Re: [OSL | CCIE_Voice] WB 1, lab 11A, task 1

2010-11-11 Thread Ilya Rubinchik
I found my error.
Disregard that.


--
Best regards,
Ilya Rubinchik
Chief UC Engineer
Mars Solutions Ltd.

Skype: im_citius
Mob: +998 (97) 712-8456
Office: +998 (71) 290-7364

From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Ilya Rubinchik
Sent: Friday, November 12, 2010 12:31 PM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] WB 1, lab 11A, task 1

Hi!

When I integrated CUC to CUCM, but got strange behavior: looks like CUC does 
not receive Calling Party Number.
When I’m trying to reach mailbox from hq or br1 phones, I’m getting “Enter your 
ID followed by the # key”.

What could be  the problem?


Thanks!

--
Best regards,
Ilya Rubinchik
Chief UC Engineer
Mars Solutions Ltd.

Skype: im_citius
Mob: +998 (97) 712-8456
Office: +998 (71) 290-7364

___
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[OSL | CCIE_Voice] WB 1, lab 11A, task 1

2010-11-11 Thread Ilya Rubinchik
Hi!

When I integrated CUC to CUCM, but got strange behavior: looks like CUC does 
not receive Calling Party Number.
When I’m trying to reach mailbox from hq or br1 phones, I’m getting “Enter your 
ID followed by the # key”.

What could be  the problem?


Thanks!

--
Best regards,
Ilya Rubinchik
Chief UC Engineer
Mars Solutions Ltd.

Skype: im_citius
Mob: +998 (97) 712-8456
Office: +998 (71) 290-7364

___
For more information regarding industry leading CCIE Lab training, please visit 
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Re: [OSL | CCIE_Voice] + Dialing issue

2010-11-11 Thread Mujaddid Ahmed
mgcp connection resets when we reset it from cucm, 
also the intelligence part to display on phone is done on cucm, gateway just 
provides information about the calling number type in isdn messages.
please correct me if i am wrong. prefix was added on cucm as per the type of 
the 
call.


 




From: ShinGei Yong 
To: Shrini ; ccie_voice@onlinestudylist.com
Cc: Mujaddid Ahmed 
Sent: Fri, November 12, 2010 9:52:00 AM
Subject: Re: [OSL | CCIE_Voice] + Dialing issue

Hi Shrini,

Can you ensure that the UCM mgcp config pump into the gateway correctly? Do a " 
sh ccm-manager" to verify.

Shingei


On Fri, Nov 12, 2010 at 2:01 AM, Shrini  wrote:

Hi Iftikhar,
>
>In the translation pattern 9.1[2-9]XX[2-9]XX --> 
>
>under calling party transformation select the check box use calling part's 
>external phone mask.
>Calling party transform mask - XXX
>Calling /called party number type : National
>Calling/Called party numbering plan : ISDN
>
>In Gateway:
>
>Under Incoming calling part settings:
>
>Change the prefix to + from Default for National numbers.
>
>You will get +1XX as calling party displayed on phone and missed calls
>
>Thx
>Shrini
>
>
>
>
>On 11/10/2010 11:55 PM, Mujaddid Ahmed wrote: 
>Hi, 
>>
>>Please confirm what can be the reason of globalization not working with cisco 
>>7965 phones.
>>I globalized it on CUCM gateway configuration by adding appropriate prefix on 
>>type of calls.
>>
>>did, no mgcp / mgcp on gateway and reset the gateway from cucm as well. 
>>
>>call lands on phone and still shows only 7 digit calling number in both 
>>ringing 
>>and missed call. 
>>
>>
>>nothing has been set so far in Device pool or on device to effect the 
>>globalization done on Gateway.
>>
>>Is there any serivce parameter or any command on router that can prevent the 
>>globalize display on phone?
>>
>>Please help, as i wasted lot of time in troubleshooting this particular 
>>issue in 
>>a very time critical environment.
>>
>>Regards,
>>Iftikhar Ahmed
>>
>> ___ For more information 
>> regarding 
>>industry leading CCIE Lab training, please visit www.ipexpert.com 
>>
>___
>For more information regarding industry leading CCIE Lab training, please 
>visit 
>www.ipexpert.com
>
>



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Re: [OSL | CCIE_Voice] + Dialing issue

2010-11-11 Thread Mujaddid Ahmed
hi,

thanks for replies, no transformations on either dp or device itself.

isdn messages received from pstn were from proper information. i made the 
prefix 
on gw pages according to same info.
but the change was not reflecting on phone after gw reset and no mgcp/ mgcp.
can u guys confirm if any command entered / not entered in gw or any service 
parameter change can effect the display on phone?

i have done it several time in my home lab and it works okay the first time.

Regards,
mujaddid





From: Shrini 
To: Mujaddid Ahmed 
Cc: ccie_voice@onlinestudylist.com
Sent: Thu, November 11, 2010 11:01:13 PM
Subject: Re: [OSL | CCIE_Voice] + Dialing issue

Hi Iftikhar,

In the translation pattern 9.1[2-9]XX[2-9]XX --> 

under calling party transformation select the check box use calling part's 
external phone mask.
Calling party transform mask - XXX
Calling /called party number type : National
Calling/Called party numbering plan : ISDN

In Gateway:

Under Incoming calling part settings:

Change the prefix to + from Default for National numbers.

You will get +1XX as calling party displayed on phone and missed calls

Thx
Shrini



On 11/10/2010 11:55 PM, Mujaddid Ahmed wrote: 
Hi, 
>
>Please confirm what can be the reason of globalization not working with cisco 
>7965 phones.
>I globalized it on CUCM gateway configuration by adding appropriate prefix on 
>type of calls.
>
>did, no mgcp / mgcp on gateway and reset the gateway from cucm as well. 
>
>call lands on phone and still shows only 7 digit calling number in both 
>ringing 
>and missed call. 
>
>
>nothing has been set so far in Device pool or on device to effect the 
>globalization done on Gateway.
>
>Is there any serivce parameter or any command on router that can prevent the 
>globalize display on phone?
>
>Please help, as i wasted lot of time in troubleshooting this particular 
>issue in 
>a very time critical environment.
>
>Regards,
>Iftikhar Ahmed
>
> ___ For more information 
> regarding 
>industry leading CCIE Lab training, please visit www.ipexpert.com 
>


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Re: [OSL | CCIE_Voice] + Dialing issue

2010-11-11 Thread ShinGei Yong
Hi Shrini,

Can you ensure that the UCM mgcp config pump into the gateway correctly? Do
a " sh ccm-manager" to verify.

Shingei

On Fri, Nov 12, 2010 at 2:01 AM, Shrini  wrote:

>  Hi Iftikhar,
>
> In the translation pattern 9.1[2-9]XX[2-9]XX -->
>
> under calling party transformation select the check box use calling part's
> external phone mask.
> Calling party transform mask - XXX
> Calling /called party number type : National
> Calling/Called party numbering plan : ISDN
>
> In Gateway:
>
> Under Incoming calling part settings:
>
> Change the prefix to + from Default for National numbers.
>
> You will get +1XX as calling party displayed on phone and missed
> calls
>
> Thx
> Shrini
>
>
>
> On 11/10/2010 11:55 PM, Mujaddid Ahmed wrote:
>
>  Hi,
>
> Please confirm what can be the reason of globalization not working with
> cisco 7965 phones.
> I globalized it on CUCM gateway configuration by adding appropriate prefix
> on type of calls.
>
> did, no mgcp / mgcp on gateway and reset the gateway from cucm as well.
>
> call lands on phone and still shows only 7 digit calling number in both
> ringing and missed call.
>
> nothing has been set so far in Device pool or on device to effect the
> globalization done on Gateway.
>
> Is there any serivce parameter or any command on router that can prevent
> the globalize display on phone?
>
> Please help, as i wasted lot of time in troubleshooting this particular
> issue in a very time critical environment.
>
> Regards,
> Iftikhar Ahmed
>
>
> ___
> For more information regarding industry leading CCIE Lab training, please 
> visit www.ipexpert.com
>
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
>
___
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Re: [OSL | CCIE_Voice] MVA Call Drops

2010-11-11 Thread anil batra
Also pls include this
 
voice service voip
allow connection h323 to h323

--- On Fri, 11/12/10, Cristobal Priego  wrote:


From: Cristobal Priego 
Subject: Re: [OSL | CCIE_Voice] MVA Call Drops
To: "Amr Sherif" 
Cc: ccie_voice@onlinestudylist.com
Date: Friday, November 12, 2010, 4:46 AM


Amr

I ran into this issue before

please try this

dial-peer voice 5999 pots
 service ccm
 incoming called-number 5999


dial-peer voice 5999 voip
 service ccm
 destination-pattern 5999
 session target ipv4:192.168.15.100

 dtmf-relay h245-alphanumeric
 codec g711ulaw
 no vad

that should do the trick


2010/11/11 Amr Sherif 


Hi Experts,
 
I have this Scenario and wish you help me in it.
 
Trying to call MVA 2123945999 from pstn Line (2123942123) using hairpin method 
here is the call flow:
 
PSTN Line (2123942123) -->  calling 2123945999 --> HQ-GW(MGCP) -->Significant 
number =4 & css-internal --->Route pattern (5999/pt-internal) ---> RL: H.323( 
10.10.110.1)
 
-->HQ IOS GW:
 
allow connection h t h
 
dial-peer voice 1 pots
 incoming called-number .
 direct-inward-dial
 
dial-peer voice 5999 voip
 service ccm
 destination-pattern 5999
 session target ipv4:192.168.15.100
 incoming called-number 5999
 dtmf-relay h245-alphanumeric
 codec g711ulaw
 no vad
 
The call hits the dial-p and hairpin back to the CCM using H323 GW 
(10.10.110.1) ---> css-mva/pt-mva ---> MVA (5999/pt-mva) . The nice lady 
answers me with no problems and ask me for my ID which is 2123942123 ,enter the 
pin 1234 and press 1 to dial the number , NOW before i forget ( i'm 
using g711ulaw in the HQ device pool within the site and g729r8 between sites 
e.g br1 & br2 and all the sites are connected to the CallManager with MGCP GW). 
 
When i dial any phone internally in HQ site ,the call pass through with no 
problem ,THE CALL ONLY DROP IF I CALL ANY OTHER SITE LIKE BR1 1002 or BR2 3002 
,so i doubted it's a codec issue ,after chaning the region for br1 and br2 to 
be g711ulaw ,suddenly i can dial all calls with no problem. 
 
So my question is there a way so to still keep dialing in internally between 
sites using g729r8 !! or i'm missing something here
 




Best regards,




Amr Sherif
Senior Network Voice Engineer
CCNA,CCNP,CCVP and CCIE Voice Written (Certified)
CCIE Voice Lab (In Progress)

___
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www.ipexpert.com



-Inline Attachment Follows-


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Re: [OSL | CCIE_Voice] H.323 Display ID

2010-11-11 Thread Prashant Patel
Hi Afzal,

Use "clid strip name" for restricting Calling Name and "clid strip" for the
number.

HTH
Prashant

On Thu, Nov 11, 2010 at 10:51 PM, Afzal Bhutta wrote:

> Hello Folks,
> I have question regarding H.323 Display ID.This is Site C H.323 gateway.I
> want to restrict calling name for 999  or Int calls.
> I tried to use clid restrict command under 900 dial-peer.But calling name
> still appear in the debug isdn Q931 output.On Phone display it shows  From
> Private.
> My question How can I restrict calling name in H.323 gateway?
> !
> dial-peer voice 900 pots
>  translation-profile outgoing int
>  destination-pattern 900T
>  clid restrict
>  port 0/0/0:15
> !
>
> Progress Ind i = 0x8183 - Origination address is non-ISDN
> Display i = 'Site C phone 2'
> Calling Party Number i = 0x11A0, '+85224044002'
> Plan:ISDN, Type:International
> Called Party Number i = 0x91, '44208555'
> Plan:ISDN, Type:International
>
>
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
>
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[OSL | CCIE_Voice] H.323 Display ID

2010-11-11 Thread Afzal Bhutta
Hello Folks,
I have question regarding H.323 Display ID.This is Site C H.323 gateway.I
want to restrict calling name for 999  or Int calls.
I tried to use clid restrict command under 900 dial-peer.But calling name
still appear in the debug isdn Q931 output.On Phone display it shows  From
Private.
My question How can I restrict calling name in H.323 gateway?
!
dial-peer voice 900 pots
 translation-profile outgoing int
 destination-pattern 900T
 clid restrict
 port 0/0/0:15
!

Progress Ind i = 0x8183 - Origination address is non-ISDN
Display i = 'Site C phone 2'
Calling Party Number i = 0x11A0, '+85224044002'
Plan:ISDN, Type:International
Called Party Number i = 0x91, '44208555'
Plan:ISDN, Type:International
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Re: [OSL | CCIE_Voice] delayed Multicast MOH streaming - Any ideas ?

2010-11-11 Thread ccieid1ot
QOS settings?  Frame-relay traffic shaping?  Just a wild stab.

On Wed, Nov 10, 2010 at 4:53 PM, Pavan  wrote:

> G711
>
>
> On Nov 10, 2010, at 3:07 PM, ccieid1ot  wrote:
>
> What's the region set to for the SB to MOH?
>
> On Tue, Nov 9, 2010 at 6:01 PM, Pavan K < 
> pav.c...@gmail.com> wrote:
>
>>
>> Multicast MOH from CCM. No Spoofing.
>>
>> For HQ site, MMOH stream perfectly
>> For Branch sites, on the Branch router i can see MMOH packets coming in
>> with the "debug ip mpacket" command but the phone doesnt play MOH until
>> about 5 mins later.
>>
>> In other words, after pressing Hold, Phone starts streaming music after
>> being on hold for 5 mins.
>>
>> Phones are SK registered to CCM. CCM counters look good (show mcast stream
>> active)
>>
>> pim is configured for sparse-dense-mode on all serial subifs, vlans &
>> loopbacks.
>>
>> IOS is 12.4 (20T2)
>>
>>
>> --
>> - Pavan
>>
>> ___
>> For more information regarding industry leading CCIE Lab training, please
>> visit www.ipexpert.com
>>
>>
>
___
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Re: [OSL | CCIE_Voice] MVA Call Drops

2010-11-11 Thread Cristobal Priego
Amr

I ran into this issue before

please try this

dial-peer voice 5999 pots
 service ccm
 incoming called-number 5999


dial-peer voice 5999 voip
 service ccm
 destination-pattern 5999
 session target ipv4:192.168.15.100

 dtmf-relay h245-alphanumeric
 codec g711ulaw
 no vad

that should do the trick

2010/11/11 Amr Sherif 

>  Hi Experts,
>
> I have this Scenario and wish you help me in it.
>
> Trying to call MVA 2123945999 from pstn Line (2123942123) using hairpin
> method here is the call flow:
>
> PSTN Line (2123942123) -->  calling 2123945999 --> HQ-GW(MGCP)
> -->Significant number =4 & css-internal --->Route pattern (5999/pt-internal)
> ---> RL: H.323( 10.10.110.1)
>
> -->HQ IOS GW:
>
> allow connection h t h
>
> dial-peer voice 1 pots
>  incoming called-number .
>  direct-inward-dial
>
> dial-peer voice 5999 voip
>  service ccm
>  destination-pattern 5999
>  session target ipv4:192.168.15.100
>  incoming called-number 5999
>  dtmf-relay h245-alphanumeric
>  codec g711ulaw
>  no vad
>
> The call hits the dial-p and hairpin back to the CCM using H323 GW
> (10.10.110.1) ---> css-mva/pt-mva ---> MVA (5999/pt-mva) . The nice lady
> answers me with no problems and ask me for my ID which is 2123942123 ,enter
> the pin 1234 and press 1 to dial the number , NOW before i forget ( i'm
> using g711ulaw in the HQ device pool within the site and g729r8 between
> sites e.g br1 & br2 and all the sites are connected to the CallManager with
> MGCP GW).
>
> When i dial any phone internally in HQ site ,the call pass through with no
> problem ,THE CALL ONLY DROP IF I CALL ANY OTHER SITE LIKE BR1 1002 or BR2
> 3002 ,so i doubted it's a codec issue ,after chaning the region for br1 and
> br2 to be g711ulaw ,suddenly i can dial all calls with no problem.
>
> So my question is there a way so to still keep dialing in internally
> between sites using g729r8 !! or i'm missing something here
>
>
>
>
>
> Best regards,
>
>
>
> Amr Sherif
> Senior Network Voice Engineer
> CCNA,CCNP,CCVP and CCIE Voice Written *(Certified)*
> CCIE Voice Lab *(In Progress)*
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
>
___
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www.ipexpert.com


[OSL | CCIE_Voice] MVA Call Drops

2010-11-11 Thread Amr Sherif

Hi Experts,
 
I have this Scenario and wish you help me in it.
 
Trying to call MVA 2123945999 from pstn Line (2123942123) using hairpin method 
here is the call flow:
 
PSTN Line (2123942123) -->  calling 2123945999 --> HQ-GW(MGCP) -->Significant 
number =4 & css-internal --->Route pattern (5999/pt-internal) ---> RL: H.323( 
10.10.110.1)
 
-->HQ IOS GW:
 
allow connection h t h
 
dial-peer voice 1 pots
 incoming called-number .
 direct-inward-dial
 
dial-peer voice 5999 voip
 service ccm
 destination-pattern 5999
 session target ipv4:192.168.15.100
 incoming called-number 5999
 dtmf-relay h245-alphanumeric
 codec g711ulaw
 no vad
 
The call hits the dial-p and hairpin back to the CCM using H323 GW 
(10.10.110.1) ---> css-mva/pt-mva ---> MVA (5999/pt-mva) . The nice lady 
answers me with no problems and ask me for my ID which is 2123942123 ,enter the 
pin 1234 and press 1 to dial the number , NOW before i forget ( i'm using 
g711ulaw in the HQ device pool within the site and g729r8 between sites e.g br1 
& br2 and all the sites are connected to the CallManager with MGCP GW). 
 
When i dial any phone internally in HQ site ,the call pass through with no 
problem ,THE CALL ONLY DROP IF I CALL ANY OTHER SITE LIKE BR1 1002 or BR2 3002 
,so i doubted it's a codec issue ,after chaning the region for br1 and br2 to 
be g711ulaw ,suddenly i can dial all calls with no problem. 
 
So my question is there a way so to still keep dialing in internally between 
sites using g729r8 !! or i'm missing something here
 




Best regards,



Amr Sherif
Senior Network Voice Engineer
CCNA,CCNP,CCVP and CCIE Voice Written (Certified)
CCIE Voice Lab (In Progress)
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Re: [OSL | CCIE_Voice] mgcp gw and T1 controller configuration

2010-11-11 Thread Ravindra Lakpriya
Thanks for the reply Joshits sorted now

On Thu, Nov 11, 2010 at 11:42 PM, Josh Kittle  wrote:
> I'm new to the list - forgive me if I don't know how things work yet.
>
> Can you show us your MGCP related configuration in Call Manager?  Also the
> output of 'show ccm'
>
>
> On Thu, Nov 11, 2010 at 2:35 PM, Ravindra Lakpriya 
> wrote:
>>
>> Hi guys,
>>
>> I'm configuring T1 controller in a MGCP gateway.
>>
>> this is my T1 controller configuration
>> --
>> controller T1 0/0/0
>> framing esf
>> linecode b8zs
>> pri-group timeslots 1-3,24 service mgcp
>>
>> MGCP Configuration
>> -
>>
>> interface Serial0/0/0:23
>>  no ip address
>>  encapsulation hdlc
>>  isdn switch-type primary-ni
>>  isdn incoming-voice voice
>>  isdn bind-l3 ccm-manager
>>  no cdp enable
>>
>> ccm-manager switchback immediate
>> ccm-manager redundant-host 210.10.10.10
>> ccm-manager mgcp
>> ccm-manager fax protocol cisco
>> !
>> mgcp call-agent 210.10.10.11 service-type mgcp version 0.1
>> mgcp dtmf-relay voip codec all mode out-of-band
>> mgcp fax t38 ecm
>> mgcp bind control source-interface Loopback0
>> mgcp bind media source-interface Loopback0
>> !
>> mgcp profile default
>>
>>
>> When execute show isdn status it give me Layer 2 error.
>>
>> Global ISDN Switchtype = primary-ni
>>
>> %Q.931 is backhauled to CCM MANAGER 0x0003 on DSL 0. Layer 3 output
>> may not apply
>>
>> ISDN Serial0/0/0:23 interface
>>        dsl 0, interface ISDN Switchtype = primary-ni
>>        L2 Protocol = Q.921 0x  L3 Protocol(s) = CCM MANAGER 0x0003
>>    Layer 1 Status:
>>        ACTIVE
>>    Layer 2 Status:
>>        TEI = 0, Ces = 1, SAPI = 0, State = TEI_ASSIGNED
>>    Layer 3 Status:
>>        0 Active Layer 3 Call(s)
>>    Active dsl 0 CCBs = 0
>>    The Free Channel Mask:  0x8007
>>    Number of L2 Discards = 0, L2 Session ID = 3
>>
>>
>> You guys have any idea due to what this happening. I execute debug
>> isdn q921 command as well. this is the out put of that.
>>
>> *Nov 11 12:36:31.651: ISDN Se0/0/0:23 Q921: User RX <- SABMEp sapi=0 tei=0
>> *Nov 11 12:36:31.651: ISDN Se0/0/0:23 Q921: S4_SABME: BACKHAULED &
>> vsc_wants_L2_up = FALSE
>> *Nov 11 12:36:32.267: ISDN Se0/0/0:23 Q921: User RX <- SABMEp sapi=0 tei=0
>> *Nov 11 12:36:32.267: ISDN Se0/0/0:23 Q921: S4_SABME: BACKHAULED &
>> vsc_wants_L2_up = FALSE
>>
>> --
>> Ravindra Lakpriya
>> ___
>> For more information regarding industry leading CCIE Lab training, please
>> visit www.ipexpert.com
>
>



-- 
Ravindra Lakpriya
+94 773 532 094
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Re: [OSL | CCIE_Voice] mgcp gw and T1 controller configuration

2010-11-11 Thread Ravindra Lakpriya
False alarm guys i was missing the mgcp bind commands :-)

in a sleepy mood.

On Thu, Nov 11, 2010 at 11:35 PM, Ravindra Lakpriya  wrote:
> Hi guys,
>
> I'm configuring T1 controller in a MGCP gateway.
>
> this is my T1 controller configuration
> --
> controller T1 0/0/0
> framing esf
> linecode b8zs
> pri-group timeslots 1-3,24 service mgcp
>
> MGCP Configuration
> -
>
> interface Serial0/0/0:23
>  no ip address
>  encapsulation hdlc
>  isdn switch-type primary-ni
>  isdn incoming-voice voice
>  isdn bind-l3 ccm-manager
>  no cdp enable
>
> ccm-manager switchback immediate
> ccm-manager redundant-host 210.10.10.10
> ccm-manager mgcp
> ccm-manager fax protocol cisco
> !
> mgcp call-agent 210.10.10.11 service-type mgcp version 0.1
> mgcp dtmf-relay voip codec all mode out-of-band
> mgcp fax t38 ecm
> mgcp bind control source-interface Loopback0
> mgcp bind media source-interface Loopback0
> !
> mgcp profile default
>
>
> When execute show isdn status it give me Layer 2 error.
>
> Global ISDN Switchtype = primary-ni
>
> %Q.931 is backhauled to CCM MANAGER 0x0003 on DSL 0. Layer 3 output
> may not apply
>
> ISDN Serial0/0/0:23 interface
>        dsl 0, interface ISDN Switchtype = primary-ni
>        L2 Protocol = Q.921 0x  L3 Protocol(s) = CCM MANAGER 0x0003
>    Layer 1 Status:
>        ACTIVE
>    Layer 2 Status:
>        TEI = 0, Ces = 1, SAPI = 0, State = TEI_ASSIGNED
>    Layer 3 Status:
>        0 Active Layer 3 Call(s)
>    Active dsl 0 CCBs = 0
>    The Free Channel Mask:  0x8007
>    Number of L2 Discards = 0, L2 Session ID = 3
>
>
> You guys have any idea due to what this happening. I execute debug
> isdn q921 command as well. this is the out put of that.
>
> *Nov 11 12:36:31.651: ISDN Se0/0/0:23 Q921: User RX <- SABMEp sapi=0 tei=0
> *Nov 11 12:36:31.651: ISDN Se0/0/0:23 Q921: S4_SABME: BACKHAULED &
> vsc_wants_L2_up = FALSE
> *Nov 11 12:36:32.267: ISDN Se0/0/0:23 Q921: User RX <- SABMEp sapi=0 tei=0
> *Nov 11 12:36:32.267: ISDN Se0/0/0:23 Q921: S4_SABME: BACKHAULED &
> vsc_wants_L2_up = FALSE
>
> --
> Ravindra Lakpriya
>



-- 
Ravindra Lakpriya
+94 773 532 094
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Re: [OSL | CCIE_Voice] mgcp gw and T1 controller configuration

2010-11-11 Thread Josh Kittle
I'm new to the list - forgive me if I don't know how things work yet.

Can you show us your MGCP related configuration in Call Manager?  Also the
output of 'show ccm'


On Thu, Nov 11, 2010 at 2:35 PM, Ravindra Lakpriya wrote:

> Hi guys,
>
> I'm configuring T1 controller in a MGCP gateway.
>
> this is my T1 controller configuration
> --
> controller T1 0/0/0
> framing esf
> linecode b8zs
> pri-group timeslots 1-3,24 service mgcp
>
> MGCP Configuration
> -
>
> interface Serial0/0/0:23
>  no ip address
>  encapsulation hdlc
>  isdn switch-type primary-ni
>  isdn incoming-voice voice
>  isdn bind-l3 ccm-manager
>  no cdp enable
>
> ccm-manager switchback immediate
> ccm-manager redundant-host 210.10.10.10
> ccm-manager mgcp
> ccm-manager fax protocol cisco
> !
> mgcp call-agent 210.10.10.11 service-type mgcp version 0.1
> mgcp dtmf-relay voip codec all mode out-of-band
> mgcp fax t38 ecm
> mgcp bind control source-interface Loopback0
> mgcp bind media source-interface Loopback0
> !
> mgcp profile default
>
>
> When execute show isdn status it give me Layer 2 error.
>
> Global ISDN Switchtype = primary-ni
>
> %Q.931 is backhauled to CCM MANAGER 0x0003 on DSL 0. Layer 3 output
> may not apply
>
> ISDN Serial0/0/0:23 interface
>dsl 0, interface ISDN Switchtype = primary-ni
>L2 Protocol = Q.921 0x  L3 Protocol(s) = CCM MANAGER 0x0003
>Layer 1 Status:
>ACTIVE
>Layer 2 Status:
>TEI = 0, Ces = 1, SAPI = 0, State = TEI_ASSIGNED
>Layer 3 Status:
>0 Active Layer 3 Call(s)
>Active dsl 0 CCBs = 0
>The Free Channel Mask:  0x8007
>Number of L2 Discards = 0, L2 Session ID = 3
>
>
> You guys have any idea due to what this happening. I execute debug
> isdn q921 command as well. this is the out put of that.
>
> *Nov 11 12:36:31.651: ISDN Se0/0/0:23 Q921: User RX <- SABMEp sapi=0 tei=0
> *Nov 11 12:36:31.651: ISDN Se0/0/0:23 Q921: S4_SABME: BACKHAULED &
> vsc_wants_L2_up = FALSE
> *Nov 11 12:36:32.267: ISDN Se0/0/0:23 Q921: User RX <- SABMEp sapi=0 tei=0
> *Nov 11 12:36:32.267: ISDN Se0/0/0:23 Q921: S4_SABME: BACKHAULED &
> vsc_wants_L2_up = FALSE
>
> --
> Ravindra Lakpriya
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
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[OSL | CCIE_Voice] mgcp gw and T1 controller configuration

2010-11-11 Thread Ravindra Lakpriya
Hi guys,

I'm configuring T1 controller in a MGCP gateway.

this is my T1 controller configuration
--
controller T1 0/0/0
framing esf
linecode b8zs
pri-group timeslots 1-3,24 service mgcp

MGCP Configuration
-

interface Serial0/0/0:23
 no ip address
 encapsulation hdlc
 isdn switch-type primary-ni
 isdn incoming-voice voice
 isdn bind-l3 ccm-manager
 no cdp enable

ccm-manager switchback immediate
ccm-manager redundant-host 210.10.10.10
ccm-manager mgcp
ccm-manager fax protocol cisco
!
mgcp call-agent 210.10.10.11 service-type mgcp version 0.1
mgcp dtmf-relay voip codec all mode out-of-band
mgcp fax t38 ecm
mgcp bind control source-interface Loopback0
mgcp bind media source-interface Loopback0
!
mgcp profile default


When execute show isdn status it give me Layer 2 error.

Global ISDN Switchtype = primary-ni

%Q.931 is backhauled to CCM MANAGER 0x0003 on DSL 0. Layer 3 output
may not apply

ISDN Serial0/0/0:23 interface
dsl 0, interface ISDN Switchtype = primary-ni
L2 Protocol = Q.921 0x  L3 Protocol(s) = CCM MANAGER 0x0003
Layer 1 Status:
ACTIVE
Layer 2 Status:
TEI = 0, Ces = 1, SAPI = 0, State = TEI_ASSIGNED
Layer 3 Status:
0 Active Layer 3 Call(s)
Active dsl 0 CCBs = 0
The Free Channel Mask:  0x8007
Number of L2 Discards = 0, L2 Session ID = 3


You guys have any idea due to what this happening. I execute debug
isdn q921 command as well. this is the out put of that.

*Nov 11 12:36:31.651: ISDN Se0/0/0:23 Q921: User RX <- SABMEp sapi=0 tei=0
*Nov 11 12:36:31.651: ISDN Se0/0/0:23 Q921: S4_SABME: BACKHAULED &
vsc_wants_L2_up = FALSE
*Nov 11 12:36:32.267: ISDN Se0/0/0:23 Q921: User RX <- SABMEp sapi=0 tei=0
*Nov 11 12:36:32.267: ISDN Se0/0/0:23 Q921: S4_SABME: BACKHAULED &
vsc_wants_L2_up = FALSE

-- 
Ravindra Lakpriya
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Re: [OSL | CCIE_Voice] + Dialing issue

2010-11-11 Thread Shrini

Hi Iftikhar,

In the translation pattern 9.1[2-9]XX[2-9]XX -->

under calling party transformation select the check box use calling 
part's external phone mask.

Calling party transform mask - XXX
Calling /called party number type : National
Calling/Called party numbering plan : ISDN

In Gateway:

Under Incoming calling part settings:

Change the prefix to + from Default for National numbers.

You will get +1XX as calling party displayed on phone and missed 
calls


Thx
Shrini



On 11/10/2010 11:55 PM, Mujaddid Ahmed wrote:

Hi,
Please confirm what can be the reason of globalization not working 
with cisco 7965 phones.
I globalized it on CUCM gateway configuration by adding appropriate 
prefix on type of calls.

did, no mgcp / mgcp on gateway and reset the gateway from cucm as well.
call lands on phone and still shows only 7 digit calling number in 
both ringing and missed call.
nothing has been set so far in Device pool or on device to effect the 
globalization done on Gateway.
Is there any serivce parameter or any command on router that can 
prevent the globalize display on phone?
Please help, as i wasted lot of time in troubleshooting this 
particular issue in a very time critical environment.

Regards,
Iftikhar Ahmed


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Re: [OSL | CCIE_Voice] + Dialing issue

2010-11-11 Thread Prashant Patel
Hi Iftikhar,

Can you recollect anything missing from the isdn messages on the gatway?


Thanks
Prashant

On Thu, Nov 11, 2010 at 9:42 AM, CCIE Voice  wrote:

>  Make sure you do not have called transformation applied on the dev pool
> or the phone. This could be localizing your display back to 7 digits.
>
> --
>
>
> On Nov 11, 2010, at 0:55, Mujaddid Ahmed  wrote:
>
>   Hi,
>
> Please confirm what can be the reason of globalization not working with
> cisco 7965 phones.
> I globalized it on CUCM gateway configuration by adding appropriate prefix
> on type of calls.
>
> did, no mgcp / mgcp on gateway and reset the gateway from cucm as well.
>
> call lands on phone and still shows only 7 digit calling number in both
> ringing and missed call.
>
> nothing has been set so far in Device pool or on device to effect the
> globalization done on Gateway.
>
> Is there any serivce parameter or any command on router that can prevent
> the globalize display on phone?
>
> Please help, as i wasted lot of time in troubleshooting this particular
> issue in a very time critical environment.
>
> Regards,
> Iftikhar Ahmed
>
>  ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
>
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Re: [OSL | CCIE_Voice] Local PSTN phone to local IP Phone (Visa Versa), can't end or answer calls.

2010-11-11 Thread Prashant Patel
Hi Mike,

If these are registered to a CME, try downgrading  the phones to an older
firmware.

HTH
Prashant

On Thu, Nov 11, 2010 at 4:08 AM, Michael Murphy  wrote:

> Dear all,
>
> Tech support asked me to post this issue.
>
> I have a 3560 POE switch connected into an 871 router with 5 IP phones
> connected.
> 871 is connected into ISP wireless router.
>
> My issues: All regarding local IP phones to me and remotely connected into
> the Vracks.
>
> 1) When I ring numbers from the PSTN IP phone, I can't answer the call on
> any IP phone or I can't end the PSTN call, (either direction). The buttons
> don't seem to work.
> The call eventually stops when the PSTN & IP phone de-registers for a few
> seconds.
>
> I don't have this issue when I make HQ/branch phones ring each other. I can
> end calls/answer calls with no issues.
>
> Regards
> Mike
>
>
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
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Re: [OSL | CCIE_Voice] + Dialing issue

2010-11-11 Thread CCIE Voice
Make sure you do not have called transformation applied on the dev pool or the 
phone. This could be localizing your display back to 7 digits. 

--


On Nov 11, 2010, at 0:55, Mujaddid Ahmed  wrote:

> Hi,
>  
> Please confirm what can be the reason of globalization not working with cisco 
> 7965 phones.
> I globalized it on CUCM gateway configuration by adding appropriate prefix on 
> type of calls.
>  
> did, no mgcp / mgcp on gateway and reset the gateway from cucm as well.
>  
> call lands on phone and still shows only 7 digit calling number in both 
> ringing and missed call.
>  
> nothing has been set so far in Device pool or on device to effect the 
> globalization done on Gateway.
>  
> Is there any serivce parameter or any command on router that can prevent the 
> globalize display on phone?
>  
> Please help, as i wasted lot of time in troubleshooting this particular issue 
> in a very time critical environment.
>  
> Regards,
> Iftikhar Ahmed
> 
> ___
> For more information regarding industry leading CCIE Lab training, please 
> visit www.ipexpert.com
___
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[OSL | CCIE_Voice] RES: + Dialing issue

2010-11-11 Thread Marcelo Alexandria
Can you look in deb isdn and verify wich type of the call?? After verify
under gw in cucm if the call type match

 

 

 

De: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] Em nome de Mujaddid Ahmed
Enviada em: quinta-feira, 11 de novembro de 2010 05:55
Para: ccie_voice@onlinestudylist.com
Assunto: [OSL | CCIE_Voice] + Dialing issue

 

Hi, 

 

Please confirm what can be the reason of globalization not working with
cisco 7965 phones.

I globalized it on CUCM gateway configuration by adding appropriate prefix
on type of calls.

 

did, no mgcp / mgcp on gateway and reset the gateway from cucm as well. 

 

call lands on phone and still shows only 7 digit calling number in both
ringing and missed call. 

 

nothing has been set so far in Device pool or on device to effect the
globalization done on Gateway.

 

Is there any serivce parameter or any command on router that can prevent the
globalize display on phone?

 

Please help, as i wasted lot of time in troubleshooting this particular
issue in a very time critical environment.

 

Regards,

Iftikhar Ahmed

 

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[OSL | CCIE_Voice] Local PSTN phone to local IP Phone (Visa Versa), can't end or answer calls.

2010-11-11 Thread Michael Murphy
Dear all,

Tech support asked me to post this issue.

I have a 3560 POE switch connected into an 871 router with 5 IP phones 
connected.
871 is connected into ISP wireless router.

My issues: All regarding local IP phones to me and remotely connected into the 
Vracks.

1) When I ring numbers from the PSTN IP phone, I can't answer the call on any 
IP phone or I can't end the PSTN call, (either direction). The buttons don't 
seem to work.
The call eventually stops when the PSTN & IP phone de-registers for a few 
seconds.

I don't have this issue when I make HQ/branch phones ring each other. I can end 
calls/answer calls with no issues.

Regards
Mike


  
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