date:20150330

Oooo, good question Brian.  It's my understanding that in order for the
below specific call flow to work, an MTP is required for DTMF inter-working
of inband to out-of-band.

PSTN Caller Pushes DTMF --- ITSP Delivers RFC2833 --- CUBE Delivers OOB
--- CUCM Devlier OOB --- UCCX CTI Port Receives OOB

Is that not the case?

On Mon, Mar 30, 2015 at 12:01 PM Brian Meade bmead...@vt.edu wrote:

 What are you trying to accomplish with the MTP that can't already be
 accomplished with media flow-through and dtmf-relay?

 On Mon, Mar 30, 2015 at 12:38 PM, Anthony Holloway 
 avholloway+cisco-v...@gmail.com wrote:

 All,

 I know the name itself, LTI, includes the word transcoding, but I'm just
 double checking that this will or will not work for registering an MTP on
 the CUBE.  All roads are leading me to the answer, but it just seems like a
 huge miss on Cisco's part to not allow us to register MTPs as well as XCODE
 via the LTI method.

 This works for me:
 dspfarm profile 1 transcode
  codec g711ulaw
  codec g729ar8
  max sessions 1
  assoc app cube
  no shut
 !

 This does not work for me (it hangs on associating to cube app):
 dapfarm profile 2 mtp
  codec g711ulaw
  max sessions software 1
  assoc app cube
  no shut
 !

 I have the required dspfarm and mode border-element commands, and
 rebooted after as well.

 Seems like with the standard requirement of rfc2833 on SIP trunks to the
 ITPS, and CTI apps in the network (I'm looking at you UCCX), MTPs play a
 large role in the success of SIP trunking for customers, and yet I cannot
 even register them locally with the LTI.

 I do have a fallback plan, so I'm not stuck.  I'm just looking for the
 optimal design scenario.  In my order of preference I would like to go:

 1. LTI
 2. SCCP via Telephony Service
 3. SCCP via CUCM

 Would you rank them differently?

 Thanks for your input in advance.

 ___
 cisco-voip mailing list
 cisco-voip@puck.nether.net
 https://puck.nether.net/mailman/listinfo/cisco-voip



___
cisco-voip mailing list
cisco-voip@puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

[cisco-voip] CUBE and LTI for MTPs

All,

I know the name itself, LTI, includes the word transcoding, but I'm just
double checking that this will or will not work for registering an MTP on
the CUBE.  All roads are leading me to the answer, but it just seems like a
huge miss on Cisco's part to not allow us to register MTPs as well as XCODE
via the LTI method.

This works for me:
dspfarm profile 1 transcode
 codec g711ulaw
 codec g729ar8
 max sessions 1
 assoc app cube
 no shut
!

This does not work for me (it hangs on associating to cube app):
dapfarm profile 2 mtp
 codec g711ulaw
 max sessions software 1
 assoc app cube
 no shut
!

I have the required dspfarm and mode border-element commands, and rebooted
after as well.

Seems like with the standard requirement of rfc2833 on SIP trunks to the
ITPS, and CTI apps in the network (I'm looking at you UCCX), MTPs play a
large role in the success of SIP trunking for customers, and yet I cannot
even register them locally with the LTI.

I do have a fallback plan, so I'm not stuck.  I'm just looking for the
optimal design scenario.  In my order of preference I would like to go:

1. LTI
2. SCCP via Telephony Service
3. SCCP via CUCM

Would you rank them differently?

Thanks for your input in advance.
___
cisco-voip mailing list
cisco-voip@puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

Re: [cisco-voip] CUBE and LTI for MTPs

What are you trying to accomplish with the MTP that can't already be
accomplished with media flow-through and dtmf-relay?

On Mon, Mar 30, 2015 at 12:38 PM, Anthony Holloway 
avholloway+cisco-v...@gmail.com wrote:

 All,

 I know the name itself, LTI, includes the word transcoding, but I'm just
 double checking that this will or will not work for registering an MTP on
 the CUBE.  All roads are leading me to the answer, but it just seems like a
 huge miss on Cisco's part to not allow us to register MTPs as well as XCODE
 via the LTI method.

 This works for me:
 dspfarm profile 1 transcode
  codec g711ulaw
  codec g729ar8
  max sessions 1
  assoc app cube
  no shut
 !

 This does not work for me (it hangs on associating to cube app):
 dapfarm profile 2 mtp
  codec g711ulaw
  max sessions software 1
  assoc app cube
  no shut
 !

 I have the required dspfarm and mode border-element commands, and rebooted
 after as well.

 Seems like with the standard requirement of rfc2833 on SIP trunks to the
 ITPS, and CTI apps in the network (I'm looking at you UCCX), MTPs play a
 large role in the success of SIP trunking for customers, and yet I cannot
 even register them locally with the LTI.

 I do have a fallback plan, so I'm not stuck.  I'm just looking for the
 optimal design scenario.  In my order of preference I would like to go:

 1. LTI
 2. SCCP via Telephony Service
 3. SCCP via CUCM

 Would you rank them differently?

 Thanks for your input in advance.

 ___
 cisco-voip mailing list
 cisco-voip@puck.nether.net
 https://puck.nether.net/mailman/listinfo/cisco-voip


___
cisco-voip mailing list
cisco-voip@puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

Re: [cisco-voip] CUBE and LTI for MTPs

dtmf-relay I believe should handle that find for you without the MTP.

On Mon, Mar 30, 2015 at 1:21 PM, Anthony Holloway 
avholloway+cisco-v...@gmail.com wrote:

 Oooo, good question Brian.  It's my understanding that in order for the
 below specific call flow to work, an MTP is required for DTMF inter-working
 of inband to out-of-band.

 PSTN Caller Pushes DTMF --- ITSP Delivers RFC2833 --- CUBE Delivers OOB
 --- CUCM Devlier OOB --- UCCX CTI Port Receives OOB

 Is that not the case?

 On Mon, Mar 30, 2015 at 12:01 PM Brian Meade bmead...@vt.edu wrote:

 What are you trying to accomplish with the MTP that can't already be
 accomplished with media flow-through and dtmf-relay?

 On Mon, Mar 30, 2015 at 12:38 PM, Anthony Holloway 
 avholloway+cisco-v...@gmail.com wrote:

 All,

 I know the name itself, LTI, includes the word transcoding, but I'm just
 double checking that this will or will not work for registering an MTP on
 the CUBE.  All roads are leading me to the answer, but it just seems like a
 huge miss on Cisco's part to not allow us to register MTPs as well as XCODE
 via the LTI method.

 This works for me:
 dspfarm profile 1 transcode
  codec g711ulaw
  codec g729ar8
  max sessions 1
  assoc app cube
  no shut
 !

 This does not work for me (it hangs on associating to cube app):
 dapfarm profile 2 mtp
  codec g711ulaw
  max sessions software 1
  assoc app cube
  no shut
 !

 I have the required dspfarm and mode border-element commands, and
 rebooted after as well.

 Seems like with the standard requirement of rfc2833 on SIP trunks to the
 ITPS, and CTI apps in the network (I'm looking at you UCCX), MTPs play a
 large role in the success of SIP trunking for customers, and yet I cannot
 even register them locally with the LTI.

 I do have a fallback plan, so I'm not stuck.  I'm just looking for the
 optimal design scenario.  In my order of preference I would like to go:

 1. LTI
 2. SCCP via Telephony Service
 3. SCCP via CUCM

 Would you rank them differently?

 Thanks for your input in advance.

 ___
 cisco-voip mailing list
 cisco-voip@puck.nether.net
 https://puck.nether.net/mailman/listinfo/cisco-voip



___
cisco-voip mailing list
cisco-voip@puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

[cisco-voip] OT: 911 App for mobile phones...


Interesting. 
http://mic.com/articles/114046/you-probably-didn-t-know-calling-911-is-a-problem-but-it-is-these-guys-are-fixing-it
 


--- 
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, N1G 2W1 

___
cisco-voip mailing list
cisco-voip@puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

Re: [cisco-voip] CUBE and LTI for MTPs

Ok, time to take a step back and appreciate the situation I just found
myself in.

You were absolutely correct.  I lab'd this up just now and my mouth is a
gape.  I've been doing and teaching the MTP method for UCCX for like 4
years now, and not once have I ever had anyone correct me.  Not in the
countless conversations I've had, TAC cases, etc.  It's like I've been in
the Truman show and everyone knew but me, but didn't want to tell me.

This is what it looked like when I had RTP-NTE on both ITSP and CUCM facing
DP's.  Note the 10.U.C.M address is because the MTP is software on CUCM.

CUBE#sh voip rtp conn | in 10\.
2 20591  20590  1643430492  10.U.B.E
 10.U.C.M

CUBE#sh call active voice | in Dtmf
tx_DtmfRelay=rtp-nte
tx_DtmfRelay=rtp-nte


I changed my CUCM facing DP's to sip-kpml and then it changes to this.
Note the change to 10.C.C.X because no MTP is needed.  And I pushed buttons
to validate, and it worked.

CUBE#sh voip rtp conn | in 10\.
2 20579  20578  1643026404  10.U.B.E
 10.C.C.X

CUBE#sh call active voice | in Dtmf
tx_DtmfRelay=rtp-nte
tx_DtmfRelay=sip-kpml

Damn.  I'm happy I posted the question, even though it wasn't the answer I
was expecting.  Thanks Brian, I owe you a beer.


On Mon, Mar 30, 2015 at 1:29 PM Brian Meade bmead...@vt.edu wrote:

 dtmf-relay I believe should handle that find for you without the MTP.

 On Mon, Mar 30, 2015 at 1:21 PM, Anthony Holloway 
 avholloway+cisco-v...@gmail.com wrote:

 Oooo, good question Brian.  It's my understanding that in order for the
 below specific call flow to work, an MTP is required for DTMF inter-working
 of inband to out-of-band.

 PSTN Caller Pushes DTMF --- ITSP Delivers RFC2833 --- CUBE Delivers OOB
 --- CUCM Devlier OOB --- UCCX CTI Port Receives OOB

 Is that not the case?

 On Mon, Mar 30, 2015 at 12:01 PM Brian Meade bmead...@vt.edu wrote:

 What are you trying to accomplish with the MTP that can't already be
 accomplished with media flow-through and dtmf-relay?

 On Mon, Mar 30, 2015 at 12:38 PM, Anthony Holloway 
 avholloway+cisco-v...@gmail.com wrote:

 All,

 I know the name itself, LTI, includes the word transcoding, but I'm
 just double checking that this will or will not work for registering an MTP
 on the CUBE.  All roads are leading me to the answer, but it just seems
 like a huge miss on Cisco's part to not allow us to register MTPs as well
 as XCODE via the LTI method.

 This works for me:
 dspfarm profile 1 transcode
  codec g711ulaw
  codec g729ar8
  max sessions 1
  assoc app cube
  no shut
 !

 This does not work for me (it hangs on associating to cube app):
 dapfarm profile 2 mtp
  codec g711ulaw
  max sessions software 1
  assoc app cube
  no shut
 !

 I have the required dspfarm and mode border-element commands, and
 rebooted after as well.

 Seems like with the standard requirement of rfc2833 on SIP trunks to
 the ITPS, and CTI apps in the network (I'm looking at you UCCX), MTPs play
 a large role in the success of SIP trunking for customers, and yet I cannot
 even register them locally with the LTI.

 I do have a fallback plan, so I'm not stuck.  I'm just looking for the
 optimal design scenario.  In my order of preference I would like to go:

 1. LTI
 2. SCCP via Telephony Service
 3. SCCP via CUCM

 Would you rank them differently?

 Thanks for your input in advance.

 ___
 cisco-voip mailing list
 cisco-voip@puck.nether.net
 https://puck.nether.net/mailman/listinfo/cisco-voip




___
cisco-voip mailing list
cisco-voip@puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

Re: [cisco-voip] CUBE and LTI for MTPs

This chart has all the interoperability that can be handled by dtmf-relay
natively on CUBE-
http://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/cube/configuration/cube-book/dtmf-relay.html#concept_264617919921874995299551391601561

Brian

On Mon, Mar 30, 2015 at 2:29 PM, Brian Meade bmead...@vt.edu wrote:

dtmf-relay I believe should handle that find for you without the MTP.

On Mon, Mar 30, 2015 at 1:21 PM, Anthony Holloway
avholloway+cisco-v...@gmail.com wrote:

Oooo, good question Brian. It's my understanding that in order for the
below specific call flow to work, an MTP is required for DTMF inter-working
of inband to out-of-band.

PSTN Caller Pushes DTMF --- ITSP Delivers RFC2833 --- CUBE Delivers OOB
--- CUCM Devlier OOB --- UCCX CTI Port Receives OOB

Is that not the case?

On Mon, Mar 30, 2015 at 12:01 PM Brian Meade bmead...@vt.edu wrote:

What are you trying to accomplish with the MTP that can't already be
accomplished with media flow-through and dtmf-relay?

On Mon, Mar 30, 2015 at 12:38 PM, Anthony Holloway
avholloway+cisco-v...@gmail.com wrote:

All,

I know the name itself, LTI, includes the word transcoding, but I'm
just double checking that this will or will not work for registering an MTP
on the CUBE. All roads are leading me to the answer, but it just seems
like a huge miss on Cisco's part to not allow us to register MTPs as well
as XCODE via the LTI method.

This works for me:
dspfarm profile 1 transcode
codec g711ulaw
codec g729ar8
max sessions 1
assoc app cube
no shut
!

This does not work for me (it hangs on associating to cube app):
dapfarm profile 2 mtp
codec g711ulaw
max sessions software 1
assoc app cube
no shut
!

I have the required dspfarm and mode border-element commands, and
rebooted after as well.

Seems like with the standard requirement of rfc2833 on SIP trunks to
the ITPS, and CTI apps in the network (I'm looking at you UCCX), MTPs play
a large role in the success of SIP trunking for customers, and yet I cannot
even register them locally with the LTI.

I do have a fallback plan, so I'm not stuck. I'm just looking for the
optimal design scenario. In my order of preference I would like to go:

1. LTI
2. SCCP via Telephony Service
3. SCCP via CUCM

Would you rank them differently?

Thanks for your input in advance.

___
cisco-voip mailing list
cisco-voip@puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

Re: [cisco-voip] OT: 911 App for mobile phones...

2015-03-30 Thread Rob Dawson

Just an FYI, there may be some NSFW content on that page . . . thankfully no 
one from HR was walking by when I scrolled down and saw the breateses ☺

Rob

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Lelio 
Fulgenzi
Sent: Monday, March 30, 2015 3:12 PM
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] OT: 911 App for mobile phones...

Interesting.

http://mic.com/articles/114046/you-probably-didn-t-know-calling-911-is-a-problem-but-it-is-these-guys-are-fixing-it
---
Lelio Fulgenzi, B.A.
Senior Analyst, Network Infrastructure
Computing and Communications Services (CCS)
University of Guelph

519‐824‐4120 Ext 56354
le...@uoguelph.camailto:le...@uoguelph.ca
www.uoguelph.ca/ccshttp://www.uoguelph.ca/ccs
Room 037, Animal Science and Nutrition Building
Guelph, Ontario, N1G 2W1

___
cisco-voip mailing list
cisco-voip@puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

Re: [cisco-voip] CUBE and LTI for MTPs

According to the table in that document, what I just did is not supported.
Great!

On Mon, Mar 30, 2015 at 1:52 PM Anthony Holloway 
avholloway+cisco-v...@gmail.com wrote:

 Ok, time to take a step back and appreciate the situation I just found
 myself in.

 You were absolutely correct.  I lab'd this up just now and my mouth is a
 gape.  I've been doing and teaching the MTP method for UCCX for like 4
 years now, and not once have I ever had anyone correct me.  Not in the
 countless conversations I've had, TAC cases, etc.  It's like I've been in
 the Truman show and everyone knew but me, but didn't want to tell me.

 This is what it looked like when I had RTP-NTE on both ITSP and CUCM
 facing DP's.  Note the 10.U.C.M address is because the MTP is software on
 CUCM.

 CUBE#sh voip rtp conn | in 10\.
 2 20591  20590  1643430492  10.U.B.E
  10.U.C.M

 CUBE#sh call active voice | in Dtmf
 tx_DtmfRelay=rtp-nte
 tx_DtmfRelay=rtp-nte


 I changed my CUCM facing DP's to sip-kpml and then it changes to this.
 Note the change to 10.C.C.X because no MTP is needed.  And I pushed buttons
 to validate, and it worked.

 CUBE#sh voip rtp conn | in 10\.
 2 20579  20578  1643026404  10.U.B.E
  10.C.C.X

 CUBE#sh call active voice | in Dtmf
 tx_DtmfRelay=rtp-nte
 tx_DtmfRelay=sip-kpml

 Damn.  I'm happy I posted the question, even though it wasn't the answer I
 was expecting.  Thanks Brian, I owe you a beer.


 On Mon, Mar 30, 2015 at 1:29 PM Brian Meade bmead...@vt.edu wrote:

 dtmf-relay I believe should handle that find for you without the MTP.

 On Mon, Mar 30, 2015 at 1:21 PM, Anthony Holloway 
 avholloway+cisco-v...@gmail.com wrote:

 Oooo, good question Brian.  It's my understanding that in order for the
 below specific call flow to work, an MTP is required for DTMF inter-working
 of inband to out-of-band.

 PSTN Caller Pushes DTMF --- ITSP Delivers RFC2833 --- CUBE Delivers
 OOB --- CUCM Devlier OOB --- UCCX CTI Port Receives OOB

 Is that not the case?

 On Mon, Mar 30, 2015 at 12:01 PM Brian Meade bmead...@vt.edu wrote:

 What are you trying to accomplish with the MTP that can't already be
 accomplished with media flow-through and dtmf-relay?

 On Mon, Mar 30, 2015 at 12:38 PM, Anthony Holloway 
 avholloway+cisco-v...@gmail.com wrote:

 All,

 I know the name itself, LTI, includes the word transcoding, but I'm
 just double checking that this will or will not work for registering an 
 MTP
 on the CUBE.  All roads are leading me to the answer, but it just seems
 like a huge miss on Cisco's part to not allow us to register MTPs as well
 as XCODE via the LTI method.

 This works for me:
 dspfarm profile 1 transcode
  codec g711ulaw
  codec g729ar8
  max sessions 1
  assoc app cube
  no shut
 !

 This does not work for me (it hangs on associating to cube app):
 dapfarm profile 2 mtp
  codec g711ulaw
  max sessions software 1
  assoc app cube
  no shut
 !

 I have the required dspfarm and mode border-element commands, and
 rebooted after as well.

 Seems like with the standard requirement of rfc2833 on SIP trunks to
 the ITPS, and CTI apps in the network (I'm looking at you UCCX), MTPs play
 a large role in the success of SIP trunking for customers, and yet I 
 cannot
 even register them locally with the LTI.

 I do have a fallback plan, so I'm not stuck.  I'm just looking for the
 optimal design scenario.  In my order of preference I would like to go:

 1. LTI
 2. SCCP via Telephony Service
 3. SCCP via CUCM

 Would you rank them differently?

 Thanks for your input in advance.

 ___
 cisco-voip mailing list
 cisco-voip@puck.nether.net
 https://puck.nether.net/mailman/listinfo/cisco-voip




___
cisco-voip mailing list
cisco-voip@puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

Re: [cisco-voip] OT: 911 App for mobile phones...


omg. my apologies. 


i just noticed that the page scrolls into other stories. 





--- 
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, N1G 2W1 

- Original Message -

From: Rob Dawson rdaw...@force3.com 
To: Lelio Fulgenzi le...@uoguelph.ca, cisco-voip@puck.nether.net 
Sent: Monday, March 30, 2015 3:51:26 PM 
Subject: RE: [cisco-voip] OT: 911 App for mobile phones... 



Just an FYI, there may be some NSFW content on that page . . . thankfully no 
one from HR was walking by when I scrolled down and saw the breateses J 

Rob 





From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Lelio 
Fulgenzi 
Sent: Monday, March 30, 2015 3:12 PM 
To: cisco-voip@puck.nether.net 
Subject: [cisco-voip] OT: 911 App for mobile phones... 



Interesting. 


http://mic.com/articles/114046/you-probably-didn-t-know-calling-911-is-a-problem-but-it-is-these-guys-are-fixing-it
 

--- 
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, N1G 2W1 

___
cisco-voip mailing list
cisco-voip@puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

Re: [cisco-voip] SRST



It's been a while, but last time I looked (and this is how our SRST routers are 
configured) you have to use COR lists. 


It was confusing at first when I started reading them, but I just looked at 
them as search spaces and partitions. The whole subset matching diagram 
confused me too. :) 





--- 
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, N1G 2W1 

- Original Message -

From: norm nicholson norm.nichol...@kitchener.ca 
To: cisco-voip@puck.nether.net 
Sent: Monday, March 30, 2015 4:21:06 PM 
Subject: [cisco-voip] SRST 







Are there ways to restrict phones or DN’s from calling out when one of our 
remote 2900’s is in SRST mode. We have a PSTN that we route 911 and 9911 calls 
to provide ANI and ALI but I am assuming anyone can seize that line by dialing 
9 while we are in SRST mode . 



Thanks 





Norm Nicholson 
Telecom Analyst 
City of Kitchener 
(519) 741-2200 x 7000 


___ 
cisco-voip mailing list 
cisco-voip@puck.nether.net 
https://puck.nether.net/mailman/listinfo/cisco-voip 

___
cisco-voip mailing list
cisco-voip@puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

Re: [cisco-voip] Temp Fail since upgrade

2015-03-30 Thread Ryan Huff

Lisa,

SIP or TDM/PRI?

Have you gandered into RTMT and taken a look at any active / recent alerts?

Thanks,

Ryan

 Original Message 
From: Lisa Notarianni lisa.notaria...@scranton.edu
Sent: Monday, March 30, 2015 04:48 PM
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] Temp Fail since upgrade

!-- /* Font Definitions */ @font-face {font-family:Cambria Math;
panose-1:2 4 5 3 5 4 6 3 2 4;} @font-face   {font-family:Calibri;   
panose-1:2 15 5 2 2 2 4 3 2 4;} /* Style Definitions */ p.MsoNormal, 
li.MsoNormal, div.MsoNormal{margin:0in;margin-bottom:.0001pt;  
font-size:12.0pt;   font-family:Times New Roman,serif;} a:link, 
span.MsoHyperlink {mso-style-priority:99; color:blue; 
text-decoration:underline;} a:visited, span.MsoHyperlinkFollowed
{mso-style-priority:99; color:purple;   text-decoration:underline;} 
p.MsoListParagraph, li.MsoListParagraph, div.MsoListParagraph   
{mso-style-priority:34; margin-top:0in; margin-right:0in;  
 margin-bottom:0in;  margin-left:.5in;   margin-bottom:.0001pt;  
font-size:12.0pt;   font-family:Times New Roman,serif;} 
span.EmailStyle17 {mso-style-type:personal-reply; 
font-family:Calibri,sans-serif;   color:#1F497D;} .MsoChpDefault  
{mso-style-type:export-only;font-family:Calibri,sans-serif;} @page 
WordSection1   {size:8.5in 11.0in; margin:1.0in 1.0in 1.0in 1.0in;} 
div.WordSection1   {page:WordSection1;} /* List Definitions */ @list l0
{mso-list-id:976649086; mso-list-type:hybrid;   
mso-list-template-ids:-1355784104 67698703 67698713 67698715 67698703 67698713 
67698715 67698703 67698713 67698715;} @list l0:level1
{mso-level-tab-stop:none;   mso-level-number-position:left; 
text-indent:-.25in;} @list l0:level2{mso-level-number-format:alpha-lower;  
 mso-level-tab-stop:none;mso-level-number-position:left; 
text-indent:-.25in;} @list l0:level3{mso-level-number-format:roman-lower;  
 mso-level-tab-stop:none;mso-level-number-position:right;
text-indent:-9.0pt;} @list l0:level4{mso-level-tab-stop:none;   
mso-level-number-position:left; text-indent:-.25in;} @list l0:level5   
 {mso-level-number-format:alpha-lower;   mso-level-tab-stop:none;
mso-level-number-position:left; text-indent:-.25in;} @list l0:level6   
 {mso-level-number-format:roman-lower;   mso-level-tab-stop:none;
mso-level-number-position:right;text-indent:-9.0pt;} @list l0:level7   
 {mso-level-tab-stop:none;   mso-level-number-position:left; 
text-indent:-.25in;} @list l0:level8{mso-level-number-format:alpha-lower;  
 mso-level-tab-stop:none;mso-level-number-position:left; 
text-indent:-.25in;} @list l0:level9{mso-level-number-format:roman-lower;  
 mso-level-tab-stop:none;mso-level-number-position:right;
text-indent:-9.0pt;} ol {margin-bottom:0in;} ul 
{margin-bottom:0in;} -- 

A few weeks ago we upgraded Call Manager and Unity Connection from 8.6.2 to 
10.5.1.

 

We have experienced 2 intermittent outbound calling issues:

 

1.   “Temp Fail” shows on phone display and only option on phone button is 
“End Call”

2.   Outbound callers dial a number and hear nothing but notice the phone 
display says “Connected”.  Caller cannot hear anything.  Person they called 
cannot hear anything.

***These issues are only intermittent.***

 

We had no issues before the upgrade.

 

Is anyone experiencing the similar issues?

 

Thank you in advance.

 

 



 

___
cisco-voip mailing list
cisco-voip@puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

Re: [cisco-voip] Cisco UCCX 10.6 demo licenses expire after activation of Finesse

2015-03-30 Thread Abhiram Kramadhati (akramadh)

Interesting. Could you grab the MCVD logs from the time of the demo license 
activation and the finesse activation when it got invalidated?

Regards,
Abhiram Kramadhati
Technical Solutions Manager, CBABU
CCIE Voice # 40065

From: Claiton Campos claitoncam...@gmail.commailto:claitoncam...@gmail.com
Date: Monday, 30 March 2015 10:52 pm
To: akramadh akram...@cisco.commailto:akram...@cisco.com, 
cisco-voip@puck.nether.netmailto:cisco-voip@puck.nether.net 
cisco-voip@puck.nether.netmailto:cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] Cisco UCCX 10.6 demo licenses expire after activation 
of Finesse

Hi Abhiram,

It is a fresh install. Yes, before the Finesse activation, the demo licenses 
were activated , could create all CSQ structure , applications upload prompts 
etc.

Em seg, 30 de mar de 2015 às 06:16, Abhiram Kramadhati (akramadh) 
akram...@cisco.commailto:akram...@cisco.com escreveu:
Hi Claiton,

Was this an upgrade or a fresh install of the UCCX 10.6? Also, did you confirm 
before the Finesse activation that the licenses did show up?

Regards,
Abhiram Kramadhati
Technical Solutions Manager, CBABU
CCIE Voice # 40065

From: Claiton Campos claitoncam...@gmail.commailto:claitoncam...@gmail.com
Date: Monday, 30 March 2015 2:36 am
To: cisco-voip@puck.nether.netmailto:cisco-voip@puck.nether.net 
cisco-voip@puck.nether.netmailto:cisco-voip@puck.nether.net
Subject: [cisco-voip] Cisco UCCX 10.6 demo licenses expire after activation of 
Finesse

Hello,

After installing UCCX 10.6 with demo licenses did the activation of Finesse and 
restarted the server to execute the configuration. But when I logged in UCCX 
server I came up with the following message:

cisco unified ccx license package is none .

Someone had a problem like this? Because of this I can not perform any 
configuration in UCCX . I've tried reinstalling the licenses but a message 
appears saying that this license doubled.
___
cisco-voip mailing list
cisco-voip@puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

[cisco-voip] Temp Fail since upgrade

2015-03-30 Thread Lisa Notarianni

A few weeks ago we upgraded Call Manager and Unity Connection from 8.6.2 to 
10.5.1.

We have experienced 2 intermittent outbound calling issues:


1.   “Temp Fail” shows on phone display and only option on phone button is 
“End Call”

2.   Outbound callers dial a number and hear nothing but notice the phone 
display says “Connected”.  Caller cannot hear anything.  Person they called 
cannot hear anything.
***These issues are only intermittent.***

We had no issues before the upgrade.

Is anyone experiencing the similar issues?

Thank you in advance.


[LNsignatureFile]

___
cisco-voip mailing list
cisco-voip@puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

Re: [cisco-voip] Temp Fail since upgrade

2015-03-30 Thread Lisa Notarianni

2 PRIs one each Communication Media Module– MGCP gateways. 2 Call Managers set 
up as redundant.

Cisco TAC looked at RTMT  with me today.  They want us to set up packet 
captures and try to duplicate the problem tomorrow.  I am just wondering if 
this is an upgrade issue that anyone else may be experiencing since we had no 
problems before the upgrade.



From: Ryan Huff [mailto:ryanh...@outlook.com]
Sent: Monday, March 30, 2015 4:52 PM
To: Lisa Notarianni; cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] Temp Fail since upgrade


Lisa,

SIP or TDM/PRI?

Have you gandered into RTMT and taken a look at any active / recent alerts?

Thanks,

Ryan


 Original Message 
From: Lisa Notarianni 
lisa.notaria...@scranton.edumailto:lisa.notaria...@scranton.edu
Sent: Monday, March 30, 2015 04:48 PM
To: cisco-voip@puck.nether.netmailto:cisco-voip@puck.nether.net
Subject: [cisco-voip] Temp Fail since upgrade
A few weeks ago we upgraded Call Manager and Unity Connection from 8.6.2 to 
10.5.1.

We have experienced 2 intermittent outbound calling issues:


1.   “Temp Fail” shows on phone display and only option on phone button is 
“End Call”

2.   Outbound callers dial a number and hear nothing but notice the phone 
display says “Connected”.  Caller cannot hear anything.  Person they called 
cannot hear anything.
***These issues are only intermittent.***

We had no issues before the upgrade.

Is anyone experiencing the similar issues?

Thank you in advance.


[LNsignatureFile]

___
cisco-voip mailing list
cisco-voip@puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

Re: [cisco-voip] Temp Fail since upgrade

2015-03-30 Thread Adam Frankel (afrankel)

Do you have any SDL Links going out of services?  Are you clustering over a WAN?

Temp Fail could be the result of two devices registered to different nodes on 
an active call when the SDL Link (tcp 8002) between those nodes goes down.

file view activelog syslog/CiscoSyslog

HTH,
—
Adam

From: Lisa Notarianni 
lisa.notaria...@scranton.edumailto:lisa.notaria...@scranton.edu
Date: Monday, March 30, 2015 at 4:48 PM
To: Cisco List cisco-voip@puck.nether.netmailto:cisco-voip@puck.nether.net
Subject: [cisco-voip] Temp Fail since upgrade

A few weeks ago we upgraded Call Manager and Unity Connection from 8.6.2 to 
10.5.1.

We have experienced 2 intermittent outbound calling issues:


1.   “Temp Fail” shows on phone display and only option on phone button is 
“End Call”

2.   Outbound callers dial a number and hear nothing but notice the phone 
display says “Connected”.  Caller cannot hear anything.  Person they called 
cannot hear anything.
***These issues are only intermittent.***

We had no issues before the upgrade.

Is anyone experiencing the similar issues?

Thank you in advance.


[LNsignatureFile]

___
cisco-voip mailing list
cisco-voip@puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

Re: [cisco-voip] Temp Fail since upgrade

2015-03-30 Thread Kevin Przybylowski

Are the mgcp gateways on a supported IOS for 10.5?

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Lisa 
Notarianni
Sent: Monday, March 30, 2015 5:02 PM
To: Ryan Huff; cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] Temp Fail since upgrade

2 PRIs one each Communication Media Module– MGCP gateways. 2 Call Managers set 
up as redundant.

Cisco TAC looked at RTMT  with me today.  They want us to set up packet 
captures and try to duplicate the problem tomorrow.  I am just wondering if 
this is an upgrade issue that anyone else may be experiencing since we had no 
problems before the upgrade.

From: Ryan Huff [mailto:ryanh...@outlook.com]
Sent: Monday, March 30, 2015 4:52 PM
To: Lisa Notarianni; 
cisco-voip@puck.nether.netmailto:cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] Temp Fail since upgrade

Lisa,

SIP or TDM/PRI?

Have you gandered into RTMT and taken a look at any active / recent alerts?

Thanks,

Ryan

 Original Message 
From: Lisa Notarianni 
lisa.notaria...@scranton.edumailto:lisa.notaria...@scranton.edu
Sent: Monday, March 30, 2015 04:48 PM
To: cisco-voip@puck.nether.netmailto:cisco-voip@puck.nether.net
Subject: [cisco-voip] Temp Fail since upgrade
A few weeks ago we upgraded Call Manager and Unity Connection from 8.6.2 to 
10.5.1.

We have experienced 2 intermittent outbound calling issues:

1.   “Temp Fail” shows on phone display and only option on phone button is 
“End Call”

2.   Outbound callers dial a number and hear nothing but notice the phone 
display says “Connected”.  Caller cannot hear anything.  Person they called 
cannot hear anything.
***These issues are only intermittent.***

We had no issues before the upgrade.

Is anyone experiencing the similar issues?

Thank you in advance.

[LNsignatureFile]

___
cisco-voip mailing list
cisco-voip@puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

Re: [cisco-voip] SRST

2015-03-30 Thread Carlos G Mendioroz


Indeed.
You can assign incoming COR lists to ranges of DNs (priviledge 
assignment) and outgoing COR lists to dial-peers (priviledge checking).


call-manager-fallback
  cor incoming emergencyOnly 1 1000 - 1100

or something like that.

-Carlos

Lelio Fulgenzi @ 30/03/2015 17:29 -0300 dixit:


It's been a while, but last time I looked (and this is how our SRST
routers are configured) you have to use COR lists.

It was confusing at first when I started reading them, but I just looked
at them as search spaces and partitions. The whole subset matching
diagram confused me too. :)



---
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, N1G 2W1


*From: *norm nicholson norm.nichol...@kitchener.ca
*To: *cisco-voip@puck.nether.net
*Sent: *Monday, March 30, 2015 4:21:06 PM
*Subject: *[cisco-voip] SRST

Are there ways to restrict phones or DN’s from calling out when one of
our remote 2900’s is in SRST mode. We have a PSTN that we route 911 and
9911 calls to provide ANI and ALI but I am assuming anyone can seize
that line by dialing 9 while we are in SRST mode .

Thanks

*Norm Nicholson*

*Telecom Analyst*

*City of Kitchener*

*(519) 741-2200 x 7000*


___
cisco-voip mailing list
cisco-voip@puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip



___
cisco-voip mailing list
cisco-voip@puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip



--
Carlos G Mendioroz  t...@huapi.ba.ar  LW7 EQI  Argentina
___
cisco-voip mailing list
cisco-voip@puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

Re: [cisco-voip] Temp Fail since upgrade

2015-03-30 Thread Ryan Huff

I'm not aware of any known caveats of that nature regarding that upgrade path.

Any recent network changes between ccm and the gateways since the upgrade?

I have seen situations with MGCP where the ccm side of the gateway config had 
to be rebuilt after a CCM upgrade.

Are you running an ED code version on the gateway? Check for caveats in your 
gateway code version with ccm 10.5.1

Have you restarted the MGCP process on the gateways (inside a maintenance 
window)?

In a more drastic approach, you could try and reintegrate the gateways as h.323 
and see if the issue persists.

Thanks,

Ryan

 Original Message 
From: Lisa Notarianni lisa.notaria...@scranton.edu
Sent: Monday, March 30, 2015 05:01 PM
To: Ryan Huff ryanh...@outlook.com,cisco-voip@puck.nether.net
Subject: RE: [cisco-voip] Temp Fail since upgrade

2 PRIs one each Communication Media Module– MGCP gateways. 2 Call Managers set 
up as redundant.

Cisco TAC looked at RTMT  with me today.  They want us to set up packet 
captures and try to duplicate the problem tomorrow.  I am just wondering if 
this is an upgrade issue that anyone else may be experiencing since we had no 
problems before the upgrade.



From: Ryan Huff [mailto:ryanh...@outlook.com]
Sent: Monday, March 30, 2015 4:52 PM
To: Lisa Notarianni; cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] Temp Fail since upgrade


Lisa,

SIP or TDM/PRI?

Have you gandered into RTMT and taken a look at any active / recent alerts?

Thanks,

Ryan


 Original Message 
From: Lisa Notarianni 
lisa.notaria...@scranton.edumailto:lisa.notaria...@scranton.edu
Sent: Monday, March 30, 2015 04:48 PM
To: cisco-voip@puck.nether.netmailto:cisco-voip@puck.nether.net
Subject: [cisco-voip] Temp Fail since upgrade
A few weeks ago we upgraded Call Manager and Unity Connection from 8.6.2 to 
10.5.1.

We have experienced 2 intermittent outbound calling issues:


1.   “Temp Fail” shows on phone display and only option on phone button is 
“End Call”

2.   Outbound callers dial a number and hear nothing but notice the phone 
display says “Connected”.  Caller cannot hear anything.  Person they called 
cannot hear anything.
***These issues are only intermittent.***

We had no issues before the upgrade.

Is anyone experiencing the similar issues?

Thank you in advance.


[LNsignatureFile]

___
cisco-voip mailing list
cisco-voip@puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

Re: [cisco-voip] CUBE High Availability Configuration Question

2015-03-30 Thread Brian V

In CUBE HA the sccp subsystem is in a standby mode on the backup router.  I
had the same thought as you at first trying to use the extra DSPs in the
standby.  The sccp system needs to be bound to the inside interface and the
DSP profiles need to use the same names on both routers.
On Mar 26, 2015 9:57 AM, Steve Siltman steve.silt...@assurant.com wrote:

 In a CUBE HA configuration, would it be wise to use a different local
 router for DSPFarm Profiles?  I would like to know what others are doing.
 It looks like I need a loopback address on each of the CUBEs so that Call
 Manager can talk to them directly without using the virtual ip interface.
 Otherwise the SCCP Media Resources will only register with Call Manager on
 the active side.  I'm going to test this but was wondering if the calls
 will fail over successfully between the CUBEs if using an MTP from another
 local router?

 Also wondering if putting CFB's on the CUBEs using loopbacks is possible.
 Knowing that conference calls will drop during a cube failure.

 My problem is that I don't have extra hardware DSP's on other routers but
 can find a router for software mtp profiles.  Anyone else playing with this?

 Thanks,

 Steve

 **
 This e-mail message and all attachments transmitted with it may contain
 legally privileged and/or confidential information intended solely for the
 use of the addressee(s). If the reader of this message is not the intended
 recipient, you are hereby notified that any reading, dissemination,
 distribution, copying, forwarding or other use of this message or its
 attachments is strictly prohibited. If you have received this message in
 error, please notify the sender immediately and delete this message and all
 copies and backups thereof. Thank you.
 ___
 cisco-voip mailing list
 cisco-voip@puck.nether.net
 https://puck.nether.net/mailman/listinfo/cisco-voip

___
cisco-voip mailing list
cisco-voip@puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

Re: [cisco-voip] Hardware for UC

2015-03-30 Thread Brian Palmer

I can tell you it is a lot easier to get through A2Q when you aren’t dealing 
with network attached storage.  The costs will be higher for a ucs C series 
chassis because it is an all in one server unlike the blades.  I prefer UCS-C 
series typically because I don’t have to depend on a network team that might or 
might not know what they are doing.  On the other hand if you already have a 
SAN setup and running the old environment it could be more cost effective to 
upgrade that than put in a  new C series setup.  In the B environments I have 
worked with I usually have a lot more communication going on between the 
various teams that support it as in many cases the network storage is used for 
all kinds of things within an environment.

This can go either way ultimately it is about requirements.




Thank you,
Brian



From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Scott 
Voll
Sent: Monday, March 30, 2015 11:43 AM
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] Hardware for UC

What hardware is everyone using to upgrade there UC enviroment to?

We have UCS Blades with netapp storage, but have had some limitations on IO 
(prior to the upgraded controllers) that I'm a little concerned about.

I like the idea of vmotion.  But I'm thinking if it's my back side on the line 
with my UC environment, Maybe the rack mount UCS might be a better bet?  We (UC 
Team) have also been thinking about the UCS mini with storage blade.

What our others doing?  What is Cisco suggesting these days?

Thanks

Scott


___
cisco-voip mailing list
cisco-voip@puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

[cisco-voip] Anyone has experience with captions for the Hearing Impaired / Hard of Hearing (HoH)

2015-03-30 Thread Perez, Edgardo

We are about to deploy Cisco IP phones for one of our remote offices and they 
are asking that we provide some type of caption or any other type of service 
for a staff member that is hearing impair/hard of hearing.

It seems like there are few companies that can provide these services, NexTalk, 
ClearCaptions and/or AcccesaPhone. Anyone has experience with any of these.

Appreciate any help or input you can provide.

Regards,

Edgardo (Ed) Perez
Assistant Director
Network Services
Southwest Research Institute, Information Technology Center,
6220 Culebra Rd., P.O. Drawer 28510, San Antonio, TX 78228-0510
voice: (210) 522-2880, fax: (210) 522-3424
e-mail: edgardo.pe...@swri.orgmailto:edgardo.pe...@swri.org

___
cisco-voip mailing list
cisco-voip@puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

Re: [cisco-voip] Hardware for UC


I'm going to be suggesting rack mount C series to my manager at this time, for 
a number of reasons. 


We have two data centres, so I have to split the equipment over two locations. 
Putting a full fledged B series solution would be cost prohibitive. 


Plus, I like that the C series has local storage. At the last CiscoLive I heard 
quite a few issues getting remote storage working properly and keeping it 
working properly. The calculations they presented to ensure it would work was 
like a first year calculus class. 


What I was hoping was that the UCSExpress module would be TRC approved at some 
point so we could run a few of those, but after finding out about how each 
server in the cluster has to be the same specs (still not sure why) not sure if 
that will fly in our environment. 


Lelio 


P.S. I'm not sure vMotion is fully supported within UC on UCS. I've read a few 
things on the list that seem to point to shutdown/copy/move only. 



--- 
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, N1G 2W1 

- Original Message -

From: Scott Voll svoll.v...@gmail.com 
To: cisco-voip@puck.nether.net 
Sent: Monday, March 30, 2015 11:42:49 AM 
Subject: [cisco-voip] Hardware for UC 


What hardware is everyone using to upgrade there UC enviroment to? 


We have UCS Blades with netapp storage, but have had some limitations on IO 
(prior to the upgraded controllers) that I'm a little concerned about. 


I like the idea of vmotion. But I'm thinking if it's my back side on the line 
with my UC environment, Maybe the rack mount UCS might be a better bet? We (UC 
Team) have also been thinking about the UCS mini with storage blade. 


What our others doing? What is Cisco suggesting these days? 


Thanks 


Scott 




___ 
cisco-voip mailing list 
cisco-voip@puck.nether.net 
https://puck.nether.net/mailman/listinfo/cisco-voip 

___
cisco-voip mailing list
cisco-voip@puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

Re: [cisco-voip] Hardware for UC