[cisco-voip] Extension that hangs up on the user?

[James Andrewartha]

Hi list,

Is there a way in CUCM to make an extension that hangs up on the other
end? Currently we have a Unity Connection AA that does that, but it's
literally the only thing CUC is being used for and I want to get rid of
it. Currently we have our AAs (and voicemail) in Exchange 2007, which is
being upgraded to 2013 soon, but as far as I can tell there's no way to
have it hang up on the caller, so I transfer to the AA in Unity.

Thanks,

-- 
James Andrewartha
Network & Projects Engineer
Christ Church Grammar School
Claremont, Western Australia
Ph. (08) 9442 1757
Mob. 0424 160 877
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Re: [cisco-voip] UK PRI MGCP

[秀王]

Sorry to dig a very old thread.

Is this method support today? Is it still limited to like maximum 5 field
where we can put them into maintenance mode?




On Thu, Dec 3, 2009 at 11:36 PM, VoiceNoob  wrote:

>  No you have done some tweaks somewhere to make a partial PRI work with
> MGCP. It does not work correctly without some changes.
>
>
>
> It may be supported by TAC but not supported  by CUCM. J
>
>
>
> *From:* cisco-voip-boun...@puck.nether.net [mailto:
> cisco-voip-boun...@puck.nether.net] *On Behalf Of *Lewis, Chris
> *Sent:* Thursday, December 03, 2009 8:52 AM
> *To:* Charles Goldsmith; Joe Martini (joemar2)
>
> *Cc:* cisco-voip@puck.nether.net
> *Subject:* Re: [cisco-voip] UK PRI MGCP
>
>
>
> To let you know, I have been using 15 channels on an ISDN30 for over 2
> years - in my UK site - with no additional tweaks on CCM or the VGW’s
>
>
>
> Plus it is supported by TAC – Spoken to Joe a few times I in the past  J
>
>
>
> Chris
>
>
>
> *From:* cisco-voip-boun...@puck.nether.net [mailto:
> cisco-voip-boun...@puck.nether.net] *On Behalf Of *Charles Goldsmith
> *Sent:* 03 December 2009 14:36
> *To:* Joe Martini (joemar2)
> *Cc:* cisco-voip@puck.nether.net
> *Subject:* Re: [cisco-voip] UK PRI MGCP
>
>
>
> *Note: *Cisco CallManager does not support the configuration or use of a
> fractional PRI when you use it with MGCP. If fractional PRI is necessary,
> you can use H.323 instead of MGCP.
>
> This can be found on
> http://www.cisco.com/en/US/tech/tk1077/technologies_tech_note09186a00806fedbe.shtml#topic1
>
> It's not supported and won't work without some tweaking.
>
> Charles
>
> On Thu, Dec 3, 2009 at 8:34 AM, Joe Martini (joemar2) 
> wrote:
>
> To do a fractional MGCP T1/E1/PRI you have to configured it as a full PRI
> on the gateway.  Then on CUCM check the CallManager service parameter,
> Change B-Channel Maintenance Status.
>
>
>
> Joe
>
>
>
>
>
>
>
> *From:* cisco-voip-boun...@puck.nether.net [mailto:
> cisco-voip-boun...@puck.nether.net] *On Behalf Of *Dan Greenway
> *Sent:* Thursday, December 03, 2009 9:25 AM
> *To:* cisco-voip@puck.nether.net
> *Subject:* Re: [cisco-voip] UK PRI MGCP
>
>
>
> I didn't think you could do MGCP with fractional E1's
>
>
>
> thanks
>
> Dan
>
>
>
>
>
> On 3 Dec 2009, at 07:10, afatsum wrote:
>
>
>
> You are missing l3 backhaul binding to ccm-manager under your serial
> interface. Add "isdn bind-l3 ccm-manager" under serial 0/0/0:15
>
> -Mus
>
> Martin Bufton wrote:
>
>
>
>
>
>  I have a Cisco router connected to UK PRI E1 with 10 channels. I
>
>  cannot get it to register.
>
>
>
>
>
>
>
>  I get the following MCGP debug message. Can you see anything wrong in
>
>  my config?
>
>
>
>
>
>
>
>
>
>
>
>  73: *Dec  2 17:55:59.251 GMT:
>
>  //-1//MGCP/mgcp_mp_get_not_entity(830):[lvl=2]Invalid
>
>  parameter (pkt 0x67654638 pkt->mgcp_parm_lines 0x)
>
>
>
>
>
>
>
>  Thanks in advance
>
>
>
>
>
>
>
>  Current configuration : 8969 bytes
>
>
>
>  !
>
>
>
>  version 12.4
>
>
>
>  no service pad
>
>
>
>  service timestamps debug datetime msec localtime show-timezone
>
>
>
>  service timestamps log datetime msec localtime show-timezone
>
>
>
>  service password-encryption
>
>
>
>  service sequence-numbers
>
>
>
>  !
>
>
>
>  hostname Cent-xxx-MGCP-GW1-RT
>
>
>
>  !
>
>
>
>  boot-start-marker
>
>
>
>  boot-end-marker
>
>
>
>  !
>
>
>
>  card type e1 0 0
>
>
>
>  logging buffered 51200 warnings
>
>
>
>  !
>
>
>
>  no aaa new-model
>
>
>
>  clock timezone GMT 0
>
>
>
>  clock summer-time BST recurring last Sun Mar 2:00 last Sun Oct 2:00
>
>
>
>  network-clock-participate wic 0
>
>
>
>  dot11 syslog
>
>
>
>  !
>
>
>
>  !
>
>
>
>  ip cef
>
>
>
>  !
>
>
>
>  !
>
>
>
>  no ip bootp server
>
>
>
>  no ip domain lookup
>
>
>
>  ip domain name
>
>
>
>  multilink bundle-name authenticated
>
>
>
>  !
>
>
>
>  isdn switch-type primary-net5
>
>
>
>  voice-card 0
>
>
>
>  no dspfarm
>
>
>
>  !
>
>
>
>  crypto pki trustpoint TP-self-signed-2117350501
>
>
>
>  enrollment selfsigned
>
>
>
>  subject-name cn=IOS-Self-Signed-Certificate-2117350501
>
>
>
>  revocation-check none
>
>
>
>  rsakeypair TP-self-signed-2117350501
>
>
>
>
>
>
>
> quit
>
>
>
>  !
>
>
>
>  !
>
>
>
>  archive
>
>
>
>  log config
>
>
>
>   hidekeys
>
>
>
>  !
>
>
>
>  !
>
>
>
>  controller E1 0/0/0
>
>
>
>  pri-group timeslots 1-10,16 service mgcp
>
>
>
>  !
>
>
>
>  controller E1 0/0/1
>
>
>
>  !
>
>
>
>  ip tcp synwait-time 10
>
>
>
>  !
>
>
>
>  !
>
>
>
>  !
>
>
>
>  !
>
>
>
>  interface Port-channel1
>
>
>
>  description EtherChannel to Cent-xxx-Core-Sw
>
>
>
>  ip address 172.16.xx.100 255.255.255.0
>
>
>
>  !
>
>
>
>  interface GigabitEthernet0/0
>
>
>
>  description Member of Etherchannel conected to Cent-xxx-Core-Sw gig
>
>  2/0/19
>
>
>
>  no ip address
>
>
>
>  duplex auto
>
>
>
>  speed auto
>
>
>
>  media-type rj45
>
>
>
>  channel-group 1
>
>
>
>  !
>
>
>
>  interface GigabitEthernet0/1
>
>
>
>  description Member of Etherchannel conected to Cent-xx-Core-Sw gig 1/0/19
>
>
>
>  no ip add

Re: [cisco-voip] Migration strategy

[秀王]

Hi Ryan,

let's say my phones are on a temp partition not reachable by other CSS. My
UDP (user device profile) are on a valid partition shared by others ( Ie.
P_Internal) but they are not logged in anywhere.

Will this confused the CUCM? Or i shall place the UDP on temp partition as
well. If so, can BAT assist me in migrating from TEMP partition to actual
partition (P_Internal)?

Cheers,
Ki Wi

On Wed, Mar 18, 2015 at 9:26 AM, Ryan Huff  wrote:

> I'm not, sure I completely understand your questions but I'll attempt to
> answer based on my understanding.
>
> Yes, you can pre-config devices and users in CCM prior to migration. If
> they are Cisco IP phones, you'll need the MAC address and model of the
> phone at a minimum.
>
> If they are non Cisco IP phones, you'll need to pre configure 3rd party
> sip devices (which is a different license requirement than a Cisco IP
> phone).
>
> Place the preconfigured dial plan that isnt migrated yet (on CCM), in a
> temp. partition that the already migrated phones cannot access. As you
> migrate, change that partition using BAT, to the correct partition for the
> portion of phones you migrated.
>
> Thanks,
>
> Ryan
>
>
>  Original Message 
> From: 秀王 
> Sent: Tuesday, March 17, 2015 09:15 PM
> To: cisco-voip@puck.nether.net
> Subject: [cisco-voip] Migration strategy
>
> Currently the client have avaya and cisco linked together using SIP.
>
> Cisco UCM cluster have users in the production environment.
>
> I'm are going to cutover more sites from avaya to cisco. Is it possible to
> preconfigure the users, extension number (let's say 87XXX range), phones
> and the user device profiles in advance?
>
> I'm thinking that if I preconfigure those information, the cucm will think
> that those extension number (87XXX) are local and unregistered.
>
> Is there a way to make CUCM thinks that in order to reach 87XXX range, it
> will still reach out to Avaya using the SIP trunk? Is there any setting in
> the route pattern can do that?
>
> I thinking that CUCM will always find a more "exact" match locally instead
> of through other source like translation pattern or route pattern.
>
>
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Re: [cisco-voip] Migration strategy

[Ryan Huff]

I'm not, sure I completely understand your questions but I'll attempt to answer 
based on my understanding.

Yes, you can pre-config devices and users in CCM prior to migration. If they 
are Cisco IP phones, you'll need the MAC address and model of the phone at a 
minimum.

If they are non Cisco IP phones, you'll need to pre configure 3rd party sip 
devices (which is a different license requirement than a Cisco IP phone).

Place the preconfigured dial plan that isnt migrated yet (on CCM), in a temp. 
partition that the already migrated phones cannot access. As you migrate, 
change that partition using BAT, to the correct partition for the portion of 
phones you migrated.

Thanks,

Ryan

 Original Message 
From: 秀王 
Sent: Tuesday, March 17, 2015 09:15 PM
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] Migration strategy

>Currently the client have avaya and cisco linked together using SIP.
>
>Cisco UCM cluster have users in the production environment.
>
>I'm are going to cutover more sites from avaya to cisco. Is it possible to
>preconfigure the users, extension number (let's say 87XXX range), phones
>and the user device profiles in advance?
>
>I'm thinking that if I preconfigure those information, the cucm will think
>that those extension number (87XXX) are local and unregistered.
>
>Is there a way to make CUCM thinks that in order to reach 87XXX range, it
>will still reach out to Avaya using the SIP trunk? Is there any setting in
>the route pattern can do that?
>
>I thinking that CUCM will always find a more "exact" match locally instead
>of through other source like translation pattern or route pattern.
>
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[cisco-voip] Migration strategy

[秀王]

Currently the client have avaya and cisco linked together using SIP.

Cisco UCM cluster have users in the production environment.

I'm are going to cutover more sites from avaya to cisco. Is it possible to
preconfigure the users, extension number (let's say 87XXX range), phones
and the user device profiles in advance?

I'm thinking that if I preconfigure those information, the cucm will think
that those extension number (87XXX) are local and unregistered.

Is there a way to make CUCM thinks that in order to reach 87XXX range, it
will still reach out to Avaya using the SIP trunk? Is there any setting in
the route pattern can do that?

I thinking that CUCM will always find a more "exact" match locally instead
of through other source like translation pattern or route pattern.
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Re: [cisco-voip] CUBE across VRFs

[Roger Wiklund]

Most providers are using SBCs facing the customer. Verizon for example
has two /25, first for SIP and second for RTP. Just point a static
route to them on the outside interface, then the default route towards
the inside interface.

On Wed, Feb 4, 2015 at 11:16 PM, Matthew Loraditch
 wrote:
> I’ve seen it come two ways, riding your same MPLS circuit, in which case if
> you have a dedicated VG you just default route that to your MPLS router and
> there you go.
>
>
>
> The other way is like you say and I’ve done that with at&t and I didn’t have
> to route with them, they NAT’d everything on their side to me. So I just
> routed their couple SBC IPs/Subnets across that handoff and my default still
> goes into my LAN.
>
>
>
> I’m sure there are other ways as well.
>
>
>
> Matthew G. Loraditch – CCNP-Voice, CCNA-R&S, CCDA
> Network Engineer
> Direct Voice: 443.541.1518
>
> Facebook | Twitter | LinkedIn | G+
>
>
>
> From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of
> Norton, Mike
> Sent: Wednesday, February 04, 2015 5:05 PM
> To: Erick
> Cc: cisco-voip@puck.nether.net
>
>
> Subject: Re: [cisco-voip] CUBE across VRFs
>
>
>
> What I’m failing to understand is... if I set the CUBE’s default route to be
> my router on my network, then how will CUBE be able to reach the SIP
> provider’s call servers on the SIP provider’s network? It seems like I will
> need a routing protocol on whichever side of the CUBE doesn’t get a default
> route. Is that a normal requirement?
>
>
>
> Just to back up a bit, I have been assuming CUBE would have two interfaces –
> one on my network, one on the SIP provider’s network. I’ve always assumed
> that this was the normal way of deploying CUBE but maybe I’m off base there
> and getting myself confused.
>
>
>
> -mn
>
>
>
>
>
> From: Erick [mailto:erick...@gmail.com]
> Sent: February-03-15 6:50 PM
> To: Norton, Mike
> Cc: Jason Aarons (AM); cisco-voip@puck.nether.net
> Subject: Re: [cisco-voip] CUBE across VRFs
>
>
>
> Only one voice vrf can be defined in IOS. Global under voice service voip.
>
>
>
> Cube-SP lets you do multiple vrf's but is EoL and way different
> configuration.
>
>
>
> If you plop a cube off your router and router interface is in a vrf and your
> separate cube is on that network then it should be fine as the cube is just
> a host then  with default route to router.
>
>
> Sent from my iPhone
>
>
> On Feb 3, 2015, at 6:08 PM, "Norton, Mike"  wrote:
>
> Doesn’t have to be two VRFs, could be one VRF and the global route table, if
> that makes a difference. This idea is no connectivity between them, other
> than the application-layer connectivity provided by CUBE. This is
> hypothetical – I’m just trying to understand how/if this would work. I’m
> looking to plop a CUBE between my network and a SIP provider’s network
> without having to participate in routing protocol on either side.
>
>
>
> -mn
>
>
>
> From: Jason Aarons (AM) [mailto:jason.aar...@dimensiondata.com]
> Sent: February-03-15 5:02 PM
> To: Norton, Mike; cisco-voip@puck.nether.net
> Subject: RE: CUBE across VRFs
>
>
>
> You have two VRFs, do they have connectivity between them?
>
>
>
> From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of
> Norton, Mike
> Sent: Tuesday, February 3, 2015 4:36 PM
> To: cisco-voip@puck.nether.net
> Subject: [cisco-voip] CUBE across VRFs
>
>
>
>
>
> Can CUBE sit across two separate VRFs? I’ve never used it, but I’m
> envisioning an ISR having a VRF-Lite with default route pointed at my
> network, and a VRF-Lite with default route pointed at the SIP provider’s
> network. I’m thinking this would be the preferred way to do it, but maybe
> I’m missing something?
>
>
>
> My Googling is dredging up a lot of really old info that I’m not sure is
> still relevant.
>
>
>
> --
>
> Mike Norton
>
>
>
> itevomcid
>
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>
>
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Re: [cisco-voip] QRT and Cisco Jabber 10.5.2

[Thomas LeMay]

Hi, Anthony,

 

Thanks for the input.  The closest thing  I have found is clicking on the upper 
right gear wheel on Jabber and then selecting the help –report a problem 
option.  From there,  I can indicate a problem report with call or video then 
select poor audio quality. I can download a report file and attach it to a TAC 
case for analysis.

 

Tom  

 

From: Anthony Holloway [mailto:avholloway+cisco-v...@gmail.com] 
Sent: Tuesday, March 17, 2015 4:47 PM
To: Thomas LeMay; cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] QRT and Cisco Jabber 10.5.2

 

I don't think Jabber qualifies for QRT support.

http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/10_0_1/ccmfeat/CUCM_BK_F3AC1C0F_00_cucm-features-services-guide-100/CUCM_BK_F3AC1C0F_00_cucm-features-services-guide-100_chapter_011.html#CUCM_RF_S6DB08DE_00

 

Speaking of which, Jabber doesn't qualify for MeetMe either; no softkey to push.

 

 

On Tue, Mar 17, 2015 at 2:17 PM Thomas LeMay  wrote:

How can one use the QRT with Cisco Jabber 10.5.2 to trouble shoot garbled audio?

 

Thank you.

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Re: [cisco-voip] QRT and Cisco Jabber 10.5.2

[Anthony Holloway]

I don't think Jabber qualifies for QRT support.

http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/10_0_1/ccmfeat/CUCM_BK_F3AC1C0F_00_cucm-features-services-guide-100/CUCM_BK_F3AC1C0F_00_cucm-features-services-guide-100_chapter_011.html#CUCM_RF_S6DB08DE_00

Speaking of which, Jabber doesn't qualify for MeetMe either; no softkey to
push.


On Tue, Mar 17, 2015 at 2:17 PM Thomas LeMay 
wrote:

> How can one use the QRT with Cisco Jabber 10.5.2 to trouble shoot garbled
> audio?
>
>
>
> Thank you.
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>
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[cisco-voip] QRT and Cisco Jabber 10.5.2

[Thomas LeMay]

How can one use the QRT with Cisco Jabber 10.5.2 to trouble shoot garbled
audio?

 

Thank you.

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Re: [cisco-voip] Setting admin password on SX20 registered to CUCM - Can't set from CUCM device page

[Ryan Ratliff (rratliff)]

Rob is right on. You must use encrypted config files for TC endpoints to take 
passwords from UCM.

Sent from my iPhone

On Mar 17, 2015, at 11:54 AM, Rob Dawson 
mailto:rdaw...@force3.com>> wrote:

It doesn’t push that down, it has to match what is provisioned on the endpoint 
for some functionality though – here is an excerpt from the “Cisco Telepresence 
Endpoints and CUCM” quick reference guide:

> Admin username and password section.
Configure the product specific configuration layout settings
as appropriate.
Admin Username: Set the username. Must be admin if
using a Secure Profile in CUCM; or must match the value
set on the endpoint if you are using Cisco TelePresence
Manager (CTS-MAN).
Admin Password: Set the password. Set to the desired
value if using a Secure Profile in CUCM; or match
the value set on the endpoint if you are using Cisco
TelePresence Manager (CTS-MAN).
NOTE: When the TelePresence endpoint is set up with
an encrypted security profile, the endpoint will read the
admin password from the CUCM. The password can not
be blank and the user name must be admin.
NOTE: The admin username and password set on CUCM
must match the system password set on the endpoint in
order for the Cisco TelePresence Manager (CTS-MAN)
to discover the endpoint and provide One Button to Push
scheduling to them.



From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Brian 
V
Sent: Monday, March 16, 2015 4:59 PM
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] Setting admin password on SX20 registered to CUCM - Can't 
set from CUCM device page


I have CUCM 10.5.2

SX20 with 7.3.1 code.

SX20 is registered successfully to CUCM. I can change the display name, the 
extension etc.. "apply changes" from CUCM and the endpoint learns of the 
changes and looks correct. So I'm guessing its properly downloading its config 
file and applying settings.

On the SX20 device page in CUCM admin there is a section about 3/4 of the way 
down called "Admin username and password"

It would seem to imply that the endpoint admin username and password can be set 
in CUCM and pushed down to the endpoint.

This is not working for me. It doesn't update.  Does anyone know if this is 
really supported ?



I can set it fine directly from the web interface of the endpoint like usual.

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Re: [cisco-voip] Setting admin password on SX20 registered to CUCM - Can't set from CUCM device page

[Rob Dawson]

It doesn’t push that down, it has to match what is provisioned on the endpoint 
for some functionality though – here is an excerpt from the “Cisco Telepresence 
Endpoints and CUCM” quick reference guide:

> Admin username and password section.
Configure the product specific configuration layout settings
as appropriate.
Admin Username: Set the username. Must be admin if
using a Secure Profile in CUCM; or must match the value
set on the endpoint if you are using Cisco TelePresence
Manager (CTS-MAN).
Admin Password: Set the password. Set to the desired
value if using a Secure Profile in CUCM; or match
the value set on the endpoint if you are using Cisco
TelePresence Manager (CTS-MAN).
NOTE: When the TelePresence endpoint is set up with
an encrypted security profile, the endpoint will read the
admin password from the CUCM. The password can not
be blank and the user name must be admin.
NOTE: The admin username and password set on CUCM
must match the system password set on the endpoint in
order for the Cisco TelePresence Manager (CTS-MAN)
to discover the endpoint and provide One Button to Push
scheduling to them.



From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Brian 
V
Sent: Monday, March 16, 2015 4:59 PM
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] Setting admin password on SX20 registered to CUCM - Can't 
set from CUCM device page


I have CUCM 10.5.2

SX20 with 7.3.1 code.

SX20 is registered successfully to CUCM. I can change the display name, the 
extension etc.. "apply changes" from CUCM and the endpoint learns of the 
changes and looks correct. So I'm guessing its properly downloading its config 
file and applying settings.

On the SX20 device page in CUCM admin there is a section about 3/4 of the way 
down called "Admin username and password"

It would seem to imply that the endpoint admin username and password can be set 
in CUCM and pushed down to the endpoint.

This is not working for me. It doesn't update.  Does anyone know if this is 
really supported ?

[CUCM Screen Capture for SX20 Device]

I can set it fine directly from the web interface of the endpoint like usual.

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Re: [cisco-voip] Unified Attendant Console - Advanced Server

[Anthony Holloway]

Can confirm.

http://www.cisco.com/c/en/us/products/collateral/unified-communications/unified-attendant-console/data_sheet_C78-731867.html

On Tue, Mar 17, 2015 at 8:28 AM Ryan Huff  wrote:

> Is there a way to tell if I have the resilience option? Looks like my SKU
> is L-CUAC10X-ADV-HA, so I assume the *HA* is the designator for
> resilience / high availability? Can anyone confirm?
>
> Thanks,
>
> Ryan
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[cisco-voip] Unified Attendant Console - Advanced Server

[Ryan Huff]

Is there a way to tell if I have the resilience option? Looks like my SKU is 
L-CUAC10X-ADV-HA, so I assume the HA is the designator for resilience / high 
availability? Can anyone confirm?

Thanks,

Ryan
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Re: [cisco-voip] ATA190 Plar SIP Dial Rule

[Barry Howser]

[SOLVED]

Here is what ended up working for me;

ATA190 firmware 1.1.2 (posted to CCO on 02-15-2015) - Specifically states
PLAR isn't supported until Version 1.1.2
SIP Dial Rule with 1 blank PLAR pattern (no button value)
Translation pattern using the *!* wildcard, not "" (blank)

Traces showed that when that ATA went off-hook, it wasn't dialing anything.
As soon as I used the *!* wildcard instead of "" (blank) in the
translation, the ATA would dial it when going off hook.


On Fri, Mar 13, 2015 at 5:25 PM, Brian Meade  wrote:

> Looks good to me.  Might want to pull the CallManager traces to see if the
> call comes in after going off-hook okay.  I can look at them if you want to
> throw them up on dropbox or something.  Sounds like it's doing something
> now at least.
>
> On Fri, Mar 13, 2015 at 5:15 PM, Barry Howser 
> wrote:
>
>> Brian,
>>
>> I have attached the screen shot of the sip dial rule.
>>
>> I have the ATA187 using the same CSS on the device and line. That CSS
>> only accesses one partition. That partition has one translation pattern,
>> with a "blank" pattern field and the digits 9911 in the "Called Party
>> Transformation" field. The translation pattern uses a CSS that has access
>> to a 9.911 route pattern (pattern discards predot).
>>
>> thanks
>>
>> On Fri, Mar 13, 2015 at 4:58 PM, Brian Meade  wrote:
>>
>>> Sorry, it was the ATA187s I tried this on.  Can you attach a screenshot
>>> of your dial rule config?
>>>
>>> On Fri, Mar 13, 2015 at 4:15 PM, Brian Meade  wrote:
>>>
 Right, that's correct.  Add 2 PLARs to the SIP Dial Rule with
 descriptions both with just a button parameter.

 I've used this for ATA 188s but haven't tested specifically on the 190.

 On Fri, Mar 13, 2015 at 3:46 PM, Barry Howser 
 wrote:

> hi Brian,
>
> So what you're saying is that in the SIP dial rule; I'll click the
> "Add Plar" button and then give my parameter a description, select 
> "Button"
> as my dial parameter then in the value box I'd enter a "1" or a "2"
> depending on if I wanted the *PLAR* working on line 1 or 2 of the ATA.
>
> I would then assume that if I wanted both ATA lines to plar, I would
> have two parameters in the SIP dial rule?
>
> O . I wish you would write Cisco docs  I can understand
> you, lol.
>
>
> On Fri, Mar 13, 2015 at 3:38 PM, Brian Meade  wrote:
>
>> For the SIP Dial Rule, all you want it to have is a PLAR with Button
>> 1 set.  Don't enter the number you want to PLAR to.  Then just set up 
>> PLAR
>> like you would for a SCCP phone with a new CSS/partition/blank 
>> translation
>> pattern.
>>
>> On Fri, Mar 13, 2015 at 3:30 PM, Barry Howser 
>> wrote:
>>
>>> Hello everyone.
>>>
>>> I have an ATA190 that needs to do a plar to 911. My dial plan uses
>>> "9" to access an outside line (including the 911 pattern).
>>>
>>> I created a SIP dial rule and added a plar pattern. I added a
>>> parameter called "911" in the description and then added 9911 in the 
>>> value
>>> field. I saved, applied config and restarted.
>>>
>>> I have applied that SIP Dial Rule to the ATA190 device's sip dial
>>> rule section and reset the ATA. When I take either of the lines off hook
>>> with an analog phone, I just get dial tone  no PLARing.
>>>
>>> What am I doing wrong?
>>>
>>> thanks
>>>
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>>>
>>
>

>>>
>>
>
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