nick Britt
> wrote:
>
>> Morning/Afternoon,
>>
>> We have a need for calls to ring a shared line (unanswered) for up
gt;
> Min-SE: 1800
>
> User-Agent: Cisco-CUCM8.6
>
> Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER,
> SUBSCRIBE, NOTIFY
>
> CSeq: 101 INVITE
>
> Expires: 180 <
>
>
> *From:* cisco-voip [mailto:ci
Hi Jason,
I have tried both ways on two separate occasions. Unfortunately I needed
two to get the result I needed (one in and one out). The Dial peer seems to
complete the first action then ignore anything else (regardless of its
positioning)
This was on 152-4 and 15.1
dial-peer voice 1 voip
d
I just did a deployment of around 2000 8945's I have tested the bluetooth
functionality which seems to work fine.
No real issues with them overall they are a tiny bit slower than the 796x
series. Overall I was pretty happy with them only 1 DOA out of the 2000. We
hadn't enabled video yet but I rec
uplinx is brilliant, I am sure Stephen Welsh is skulking around here
practicing his modesty. I will continue to push customers to buy it.
On Sat, Aug 3, 2019 at 12:46 AM Fares Alsaafani wrote:
> Hi Matthew, I have used remote control for Cisco phone software was great
> saved my day on remote si
Hey Ryan - sorry to grave dig.
Can you remember the reference for the common phone profile by chance? We
have 50 or so ATA's that we will need to replace with 191's in the meantime
i'd like to try the common phone profile change.
Was it just the name of the common profile or is it should I change
A customer has had a domain name, this includes the DNS and the active
directory integration. I am trying to pull together the necessary steps for
each application.
Below is what I have deduced from the documentation so far
*Change Domain name CUCM, Pub and Sub*
The CUCM processNode name is the
gt;
> On Nov 11, 2019, at 18:21, Nick Britt wrote:
>
>
> A customer has had a domain name, this includes the DNS and the active
> directory integration. I am trying to pull together the necessary steps for
> each application.
>
> Below is what I have deduced from the docum
What's the call volume? is MTP required ticked on the CUCM SIP trunk?
In RTMT can you check your software MTP resources?
On Fri, Jan 17, 2020 at 1:48 PM Jonatan Quezada <
jonatan.quez...@chemeketa.edu> wrote:
> UCCX version: 10.6.1.11003-29 (ES02-11)
>
> CUCM Version: 11.5.1.15900-18
>
>
> calle
UCM.Now if you see finesse
>> send a CTI command to hang up the phone, thats a different issue.
>>
>> Most of the time on answer drops can be a sign of codec mismatch.
>>
>> Post some cucm logs, with call examples.
>>
>> On Jan 17, 2020, at 2:52 PM, Nic
Hi
What sort of testing, load testing?
Chrs
On Wed, Jul 8, 2020 at 7:54 AM UC Penguin wrote:
> I’m curious if anyone has any experience with SIP testing tools.
>
> I’m looking for a tool to be able to script testing valid configurations.
> Ex. Does MS Teams actually accept a call to this URI o
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