What load on the 51s and is that load version present on all nodes?
Thanks,
Ryan
Original Message
From: Jonathan Charles
Sent: Thursday, April 23, 2015 08:44 PM
To: Ryan Huff
Subject: Re: [cisco-voip] Cisco 8851 not failing over to backup circuit...
CC: cisco-voip
Could also be a CTL issue if your doing secure ...
Thanks,
Ryan
Original Message
From: Ryan Huff
Sent: Thursday, April 23, 2015 08:48 PM
To: Jonathan Charles
Subject: Re: [cisco-voip] Cisco 8851 not failing over to backup circuit...
CC: cisco-voip@puck.nether.net
>What l
Are you able to see the physical phone during the event? What is the physical
phone's behavior?
Does the physical phone show un regestered or does the phone show the new
ccm/tftps addresses eventhough it doesn't register?
Whst load version on the 8851's?
Thanks,
Ryan
Original Messa
So you can add the node without causing a phone reboot. Just keep it out of the
ccm group in the cluster. Then add it in during a maintrnance window.
Thanks,
Ryan
Original Message
From: Tommy Schlotterer
Sent: Monday, April 27, 2015 02:07 PM
To: cisco-voip@puck.nether.net
Sub
I'll have to lab this up and see what I am missing; last week I added a 9.1.x
cluster node and I didnt get any phone bounces during the install.
Strange.
Thanks,
Ryan
Original Message
From: "Jason Aarons (AM)"
Sent: Monday, April 27, 2015 02:54 PM
To: Brian Meade ,Tommy Schl
The below OUT map, applied in the output direction on a WAN(mpls) facing
interface, should put RTP, Signaling and anything from access-list 51 at the
top of the heap and give everything else best effort.
Not that anything isn't working, I just want to make sure I'm not making
something up ... e
-51 - I couldn't tell from
your initial email if that was your intent. If you want just either type of
traffic to get the VOICE treatment then you need match-any.
Also agree with John re: using priority instead so that it kicks in LLQ for
those packets.
On Sat, May 2, 2015 at 9:55 AM, Ryan
I have a service policy-map that I am trying to automate when it is actually
applied to the interface. So between the hours of 8AM and 5PM, I want to apply
this service policy to the interface and between the hours of 5PM and 8AM I do
not want the service policy applied.
The policy application
I have a situation, where, in some case the ingress pstn call leg is trying to
use g.729 (when there is nothing in the gateway that would indicate it's
preference).
Call path when G729 is negotiated:
PSTN -> h.323(PRI)
dial-peer match -> SIP trunk to Unified Proxy Server - > SIP
Customer Voic
on the H.323 GW to see if it's offering anything
other than G.711ulaw in the Invite to CUSP? If it's only G.711 there, it must
be getting changed by CUSP. I'm not familiar enough with it to know what to
check though.
On Wed, May 6, 2015 at 5:40 PM, Ryan Huff wrote:
I have a s
issues when an incoming call wasn’t
properly matching a dial peer – it was hitting a “default” which I believe was
g.729. I’d verify it’s actually hitting the dial-peer you think it should be.
From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Ryan
Huff
Sent: Wednesday, M
Might be easier to start the other way around.
What is actually wrong?
RTMT does work with registered phones, and presumably, your wireless phones
are registered?
Are the wireless phones not registering to call manager?
Thanks,
Ryan
Original Message
From: Scott Voll
Sent
So if ccm doesnt show phone registration AT ALL for your wireless phones, and
auto registration is enabled/works or you have pre-built devices in ccm for the
phones, I would start troubleshooting you're wireless gear.
Thanks,
Ryan
Original Message
From: Scott Voll
Sent: Thu
e to very
different things.
Thanks,
Ryan
Original Message
From: Scott Voll
Sent: Thursday, May 7, 2015 05:46 PM
To: Ryan Huff
Subject: Re: [cisco-voip] RTMT??? or some other way to monitor a phone
CC: Brian Meade ,cisco-voip@puck.nether.net
>my 8861 phones are working wirele
lpful in either
eliminating or proving the router as the source of the issue?
As I understand it, this should allow sdp to pass to the other call leg without
media negotiations.
Thanks,
Ryan
Original Message
From: Brian Meade
Sent: Wednesday, May 6, 2015 05:54 PM
To: Ryan Hu
nether.net
That may help but it's probably easier to just start with the "debug ccsip
messages" to see what the router is sending in the outgoing Invite to CUSP.
On Thu, May 7, 2015 at 7:50 PM, Ryan Huff wrote:
Brian,
I'm guessing that the router in this case is somehow, messi
That's what I was reading. I worked this out with TAC, Luis (a pretty sharp IE)
is trying to raise a bug evaluation.
Thanks,
Ryan
Original Message
From: Brian Meade
Sent: Friday, May 8, 2015 01:47 PM
To: Ryan Huff
Subject: Re: [cisco-voip] Codec negotiation issue, a l
Codec transparent just passes sdp through to the other call leg without trying
to do media negotiations.
So without codec transparent, what happens?
Thanks,
Ryan
Original Message
From: s m
Sent: Sunday, May 10, 2015 01:19 AM
To: cisco-voip@puck.nether.net
Subject: [cisco-vo
The reason that is happening is due to media negotiation failure as you mention
(both call legs are not offering the same codec capabilities). In that exact
configuration, you would need a transcoder (which you could run on the router
if you have enough DSP).
Are you sold on h323 or can you do
You can try this technique
http://ryanthomashuff.com/2014/11/call-blocking-by-caller-id/ which uses the
"route next hop by calling party number". Works well for ANI call blocking; I'm
using it to keep spam numbers off of uccx call center triggers. You have to
apply it at the top of the dial pla
Right, flip all incoming into a filter partition that 'screens' ani and let
everything pass that doesnt match 'bad' ani patterns is how I do it too.
But with ani spoofing, it is becoming a less effective measure.
When I worked for a service provider, this how we did it. Anything more
aggressive
Another common source of this is codec mismatch.
So if your ingress region isn't related to the region that MoH is in with the
G.711/G.722 bandwidth profile, you'll get this issue. If you get tone-on-hold
then it is usally partition/css/tftp related but dead silence is usually codec
related.
5 node 10.5.2 ccm cluster, each node based on the 2,500 user OVA
The OVA deploys with one vCPU however, the docwiki
(http://docwiki.cisco.com/wiki/Virtualization_for_Cisco_Unified_Communications_Manager_(CUCM)#Notes_on_2500_user_VM_configurations)
Shows that it may be advisable to deploy with 2
I have CCX 8.5.1; I am going to 10.6.1.
I am NOT doing an in-place upgrade. I'm having issues sourcing the .ISO for
their current version, so I can restore/jump upgrade it to 10.6.1 and get a
same version DRS to import.
Is it possible to extract the historical reporting data *only*, and move th
Too add-on to Kevin; if there is an annun in the MRGL, is the anunn's region
related to the region of the CTI RP with the same codec bandwidth profile?
What type of ingress gateway SIP ... h.323 ... ?
What is the value for the "Send H225 User Info Message" under Service
Parameters->Call Manager
In 10.5 call manager can have local and LDAP users simultaneously.
Unity connections 10.5 can have local, LDAP imported and AXL imported users
simultaneously.
By removing an LDAP user from your directory (and then subsequently performing
a LDAP sync) the user will not delete itself but rather
So where are you seeing the out of service message? In call manager's voicemail
port page?
Thanks,
Ryan
Original Message
From: norm.nichol...@kitchener.ca
Sent: Saturday, June 6, 2015 12:18 PM
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] Unity 10.5 error
>
>I have a n
there.
Thanks,
Ryan
Original Message
From: Ryan Huff
Sent: Saturday, June 6, 2015 02:11 PM
To: norm.nichol...@kitchener.ca,cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] Unity 10.5 error
>So where are you seeing the out of service message? In call manager's
&g
Has anyone been noticing that in this environment (UCCX 10.6 | CUCM 10.5.2 |
CAD); extensions will sometimes report as OutOfService (and subsequently cause
CAD login issues) when in fact, the phone is registered, associated to the
jTAPI/RMCMSubsys user, has CTI Control enabled (and is setup corr
et
Are you on UCM 10.5.2su1 ? I haven't seen that issue at all.
Is your VMware LRO disabled ?
Do your UCMs have one or two vCPU ?
On Jun 9, 2015 12:42 AM, "Ryan Huff" wrote:
Has anyone been noticing that in this environment (UCCX 10.6 | CUCM 10.5.2 |
CAD); extensions will some
Experiencing some strange off-hook delay when dialing.
Here are the tests I did:
When dialing an exact match (another DN) from the off-hook position, I get a
full T302 delay before the digits are sent. If however, I mark the target DN as
urgent priority, it routes immediately (as I would expect
CBABU
CCIE Collaboration # 40065
From: Ryan Huff
Date: Tuesday, 9 June 2015 10:40 pm
To: Justin Steinberg
Cc: Cisco VOIP
Subject: Re: [cisco-voip] UCCX 10.6 | CUCM 10.5.2 | CAD Extension out of service
Hi Justin, thank you for your reply.
CUCM 10.5.2su1 is in the acti
Date: Wed, 10 Jun 2015 06:05:38 -0700
Subject: RE: [cisco-voip] Interesting off hook dial delay
Is this potentially matching something in the “none” partition? From:
cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Ryan Huff
Sent: Wednesday, June 10, 2015 7:48 AM
To:
t Analysis Complexity
CallManager Service Parameter to "TranslationAndAlternatePatternAnalysis" on
each node then pull CallManager traces for a test call. The Digit Analysis
section in the traces will now show the overlapping pattern.
On Wed, Jun 10, 2015 at 8:48 AM, Ryan Huff wrote:
Exper
the overlapping pattern.
On Wed, Jun 10, 2015 at 8:48 AM, Ryan Huff
wrote:
Experiencing some strange off-hook delay when dialing.
Here are the tests I did:
When dialing an exact match (another DN) from the off-hook position, I get a
full T302 delay before the digits are sent. If ho
Helpful bash script for linux user's CLI that replicates Cisco IOS feature of
typing in an IP address to invoke the telnet process without having to type the
word "telnet".
http://ryanthomashuff.com/2015/06/linux-bash-script-for-automating-telnet-login/
- Check that forward and reverse DNS is working for ALL UC server FQDNs on your
DNS server
- Have a look in CUCM Unified reporting and make sure you don't have any DB
replication issues
- Look at the local voice network the phone is on; is it getting a DNS server
that is able to resolve the FQDN
I would replace the copper pair between the FXS port and the analog phone and
then test again. If you are using a custom copper pair, try putting a few
twists in the pair and then re-crimp/punch-down both ends. Is this FXS port
under IOS control, stcapp or mgcpapp?
If it truly is an audible art
Ctrl-F7 (in the view menu) will turn off the salutation popup.
Thanks,
Ryan
Original Message
From: James Dust
Sent: Monday, June 22, 2015 02:51 PM
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] FW: Unified Attendant Console ver 9.1.1.20 Question
>Morning all,
>
>We have
I don't believe so, at least not on the CPE side.
The presence of all channels being busy would indicate that there aren't any
channels available for the next call (or the next call after all channels are
busy) to ring through the PRI for you to forward to voicemail, even if there
were a way f
utlook.com; cisco-voip@puck.nether.net
Subject: RE: [cisco-voip] isdn channels full message
Date: Tue, 23 Jun 2015 13:55:42 +0100
Hi just to add it to. We need it only for the outbound direction. From: Ryan
Huff [mailto:ryanh...@outlook.com]
Sent: 23 June 2015 13:51
To: abbas wali; cisco-voip@puck.
Justin,
For clarification, you have a user(s) that makes an outbound call from call
manager to the pstn (via a sip trunk to an itsp?) And some of them want to
block their Caller ID or mask it to anon?
Thanks,
Ryan
Original Message
From: Justin Steinberg
Sent: Tuesday, June
Wes,
I tried something like this before with h.323 (full disclosure, it was ccm
8.0.3) and the behavior I experienced was that on the busy out of the first
gateway (the primary), call manager did not the recall the call and try the
next member in the group, it just busied on the first h.323 gat
That is probably it. Thanks Brian!
Thanks,
Ryan
Original Message
From: Brian Meade
Sent: Tuesday, June 23, 2015 03:34 PM
To: Ryan Huff
Subject: Re: [cisco-voip] isdn channels full message
CC: "Wes Sisk (wsisk)" ,abbas Wali
,cisco-voip@puck.nether.net
>
That is probably it. Thanks Brian!
Thanks,
Ryan
Original Message
From: Brian Meade
Sent: Tuesday, June 23, 2015 03:34 PM
To: Ryan Huff
Subject: Re: [cisco-voip] isdn channels full message
CC: "Wes Sisk (wsisk)" ,abbas Wali
,cisco-voip@puck.nether.net
>
HA! Thanks for this Wes!
From: ws...@cisco.com
To: dpa...@fidelus.com
Date: Thu, 25 Jun 2015 17:45:44 +
Subject: Re: [cisco-voip] Strange Number
CC: cisco-voip@puck.nether.net; aj...@buffalo.edu
Thanks for pointing this out! I’ll get the doc updated.
The rest of the list:
"**##*1",
Is this a greenfield or a break/fix?
I would first check the obvious stuff;
Correct Device Pool/Region/Location/PT/CSS used on the CTI Ports?Does the call
drop as soon as it is parked/transferred OR only once the call is connected to
the called phone? If it is the latter, may be a regional/cod
I imagine there are a few ways to attack this but I thought I would share the
method I just used. I needed to reset the PIN for ALOT of CUE accounts and
didn't have much time to research so I just cooked up a PHP script for it.
http://ryanthomashuff.com/2015/07/bulk-reset-pin-and-password-cisco-
You are correct about LDAP Authentication, needs the publisher to be up.
I think SAML SSO is just CUCM and CUIM&P and it rides on top of LDAP
syncronization but I could be wrong brcause I don't play with SAML SSO that
often.
Thanks,
Ryan
Original Message
From: Matthew Collin
My suspicion is it has to do with controlling the number of queries being
issued and from where or perhaps and more specifically, tracking the failover
itself.
Once the failover occurred, the identity of the cucm-side ldap sync would
change and AD servers might not handle that gracefully. I
Hi Dan!
Thanks for the clarification/correction I just happen to have a few 3-node
cluster hanging around and I just tried this 5 times in a mix of 9.1.1, 10.0
and 10.5 and here is what I found:
3 times LDAP auth was a seamless failover to the sub
2 times LDAP auth did not work on the s
d desktop agent access to the pub and see what happens.
And then another test where the ACL blocks access to the LDAP server
temporarily.
Sent from my iPhone
On Jul 6, 2015, at 10:04 AM, Ryan Huff wrote:
Hi Dan!
Thanks for the clarification/correction I just happen to have a few
Are you sip to cuc ab d the pstn? If so, what is your dtmf support in your
trunks?
Thanks,
Ryan
Original Message
From: Jonathan Charles
Sent: Monday, July 6, 2015 04:11 PM
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] Cisco 8841/51 not sending DTMF...
>We have a site
, July 7, 2015 07:44 PM
To: Ryan Huff
Subject: Re: [cisco-voip] Cisco 8841/51 not sending DTMF...
CC: cisco-voip@puck.nether.net
>I am SCCP to CUC; SIP to SIP for PSTN (SIP trunk to CUBE, SIP trunk to CL)
>
>On Mon, Jul 6, 2015 at 3:27 PM, Ryan Huff wrote:
>
>> Are you sip to cuc
2015 08:20 PM
To: Ryan Huff
Subject: Re: [cisco-voip] Cisco 8841/51 not sending DTMF...
CC: cisco-voip@puck.nether.net
>Just RTP-NTE on the gateway... verified dial peer matched...
>
>On Tue, Jul 7, 2015 at 7:03 PM, Ryan Huff wrote:
>
>> On the intetnal side, do your skinny port
Have you enabled "always use prime line" for the phone?
Thanks,
Ryan
Original Message
From: "Haas, Neal"
Sent: Thursday, July 9, 2015 10:32 AM
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] New 8851 Phones
>So we got some 8851 phones into test. Only thing we don't like
Have you tried:
Set calendar integration in the client "none"
Sign out/Close the client.
Remove the jabber cache again ~/appdata/local , /appdata/roaming
Log the client back in and set your calendar integration back to its original
setting.
Thanks,
Ryan
Original Message --
Here is a good explanation of the issue and how to work around it:
http://eltonoverip.com/blog/2015/07/firefox-39-0-ssl-error-weak-ephemeral-diffie-hellman-key/
From: dennis.h...@wwt.com
To: wo...@justfamily.org; cisco-voip@puck.nether.net
Date: Thu, 9 Jul 2015 19:53:09 +
Subject: Re: [cisco-
Here is the Cisco security advisory for the OpenSSL flaw found in June/2015
Long, Long list of products affected:
http://tools.cisco.com/security/center/content/CiscoSecurityAdvisory/cisco-sa-20150612-openssl
-Thanks,
Ryan
From: ryanh...@outlook.com
To: dennis.h...@wwt.com; wo...@justfam
If you are talking about line side, I don't believe so.
Thanks,
Ryan
Original Message
From: Mark Holloway
Sent: Friday, July 10, 2015 07:11 PM
To: Cisco VoIP Group
Subject: [cisco-voip] Shared Line Appearances on SIP Phones
>By default phones with SIP firmware using shared l
Sounds like a codec issue. Are both sides negotiating the same codec?
Thanks,
Ryan
Original Message
From: Robert Schuknecht
Sent: Wednesday, July 15, 2015 09:37 AM
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] External Call from Movi Client to Conductor fails after 5
s
So this bug cost me 2 hours last night .. one of the more ridiculous ones I
have seen in awhile. 10.5.2; the Subject Alternative Name in the CSR can't be
the same as the CN (although that is how most CA's will generate).
The work around? Add a space character at the end of the CN when you genera
Wrong and Visionary. Unfortunately, you have to
be a visionary to see it." – Sheldon Cooper
Click here to join me in my Collaboration Meeting Room
From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net]
On Behalf Of Ryan Huff
Sent: Friday, July 17, 2015 4:10 PM
To: cisco-voip@puck.neth
Nate,
I am not by my Linux machine (refuse to use winblows) to vet this but could you
do:
*5XXX with a CPTM of XXX3101 with a prefix of 8?
Thanks,
Ryan
Original Message
From: NateCCIE
Sent: Monday, July 20, 2015 08:08 PM
To: 'Cisco VOIP'
Subject: [cisco-voip] CUCM translat
From: Ryan Huff
Sent: Monday, July 20, 2015 08:27 PM
To: natec...@gmail.com,cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] CUCM translation pattern postfix digits
>Nate,
>
>I am not by my Linux machine (refuse to use winblows) to vet this but could
>you do:
>
>*5
Xlate : *5XXX (CPTM: 8XXX)
Route Pattern: 8XXX (CPTM: 3101 Prefix: 8XXX) <-> Route list/group to h.323
gateway that uses a ccm call processing node as the ip address of the gateway.
It could be the IPA talking but that sounds like it should work?
Thanks,
Ryan___
Well Lelio, that is what you get for trying to do digit manipulation on a
napkin in between wings! Lol
Thanks,
Ryan
Original Message
From: Lelio Fulgenzi
Sent: Monday, July 20, 2015 09:22 PM
To: Ryan Huff ,natec...@gmail.com
Subject: Re: [cisco-voip] CUCM translation pattern
Scot, It comes pre-installed on the BE6/7K. The other option is to get it
e-delivered from the PUT tool (using a valid contract). There are alternative
methods to getting non-bootable ISOs to work but I cannot endorse them as the
are not supported.
Thanks,
Ryan
Date: Tue, 21 Jul 2015 09:01:50
Can anyone tell me what metrics CUC looks at when generating the Unused
Voicemail Account Report?
Thanks,
Ryan
___
cisco-voip mailing list
cisco-voip@puck.nether.net
https://puck.nether.net/mailman/listinfo/cis
Lelio,
Yes you need an external instance of postgresql. I find that Ubuntu works best.
Thanks,
Ryan
Original Message
From: Lelio Fulgenzi
Sent: Tuesday, July 21, 2015 12:37 PM
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] CUCM IM&P and postgress database "instances"
>
y use a different table in
the database?
---
Lelio Fulgenzi, B.A.
Senior Analyst, Network Infrastructure
Computing and Communications Services (CCS)
University of Guelph
519‐824‐4120 Ext 56354
le...@uoguelph.ca
www.uoguelph.ca/ccs
Room 037, Animal Science and Nutrition Building
Guelph, Ontario, N1
B.A.
Senior Analyst, Network Infrastructure
Computing and Communications Services (CCS)
University of Guelph
519‐824‐4120 Ext 56354
le...@uoguelph.ca
www.uoguelph.ca/ccs
Room 037, Animal Science and Nutrition Building
Guelph, Ontario, N1G 2W1
From: "Ryan Huff"
To: "Lelio Fulgenzi"
Cc
If you need/want any additional assistance just unicast me.
Thanks,
Ryan
Original Message
From: Lelio Fulgenzi
Sent: Tuesday, July 21, 2015 02:52 PM
To: Ryan Huff
Subject: Re: [cisco-voip] CUCM IM&P and postgress database "instances"
CC: cisco-voip@puck.nethe
If you need/want any additional assistance just unicast me.
Thanks,
Ryan
Original Message
From: Lelio Fulgenzi
Sent: Tuesday, July 21, 2015 02:52 PM
To: Ryan Huff
Subject: Re: [cisco-voip] CUCM IM&P and postgress database "instances"
CC: cisco-voip@puck.nethe
Had a healthy, licensed 5 node cluster. Some Telepresence units were added and
that put the PLM out of compliance, clearly. So the TP licenses were added
(using the built in auto-pak fill tool)and that brought the TP license shortage
back into compliance BUT put basic messaging, essential and en
If it is not from an LDAP integration, it maybe from an AXL integration. You
could get a COBRAS snapshot of the account (to get the recorded name and
messages ... etc). Then delete the account and restore.
From: norm.nichol...@kitchener.ca
To: cisco-voip@puck.nether.net
Date: Fri, 24 Jul 2015 15
Trying to use the answer file generator for CUCM 8.6 and it generates the
files, but I don't see a pre-generated license MAC any ideas?
___
cisco-voip mailing list
cisco-voip@puck.nether.net
https://puck.ne
Looking for a concise compatibility matrix for Expressway c/e and cucm. I've
found various deployment scenario docs which specify CUCM versions in the
prereq's leading me to believe there are compatibility requirments but I have
not found them yet.
Any ideas?
You may try a hunt group/broadcast and put both DNs in seperate partitions.
Thanks,
Ryan
Original Message
From: Jacky Cai
Sent: Saturday, August 15, 2015 09:49 PM
To: VOIP Group
Subject: [cisco-voip] route to two dial peers simultaneously in ios voice
gateway
>Is ther
My experince is that analog faxing over SIP is sometimes a dance.
What I have found to work consistently is to disable SuperG3 and ECM on the fax
modem and restrict the rx/tx of the modem to 14.4 Kbps.
Thanks,
Ryan
Original Message
From: "Haas, Neal"
Sent: Wednesday, August
My thoughts ...
Maybe switch to the inactive load and test as a quick-fix.
Is this the only sip endpoint in this CME that is impacted? Are there SCCP
endpoints and if so, do they have the same issue?
Can customer complete a dial and does the issue continue once connected?
Thanks,
Ryan
-
I believe that is correct, the actual audible tone comes from the phone,
triggered to play through signaling from the call control server.
Thanks,
Ryan
Original Message
From: NateCCIE
Sent: Thursday, August 20, 2015 07:21 PM
To: 'Ryan Huff' ,"'Jason Aa
Ah . derivitive of the studder good stuff Brian, totally forgot
about that.
Thanks,
Ryan
Original Message
From: Brian Meade
Sent: Thursday, August 20, 2015 10:55 PM
To: "Jason Aarons (AM)"
Subject: Re: [cisco-voip] 3905 dial tone problem
CC: "cisco-voip (cisco-v
Not sure which post you are referencing but here is what I have used in the
past and it worked for me;
voice class sip-profiles XXX
request INVITE sdp-header Video-Attribute remove
request INVITE sdp-header Video-Media modify "m=video(.*)" ""
request INVITE sdp-header Video-Bandwidth-Info remo
The flavor of VMWare that comes with the BE6/7k only has the ability to use a
vSwitch, not a distributed vSwitch due the differences in licensing (Enterprise
Plus being what enables the ability to use a distributed vSwitch).
According to this VMWare KB article;
http://kb.vmware.com/selfservice
deo says to use On as well:
https://www.youtube.com/watch?v=3dhxqlH_0_I
On Tue, Aug 25, 2015 at 9:04 AM Ryan Huff wrote:
The flavor of VMWare that comes with the BE6/7k only has the ability to use a
vSwitch, not a distributed vSwitch due the differences in licensing (Enterprise
Plus being w
Is it just me not finding 7970 listed on CCO?
___
cisco-voip mailing list
cisco-voip@puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip
From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net]
On Behalf Of Damisch, Kevin
Sent: Wednesday, August 26, 2015 9:13 AM
To: Ryan Huff ; cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] IP Phone 7970 Firmware
Strange. You can’t browse for it, but Google found i
Sounds like a codec/media issue. Are you supporting early offer?
Thanks,
Ryan
Original Message
From: Aaron Banks
Sent: Friday, August 28, 2015 03:35 PM
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] Service Provider SIP Trunks
>
>
>I have a strange problem with SIP trun
Thanks,
Ryan
Original Message
From: Ryan Huff
Sent: Friday, August 28, 2015 05:12 PM
To: amichaelba...@hotmail.com
Subject: RE: [cisco-voip] Service Provider SIP Trunks
>As long as you didn't change anything on the cpe side, it may be more likely
>your itsp c
I have an 8.4 HA Presence cluster that I am doing a DRS restore on. The
question I have is regarding contact lists. Will I recover all the user contact
lists and data by restoring just the primary node or do I need to also restore
the failover node (rather than just rebuild a new failover) to re
So this is less a CUCM error as it is actually an issue with RedHat (OS).
Basically, your platform administrator account is roached (doesn't have the
right permissions) and when you try to log in -SELinux is spanking you.
When you installed the patch, did you install with out a switch-version o
Also, if you are able to login into OS Administration (GUI) with the platform
administrator account, but just can't SSH then you may be missing the
/etc/selinux/targeted/contexts/users/sshout_u file. If you can get into the
root account (had it previously enabled before the failed patch), you ca
@puck.nether.net
I am able to login to the OS Admin page with that account... so I should be
able to make a remote support account and get tac to fix this...
Thanks!
Jonathan
On Thu, Sep 3, 2015 at 2:02 PM, Ryan Huff wrote:
Also, if you are able to login into OS Administration (GUI) with the platform
A word on using the mobile jabber clients with mra when webex connect is
configured on the domain (but isn't being used and can't be turned off for one
reason or another) ...
The only way I was able to get the provisioning URL to work for the mobile
clients (so the clients will exclude WebEx se
Brian I had this issue this weekend in 8.6. My original issue was the "no
home uds cluster" but I had issues with the proxy protocol violation.
Tac's response was go to 8.6.1 (released 9/11/15 ... yikes) or roll back to 8.5
Thanks,
Ryan
Original Message
From: Brian Mead
What I have done in the past that that has worked is delete the cert on the pub
first and then restart the tomcat service. Then do each sub node the same.
Thanks,
Ryan
Original Message
From: James Andrewartha
Sent: Tuesday, September 15, 2015 01:22 AM
To: cisco-voip voip list
Since you mention using extension mobility
When the Agent logs in with their Ex. mobility profile, does the DN happen to
be on another IP phone? The only way to "share" an ACD extension between
multiple devices is to
assign it to a Device Profile exclusively and then login using Extension
e. does anyone know, if want to get traces from RTMT which option
should I use i.e. Cisco Call Manager will suffice ? thanks From: Ryan Huff
[mailto:ryanh...@outlook.com]
Sent: 15 September 2015 14:35
To: abbas wali ; cisco-voip@puck.nether.net
Subject: RE: [cisco-voip] UCCX 9 EM agents cant
any issues. Even I can do it on my CIPC But these new 3 agents cant.
From: Ryan Huff [mailto:ryanh...@outlook.com]
Sent: 15 September 2015 15:05
To: abbas wali ; cisco-voip@puck.nether.net
Subject: RE: [cisco-voip] UCCX 9 EM agents cant login Assuming this isn't a
new/upgrade UCCX and it is
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