Thanks Nick! This clears up the mystery of why the router was behaving
different from default.
-sreekanth
From: Nick Barnett
Sent: Tuesday, April 27, 2021 9:11 PM
To: Sreekanth Narayanan (sreenara) ; cisco-voip
Subject: Re: [cisco-voip] Outbound SIP connection
Nick,
What's the disconnect cause from the CUBE? 102?
Do you have logs for this call? Would be clear which timers are expiring,
causing the problem.
debug ccsip message
debug ccsip error
debug ccsip info
-sreekanth
From: cisco-voip on behalf of Nick Barnett
S
Hey All!
I'm reaching out to understand how call issues are troubleshot today end-to-end
within an enterprise. I wanted to get a sense of whether you all use any
applications that act as Central Logging servers for all the applications that
provide Collaboration services.
For example, CUCM, UC
This behavior sounds as though the phone thinks it’s registered to the CME, but
isn’t..
What are the outputs of these commands?
show voice register pool all brief
show voice register global
show voice register pool
Also, take a packet capture from the CME if you can to see if any SIP packets
ar
Yes, VXML is no longer supported on the ISR4K so these routers cannot be used
in a CVP deployment as VXML gateways.
The VVB is the alternative.
I’m not sure the fax TCL script has been tested on ISR4K, but BACD has been
tested and works fine.
Regards
Sreekanth
From: cisco-voip On Behalf Of Le
Yes this is tested in BE4K. BACD is implemented for queuing.
This script should work fine.
https://software.cisco.com/download/home/286322812/type/282786567/release/3.0.0.6
The bug wasn’t reproducible in the customer environment.
Regards
Sreekanth
From: cisco-voip On Behalf Of Brian Meade
Sent
This happens when the CUBE receives the prompt from the ITSP over RTP and is
not aware of where it has to stream this prompt because CUCM does Delayed Offer
by default. So Cube doesnt have the phone's IP address.
You can fix this by forcing CUCM to send provisional acknowöedgement by:
1. putting
What IOS version are you running on the CUBE? I can think of a couple of things.
1. In 15.6(2)T, a new feature has been introduced called multi-tenant where you
can configure separate voice class tenants. Each tenant can have separate
authentication mutually exclusive to one another and can be bo
Hey Scott,
Unfortunately, there is no way you can configure the RTMT or syslogs to catch
slips on the lines. You would need something along the lines of an EEM applet
running on the router which runs the show controller t1/e1 x/y/z output at
regular intervals and looks for increases in the slip
?The BACD script has not been updated for Voice Hunt Groups yet. This is in the
roadmap for July 2016.?
Thanks
Sreekanth
From: cisco-voip on behalf of Brian V
Sent: Monday, January 4, 2016 8:44 AM
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] CME - b
Hi Frank,
Based on the working faxes between the VG224s and the RF, we can say that the
network is stable. And you don’t seem to have seen the RTN with this flow.
We can also say that both devices in this call flow work fine.
Now when you make calls from the VG224s to the PSTN, the fax machines
Try this.
voice class sip-profiles 1
request INVITE sip-header allow-header modify ", UPDATE" ""
apply this profile under voice service voip -> sip, or under the dial-peer
pointing to the ITSP.
/S
From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Ahmed
Abd EL-Rahman
Se
Hey Guys!
Just a follow up, the video is now on Youtube.
https://www.youtube.com/watch?v=HkPjTBvx_YA
Sreekanth
From: Sreekanth Narayanan (sreenara)
Sent: Tuesday, November 17, 2015 1:20 PM
To: cisco-voip@puck.nether.net
Subject: Video on PCM captures, Packet captures and DS0 Dumps on IOS
: Sreekanth Narayanan (sreenara)
Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] 7925s and 7921s losing registration
They were never on a secure cluster... it was 1.4.4.3 before... the issue was
there before. We upgraded to try to resolve this issue...
Jonathan
On Thu, Nov 19, 2015 at 12:33
Did you start seeing this issue only after upgrading the firmware?
What was the firmware they were running before?
Were these phones moved from a secure cluster to this one?
From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of
Jonathan Charles
Sent: Thursday, November 19, 20
Hey Guys!
I created a video on CSC for instructing customers on taking PCM captures,
Packet captures and DS0 Dumps on the IOS routers.
Here's the link for this video. http://cs.co/6180BuNir
Please share this with people who can benefit from the video! These captures
can be used in troubleshooti
Thanks Dan! Glad you liked the video!
Sreekanth
From: Daniel Pagan [mailto:dpa...@fidelus.com]
Sent: Monday, November 16, 2015 2:14 AM
To: Ahmed Abd EL-Rahman ; Sreekanth Narayanan
(sreenara) ; cisco-voip@puck.nether.net
Subject: RE: call tracking question
Sreekanth: Really enjoyed your LUA
You'll firstly need to find the call using the called or calling number, and
then you can track it using the Global call identifier in the CCAPI logs.
For example, in this line:
8261462: Sep 29 14:31:10.720: //640449/14CF2490BDF1/CCAPI/cc_api_call_connected:
Call Entry(Connected=TRUE, Responsed
ange 2010 shortly
>
>
> Jorge Rodriguez, CCNP, CCNP-V
> Senior Voice/Data Network Consultant
> Netxar Technologies, a Digicel Company
> Cel 7876888530
> Office 7877650058
> jorge.rodrig...@netxar.com
>
>
>
>
> On Sep 24, 2015, at 9:08 AM, Sreekanth
I'm guessing you're using Unity?
www.cisco.com/c/en/us/td/docs/voice_ip_comm/unity/7x/using_exch2010/usingex2010withcu7x5x.pdf
On 24 September 2015 at 18:33, Jorge L. Rodriguez Aguila <
jorge.rodrig...@netxar.com> wrote:
> Is this basically breaking whats done and starting over or is there som
You could switch on 'debug vpm signal' and log the debugs to a syslog server or to the buffer. You'll need to increase the buffer size. Another trace that you can use after the call completes is 'show voice trace
Hi Quenten,
Where does the BACD script fail? Are you able to hit the application and
hear the prompts at least?
Could you share your configuration?
Thanks
sreekanth
On 8 July 2015 at 09:50, Quenten Grasso wrote:
> Hi Everyone,
>
>
>
> I’m trying to setup an Auto Attendant on a Cisco Call mana
Hey Euan,
I don't think that's possible. I'm not sure where this information is
present.
In the datasheet, it does say that you can mount the camera on a PC and
connect it to the phone via an extended usb cable. It doesn't mention
directly connecting this camera to the PC.
http://www.cisco.com/c/
It should be ok, but you cannot keep a live audio source on the router,
multicast that, and expect the cube component on the same router to listen
to this stream in a hold scenario.
On 28 March 2015 at 17:27, Barry Howser wrote:
> I am migrating a customer's PRI off an old unsupported 3745 route
Hi Shabbar,
What is the CUCM version?
So the users go into Inactive mode every 6 hours? Or once everyday? If
once, what time does that happen and is that during a sync?
Have you taken a look at the DirSync logs during the period of failure?
What about a packet capture to see if this could be an i
Martin,
Is there a reason you'd like to do this? What is the use case?
By default, the older traces will get overwritten when there is no space
left for newer traces.
You can manually delete the files from the CLI or the Remote Browse on RTMT.
Thanks
Sreekanth
On 23 September 2014 15:43, Martin
offsite - will probably just move the whole call tree to UCCX for IVR at
> that point.
>
>
> Thanks for your ideas!!
>
> Ed
>
>
> On Mon, Sep 1, 2014 at 3:04 AM, Sreekanth Narayanan
> wrote:
>
>> Hi Ed,
>>
>> Could it be possible that the answerin
Hi Ed,
Could it be possible that the answering service does not answer the call
sometimes? This may be causing the timeout.
>From the POV of the CUCM, this is just another regular outgoing call over
PRI starting from a VM port (which acts like an SCCP phone), which would
then do the transfer to th
Hi Cos,
I'm unable to see an attachment in the mail. Not sure if it got filtered at
my end.
Voice quality on this phone has been seen on the 9.3 firmware by an other
customer too. I suggest moving to 9.4(1), where there have been
improvements in this component on the phone and checking if you are
Hi Dave,
1. I'd take packet captures from the switch where the ATA is connected to
find out what it's doing. Whether it reaches out to the tftp server for its
configuration file. Then check if it downloads its load and contacts the
CUCM over SIP.
2. You can also manually set the IP, tftp and defau
Also, what's different about this device? Have you confirmed with the
customer that this is the only device that's facing the problem.
If the call in made to the device instead of from it, does the other side
still see the issue?
Sreekanth
On 30 June 2014 22:21, Ryan Ratliff (rratliff) wrote:
You'll have to install the new locale for the same country on 8.6. Since
there are separate locale files for each version, there could be
differences, causing incompatibilities.
http://software.cisco.com/download/release.html?mdfid=283782839&flowid=45898&softwareid=282074333&release=8.6(2.3000-1)&r
M and check the
following box, and try the call again.
Early Offer support for voice and video calls (insert MTP if needed)
Thanks
Sreekanth
From: Dana Tong [mailto:dana_t...@bridgepoint.com.au]
Sent: Wednesday, April 30, 2014 2:01 PM
To: Sreekanth Narayanan (sreenara)
Subject: RE: One-way audio after
Hi Dana,
Have you tried to take packet captures from the back of the phone and see if
the stream is reaching the phone from the remote-end?
If there is no MTP/Xcoder invoked, you will see the RTP stream coming directly
from the CUBE (assuming that cube is in flow-through mode).
If the SIP trace
What's this dial-peer below?
dial-peer voice 3000 voip
destination-pattern 3...$ <<< is this r8 or some other flavor?*
dtmf-relay h245-alphanumeric
no vad
I'd try 2 things:
1. create
voice class codec 1
codec preference 1 g729r8
codec preference 2 g729br8
codec preference 3 g711ulaw
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