Re: [cisco-voip] call-forward

[Quenten Grasso]

Thanks Pavan and Ryan for your suggestions the template idea worked perfectly, 
thanks again.

Regards,
Quenten Grasso

From: Ryan Huff [mailto:ryanh...@outlook.com]
Sent: Thursday, 5 May 2016 9:50 AM
To: Quenten Grasso
Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] call-forward

You could use the call-forward max-length configuration (applied under 
ephone-dn) and specify a length lower than the number of digits they are 
forwarding to.

You could also remove the Call Forward All key from the soft key template and 
not make it an option on the phone.

Thanks,

Ryan

On May 4, 2016, at 7:40 PM, Quenten Grasso 
> wrote:
Hi,

I'm trying to work out if it's possible to stop the user of a 7960 phone from 
using the call-forward feature one CCME.

as they seem to keep setting a call-forward by accident and not realising.

Regards,
Quenten Grasso
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Re: [cisco-voip] call-forward

[Ryan Huff]

You could use the call-forward max-length configuration (applied under 
ephone-dn) and specify a length lower than the number of digits they are 
forwarding to.

You could also remove the Call Forward All key from the soft key template and 
not make it an option on the phone.

Thanks,

Ryan

On May 4, 2016, at 7:40 PM, Quenten Grasso 
> wrote:

Hi,

I'm trying to work out if it's possible to stop the user of a 7960 phone from 
using the call-forward feature one CCME.

as they seem to keep setting a call-forward by accident and not realising.

Regards,
Quenten Grasso
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Re: [cisco-voip] call-forward

[Pavan K]

I believe you can remove the callfwdall softkey using an ephone template
On May 4, 2016 6:40 PM, "Quenten Grasso"  wrote:

> Hi,
>
>
>
> I’m trying to work out if it’s possible to stop the user of a 7960 phone
> from using the call-forward feature one CCME.
>
>
>
> as they seem to keep setting a call-forward by accident and not realising.
>
>
>
> Regards,
>
> Quenten Grasso
>
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>
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[cisco-voip] call-forward

[Quenten Grasso]

Hi,

I'm trying to work out if it's possible to stop the user of a 7960 phone from 
using the call-forward feature one CCME.

as they seem to keep setting a call-forward by accident and not realising.

Regards,
Quenten Grasso
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Re: [cisco-voip] Authenticating sip trunk to ITSP from CUBE?

[Dave Goodwin]

Is there anything wrong with adding voice-class sip bind commands to ALL
the voip dial-peers, and then set the global binding to the interface that
faces the ITSP requiring authentication (since it seems sip-ua REGISTER
messages use the global bind)?

-Dave

On Wed, May 4, 2016 at 4:22 PM, Nick Barnett  wrote:

> Thanks for everybody's ideas.
>
> Unfortunately, 15.6 is OUT because it is not on the CVP 10.0 compatibility
> matrix :(
>
> I'm going to look at using multiple registrars and see if I can trick it
> into behaving... if that doesn't work, I guess I'll have to remove my
> global binding...
>
>
> On Wed, May 4, 2016 at 11:35 AM, Sreekanth  wrote:
>
>> Yes, sip-ua tells CUBE to send REGISTER messages towards a Registrar
>> server globally with the authentication and credential parameters. These
>> REGISTER messages will be bound to the interface that is bound under voice
>> service voip -> sip. However, in the 15.6(2)T version, the tenant
>> configurations under the dial-peers will instruct the CUBE to send out
>> REGISTER messages.
>>
>> I just checked with the router in my lab and actually, option 2 won't be
>> possible. It won't instruct the CUBE to send out REGISTER messages. It will
>> only instruct the CUBE to add authentication credentials and realm settings
>> when sending out the INVITE messages towards the session target configured
>> under the dial-peer.
>>
>> You will have to go with option 1.
>> *1. Create the voice class tenant for the SIP trunk to ITSP and bind it
>> with the right interface.*
>> voice class tenant 1
>>   registrar dns:cisco.com expires 3600
>>   credentials username cisco password cisco realm cisco.com
>>   authentication username cisco123 password 7 cisco123
>>   sip-server dns:cisco.com
>>   bind control source-interface GigabitEthernet0/2
>>   bind media source-interface GigabitEthernet0/2
>>   early-offer forced
>>
>> *2. Apply the voice class tenant to the dial-peer. Create specific
>> dial-peers towards ITSP.*
>> dial-peer voice X voip
>>  voice-class sip tenant 1
>>
>> When this is done, CUBE will send REGISTER messages as well towards this
>> ITSP with the traffic bound to gig0/2.
>> This way you can have multiple ITSP trunks on 1 CUBE.
>>
>> Sreekanth
>>
>>
>>
>>
>> On Wednesday 04 May 2016 09:29 PM, Nick Barnett wrote:
>>
>> I'm currently on c3900e-universalk9-mz.SPA.153-3.M6, but can totally
>> upgrade. Was actually planning on going to 15.4 this weekend. Jumping 3
>> versions kind of scares me, so maybe staging is in order.
>>
>> *I do have some limited auth commands on the dial peer, if this is what
>> you were talking about... but I don't think it applies in this scenario. I
>> don't have any options for credentials:*
>> CUBE(config-dial-peer)#voice-class sip authenticate ?
>>   redirecting-number  Use redirecting number credentials while
>> authenticating
>>
>> CUBE(config-dial-peer)#voice-class sip cred
>> CUBE(config-dial-peer)#voice-class sip c?
>>   call-route  calltype-video  conn-reuse  copy-list
>>
>> *There is also the registration commands:*
>> CUBE(config-dial-peer)#voice-class sip registration ?
>>   passthrough  SIP Registration Passthrough Options
>>
>> CUBE(config-dial-peer)#voice-class sip registration passthrough ?
>>   dynamic  SIP Registration Use dynamic Registrar Details
>> (default)
>>   local-fallback   Local Fallback - (e2e)
>>   rate-limit   SIP Registration pass-through rate-limit Options
>>   reg-sync Registration Sync - send REGISTER when registrar up
>> (p2p)
>>   registrar-index  Registrar Index(s) used for registration passthrough
>>   static   SIP Registration Use static Registrar Details
>>   system   Use global registration passthrough CLI setting
>>   
>>
>> *I tried using the system passthrough setting, but it did not work.*
>>
>> *I need to make sure I understand what is actually happening.*
>>
>> *I don't think the CUBE is even looking at dial-peers for REGISTER
>> messages. Am I correct?  If so, no amount of dial peer settings is going to
>> make any difference here... unless there is a way to create a dial-peer
>> that will intercept REGISTER messages. I believe it is using the REALM
>> settings in the credentials and authentication strings (that I entered into
>> sip-ua). And sip-ua is using the global bind settings I set within voice
>> service voip -> SIP (which are set to the inside interface).*
>>
>> *Please set me straight!*
>>
>> Thanks,
>> Nick
>>
>> On Wed, May 4, 2016 at 10:37 AM, Sreekanth Narayanan (sreenara) <
>> sreen...@cisco.com> wrote:
>>
>>> 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 bound
>>> to different interfaces.
>>>
>>> 2. In 

Re: [cisco-voip] Looking for beta testers for UCCX Call Flow Designer

[Erick Bergquist]

Tanner,

I would love to try this out and put it through it's paces.

Erick


On Wed, May 4, 2016 at 1:42 PM, Aaron Banks  wrote:
> I would love to try out that software!
>
>
>
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Re: [cisco-voip] Authenticating sip trunk to ITSP from CUBE?

[Nick Barnett]

Thanks for everybody's ideas.

Unfortunately, 15.6 is OUT because it is not on the CVP 10.0 compatibility
matrix :(

I'm going to look at using multiple registrars and see if I can trick it
into behaving... if that doesn't work, I guess I'll have to remove my
global binding...


On Wed, May 4, 2016 at 11:35 AM, Sreekanth  wrote:

> Yes, sip-ua tells CUBE to send REGISTER messages towards a Registrar
> server globally with the authentication and credential parameters. These
> REGISTER messages will be bound to the interface that is bound under voice
> service voip -> sip. However, in the 15.6(2)T version, the tenant
> configurations under the dial-peers will instruct the CUBE to send out
> REGISTER messages.
>
> I just checked with the router in my lab and actually, option 2 won't be
> possible. It won't instruct the CUBE to send out REGISTER messages. It will
> only instruct the CUBE to add authentication credentials and realm settings
> when sending out the INVITE messages towards the session target configured
> under the dial-peer.
>
> You will have to go with option 1.
> *1. Create the voice class tenant for the SIP trunk to ITSP and bind it
> with the right interface.*
> voice class tenant 1
>   registrar dns:cisco.com expires 3600
>   credentials username cisco password cisco realm cisco.com
>   authentication username cisco123 password 7 cisco123
>   sip-server dns:cisco.com
>   bind control source-interface GigabitEthernet0/2
>   bind media source-interface GigabitEthernet0/2
>   early-offer forced
>
> *2. Apply the voice class tenant to the dial-peer. Create specific
> dial-peers towards ITSP.*
> dial-peer voice X voip
>  voice-class sip tenant 1
>
> When this is done, CUBE will send REGISTER messages as well towards this
> ITSP with the traffic bound to gig0/2.
> This way you can have multiple ITSP trunks on 1 CUBE.
>
> Sreekanth
>
>
>
>
> On Wednesday 04 May 2016 09:29 PM, Nick Barnett wrote:
>
> I'm currently on c3900e-universalk9-mz.SPA.153-3.M6, but can totally
> upgrade. Was actually planning on going to 15.4 this weekend. Jumping 3
> versions kind of scares me, so maybe staging is in order.
>
> *I do have some limited auth commands on the dial peer, if this is what
> you were talking about... but I don't think it applies in this scenario. I
> don't have any options for credentials:*
> CUBE(config-dial-peer)#voice-class sip authenticate ?
>   redirecting-number  Use redirecting number credentials while
> authenticating
>
> CUBE(config-dial-peer)#voice-class sip cred
> CUBE(config-dial-peer)#voice-class sip c?
>   call-route  calltype-video  conn-reuse  copy-list
>
> *There is also the registration commands:*
> CUBE(config-dial-peer)#voice-class sip registration ?
>   passthrough  SIP Registration Passthrough Options
>
> CUBE(config-dial-peer)#voice-class sip registration passthrough ?
>   dynamic  SIP Registration Use dynamic Registrar Details (default)
>   local-fallback   Local Fallback - (e2e)
>   rate-limit   SIP Registration pass-through rate-limit Options
>   reg-sync Registration Sync - send REGISTER when registrar up
> (p2p)
>   registrar-index  Registrar Index(s) used for registration passthrough
>   static   SIP Registration Use static Registrar Details
>   system   Use global registration passthrough CLI setting
>   
>
> *I tried using the system passthrough setting, but it did not work.*
>
> *I need to make sure I understand what is actually happening.*
>
> *I don't think the CUBE is even looking at dial-peers for REGISTER
> messages. Am I correct?  If so, no amount of dial peer settings is going to
> make any difference here... unless there is a way to create a dial-peer
> that will intercept REGISTER messages. I believe it is using the REALM
> settings in the credentials and authentication strings (that I entered into
> sip-ua). And sip-ua is using the global bind settings I set within voice
> service voip -> SIP (which are set to the inside interface).*
>
> *Please set me straight!*
>
> Thanks,
> Nick
>
> On Wed, May 4, 2016 at 10:37 AM, Sreekanth Narayanan (sreenara) <
> sreen...@cisco.com> wrote:
>
>> 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 bound
>> to different interfaces.
>>
>> 2. In your current IOS, check if you are able to configure the
>> authentication and credential commands at the dial peer level. I am not
>> sure which IOS had this introduced but it is worth a try.
>>
>>
>>
>> Sreekanth
>>
>> Sent from a phone.
>>
>>
>>  Original message 
>> From: Nick Barnett 
>> Date: 5/4/16 8:03 PM (GMT+05:30)
>> To: Brian Meade 
>> Cc: Cisco VoIP Group 
>> 

Re: [cisco-voip] Looking for beta testers for UCCX Call Flow Designer

[Aaron Banks]

I would love to try out that software!


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Re: [cisco-voip] Authenticating sip trunk to ITSP from CUBE?

[Sreekanth]

Yes, sip-ua tells CUBE to send REGISTER messages towards a Registrar 
server globally with the authentication and credential parameters. These 
REGISTER messages will be bound to the interface that is bound under 
voice service voip -> sip. However, in the 15.6(2)T version, the tenant 
configurations under the dial-peers will instruct the CUBE to send out 
REGISTER messages.


I just checked with the router in my lab and actually, option 2 won't be 
possible. It won't instruct the CUBE to send out REGISTER messages. It 
will only instruct the CUBE to add authentication credentials and realm 
settings when sending out the INVITE messages towards the session target 
configured under the dial-peer.


You will have to go with option 1.
*1. Create the voice class tenant for the SIP trunk to ITSP and bind it 
with the right interface.*

voice class tenant 1
  registrar dns:cisco.com expires 3600
  credentials username cisco password cisco realm cisco.com
  authentication username cisco123 password 7 cisco123
  sip-server dns:cisco.com
  bind control source-interface GigabitEthernet0/2
  bind media source-interface GigabitEthernet0/2
  early-offer forced

*2. Apply the voice class tenant to the dial-peer. Create specific 
dial-peers towards ITSP.*

dial-peer voice X voip
 voice-class sip tenant 1

When this is done, CUBE will send REGISTER messages as well towards this 
ITSP with the traffic bound to gig0/2.

This way you can have multiple ITSP trunks on 1 CUBE.

Sreekanth




On Wednesday 04 May 2016 09:29 PM, Nick Barnett wrote:
I'm currently on c3900e-universalk9-mz.SPA.153-3.M6, but can totally 
upgrade. Was actually planning on going to 15.4 this weekend. Jumping 
3 versions kind of scares me, so maybe staging is in order.


*I do have some limited auth commands on the dial peer, if this is 
what you were talking about... but I don't think it applies in this 
scenario. I don't have any options for credentials:*

CUBE(config-dial-peer)#voice-class sip authenticate ?
  redirecting-number  Use redirecting number credentials while 
authenticating


CUBE(config-dial-peer)#voice-class sip cred
CUBE(config-dial-peer)#voice-class sip c?
  call-route  calltype-video  conn-reuse  copy-list
*
*
*There is also the registration commands:*
CUBE(config-dial-peer)#voice-class sip registration ?
  passthrough  SIP Registration Passthrough Options

CUBE(config-dial-peer)#voice-class sip registration passthrough ?
  dynamic  SIP Registration Use dynamic Registrar Details 
(default)

  local-fallback   Local Fallback - (e2e)
  rate-limit   SIP Registration pass-through rate-limit Options
  reg-sync Registration Sync - send REGISTER when registrar up 
(p2p)

  registrar-index  Registrar Index(s) used for registration passthrough
  static   SIP Registration Use static Registrar Details
  system   Use global registration passthrough CLI setting
  

*I tried using the system passthrough setting, but it did not work.*
*
*
*I need to make sure I understand what is actually happening.*
*
*
*I don't think the CUBE is even looking at dial-peers for REGISTER 
messages. Am I correct?  If so, no amount of dial peer settings is 
going to make any difference here... unless there is a way to create a 
dial-peer that will intercept REGISTER messages. I believe it is using 
the REALM settings in the credentials and authentication strings (that 
I entered into sip-ua). And sip-ua is using the global bind settings I 
set within voice service voip -> SIP (which are set to the inside 
interface).*

*
*
*Please set me straight!*

Thanks,
Nick

On Wed, May 4, 2016 at 10:37 AM, Sreekanth Narayanan (sreenara) 
> wrote:


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 bound to different interfaces.

2. In your current IOS, check if you are able to configure the
authentication and credential commands at the dial peer level. I
am not sure which IOS had this introduced but it is worth a try.



Sreekanth

Sent from a phone.


 Original message 
From: Nick Barnett >
Date: 5/4/16 8:03 PM (GMT+05:30)
To: Brian Meade >
Cc: Cisco VoIP Group >
Subject: Re: [cisco-voip] Authenticating sip trunk to ITSP from CUBE?

I'm binding control and media to my inside interface:

sip

  bind control source-interface GigabitEthernet0/0
  bind media source-interface GigabitEthernet0/0

I suspect this is the issue... is there any way to make the
REGISTER messages come from the 

Re: [cisco-voip] Looking for beta testers for UCCX Call Flow Designer

[Terry Oakley]

Tanner I can see the benefit of this and have been looking for a tool to do 
this for a long time.  Can you add me to your beta list?  We are just getting 
ready to consolidate our service areas into one call location so would like to 
test your product for that change.

Thanks

Terry


Terry Oakley
Telecommunications Coordinator | Information Technology Services
Red Deer College |100 College Blvd. | Box 5005 | Red Deer | Alberta | T4N 5H5
work (403) 342-3521   |  FAX (403) 343-4034



From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of 
Tanner Ezell
Sent: May 3, 2016 7:24 PM
To: Cisco VOIP 
Subject: Re: [cisco-voip] Looking for beta testers for UCCX Call Flow Designer

I now realize I forgot to include the video!  
https://www.youtube.com/watch?v=CUwGGbPjmWY

On Tue, May 3, 2016 at 3:34 PM, Tanner Ezell 
> wrote:
As many of you know I've been developing a software solution to bring visual 
call flow development to UCCX (you can see a demo video I put together 
showcasing functionality, including export to Visio and AEF). After much time 
I'm extremely pleased to announce we're looking for beta testers to play with 
the software and provide feedback.

Ideally I'm looking for folks who are able to apply real world needs to the 
application to make recommendations for improvement and enhancements.

Anyone interested please feel free to email me directly or on thread to discuss 
further.

The software is Web based, so if you're a Mac or Linux guy, you can still apply!

I'm also happy to answer any questions out there regarding the software.

Regards,
Tanner Ezell

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Re: [cisco-voip] Authenticating sip trunk to ITSP from CUBE?

[Nick Barnett]

I'm currently on c3900e-universalk9-mz.SPA.153-3.M6, but can totally
upgrade. Was actually planning on going to 15.4 this weekend. Jumping 3
versions kind of scares me, so maybe staging is in order.

*I do have some limited auth commands on the dial peer, if this is what you
were talking about... but I don't think it applies in this scenario. I
don't have any options for credentials:*
CUBE(config-dial-peer)#voice-class sip authenticate ?
  redirecting-number  Use redirecting number credentials while
authenticating

CUBE(config-dial-peer)#voice-class sip cred
CUBE(config-dial-peer)#voice-class sip c?
  call-route  calltype-video  conn-reuse  copy-list

*There is also the registration commands:*
CUBE(config-dial-peer)#voice-class sip registration ?
  passthrough  SIP Registration Passthrough Options

CUBE(config-dial-peer)#voice-class sip registration passthrough ?
  dynamic  SIP Registration Use dynamic Registrar Details (default)
  local-fallback   Local Fallback - (e2e)
  rate-limit   SIP Registration pass-through rate-limit Options
  reg-sync Registration Sync - send REGISTER when registrar up (p2p)
  registrar-index  Registrar Index(s) used for registration passthrough
  static   SIP Registration Use static Registrar Details
  system   Use global registration passthrough CLI setting
  

*I tried using the system passthrough setting, but it did not work.*

*I need to make sure I understand what is actually happening.*

*I don't think the CUBE is even looking at dial-peers for REGISTER
messages. Am I correct?  If so, no amount of dial peer settings is going to
make any difference here... unless there is a way to create a dial-peer
that will intercept REGISTER messages. I believe it is using the REALM
settings in the credentials and authentication strings (that I entered into
sip-ua). And sip-ua is using the global bind settings I set within voice
service voip -> SIP (which are set to the inside interface).*

*Please set me straight!*

Thanks,
Nick

On Wed, May 4, 2016 at 10:37 AM, Sreekanth Narayanan (sreenara) <
sreen...@cisco.com> wrote:

> 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 bound
> to different interfaces.
>
> 2. In your current IOS, check if you are able to configure the
> authentication and credential commands at the dial peer level. I am not
> sure which IOS had this introduced but it is worth a try.
>
>
>
> Sreekanth
>
> Sent from a phone.
>
>
>  Original message 
> From: Nick Barnett 
> Date: 5/4/16 8:03 PM (GMT+05:30)
> To: Brian Meade 
> Cc: Cisco VoIP Group 
> Subject: Re: [cisco-voip] Authenticating sip trunk to ITSP from CUBE?
>
> I'm binding control and media to my inside interface:
>
> sip
>
>   bind control source-interface GigabitEthernet0/0
>   bind media source-interface GigabitEthernet0/0
>
> I suspect this is the issue... is there any way to make the REGISTER
> messages come from the outside gi0/1 interface?
>
> The reason I'm binding to inside is that we have a a very fluid internal
> network. I have to make and modify internal dial peers almost daily.  When
> I need to create a dial peer and put the bind statements on the dial peer,
> it won't bind properly since there are active SIP calls on the CUBE... so I
> bound it globally. My external dial peers rarely change, so I bind those
> directly to gi0/1 (on the DP).
>
> I was under the impression that REGISTER events can take place without a
> dial peer... but is there a way to, i dunno, make a dial peer for register
> messages?  Can I use SIP profile magic to get it working as is?
>
> I found this article which is pretty much exactly what I'm dealing with,
> but it doesn't mention REGISTER at all...
>
>https://supportforums.cisco.com/blog/154506
> 
>
>
>
>
> On Wed, May 4, 2016 at 9:06 AM, Brian Meade  wrote:
>
>> Do you already have the SIP bind under voice service voip?
>> voice service voice
>>  sip
>>   bind all source-interface FastEthernet0
>>
>> On Wed, May 4, 2016 at 9:58 AM, Nick Barnett 
>> wrote:
>>
>>> I've never dealt with an authenticated SIP trunk before and I'm having
>>> some issues. I was wondering if anyone has had a similar experience. I
>>> already have 2 SIP trunks from ITSP-1 that do NOT require authentication.
>>> These are working fine and have been for years.
>>>
>>> We are adding ITSP-2 and 

Re: [cisco-voip] Authenticating sip trunk to ITSP from CUBE?

[Sreekanth Narayanan (sreenara)]

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 bound to different 
interfaces.

2. In your current IOS, check if you are able to configure the authentication 
and credential commands at the dial peer level. I am not sure which IOS had 
this introduced but it is worth a try.



Sreekanth

Sent from a phone.


 Original message 
From: Nick Barnett 
Date: 5/4/16 8:03 PM (GMT+05:30)
To: Brian Meade 
Cc: Cisco VoIP Group 
Subject: Re: [cisco-voip] Authenticating sip trunk to ITSP from CUBE?


I'm binding control and media to my inside interface:

sip

  bind control source-interface GigabitEthernet0/0
  bind media source-interface GigabitEthernet0/0

I suspect this is the issue... is there any way to make the REGISTER messages 
come from the outside gi0/1 interface?

The reason I'm binding to inside is that we have a a very fluid internal 
network. I have to make and modify internal dial peers almost daily.  When I 
need to create a dial peer and put the bind statements on the dial peer, it 
won't bind properly since there are active SIP calls on the CUBE... so I bound 
it globally. My external dial peers rarely change, so I bind those directly to 
gi0/1 (on the DP).

I was under the impression that REGISTER events can take place without a dial 
peer... but is there a way to, i dunno, make a dial peer for register messages? 
 Can I use SIP profile magic to get it working as is?

I found this article which is pretty much exactly what I'm dealing with, but it 
doesn't mention REGISTER at all...

   
https://supportforums.cisco.com/blog/154506



On Wed, May 4, 2016 at 9:06 AM, Brian Meade 
> wrote:
Do you already have the SIP bind under voice service voip?
voice service voice
 sip
  bind all source-interface FastEthernet0

On Wed, May 4, 2016 at 9:58 AM, Nick Barnett 
> wrote:
I've never dealt with an authenticated SIP trunk before and I'm having some 
issues. I was wondering if anyone has had a similar experience. I already have 
2 SIP trunks from ITSP-1 that do NOT require authentication. These are working 
fine and have been for years.

We are adding ITSP-2 and their SIP service DOES require auth.  I've followed 
their integration guide (which left a lot to be desired) and their acceptance 
team is telling me my auth is coming from our private class A address.

Our CUBE is in HA with an inside (10.x.x.x) and outside (public) IP address. 
They are seeing REGISTER messages sourcing the inside VIP.

I was looking around for an auth BIND statement or something like that, but I 
haven't had any luck. Any pointers?

Thanks,
Nick

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Re: [cisco-voip] Authenticating sip trunk to ITSP from CUBE?

[NateCCIE]

BINDs on CUBEs are evil.  It will use the egress interface automatically 
without binds, and as you have found, you cannot change binds with calls up.  

 

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Nick 
Barnett
Sent: Wednesday, May 04, 2016 8:32 AM
To: Brian Meade 
Cc: Cisco VoIP Group 
Subject: Re: [cisco-voip] Authenticating sip trunk to ITSP from CUBE?

 

I'm binding control and media to my inside interface:

sip  

  bind control source-interface GigabitEthernet0/0
  bind media source-interface GigabitEthernet0/0

I suspect this is the issue... is there any way to make the REGISTER messages 
come from the outside gi0/1 interface?

The reason I'm binding to inside is that we have a a very fluid internal 
network. I have to make and modify internal dial peers almost daily.  When I 
need to create a dial peer and put the bind statements on the dial peer, it 
won't bind properly since there are active SIP calls on the CUBE... so I bound 
it globally. My external dial peers rarely change, so I bind those directly to 
gi0/1 (on the DP).

I was under the impression that REGISTER events can take place without a dial 
peer... but is there a way to, i dunno, make a dial peer for register messages? 
 Can I use SIP profile magic to get it working as is?

I found this article which is pretty much exactly what I'm dealing with, but it 
doesn't mention REGISTER at all...

   https://supportforums.cisco.com/blog/154506 

 

 

 

On Wed, May 4, 2016 at 9:06 AM, Brian Meade  > wrote:

Do you already have the SIP bind under voice service voip?

voice service voice

 sip

  bind all source-interface FastEthernet0

 

On Wed, May 4, 2016 at 9:58 AM, Nick Barnett  > wrote:

I've never dealt with an authenticated SIP trunk before and I'm having some 
issues. I was wondering if anyone has had a similar experience. I already have 
2 SIP trunks from ITSP-1 that do NOT require authentication. These are working 
fine and have been for years.

 

We are adding ITSP-2 and their SIP service DOES require auth.  I've followed 
their integration guide (which left a lot to be desired) and their acceptance 
team is telling me my auth is coming from our private class A address.

 

Our CUBE is in HA with an inside (10.x.x.x) and outside (public) IP address. 
They are seeing REGISTER messages sourcing the inside VIP.

 

I was looking around for an auth BIND statement or something like that, but I 
haven't had any luck. Any pointers?

 

Thanks,

Nick

 

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Re: [cisco-voip] Authenticating sip trunk to ITSP from CUBE?

[Nick Barnett]

I'm binding control and media to my inside interface:

sip

  bind control source-interface GigabitEthernet0/0
  bind media source-interface GigabitEthernet0/0

I suspect this is the issue... is there any way to make the REGISTER
messages come from the outside gi0/1 interface?

The reason I'm binding to inside is that we have a a very fluid internal
network. I have to make and modify internal dial peers almost daily.  When
I need to create a dial peer and put the bind statements on the dial peer,
it won't bind properly since there are active SIP calls on the CUBE... so I
bound it globally. My external dial peers rarely change, so I bind those
directly to gi0/1 (on the DP).

I was under the impression that REGISTER events can take place without a
dial peer... but is there a way to, i dunno, make a dial peer for register
messages?  Can I use SIP profile magic to get it working as is?

I found this article which is pretty much exactly what I'm dealing with,
but it doesn't mention REGISTER at all...

   https://supportforums.cisco.com/blog/154506





On Wed, May 4, 2016 at 9:06 AM, Brian Meade  wrote:

> Do you already have the SIP bind under voice service voip?
> voice service voice
>  sip
>   bind all source-interface FastEthernet0
>
> On Wed, May 4, 2016 at 9:58 AM, Nick Barnett 
> wrote:
>
>> I've never dealt with an authenticated SIP trunk before and I'm having
>> some issues. I was wondering if anyone has had a similar experience. I
>> already have 2 SIP trunks from ITSP-1 that do NOT require authentication.
>> These are working fine and have been for years.
>>
>> We are adding ITSP-2 and their SIP service DOES require auth.  I've
>> followed their integration guide (which left a lot to be desired) and their
>> acceptance team is telling me my auth is coming from our private class A
>> address.
>>
>> Our CUBE is in HA with an inside (10.x.x.x) and outside (public) IP
>> address. They are seeing REGISTER messages sourcing the inside VIP.
>>
>> I was looking around for an auth BIND statement or something like that,
>> but I haven't had any luck. Any pointers?
>>
>> Thanks,
>> Nick
>>
>> ___
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>> cisco-voip@puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>>
>
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Re: [cisco-voip] Authenticating sip trunk to ITSP from CUBE?

[Ryan Huff]

Nick,

What are you binding SIP to?

Thanks,

Ryan

> On May 4, 2016, at 9:59 AM, Nick Barnett  wrote:
> 
> I've never dealt with an authenticated SIP trunk before and I'm having some 
> issues. I was wondering if anyone has had a similar experience. I already 
> have 2 SIP trunks from ITSP-1 that do NOT require authentication. These are 
> working fine and have been for years.
> 
> We are adding ITSP-2 and their SIP service DOES require auth.  I've followed 
> their integration guide (which left a lot to be desired) and their acceptance 
> team is telling me my auth is coming from our private class A address.
> 
> Our CUBE is in HA with an inside (10.x.x.x) and outside (public) IP address. 
> They are seeing REGISTER messages sourcing the inside VIP.
> 
> I was looking around for an auth BIND statement or something like that, but I 
> haven't had any luck. Any pointers?
> 
> Thanks,
> Nick
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Re: [cisco-voip] Authenticating sip trunk to ITSP from CUBE?

[Brian Meade]

Do you already have the SIP bind under voice service voip?
voice service voice
 sip
  bind all source-interface FastEthernet0

On Wed, May 4, 2016 at 9:58 AM, Nick Barnett  wrote:

> I've never dealt with an authenticated SIP trunk before and I'm having
> some issues. I was wondering if anyone has had a similar experience. I
> already have 2 SIP trunks from ITSP-1 that do NOT require authentication.
> These are working fine and have been for years.
>
> We are adding ITSP-2 and their SIP service DOES require auth.  I've
> followed their integration guide (which left a lot to be desired) and their
> acceptance team is telling me my auth is coming from our private class A
> address.
>
> Our CUBE is in HA with an inside (10.x.x.x) and outside (public) IP
> address. They are seeing REGISTER messages sourcing the inside VIP.
>
> I was looking around for an auth BIND statement or something like that,
> but I haven't had any luck. Any pointers?
>
> Thanks,
> Nick
>
> ___
> cisco-voip mailing list
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>
>
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[cisco-voip] Authenticating sip trunk to ITSP from CUBE?

[Nick Barnett]

I've never dealt with an authenticated SIP trunk before and I'm having some
issues. I was wondering if anyone has had a similar experience. I already
have 2 SIP trunks from ITSP-1 that do NOT require authentication. These are
working fine and have been for years.

We are adding ITSP-2 and their SIP service DOES require auth.  I've
followed their integration guide (which left a lot to be desired) and their
acceptance team is telling me my auth is coming from our private class A
address.

Our CUBE is in HA with an inside (10.x.x.x) and outside (public) IP
address. They are seeing REGISTER messages sourcing the inside VIP.

I was looking around for an auth BIND statement or something like that, but
I haven't had any luck. Any pointers?

Thanks,
Nick
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Re: [cisco-voip] LDAP, Sync, Filters and CUCM

[Ed Leatherman]

We just ran into the behavior with some agent username changes last week,
and it was not even LDAP related - the technician was just trying to be
helpful and update some local end user id's for consistency and didn't
think that change needed any pre-testing. When he noticed it, he changed
all the names back but their skills such were still cleared out. Luckily it
was a small number of accounts and we were able to get skills re-assigned
quickly, but I think it will still have a minor impact on the agent
reporting.


On Tue, May 3, 2016 at 4:41 PM, Justin Steinberg 
wrote:

> CUCM doesn't delete the users when they are marked inactive.   you can fix
> the LDAP agreement, resync and get them back if you get it fixed before the
> garbage cycle clears them out.
>
> this is really an issue for the UCCX team, they should handle user changes
> in CUCM more gracefully.  This is also the case for situations where the
> username changes.  If the account changes in the LDAP directory, CUCM will
> usually see the change and update the username.  however, UCCX doesn't
> track the same way and will delete the agent's old account and create a new
> account for the new username.  the new UCCX account will have no skills, no
> team, no associated historical data, etc.I haven't tested this with
> UCCX 11, but this is how it has been on UCCX when I last tested it.
>
> On Tue, May 3, 2016 at 3:04 PM, Anthony Holloway <
> avholloway+cisco-v...@gmail.com> wrote:
>
>> This is related to my post I just made on UCCX and LDAP via CUCM.
>>
>> I also just found out that a CUCM with an already synced user database
>> behaves in the following two ways:
>>
>>1. If you modify the filter such that it matches 0 records, the sync
>>doesn't happen at all.  No users are marked as Inactive, no users are
>>pulled in, and no users are updated
>>
>>You will see this in the DirSync log
>>Dirsync synched zero users. Please verify the custom LDAP filter
>>configured for this agreement
>>
>>2. However, if you modify the filter such that it matches a single
>>record, the sync does happen.  All of the non-matched users will become
>>Inactive.
>>
>>You will see this in the DirSync log (the value 1660 will vary by
>>scenario)
>>DSDBInterface.setUserInactive Found 1660 users in database needing
>>update
>>
>> For #1, it seems like this might be a protection mechanism, preventing
>> you from destroying your entire corporate directory.  Because, recall that
>> EM, Jabber, Finesse, etc., all require your account to be Active Synced in
>> order to authenticate you; therefore, making 1660 people go Inactive will
>> have a large impact.  Or perhaps it was a coding error, and they should
>> have made all users go Inactive?
>>
>> For #2, if we're thinking #1 could be a protective mode, then wouldn't
>> 100% user loss be just as bad as 99%?  Perhaps the protection mechanism
>> should look for a smaller percentage drop in Active users and prohibit an
>> LDAP update at that time and display a warning on the page (I.e., Like the
>> last known good backup now shows up on the About page).
>>
>> What do you think?  Have you seen this before?  Has it bit you?  Am I
>> missing something obvious?  Let me know.  Thanks.
>>
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>>
>>
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>


-- 
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Re: [cisco-voip] Looking for beta testers for UCCX Call Flow Designer

[Pavan K]

Aah nice. Didn't realize we could generate visios from aef. Sign me up
please.

Thanks.
On May 3, 2016 10:56 PM, "Anthony Holloway" 
wrote:

> Value is hard to place properly, as it's so subjective.
>
> What makes this an even harder value proposition is that it's from a
> smaller third party company (sorry, no offense), and a customer who invests
> in this tool, will have to bet on it being around for the life of the
> system.  Else, it was simply just a good idea at the time.
>
> Where I see this really adding value is in green field deployments where
> you can involve BAs in the documentation process, all the while they are
> piecing together the building blocks for the actual script.  Or also in
> daily operations, where an update to the documentation, translates directly
> into the change itself.  This eliminates documentation of call flows
> becoming out of date.  And since you can use this tool to simply create
> images or Visio diagrams, without the need to export AEF files, you can
> technically use it for any and all call flow diagrams (I.e., Hunt Groups)
> keeping a consistent look and feel to your documents.
>
> I'd like to try this tool out, and I have an upcoming project that's not
> too large where I think this could be a good fit.  Reach out to me Tanner.
>
> On Tue, May 3, 2016 at 9:41 PM, Kevin Przybylowski  > wrote:
>
>> I can see value if it will take existing AEFs and export them to a
>> functional Visio diagram.
>>
>>
>>
>> *From:* cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] *On
>> Behalf Of *Pavan K
>> *Sent:* Tuesday, May 3, 2016 9:31 PM
>> *To:* Tanner Ezell ; Cisco VOIP <
>> cisco-voip@puck.nether.net>
>> *Subject:* Re: [cisco-voip] Looking for beta testers for UCCX Call Flow
>> Designer
>>
>>
>>
>> I watched the video and am curious. What is the value add to uccx editor ?
>>
>>
>>
>>
>>
>> On Tue, May 3, 2016, 8:24 PM Tanner Ezell  wrote:
>>
>> I now realize I forgot to include the video!
>> https://www.youtube.com/watch?v=CUwGGbPjmWY
>>
>>
>>
>> On Tue, May 3, 2016 at 3:34 PM, Tanner Ezell 
>> wrote:
>>
>> As many of you know I've been developing a software solution to bring
>> visual call flow development to UCCX (you can see a demo video I put
>> together showcasing functionality, including export to Visio and AEF).
>> After much time I'm extremely pleased to announce we're looking for beta
>> testers to play with the software and provide feedback.
>>
>>
>>
>> Ideally I'm looking for folks who are able to apply real world needs to
>> the application to make recommendations for improvement and enhancements.
>>
>>
>>
>> Anyone interested please feel free to email me directly or on thread to
>> discuss further.
>>
>>
>>
>> The software is Web based, so if you're a Mac or Linux guy, you can still
>> apply!
>>
>>
>>
>> I'm also happy to answer any questions out there regarding the software.
>>
>>
>>
>> Regards,
>>
>> Tanner Ezell
>>
>>
>>
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