date:20101005

Re: [OSL | CCIE_Voice] MoH SRST (Stream from Flash)`

2010-10-05 Thread Mark Holloway

My gateway is H323. I can do the same exact configuration on an MGCP gateway 
and it works.  I always need to do no mgcp/mgcp as well.  Have you run into 
this situation on an H323 gateway?


On Oct 5, 2010, at 6:53 PM, Kalyan iyer wrote:

> Hey Mark,
> 
> I ran into the same problem with MOH. You have the correct configuration.
> 
>  However, If your BR1 RTR is a MGCP GW, like I had you will need to do a "no 
> mgcp" / "mgcp" to make the MOH work.
> 
> Thanks
> Kalyan
> 
> On Sun, Oct 3, 2010 at 9:39 PM, David Lee  wrote:
> Hey Mark,
> 
> Check the MRGL of the voice gateway.  The phone where you press hold <-- from 
> this phone is the source determined.  But the MOH is taken from the MRGL 
> configured on the holdee, in this case the VG.
> 
> Thanks,
> 
> -Dave
> 
> On Sun, Oct 3, 2010 at 9:08 PM,  
> wrote:
> Send CCIE_Voice mailing list submissions to
>ccie_voice@onlinestudylist.com
> 
> To subscribe or unsubscribe via the World Wide Web, visit
>http://onlinestudylist.com/mailman/listinfo/ccie_voice
> or, via email, send a message with subject or body 'help' to
>ccie_voice-requ...@onlinestudylist.com
> 
> You can reach the person managing the list at
>ccie_voice-ow...@onlinestudylist.com
> 
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of CCIE_Voice digest..."
> 
> 
> Today's Topics:
> 
>   1. MoH SRST (Stream from Flash)` (Mark Holloway)
>   2. Re: MoH SRST (Stream from Flash)` (Prashant Patel)
>   3. Re: MoH SRST (Stream from Flash)` (James Key)
>   4. Re: MoH SRST (Stream from Flash)` (Mark Holloway)
> 
> 
> --
> 
> Message: 1
> Date: Sun, 3 Oct 2010 17:17:44 -0700
> From: Mark Holloway 
> To: CCIE Voice Maillist 
> Subject: [OSL | CCIE_Voice] MoH SRST (Stream from Flash)`
> Message-ID: <85912468-288c-4ecc-9e9e-f9d9d22a3...@markholloway.com>
> Content-Type: text/plain; charset="us-ascii"
> 
> I thought I had this figured out but I'm slipping up somewhere.  Could use 
> some help. :)
> 
> I'm configuring multicast moh at BR1 using G.711 and streaming from BR1 
> router flash.  BR1 is an H323 gateway.
> 
> call-manager-fallback
> max-dn 24
> max-ephones 2
> ip source address 10.20.30.254 < this is the voice vlan default gateway
> moh music-on-hold.au < piano music file in flash
> multicast moh 239.1.1.1 port 16384 route  
> 
> 
> ip multicast-routing is enabled
> ip pim dense mode is configured on voice vlan interface and loop0 interface
> 
> cucm > moh audio source and PUB are configured for multicast routing (1 hop) 
> and assigned to mrg br1_mcast_moh which is assigned to mrgl br1 which is 
> assigned to br1 device pool
> 
> I have a device pool 'MoH' that has a region 'MoH' and is assigned G711 to 
> all other regions.  This region is assign to device pool MoH, and device pool 
> MoH is assign to the MoH servers.
> 
> 
> When HQ calls BR1 and HQ presses hold, the BR1 phone hears piano music.
> 
> When PSTN calls BR1 and BR1 presses hold, PSTN hears "beep beep beep"
> 
> r2# debug ephone moh
> EPHONE music-on-hold debugging is enabled
> Oct  4 00:09:50.794: MoH route If Vlan302 ETHERNET 192.168.65.254 via ARP
> Oct  4 00:09:50.794: MoH route If Loopback0 46 192.1.65.254 via 192.1.65.254
> 
> 
> 
> r2#debug ccm-m music-on-hold all
> Call Manager music-on-hold all debugging is on
> r2#
> Oct  4 00:16:08.156: moh_update_rtp: callID 12 dstCallID -1
> Oct  4 00:16:08.156: moh_process_ccb: dstadr 192.168.65.30, callid -1, port 
> 21836,
>codec 16, moh_en 0, moh_addr 0.0.0.0
> Oct  4 00:16:08.160: moh_update_rtp: callID 12 dstCallID -1
> Oct  4 00:16:08.160: moh_update_rtp: callID 12 dstCallID -1
> Oct  4 00:16:08.164: moh_process_ccb: dstadr 192.168.65.30, callid 11, port 
> 21836,
>codec 16, moh_en 0, moh_addr 0.0.0.0
> Oct  4 00:16:08.164: moh_update_rtp: callID 12 dstCallID 11
> Oct  4 00:16:08.180: %ISDN-6-CONNECT: Interface Serial0/0/0:0 is now 
> connected to 911 N/A
> Oct  4 00:16:08.180: %ISDN-6-CONNECT: Interface Serial0/0/0:0 is now 
> connected to 911 N/A
> Oct  4 00:16:09.028: moh_update_rtp: callID 12 dstCallID 11
> Oct  4 00:16:09.028: moh_update_rtp: callID 12 dstCallID 11
> Oct  4 00:16:09.032: moh_update_rtp: callID 12 dstCallID 11
> Oct  4 00:16:17.264: %ISDN-6-DISCONNECT: Interface Serial0/0/0:0  
> disconnected from 911 , call lasted 9 seconds
> Oct  4 00:16:17.268: moh_update_rtp: callID 12 dstCallID 11
> 
> 
> -- next part --
> An HTML attachment was scrubbed...
> URL: 
> 
> --
> 
> Message: 2
> Date: Sun, 3 Oct 2010 20:20:43 -0400
> From: Prashant Patel 
> To: Mark Holloway 
> Cc: CCIE Voice Maillist 
> Subject: Re: [OSL | CCIE_Voice] MoH SRST (Stream from Flash)`
> Message-ID:
>
> Content-Type: text/plain; charset="iso-8859-1"
> 
> Hi Mark,
> 
> Try adding tftp-server flash:music-on-hold.au
> 
> Also reload may help :)
> 
> Thanks,
> Prashant
> 
>

[OSL | CCIE_Voice] Vol2 Lab2 5.1 - 4digit GK calls

2010-10-05 Thread Mann Chaddha

Hi David

I reckon that by providing Voice Class Codec at the Outbound DP on
CME, you have allowed the call to proceed with G711 to the GK.
Ideally, if the Inbound DP (SIP Voice Pool in this case) and the
Outbound DP (DP to HQ/BR1) have been hard-coded to different codec
values, they should invoke a local XCoder. In your case, that doesn't
happen as your outbound DP has a Voice Class Codec assigned to it.

Why don't you hard-code the Outbound DP with G729 & Inbound DP with
G711 and then test the same?

HTH
Mann
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Re: [OSL | CCIE_Voice] Order for arranging resources in MRGL

2010-10-05 Thread CCIE Voice GMAIL

Can you please elaborate on this statement "if you only have a single CFB,
MOH, and MTP, you can list them in the same MRG because UCM won't can't use
a CFB if you need to insert an MTP."

 

Are you saying that when in a MRG, having both a CFB and an MTP would result
in only one resource being usable at a time?  Or are you saying that having
a CFB and MTP in the same MRGL will result in one resource being usable at a
time?

 

I'm just confused at the wording.

 

Thanks for your help,

Jeff

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of groganhockey
Sent: Tuesday, October 05, 2010 7:14 PM
To: Pithog Oil
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] Order for arranging resources in MRGL

 

Keep in mind these guidelines when configuring MRGL:
Resources are searched in the order listed in the MRGL, so if you insert
three MRGs into the MRGL, UCM will search the first MRG for an appropriate
resource, then search the second MRG, then search the third.

Resources within an MRG are selected in a round-robin fashion.

Ordering within the MRGL is only applicable to like-type of resources, so if
you only have a single CFB, MOH, and MTP, you can list them in the same MRG
because UCM won't can't use a CFB if you need to insert an MTP.

mike



On Tue, Oct 5, 2010 at 1:51 PM, Pithog Oil  wrote:


Hi experts

 

I know there is an Order for arranging arranging resources in the MRGL in a
scenario where i have multiple resoucres in a site , but i need to figure
out where Conference bridge fits in , in the order.

 

Please correct me if wrong

 

MOH first

Transcoder second

MTP third

 

Also i while like to know if its possible to have a resources Glut.

 



___
For more information regarding industry leading CCIE Lab training, please
visit www.ipexpert.com

 

___
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Re: [OSL | CCIE_Voice] Order for arranging resources in MRGL

2010-10-05 Thread groganhockey

Keep in mind these guidelines when configuring MRGL:
Resources are searched in the order listed in the MRGL, so if you insert
three MRGs into the MRGL, UCM will search the first MRG for an appropriate
resource, then search the second MRG, then search the third.

Resources within an MRG are selected in a round-robin fashion.

Ordering within the MRGL is only applicable to like-type of resources, so if
you only have a single CFB, MOH, and MTP, you can list them in the same MRG
because UCM won't can't use a CFB if you need to insert an MTP.

mike

On Tue, Oct 5, 2010 at 1:51 PM, Pithog Oil  wrote:

> Hi experts
>
> I know there is an Order for arranging arranging resources in the MRGL in a
> scenario where i have multiple resoucres in a site , but i need to figure
> out where Conference bridge fits in , in the order.
>
> Please correct me if wrong
>
> MOH first
> Transcoder second
> MTP third
>
> Also i while like to know if its possible to have a resources Glut.
>
>
>
>
> ___
> 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

Re: [OSL | CCIE_Voice] MoH SRST (Stream from Flash)`

2010-10-05 Thread Kalyan iyer

Hey Mark,

I ran into the same problem with MOH. You have the correct configuration.

 However, If your BR1 RTR is a MGCP GW, like I had you will need to do a "no
mgcp" / "mgcp" to make the MOH work.

Thanks
Kalyan

On Sun, Oct 3, 2010 at 9:39 PM, David Lee  wrote:

> Hey Mark,
>
> Check the MRGL of the voice gateway.  The phone where you press hold <--
> from this phone is the source determined.  But the MOH is taken from the
> MRGL configured on the holdee, in this case the VG.
>
> Thanks,
>
> -Dave
>
> On Sun, Oct 3, 2010 at 9:08 PM, wrote:
>
>> Send CCIE_Voice mailing list submissions to
>>ccie_voice@onlinestudylist.com
>>
>> To subscribe or unsubscribe via the World Wide Web, visit
>>http://onlinestudylist.com/mailman/listinfo/ccie_voice
>> or, via email, send a message with subject or body 'help' to
>>ccie_voice-requ...@onlinestudylist.com
>>
>> You can reach the person managing the list at
>>ccie_voice-ow...@onlinestudylist.com
>>
>> When replying, please edit your Subject line so it is more specific
>> than "Re: Contents of CCIE_Voice digest..."
>>
>>
>> Today's Topics:
>>
>>   1. MoH SRST (Stream from Flash)` (Mark Holloway)
>>   2. Re: MoH SRST (Stream from Flash)` (Prashant Patel)
>>   3. Re: MoH SRST (Stream from Flash)` (James Key)
>>   4. Re: MoH SRST (Stream from Flash)` (Mark Holloway)
>>
>>
>> --
>>
>> Message: 1
>> Date: Sun, 3 Oct 2010 17:17:44 -0700
>> From: Mark Holloway 
>> To: CCIE Voice Maillist 
>> Subject: [OSL | CCIE_Voice] MoH SRST (Stream from Flash)`
>> Message-ID: <85912468-288c-4ecc-9e9e-f9d9d22a3...@markholloway.com>
>> Content-Type: text/plain; charset="us-ascii"
>>
>> I thought I had this figured out but I'm slipping up somewhere.  Could use
>> some help. :)
>>
>> I'm configuring multicast moh at BR1 using G.711 and streaming from BR1
>> router flash.  BR1 is an H323 gateway.
>>
>> call-manager-fallback
>> max-dn 24
>> max-ephones 2
>> ip source address 10.20.30.254 < this is the voice vlan default gateway
>> moh music-on-hold.au < piano music file in flash
>> multicast moh 239.1.1.1 port 16384 route 
>> 
>>
>> ip multicast-routing is enabled
>> ip pim dense mode is configured on voice vlan interface and loop0
>> interface
>>
>> cucm > moh audio source and PUB are configured for multicast routing (1
>> hop) and assigned to mrg br1_mcast_moh which is assigned to mrgl br1 which
>> is assigned to br1 device pool
>>
>> I have a device pool 'MoH' that has a region 'MoH' and is assigned G711 to
>> all other regions.  This region is assign to device pool MoH, and device
>> pool MoH is assign to the MoH servers.
>>
>>
>> When HQ calls BR1 and HQ presses hold, the BR1 phone hears piano music.
>>
>> When PSTN calls BR1 and BR1 presses hold, PSTN hears "beep beep beep"
>>
>> r2# debug ephone moh
>> EPHONE music-on-hold debugging is enabled
>> Oct  4 00:09:50.794: MoH route If Vlan302 ETHERNET 192.168.65.254 via ARP
>> Oct  4 00:09:50.794: MoH route If Loopback0 46 192.1.65.254 via
>> 192.1.65.254
>>
>>
>>
>> r2#debug ccm-m music-on-hold all
>> Call Manager music-on-hold all debugging is on
>> r2#
>> Oct  4 00:16:08.156: moh_update_rtp: callID 12 dstCallID -1
>> Oct  4 00:16:08.156: moh_process_ccb: dstadr 192.168.65.30, callid -1,
>> port 21836,
>>codec 16, moh_en 0, moh_addr 0.0.0.0
>> Oct  4 00:16:08.160: moh_update_rtp: callID 12 dstCallID -1
>> Oct  4 00:16:08.160: moh_update_rtp: callID 12 dstCallID -1
>> Oct  4 00:16:08.164: moh_process_ccb: dstadr 192.168.65.30, callid 11,
>> port 21836,
>>codec 16, moh_en 0, moh_addr 0.0.0.0
>> Oct  4 00:16:08.164: moh_update_rtp: callID 12 dstCallID 11
>> Oct  4 00:16:08.180: %ISDN-6-CONNECT: Interface Serial0/0/0:0 is now
>> connected to 911 N/A
>> Oct  4 00:16:08.180: %ISDN-6-CONNECT: Interface Serial0/0/0:0 is now
>> connected to 911 N/A
>> Oct  4 00:16:09.028: moh_update_rtp: callID 12 dstCallID 11
>> Oct  4 00:16:09.028: moh_update_rtp: callID 12 dstCallID 11
>> Oct  4 00:16:09.032: moh_update_rtp: callID 12 dstCallID 11
>> Oct  4 00:16:17.264: %ISDN-6-DISCONNECT: Interface Serial0/0/0:0
>>  disconnected from 911 , call lasted 9 seconds
>> Oct  4 00:16:17.268: moh_update_rtp: callID 12 dstCallID 11
>>
>>
>> -- next part --
>> An HTML attachment was scrubbed...
>> URL:
>> 
>>
>> --
>>
>> Message: 2
>> Date: Sun, 3 Oct 2010 20:20:43 -0400
>> From: Prashant Patel 
>> To: Mark Holloway 
>> Cc: CCIE Voice Maillist 
>> Subject: Re: [OSL | CCIE_Voice] MoH SRST (Stream from Flash)`
>> Message-ID:
>>
>> Content-Type: text/plain; charset="iso-8859-1"
>>
>> Hi Mark,
>>
>> Try adding tftp-server flash:music-on-hold.au
>>
>> Also reload may help :)
>>
>> Thanks,
>> Prashant
>>
>>
>>
>> On Sun, Oct 3, 2010 at 8:17 PM, Mark Holloway 
>> wrote:
>>
>> > I thought I had this figured out but I'm slipping up somewhere.  Could
>> use
>> > some help. :)
>> >
>> > I'm c

Re: [OSL | CCIE_Voice] Vol2 Lab2 5.1 - 4digit GK calls

2010-10-05 Thread David A

yep its registered but never invoked when calling the GK from the CME SIP
phone. It is invoked when a call comes into the CME phone from CUCM and I
can see it in "sh sccp conn". I am using a 7975 phone as the SIP phone on
CME.

Thanks,
DA




On Tue, Oct 5, 2010 at 6:49 PM, Stutz, Bernhard  wrote:

>  Are you sure that your transcoder on cme is been registered?
> "show sdspfarm units" will show you that.
>
> as far as i know you don't need any special command on the voice
> register global to have the dspfarm resources beeing invoked.
>
> hth,
> Bernhard
>
> --
>  *Von:* David A [mailto:david.a...@gmail.com]
> *Gesendet:* Di 05.10.2010 21:08
>
> *An:* Stutz, Bernhard
> *Cc:* CCIE Voice GMAIL; osl osl
> *Betreff:* Re: [OSL | CCIE_Voice] Vol2 Lab2 5.1 - 4digit GK calls
>
>   Yes Bernhard, When I change the codec on the voice register pool to g729
> (default) it works fine. But I have a transcoder configured on the CME on
> telephony service which should be invoked if needed. The voice-class codec
> is already on the outgoing dialpeer towards gk but still it does not invoke
> a transcoder. I am not sure but do I need any special command on the voice
> register global to invoke the transcoder?
>
> Thanks,
> DA
>
> On Tue, Oct 5, 2010 at 1:40 PM, Stutz, Bernhard  wrote:
>
>>  hm sounds like an codec issue...
>> you have g711ulaw hardcoded at your cme sip phone. try to use there the
>> voice class codec aswell or if this doesn't help add a transcoder at cme
>> site aswell.
>> or try to use hardcoded g729 on the sip phone pool
>> don't forget to do always create prof and reset at voice register global
>> after a change
>>
>> hth,
>> Bernhard
>>
>> --
>>  *Von:* David A [mailto:david.a...@gmail.com]
>> *Gesendet:* Di 05.10.2010 18:16
>>
>> *An:* Stutz, Bernhard
>> *Cc:* CCIE Voice GMAIL; osl osl
>> *Betreff:* Re: [OSL | CCIE_Voice] Vol2 Lab2 5.1 - 4digit GK calls
>>
>>   Hi Bernhard,
>>
>> I added an MTP and a Transcoder on HQ. Added these to all mrgls. Checked
>> the MTP required box and inbound faststart. When I answer the call it just
>> disconnects.
>> I still see the call on the GK with 16kbps coming in.
>>
>> Thanks,
>> DA
>>
>>
>>
>>
>> On Tue, Oct 5, 2010 at 11:54 AM, Stutz, Bernhard 
>> wrote:
>>
>>>  Hi David,
>>>
>>> Do you have MTP on the gk trunk enabled and inbound faststart?
>>> You need to use  the IOS MTP as the CUCM MTP doesn't support G.729
>>>
>>> hth,
>>> Bernhard
>>>
>>> --
>>> *Von:* David A [mailto:david.a...@gmail.com]
>>> *Gesendet:* Di 05.10.2010 17:42
>>> *An:* Stutz, Bernhard
>>> *Cc:* CCIE Voice GMAIL; osl osl
>>>
>>> *Betreff:* Re: [OSL | CCIE_Voice] Vol2 Lab2 5.1 - 4digit GK calls
>>>
>>>   Hi Bernhard,
>>>
>>> The outboud call from the CME SIP phone is using the dial-peer
>>>
>>> dial-peer voice 15 voip
>>>  destination-pattern [15]...$
>>>  voice-class codec 1
>>>  session target ras
>>>  incoming called-number .
>>>  tech-prefix 1#
>>>  dtmf-relay h245-alphanumeric rtp-nte
>>> When I place a call I get this
>>>
>>> 3845-CME-SiteC#show call active voice compact
>>>A/O FAX T Codec   typePeer Address   IP
>>> R:
>>> Total call-legs: 2
>>>290 ANS T4 g711ulawVOIPP3002
>>> 10.10.202.54:25500
>>>291 ORG T4 g729r8 pre- VOIPP1#1001
>>> 0.0.0.0:0
>>>
>>> The other end is the GK and call ends on the SiteB phone. I dont think I
>>> need a dial-peer on the GK to route to CUCM as it is done through the GK
>>> trunk.
>>>  I can answer the call and see it on GK
>>>
>>> 2811-HQ-GW#sh gatekeeper call
>>> Total number of active calls = 1.
>>>  GATEKEEPER CALL INFO
>>>  
>>> LocalCallIDAge(secs)   BW
>>> 51-29194   7   16(Kbps)
>>>  Endpt(s): Alias E.164Addr
>>>src EP: SiteC-GW  3002
>>>CallSignalAddr  Port  RASSignalAddr   Port
>>>10.10.110.3 1720  10.10.110.3 58555
>>>  Endpt(s): Alias E.164Addr
>>>dst EP: gk-trunk_21#1001
>>>CallSignalAddr  Port  RASSignalAddr   Port
>>>10.137.151.26   1720  10.137.151.26   32796
>>> But after answering there is no audio and call drops after a few seconds.
>>>
>>>
>>> Thanks,
>>> DA
>>>
>>>
>>>
>>>
>>> On Tue, Oct 5, 2010 at 4:14 AM, Stutz, Bernhard 
>>> wrote:
>>>
  Hi,



 You are probably hitting the 0 dial-peer. Make sure you have a inbound
 dial-peer on the other end.

 Have a look which dial-peers you are using:



 sh call active voice compact

 or

 sh call active voice brief



 hth,

 Bernhard



 *Von:* ccie_voice-boun...@onlinestudylist.com [mailto:
 ccie_voice-boun...@onlinestudylist.com] *Im Auftrag von *CCIE Voice
 GMAIL
 *Gesendet:* Montag,

Re: [OSL | CCIE_Voice] Vol2 Lab2 5.1 - 4digit GK calls

2010-10-05 Thread Stutz, Bernhard

Are you sure that your transcoder on cme is been registered?
"show sdspfarm units" will show you that.
 
as far as i know you don't need any special command on the voice register 
global to have the dspfarm resources beeing invoked.
 
hth,
Bernhard 



Von: David A [mailto:david.a...@gmail.com]
Gesendet: Di 05.10.2010 21:08
An: Stutz, Bernhard
Cc: CCIE Voice GMAIL; osl osl
Betreff: Re: [OSL | CCIE_Voice] Vol2 Lab2 5.1 - 4digit GK calls


Yes Bernhard, When I change the codec on the voice register pool to g729 
(default) it works fine. But I have a transcoder configured on the CME on 
telephony service which should be invoked if needed. The voice-class codec is 
already on the outgoing dialpeer towards gk but still it does not invoke a 
transcoder. I am not sure but do I need any special command on the voice 
register global to invoke the transcoder?
 
Thanks,
DA


On Tue, Oct 5, 2010 at 1:40 PM, Stutz, Bernhard  wrote:


hm sounds like an codec issue...
you have g711ulaw hardcoded at your cme sip phone. try to use there the 
voice class codec aswell or if this doesn't help add a transcoder at cme site 
aswell.
or try to use hardcoded g729 on the sip phone pool
don't forget to do always create prof and reset at voice register 
global after a change
 
hth,
Bernhard




Von: David A [mailto:david.a...@gmail.com]

Gesendet: Di 05.10.2010 18:16 

An: Stutz, Bernhard
Cc: CCIE Voice GMAIL; osl osl
Betreff: Re: [OSL | CCIE_Voice] Vol2 Lab2 5.1 - 4digit GK calls


Hi Bernhard,
 
I added an MTP and a Transcoder on HQ. Added these to all mrgls. 
Checked the MTP required box and inbound faststart. When I answer the call it 
just disconnects.
I still see the call on the GK with 16kbps coming in.
 
Thanks,
DA
 


 
On Tue, Oct 5, 2010 at 11:54 AM, Stutz, Bernhard  
wrote:


Hi David,
 
Do you have MTP on the gk trunk enabled and inbound faststart?
You need to use  the IOS MTP as the CUCM MTP doesn't support 
G.729
 
hth,
Bernhard



Von: David A [mailto:david.a...@gmail.com]
Gesendet: Di 05.10.2010 17:42
An: Stutz, Bernhard
Cc: CCIE Voice GMAIL; osl osl 

Betreff: Re: [OSL | CCIE_Voice] Vol2 Lab2 5.1 - 4digit GK calls


Hi Bernhard,
 
The outboud call from the CME SIP phone is using the dial-peer
 
dial-peer voice 15 voip
 destination-pattern [15]...$
 voice-class codec 1
 session target ras
 incoming called-number .
 tech-prefix 1#
 dtmf-relay h245-alphanumeric rtp-nte

When I place a call I get this
 
3845-CME-SiteC#show call active voice compact
   A/O FAX T Codec   typePeer Address  
 IP R:
Total call-legs: 2
   290 ANS T4 g711ulawVOIPP3002 
10.10.202.54:25500  
   291 ORG T4 g729r8 pre- VOIPP1#1001   
   0.0.0.0:0  
 
The other end is the GK and call ends on the SiteB phone. I 
dont think I need a dial-peer on the GK to route to CUCM as it is done through 
the GK trunk.
 I can answer the call and see it on GK
 
2811-HQ-GW#sh gatekeeper call
Total number of active calls = 1.
 GATEKEEPER CALL INFO
 
LocalCallIDAge(secs)   BW
51-29194   7   16(Kbps)
 Endpt(s): Alias E.164Addr
   src EP: SiteC-GW  3002
   CallSignalAddr  Port  RASSignalAddr   Port
   10.10.110.3 1720  10.10.110.3 58555
 Endpt(s): Alias E.164Addr
   dst EP: gk-trunk_21#1001
   CallSignalAddr  Port  RASSignalAddr   Port
   10.137.151.26   1720  10.137.151.26   32796

But after answering there is no audio and call drops after a 
few seconds.
 
 
Thanks,
DA
 


 
On Tue, Oct 5, 2010 at 4:14 AM, Stutz, Be

Re: [OSL | CCIE_Voice] Vol2 Lab2 5.1 - 4digit GK calls

2010-10-05 Thread David A

Yes Bernhard, When I change the codec on the voice register pool to g729
(default) it works fine. But I have a transcoder configured on the CME on
telephony service which should be invoked if needed. The voice-class codec
is already on the outgoing dialpeer towards gk but still it does not invoke
a transcoder. I am not sure but do I need any special command on the voice
register global to invoke the transcoder?

Thanks,
DA

On Tue, Oct 5, 2010 at 1:40 PM, Stutz, Bernhard  wrote:

>  hm sounds like an codec issue...
> you have g711ulaw hardcoded at your cme sip phone. try to use there the
> voice class codec aswell or if this doesn't help add a transcoder at cme
> site aswell.
> or try to use hardcoded g729 on the sip phone pool
> don't forget to do always create prof and reset at voice register global
> after a change
>
> hth,
> Bernhard
>
> --
>  *Von:* David A [mailto:david.a...@gmail.com]
> *Gesendet:* Di 05.10.2010 18:16
>
> *An:* Stutz, Bernhard
> *Cc:* CCIE Voice GMAIL; osl osl
> *Betreff:* Re: [OSL | CCIE_Voice] Vol2 Lab2 5.1 - 4digit GK calls
>
>   Hi Bernhard,
>
> I added an MTP and a Transcoder on HQ. Added these to all mrgls. Checked
> the MTP required box and inbound faststart. When I answer the call it just
> disconnects.
> I still see the call on the GK with 16kbps coming in.
>
> Thanks,
> DA
>
>
>
>
> On Tue, Oct 5, 2010 at 11:54 AM, Stutz, Bernhard 
> wrote:
>
>>  Hi David,
>>
>> Do you have MTP on the gk trunk enabled and inbound faststart?
>> You need to use  the IOS MTP as the CUCM MTP doesn't support G.729
>>
>> hth,
>> Bernhard
>>
>> --
>> *Von:* David A [mailto:david.a...@gmail.com]
>> *Gesendet:* Di 05.10.2010 17:42
>> *An:* Stutz, Bernhard
>> *Cc:* CCIE Voice GMAIL; osl osl
>>
>> *Betreff:* Re: [OSL | CCIE_Voice] Vol2 Lab2 5.1 - 4digit GK calls
>>
>>   Hi Bernhard,
>>
>> The outboud call from the CME SIP phone is using the dial-peer
>>
>> dial-peer voice 15 voip
>>  destination-pattern [15]...$
>>  voice-class codec 1
>>  session target ras
>>  incoming called-number .
>>  tech-prefix 1#
>>  dtmf-relay h245-alphanumeric rtp-nte
>> When I place a call I get this
>>
>> 3845-CME-SiteC#show call active voice compact
>>A/O FAX T Codec   typePeer Address   IP
>> R:
>> Total call-legs: 2
>>290 ANS T4 g711ulawVOIPP3002
>> 10.10.202.54:25500
>>291 ORG T4 g729r8 pre- VOIPP1#1001
>> 0.0.0.0:0
>>
>> The other end is the GK and call ends on the SiteB phone. I dont think I
>> need a dial-peer on the GK to route to CUCM as it is done through the GK
>> trunk.
>>  I can answer the call and see it on GK
>>
>> 2811-HQ-GW#sh gatekeeper call
>> Total number of active calls = 1.
>>  GATEKEEPER CALL INFO
>>  
>> LocalCallIDAge(secs)   BW
>> 51-29194   7   16(Kbps)
>>  Endpt(s): Alias E.164Addr
>>src EP: SiteC-GW  3002
>>CallSignalAddr  Port  RASSignalAddr   Port
>>10.10.110.3 1720  10.10.110.3 58555
>>  Endpt(s): Alias E.164Addr
>>dst EP: gk-trunk_21#1001
>>CallSignalAddr  Port  RASSignalAddr   Port
>>10.137.151.26   1720  10.137.151.26   32796
>> But after answering there is no audio and call drops after a few seconds.
>>
>>
>> Thanks,
>> DA
>>
>>
>>
>>
>> On Tue, Oct 5, 2010 at 4:14 AM, Stutz, Bernhard 
>> wrote:
>>
>>>  Hi,
>>>
>>>
>>>
>>> You are probably hitting the 0 dial-peer. Make sure you have a inbound
>>> dial-peer on the other end.
>>>
>>> Have a look which dial-peers you are using:
>>>
>>>
>>>
>>> sh call active voice compact
>>>
>>> or
>>>
>>> sh call active voice brief
>>>
>>>
>>>
>>> hth,
>>>
>>> Bernhard
>>>
>>>
>>>
>>> *Von:* ccie_voice-boun...@onlinestudylist.com [mailto:
>>> ccie_voice-boun...@onlinestudylist.com] *Im Auftrag von *CCIE Voice
>>> GMAIL
>>> *Gesendet:* Montag, 4. Oktober 2010 23:24
>>> *An:* 'osl osl'
>>> *Betreff:* Re: [OSL | CCIE_Voice] Vol2 Lab2 5.1 - 4digit GK calls
>>>
>>>
>>>
>>> For the first issue, if you add the CME router as an H323 gateway in CUCM
>>> the correct bandwidth will show.  Make sure that the CSS includes the
>>> partition that contains the phones.
>>>
>>
>>
>>
>>>
>>>
>>> *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
>>> ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *David A
>>> *Sent:* Monday, October 04, 2010 1:43 PM
>>> *To:* ccie_voice@onlinestudylist.com
>>> *Subject:* [OSL | CCIE_Voice] Vol2 Lab2 5.1 - 4digit GK calls
>>>
>>>
>>>
>>> Hi All,
>>>
>>>
>>>
>>> I am doing the Vol2 Lab2 GK scenario and running into a couple of issues.
>>>
>>>
>>>
>>> issue 1 - Call from CME SCCP phone to CUCM phones is using 128kbps. When
>>> I check the codec used on the call on both phones it says g729. The gk-tunk
>>> is in DP GK with region g729 to everyone.
>>>
>>>
>>>
>>> 281

[OSL | CCIE_Voice] Order for arranging resources in MRGL

2010-10-05 Thread Pithog Oil

Hi experts
I know there is an Order for arranging arranging resources in the MRGL in a 
scenario where i have multiple resoucres in a site , but i need to figure out 
where Conference bridge fits in , in the order.
Please correct me if wrong
MOH firstTranscoder secondMTP third
Also i while like to know if its possible to have a resources Glut.




  ___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

Re: [OSL | CCIE_Voice] Vol2 Lab2 5.1 - 4digit GK calls

2010-10-05 Thread Stutz, Bernhard

hm sounds like an codec issue...
you have g711ulaw hardcoded at your cme sip phone. try to use there the voice 
class codec aswell or if this doesn't help add a transcoder at cme site aswell.
or try to use hardcoded g729 on the sip phone pool
don't forget to do always create prof and reset at voice register global after 
a change
 
hth,
Bernhard



Von: David A [mailto:david.a...@gmail.com]
Gesendet: Di 05.10.2010 18:16
An: Stutz, Bernhard
Cc: CCIE Voice GMAIL; osl osl
Betreff: Re: [OSL | CCIE_Voice] Vol2 Lab2 5.1 - 4digit GK calls


Hi Bernhard,
 
I added an MTP and a Transcoder on HQ. Added these to all mrgls. Checked the 
MTP required box and inbound faststart. When I answer the call it just 
disconnects.
I still see the call on the GK with 16kbps coming in.
 
Thanks,
DA
 


 
On Tue, Oct 5, 2010 at 11:54 AM, Stutz, Bernhard  wrote:


Hi David,
 
Do you have MTP on the gk trunk enabled and inbound faststart?
You need to use  the IOS MTP as the CUCM MTP doesn't support G.729
 
hth,
Bernhard



Von: David A [mailto:david.a...@gmail.com]
Gesendet: Di 05.10.2010 17:42
An: Stutz, Bernhard
Cc: CCIE Voice GMAIL; osl osl 

Betreff: Re: [OSL | CCIE_Voice] Vol2 Lab2 5.1 - 4digit GK calls


Hi Bernhard,
 
The outboud call from the CME SIP phone is using the dial-peer
 
dial-peer voice 15 voip
 destination-pattern [15]...$
 voice-class codec 1
 session target ras
 incoming called-number .
 tech-prefix 1#
 dtmf-relay h245-alphanumeric rtp-nte

When I place a call I get this
 
3845-CME-SiteC#show call active voice compact
   A/O FAX T Codec   typePeer Address   IP 
R:
Total call-legs: 2
   290 ANS T4 g711ulawVOIPP3002 
10.10.202.54:25500  
   291 ORG T4 g729r8 pre- VOIPP1#1001  
0.0.0.0:0  
 
The other end is the GK and call ends on the SiteB phone. I dont think 
I need a dial-peer on the GK to route to CUCM as it is done through the GK 
trunk.
 I can answer the call and see it on GK
 
2811-HQ-GW#sh gatekeeper call
Total number of active calls = 1.
 GATEKEEPER CALL INFO
 
LocalCallIDAge(secs)   BW
51-29194   7   16(Kbps)
 Endpt(s): Alias E.164Addr
   src EP: SiteC-GW  3002
   CallSignalAddr  Port  RASSignalAddr   Port
   10.10.110.3 1720  10.10.110.3 58555
 Endpt(s): Alias E.164Addr
   dst EP: gk-trunk_21#1001
   CallSignalAddr  Port  RASSignalAddr   Port
   10.137.151.26   1720  10.137.151.26   32796

But after answering there is no audio and call drops after a few 
seconds.
 
 
Thanks,
DA
 


 
On Tue, Oct 5, 2010 at 4:14 AM, Stutz, Bernhard  
wrote:


Hi,

 

You are probably hitting the 0 dial-peer. Make sure you have a 
inbound dial-peer on the other end.

Have a look which dial-peers you are using:

 

sh call active voice compact

or

sh call active voice brief

 

hth,

Bernhard

 

Von: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] Im Auftrag von CCIE Voice GMAIL
Gesendet: Montag, 4. Oktober 2010 23:24
An: 'osl osl'
Betreff: Re: [OSL | CCIE_Voice] Vol2 Lab2 5.1 - 4digit GK calls

 

For the first issue, if you add the CME router as an H323 
gateway in CUCM the correct bandwidth will show.  Make sure that the CSS 
includes the partition that contains the phones.

 
 

 

From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of David A
Sent: Monday, October 04, 2010 1:43 PM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] Vol2 Lab2 5.1 - 4digit GK calls

 

Hi All,

 

I am doing the Vol2 Lab2 GK scenario and running into a couple 
of issues.

 

issue 1 - Call from CME SCCP phone to CUCM phones is using 
128kbps. When I check the codec used on the call on both phones i

Re: [OSL | CCIE_Voice] Vol2 Lab2 5.1 - 4digit GK calls

2010-10-05 Thread David A

Hi Bernhard,

I added an MTP and a Transcoder on HQ. Added these to all mrgls. Checked the
MTP required box and inbound faststart. When I answer the call it just
disconnects.
I still see the call on the GK with 16kbps coming in.

Thanks,
DA




On Tue, Oct 5, 2010 at 11:54 AM, Stutz, Bernhard  wrote:

>  Hi David,
>
> Do you have MTP on the gk trunk enabled and inbound faststart?
> You need to use  the IOS MTP as the CUCM MTP doesn't support G.729
>
> hth,
> Bernhard
>
> --
> *Von:* David A [mailto:david.a...@gmail.com]
> *Gesendet:* Di 05.10.2010 17:42
> *An:* Stutz, Bernhard
> *Cc:* CCIE Voice GMAIL; osl osl
>
> *Betreff:* Re: [OSL | CCIE_Voice] Vol2 Lab2 5.1 - 4digit GK calls
>
>   Hi Bernhard,
>
> The outboud call from the CME SIP phone is using the dial-peer
>
> dial-peer voice 15 voip
>  destination-pattern [15]...$
>  voice-class codec 1
>  session target ras
>  incoming called-number .
>  tech-prefix 1#
>  dtmf-relay h245-alphanumeric rtp-nte
> When I place a call I get this
>
> 3845-CME-SiteC#show call active voice compact
>A/O FAX T Codec   typePeer Address   IP
> R:
> Total call-legs: 2
>290 ANS T4 g711ulawVOIPP3002
> 10.10.202.54:25500
>291 ORG T4 g729r8 pre- VOIPP1#1001
> 0.0.0.0:0
>
> The other end is the GK and call ends on the SiteB phone. I dont think I
> need a dial-peer on the GK to route to CUCM as it is done through the GK
> trunk.
>  I can answer the call and see it on GK
>
> 2811-HQ-GW#sh gatekeeper call
> Total number of active calls = 1.
>  GATEKEEPER CALL INFO
>  
> LocalCallIDAge(secs)   BW
> 51-29194   7   16(Kbps)
>  Endpt(s): Alias E.164Addr
>src EP: SiteC-GW  3002
>CallSignalAddr  Port  RASSignalAddr   Port
>10.10.110.3 1720  10.10.110.3 58555
>  Endpt(s): Alias E.164Addr
>dst EP: gk-trunk_21#1001
>CallSignalAddr  Port  RASSignalAddr   Port
>10.137.151.26   1720  10.137.151.26   32796
> But after answering there is no audio and call drops after a few seconds.
>
>
> Thanks,
> DA
>
>
>
>
> On Tue, Oct 5, 2010 at 4:14 AM, Stutz, Bernhard  wrote:
>
>>  Hi,
>>
>>
>>
>> You are probably hitting the 0 dial-peer. Make sure you have a inbound
>> dial-peer on the other end.
>>
>> Have a look which dial-peers you are using:
>>
>>
>>
>> sh call active voice compact
>>
>> or
>>
>> sh call active voice brief
>>
>>
>>
>> hth,
>>
>> Bernhard
>>
>>
>>
>> *Von:* ccie_voice-boun...@onlinestudylist.com [mailto:
>> ccie_voice-boun...@onlinestudylist.com] *Im Auftrag von *CCIE Voice GMAIL
>> *Gesendet:* Montag, 4. Oktober 2010 23:24
>> *An:* 'osl osl'
>> *Betreff:* Re: [OSL | CCIE_Voice] Vol2 Lab2 5.1 - 4digit GK calls
>>
>>
>>
>> For the first issue, if you add the CME router as an H323 gateway in CUCM
>> the correct bandwidth will show.  Make sure that the CSS includes the
>> partition that contains the phones.
>>
>
>
>
>>
>>
>> *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
>> ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *David A
>> *Sent:* Monday, October 04, 2010 1:43 PM
>> *To:* ccie_voice@onlinestudylist.com
>> *Subject:* [OSL | CCIE_Voice] Vol2 Lab2 5.1 - 4digit GK calls
>>
>>
>>
>> Hi All,
>>
>>
>>
>> I am doing the Vol2 Lab2 GK scenario and running into a couple of issues.
>>
>>
>>
>> issue 1 - Call from CME SCCP phone to CUCM phones is using 128kbps. When I
>> check the codec used on the call on both phones it says g729. The gk-tunk is
>> in DP GK with region g729 to everyone.
>>
>>
>>
>> 2811-HQ-GW#sh gatekeeper call
>> Total number of active calls = 1.
>>  GATEKEEPER CALL INFO
>>  
>> LocalCallIDAge(secs)   BW
>> 25-49659   21  128(Kbps) <--- should
>> be 16kbps as per the requirement
>>  Endpt(s): Alias E.164Addr
>>src EP: SiteC-GW  3003
>>CallSignalAddr  Port  RASSignalAddr   Port
>>10.10.110.3 1720  10.10.110.3 58555
>>  Endpt(s): Alias E.164Addr
>>dst EP: gk-trunk_21#1002
>>CallSignalAddr  Port  RASSignalAddr   Port
>>10.137.151.26   1720  10.137.151.26   33447
>>
>>
>>
>>
>>
>> issue 2 - Call from CUCM to CME SIP phone(g711)  works fine and invokes
>> the transcoder and i see a 16kbps GK call. However when I call from CME SIP
>> phone to any CUCM phone, CUCM phone rings and I can answer it. However it
>> drops after a few seconds and I see no transcoder being used. Here are my
>> configs
>>
>>
>>
>> Site C -
>>
>>
>>
>> voice register pool  1
>>  id mac 0025.4593.0368
>>  type 7975
>>  number 1 dn 1
>>  number 2 dn 2
>>  template 1
>>  description 32143002
>>  codec g711ulaw
>>
>>

Re: [OSL | CCIE_Voice] Vol2 Lab2 5.1 - 4digit GK calls

2010-10-05 Thread Stutz, Bernhard

Hi David,
 
Do you have MTP on the gk trunk enabled and inbound faststart?
You need to use  the IOS MTP as the CUCM MTP doesn't support G.729
 
hth,
Bernhard



Von: David A [mailto:david.a...@gmail.com]
Gesendet: Di 05.10.2010 17:42
An: Stutz, Bernhard
Cc: CCIE Voice GMAIL; osl osl
Betreff: Re: [OSL | CCIE_Voice] Vol2 Lab2 5.1 - 4digit GK calls


Hi Bernhard,
 
The outboud call from the CME SIP phone is using the dial-peer
 
dial-peer voice 15 voip
 destination-pattern [15]...$
 voice-class codec 1
 session target ras
 incoming called-number .
 tech-prefix 1#
 dtmf-relay h245-alphanumeric rtp-nte

When I place a call I get this
 
3845-CME-SiteC#show call active voice compact
   A/O FAX T Codec   typePeer Address   IP 
R:
Total call-legs: 2
   290 ANS T4 g711ulawVOIPP3002 10.10.202.54:25500 
 
   291 ORG T4 g729r8 pre- VOIPP1#1001  0.0.0.0:0 
 
 
The other end is the GK and call ends on the SiteB phone. I dont think I need a 
dial-peer on the GK to route to CUCM as it is done through the GK trunk.
 I can answer the call and see it on GK
 
2811-HQ-GW#sh gatekeeper call
Total number of active calls = 1.
 GATEKEEPER CALL INFO
 
LocalCallIDAge(secs)   BW
51-29194   7   16(Kbps)
 Endpt(s): Alias E.164Addr
   src EP: SiteC-GW  3002
   CallSignalAddr  Port  RASSignalAddr   Port
   10.10.110.3 1720  10.10.110.3 58555
 Endpt(s): Alias E.164Addr
   dst EP: gk-trunk_21#1001
   CallSignalAddr  Port  RASSignalAddr   Port
   10.137.151.26   1720  10.137.151.26   32796

But after answering there is no audio and call drops after a few seconds.
 
 
Thanks,
DA
 


 
On Tue, Oct 5, 2010 at 4:14 AM, Stutz, Bernhard  wrote:


Hi,

 

You are probably hitting the 0 dial-peer. Make sure you have a inbound 
dial-peer on the other end.

Have a look which dial-peers you are using:

 

sh call active voice compact

or

sh call active voice brief

 

hth,

Bernhard

 

Von: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] Im Auftrag von CCIE Voice GMAIL
Gesendet: Montag, 4. Oktober 2010 23:24
An: 'osl osl'
Betreff: Re: [OSL | CCIE_Voice] Vol2 Lab2 5.1 - 4digit GK calls

 

For the first issue, if you add the CME router as an H323 gateway in 
CUCM the correct bandwidth will show.  Make sure that the CSS includes the 
partition that contains the phones.

 
 

 

From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of David A
Sent: Monday, October 04, 2010 1:43 PM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] Vol2 Lab2 5.1 - 4digit GK calls

 

Hi All,

 

I am doing the Vol2 Lab2 GK scenario and running into a couple of 
issues.

 

issue 1 - Call from CME SCCP phone to CUCM phones is using 128kbps. 
When I check the codec used on the call on both phones it says g729. The 
gk-tunk is in DP GK with region g729 to everyone.

 

2811-HQ-GW#sh gatekeeper call
Total number of active calls = 1.
 GATEKEEPER CALL INFO
 
LocalCallIDAge(secs)   BW
25-49659   21  128(Kbps) <--- 
should be 16kbps as per the requirement
 Endpt(s): Alias E.164Addr
   src EP: SiteC-GW  3003
   CallSignalAddr  Port  RASSignalAddr   Port
   10.10.110.3 1720  10.10.110.3 58555
 Endpt(s): Alias E.164Addr
   dst EP: gk-trunk_21#1002
   CallSignalAddr  Port  RASSignalAddr   Port
   10.137.151.26   1720  10.137.151.26   33447

 

 

issue 2 - Call from CUCM to CME SIP phone(g711)  works fine and invokes 
the transcoder and i see a 16kbps GK call. However when I call from CME SIP 
phone to any CUCM phone, CUCM phone rings and I can answer it. However it drops 
after a few seconds and I see no transcoder being used. Here are my configs

 

Site C - 

 

voice register pool  1
 id mac 0025.4593.0368
 type 7975
 number 1 dn 1
 number 2 dn 2
 template 1
 description 32143002
 codec g711ulaw

 

!
dial-peer voice 15 voip
 destination-pattern [15]...$
 session target ras
 incoming called-nu

Re: [OSL | CCIE_Voice] Vol2 Lab2 5.1 - 4digit GK calls

2010-10-05 Thread David A

Hi Bernhard,

The outboud call from the CME SIP phone is using the dial-peer

dial-peer voice 15 voip
 destination-pattern [15]...$
 voice-class codec 1
 session target ras
 incoming called-number .
 tech-prefix 1#
 dtmf-relay h245-alphanumeric rtp-nte
When I place a call I get this

3845-CME-SiteC#show call active voice compact
   A/O FAX T Codec   typePeer Address   IP
R:
Total call-legs: 2
   290 ANS T4 g711ulawVOIPP3002
10.10.202.54:25500
   291 ORG T4 g729r8 pre- VOIPP1#1001  0.0.0.0:0

The other end is the GK and call ends on the SiteB phone. I dont think I
need a dial-peer on the GK to route to CUCM as it is done through the GK
trunk.
 I can answer the call and see it on GK

2811-HQ-GW#sh gatekeeper call
Total number of active calls = 1.
 GATEKEEPER CALL INFO
 
LocalCallIDAge(secs)   BW
51-29194   7   16(Kbps)
 Endpt(s): Alias E.164Addr
   src EP: SiteC-GW  3002
   CallSignalAddr  Port  RASSignalAddr   Port
   10.10.110.3 1720  10.10.110.3 58555
 Endpt(s): Alias E.164Addr
   dst EP: gk-trunk_21#1001
   CallSignalAddr  Port  RASSignalAddr   Port
   10.137.151.26   1720  10.137.151.26   32796
But after answering there is no audio and call drops after a few seconds.


Thanks,
DA




On Tue, Oct 5, 2010 at 4:14 AM, Stutz, Bernhard  wrote:

>  Hi,
>
>
>
> You are probably hitting the 0 dial-peer. Make sure you have a inbound
> dial-peer on the other end.
>
> Have a look which dial-peers you are using:
>
>
>
> sh call active voice compact
>
> or
>
> sh call active voice brief
>
>
>
> hth,
>
> Bernhard
>
>
>
> *Von:* ccie_voice-boun...@onlinestudylist.com [mailto:
> ccie_voice-boun...@onlinestudylist.com] *Im Auftrag von *CCIE Voice GMAIL
> *Gesendet:* Montag, 4. Oktober 2010 23:24
> *An:* 'osl osl'
> *Betreff:* Re: [OSL | CCIE_Voice] Vol2 Lab2 5.1 - 4digit GK calls
>
>
>
> For the first issue, if you add the CME router as an H323 gateway in CUCM
> the correct bandwidth will show.  Make sure that the CSS includes the
> partition that contains the phones.
>



>
>
> *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
> ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *David A
> *Sent:* Monday, October 04, 2010 1:43 PM
> *To:* ccie_voice@onlinestudylist.com
> *Subject:* [OSL | CCIE_Voice] Vol2 Lab2 5.1 - 4digit GK calls
>
>
>
> Hi All,
>
>
>
> I am doing the Vol2 Lab2 GK scenario and running into a couple of issues.
>
>
>
> issue 1 - Call from CME SCCP phone to CUCM phones is using 128kbps. When I
> check the codec used on the call on both phones it says g729. The gk-tunk is
> in DP GK with region g729 to everyone.
>
>
>
> 2811-HQ-GW#sh gatekeeper call
> Total number of active calls = 1.
>  GATEKEEPER CALL INFO
>  
> LocalCallIDAge(secs)   BW
> 25-49659   21  128(Kbps) <--- should be
> 16kbps as per the requirement
>  Endpt(s): Alias E.164Addr
>src EP: SiteC-GW  3003
>CallSignalAddr  Port  RASSignalAddr   Port
>10.10.110.3 1720  10.10.110.3 58555
>  Endpt(s): Alias E.164Addr
>dst EP: gk-trunk_21#1002
>CallSignalAddr  Port  RASSignalAddr   Port
>10.137.151.26   1720  10.137.151.26   33447
>
>
>
>
>
> issue 2 - Call from CUCM to CME SIP phone(g711)  works fine and invokes the
> transcoder and i see a 16kbps GK call. However when I call from CME SIP
> phone to any CUCM phone, CUCM phone rings and I can answer it. However it
> drops after a few seconds and I see no transcoder being used. Here are my
> configs
>
>
>
> Site C -
>
>
>
> voice register pool  1
>  id mac 0025.4593.0368
>  type 7975
>  number 1 dn 1
>  number 2 dn 2
>  template 1
>  description 32143002
>  codec g711ulaw
>
>
>
> !
> dial-peer voice 15 voip
>  destination-pattern [15]...$
>  session target ras
>  incoming called-number .
>  tech-prefix 1#
>  dtmf-relay h245-alphanumeric rtp-nte
> !
> dial-peer voice 3000 voip
>  incoming called-number 3...$
>  dtmf-relay h245-alphanumeric
> !
>
>
>
> Any clues?
>
>
>
> Thanks,
>
> DA
>
> ___
> 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

Re: [OSL | CCIE_Voice] Vol2 Lab2 5.1 - 4digit GK calls

2010-10-05 Thread David A

Thanks Roger. I will try the workaround.

2010/10/5 Roger Källberg 

>  Region will overwrite the default, so to work around the "fix" for the
> bug you need to specify the codec to G711 to be used within the region local
> to the phone/VGW/and so on.
>
> Sincerely
>
>  *Roger Källberg*
> CCIE #26199 (Voice)
> Consultant
> Cygate AB
>  --
> *Från:* Stutz, Bernhard [st...@pandacom.de]
> *Skickat:* den 5 oktober 2010 16:09
> *Till:* Roger Källberg; David A; ccie_voice@onlinestudylist.com
> *Ämne:* AW: [OSL | CCIE_Voice] Vol2 Lab2 5.1 - 4digit GK calls
>
>If you are changing the IntraAudioRegionDefault to G.729 you will fix
> that but you will then break the requirement to have G.711 for intra
> region calls. isn't it?
> Or will in this case the Region Setting overwrite the default setting?
>
> cheers,
> Bernhard
>
> --
> *Von:* ccie_voice-boun...@onlinestudylist.com im Auftrag von Roger
> Källberg
> *Gesendet:* Di 05.10.2010 14:21
> *An:* David A; ccie_voice@onlinestudylist.com
> *Betreff:* Re: [OSL | CCIE_Voice] Vol2 Lab2 5.1 - 4digit GK calls
>
>  Your hitting bug CSCsl74701. This is a well known bug that you should be
> really familiar with. There are many posts on the OSL about this and also
> Matthew Barry has an excellent post on his blog about this. See this url,
> http://ciscovoiceguru.com/382/cscsl74701-bug-details/
>
> Sincerely
>
>  *Roger Källberg*
> CCIE #26199 (Voice)
> Consultant
> Cygate AB
>  --
> *Från:* David A [david.a...@gmail.com]
> *Skickat:* den 4 oktober 2010 22:43
> *Till:* ccie_voice@onlinestudylist.com
> *Ämne:* [OSL | CCIE_Voice] Vol2 Lab2 5.1 - 4digit GK calls
>
>  Hi All,
>
> I am doing the Vol2 Lab2 GK scenario and running into a couple of issues.
>
> issue 1 - Call from CME SCCP phone to CUCM phones is using 128kbps. When I
> check the codec used on the call on both phones it says g729. The gk-tunk is
> in DP GK with region g729 to everyone.
>
> 2811-HQ-GW#sh gatekeeper call
> Total number of active calls = 1.
>  GATEKEEPER CALL INFO
>  
> LocalCallIDAge(secs)   BW
> 25-49659   21  128(Kbps) <--- should be
> 16kbps as per the requirement
>  Endpt(s): Alias E.164Addr
>src EP: SiteC-GW  3003
>CallSignalAddr  Port  RASSignalAddr   Port
>10.10.110.3 1720  10.10.110.3 58555
>  Endpt(s): Alias E.164Addr
>dst EP: gk-trunk_21#1002
>CallSignalAddr  Port  RASSignalAddr   Port
>10.137.151.26   1720  10.137.151.26   33447
>
>
> issue 2 - Call from CUCM to CME SIP phone(g711)  works fine and invokes the
> transcoder and i see a 16kbps GK call. However when I call from CME SIP
> phone to any CUCM phone, CUCM phone rings and I can answer it. However it
> drops after a few seconds and I see no transcoder being used. Here are my
> configs
>
> Site C -
>
> voice register pool  1
>  id mac 0025.4593.0368
>  type 7975
>  number 1 dn 1
>  number 2 dn 2
>  template 1
>  description 32143002
>  codec g711ulaw
>
> !
> dial-peer voice 15 voip
>  destination-pattern [15]...$
>  session target ras
>  incoming called-number .
>  tech-prefix 1#
>  dtmf-relay h245-alphanumeric rtp-nte
> !
> dial-peer voice 3000 voip
>  incoming called-number 3...$
>  dtmf-relay h245-alphanumeric
> !
>
> Any clues?
>
> Thanks,
> DA
>
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

Re: [OSL | CCIE_Voice] Vol2 Lab2 5.1 - 4digit GK calls

2010-10-05 Thread Roger Källberg

Region will overwrite the default, so to work around the "fix" for the bug you 
need to specify the codec to G711 to be used within the region local to the 
phone/VGW/and so on.

Sincerely

Roger Källberg
CCIE #26199 (Voice)
Consultant
Cygate AB

Från: Stutz, Bernhard [st...@pandacom.de]
Skickat: den 5 oktober 2010 16:09
Till: Roger Källberg; David A; ccie_voice@onlinestudylist.com
Ämne: AW: [OSL | CCIE_Voice] Vol2 Lab2 5.1 - 4digit GK calls

If you are changing the IntraAudioRegionDefault to G.729 you will fix that but 
you will then break the requirement to have G.711 for intra region calls. isn't 
it?
Or will in this case the Region Setting overwrite the default setting?

cheers,
Bernhard


Von: ccie_voice-boun...@onlinestudylist.com im Auftrag von Roger Källberg
Gesendet: Di 05.10.2010 14:21
An: David A; ccie_voice@onlinestudylist.com
Betreff: Re: [OSL | CCIE_Voice] Vol2 Lab2 5.1 - 4digit GK calls

Your hitting bug CSCsl74701. This is a well known bug that you should be really 
familiar with. There are many posts on the OSL about this and also Matthew 
Barry has an excellent post on his blog about this. See this url, 
http://ciscovoiceguru.com/382/cscsl74701-bug-details/

Sincerely

Roger Källberg
CCIE #26199 (Voice)
Consultant
Cygate AB

Från: David A [david.a...@gmail.com]
Skickat: den 4 oktober 2010 22:43
Till: ccie_voice@onlinestudylist.com
Ämne: [OSL | CCIE_Voice] Vol2 Lab2 5.1 - 4digit GK calls

Hi All,

I am doing the Vol2 Lab2 GK scenario and running into a couple of issues.

issue 1 - Call from CME SCCP phone to CUCM phones is using 128kbps. When I 
check the codec used on the call on both phones it says g729. The gk-tunk is in 
DP GK with region g729 to everyone.

2811-HQ-GW#sh gatekeeper call
Total number of active calls = 1.
 GATEKEEPER CALL INFO
 
LocalCallIDAge(secs)   BW
25-49659   21  128(Kbps) <--- should be 
16kbps as per the requirement
 Endpt(s): Alias E.164Addr
   src EP: SiteC-GW  3003
   CallSignalAddr  Port  RASSignalAddr   Port
   10.10.110.3 1720  10.10.110.3 58555
 Endpt(s): Alias E.164Addr
   dst EP: gk-trunk_21#1002
   CallSignalAddr  Port  RASSignalAddr   Port
   10.137.151.26   1720  10.137.151.26   33447


issue 2 - Call from CUCM to CME SIP phone(g711)  works fine and invokes the 
transcoder and i see a 16kbps GK call. However when I call from CME SIP phone 
to any CUCM phone, CUCM phone rings and I can answer it. However it drops after 
a few seconds and I see no transcoder being used. Here are my configs

Site C -

voice register pool  1
 id mac 0025.4593.0368
 type 7975
 number 1 dn 1
 number 2 dn 2
 template 1
 description 32143002
 codec g711ulaw

!
dial-peer voice 15 voip
 destination-pattern [15]...$
 session target ras
 incoming called-number .
 tech-prefix 1#
 dtmf-relay h245-alphanumeric rtp-nte
!
dial-peer voice 3000 voip
 incoming called-number 3...$
 dtmf-relay h245-alphanumeric
!

Any clues?

Thanks,
DA___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

Re: [OSL | CCIE_Voice] BACD drop through not working on 2821

2010-10-05 Thread Tamer Ismail

Hello,

I found the solution...

Removing en from this command [param drop-through-prompt en_bacd_welcome.au]

So, the command should be param drop-through-prompt _bacd_welcome.au

 

Enjoy Cisco J

 

From: Roger Källberg [mailto:roger.kallb...@cygate.se] 
Sent: Tuesday, October 05, 2010 2:48 PM
To: Tamer Ismail; CCIE_Voice@onlinestudylist.com
Subject: SV: [OSL | CCIE_Voice] BACD drop through not working on 2821

 

Your missing some needed parameters, see this config example from the
document "Cisco Unified CME B-ACD and Tcl Call-Handling Applications". This
can be found on Cisco support and will be available to you in the lab.


Cisco Unified CME B-ACD with Drop-Through Option: Example
The following example sets parameters for an AA service called aa and a
call-queue service called callq. The direct-dial number to reach the AA
service is 800 555-0100. Callers to this number drop through to the ephone
hunt group that has a pilot number of 5071 after hearing the initial prompt
from the file en_dt_prompt.au.


ephone-hunt 10 sequential
 pilot 5071
 list 5011, 5012, 5013, 5014, 5015
 timeout 10
 
application
 service callq tftp://192.168.254.254/user1/CallQ/B-ACD/app-b-acd.tcl
  param queue-manager-debugs 1
  param aa-hunt1 5071
  param number-of-hunt-grps 1
  param queue-len 10


 service aa tftp://192.168.254.254/user1/CallQ/B-ACD/app-b-acd-aa.tcl
  paramspace english location tftp://192.168.254.254/user1/prompts/ 
  paramspace english index 0
  paramspace english language en
  param aa-pilot 8005550100
  param number-of-hunt-grps 1
  param service-name callq
  param handoff-string aa
  param second-greeting-time 60
  param drop-through-option 1
  param drop-through-prompt _dt_prompt.au
  param call-retry-timer 15
  param max-time-call-retry 700
  param voice-mail 5000
  param max-time-vm-retry 2 


Sincerely

Roger Källberg
CCIE #26199 (Voice)
Consultant
Cygate AB

Från: Tamer Ismail [tih...@gmail.com]
Skickat: den 5 oktober 2010 12:26
Till: CCIE_Voice@onlinestudylist.com
Ämne: [OSL | CCIE_Voice] BACD drop through not working on 2821

Hello,
I want to configure BACD drop through on router 2821, when I call the
pilot number, the call is drops.
Can someone check and tell me, may I made something wrong.

Router:
2800 Software (C2800NM-IPVOICEK9-M), c2800nm-ipvoicek9-mz.124-22.YB7.bin

Flash:
10   30421 Oct 04 2010 11:37:44   app-b-acd-3.0.0.2.tcl
11   55599 Oct 04 2010 11:37:46   app-b-acd-aa-3.0.0.2.tcl
12   75650 Oct 04 2010 11:37:48   en_bacd_allagentsbusy.au
13   83291 Oct 04 2010 11:37:50   en_bacd_disconnect.au
14   63055 Oct 04 2010 11:37:50   en_bacd_enter_dest.au
15   37952 Oct 04 2010 11:37:52   en_bacd_invalidoption.au
16  496521 Oct 04 2010 11:38:04  en_bacd_music_on_hold.au
17  123446 Oct 04 2010 11:38:08  en_bacd_options_menu.au
18   42978 Oct 04 2010 11:38:10   en_bacd_welcome.au

Configuration:
!
application
 service aa flash:app-b-acd-aa-3.0.0.2.tcl
  param number-of-hunt-grps 1
  paramspace english index 0
  param drop-through-option 1
  param handoff-string aa
  paramspace english language en
  param max-time-vm-retry 2
  param aa-pilot 6299
  paramspace english location flash:
  param second-greeting-time 30
  param drop-through-prompt en_bacd_welcome.au
  param call-retry-timer 15
  param max-time-call-retry 700
  param service-name callq
 !
 service callq flash:app-b-acd-3.0.0.2.tcl
  param queue-len 10
  param aa-hunt1 6300
  param queue-manager-debugs 1
  param number-of-hunt-grps 1
 !
dial-peer voice 6299 voip
 service aa
 destination-pattern 6299
 session target ipv4:10.5.21.2
 incoming called-number 6299
 dtmf-relay h245-alphanumeric
 codec g711ulaw
 no vad
!
ephone-hunt 10 longest-idle
 pilot 6300
 list 6201, 6202
 timeout 10, 10
!

Thanks for help.
Tamer,

___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

Re: [OSL | CCIE_Voice] Vol2 Lab2 5.1 - 4digit GK calls

2010-10-05 Thread Stutz, Bernhard

If you are changing the IntraAudioRegionDefault to G.729 you will fix that but 
you will then break the requirement to have G.711 for intra region calls. isn't 
it?
Or will in this case the Region Setting overwrite the default setting?
 
cheers,
Bernhard



Von: ccie_voice-boun...@onlinestudylist.com im Auftrag von Roger Källberg
Gesendet: Di 05.10.2010 14:21
An: David A; ccie_voice@onlinestudylist.com
Betreff: Re: [OSL | CCIE_Voice] Vol2 Lab2 5.1 - 4digit GK calls


Your hitting bug CSCsl74701. This is a well known bug that you should be really 
familiar with. There are many posts on the OSL about this and also Matthew 
Barry has an excellent post on his blog about this. See this url, 
http://ciscovoiceguru.com/382/cscsl74701-bug-details/
 
Sincerely
 
Roger Källberg
CCIE #26199 (Voice)
Consultant
Cygate AB



Från: David A [david.a...@gmail.com]
Skickat: den 4 oktober 2010 22:43
Till: ccie_voice@onlinestudylist.com
Ämne: [OSL | CCIE_Voice] Vol2 Lab2 5.1 - 4digit GK calls


Hi All,
 
I am doing the Vol2 Lab2 GK scenario and running into a couple of issues.
 
issue 1 - Call from CME SCCP phone to CUCM phones is using 128kbps. When I 
check the codec used on the call on both phones it says g729. The gk-tunk is in 
DP GK with region g729 to everyone.
 
2811-HQ-GW#sh gatekeeper call
Total number of active calls = 1.
 GATEKEEPER CALL INFO
 
LocalCallIDAge(secs)   BW
25-49659   21  128(Kbps) <--- should be 
16kbps as per the requirement
 Endpt(s): Alias E.164Addr
   src EP: SiteC-GW  3003
   CallSignalAddr  Port  RASSignalAddr   Port
   10.10.110.3 1720  10.10.110.3 58555
 Endpt(s): Alias E.164Addr
   dst EP: gk-trunk_21#1002
   CallSignalAddr  Port  RASSignalAddr   Port
   10.137.151.26   1720  10.137.151.26   33447
 
 
issue 2 - Call from CUCM to CME SIP phone(g711)  works fine and invokes the 
transcoder and i see a 16kbps GK call. However when I call from CME SIP phone 
to any CUCM phone, CUCM phone rings and I can answer it. However it drops after 
a few seconds and I see no transcoder being used. Here are my configs
 
Site C - 
 
voice register pool  1
 id mac 0025.4593.0368
 type 7975
 number 1 dn 1
 number 2 dn 2
 template 1
 description 32143002
 codec g711ulaw

 
!
dial-peer voice 15 voip
 destination-pattern [15]...$
 session target ras
 incoming called-number .
 tech-prefix 1#
 dtmf-relay h245-alphanumeric rtp-nte
!
dial-peer voice 3000 voip
 incoming called-number 3...$
 dtmf-relay h245-alphanumeric
!

 
Any clues?
 
Thanks,
DA

___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

Re: [OSL | CCIE_Voice] BACD drop through not working on 2821

2010-10-05 Thread Roger Källberg

You can find the document I referred to on cisco.com -> support - Voice and 
Unified Communication -> IP Telephony -> Call Control -> Cisco Unified 
Communication Manager Express -> Configuration Guides

Roger Källberg
CCIE #26199 (Voice)
Consultant
Cygate AB

Från: Tamer Ismail [tih...@gmail.com]
Skickat: den 5 oktober 2010 12:26
Till: CCIE_Voice@onlinestudylist.com
Ämne: [OSL | CCIE_Voice] BACD drop through not working on 2821

Hello,
I want to configure BACD drop through on router 2821, when I call the
pilot number, the call is drops.
Can someone check and tell me, may I made something wrong.

Router:
2800 Software (C2800NM-IPVOICEK9-M), c2800nm-ipvoicek9-mz.124-22.YB7.bin

Flash:
10   30421 Oct 04 2010 11:37:44   app-b-acd-3.0.0.2.tcl
11   55599 Oct 04 2010 11:37:46   app-b-acd-aa-3.0.0.2.tcl
12   75650 Oct 04 2010 11:37:48   en_bacd_allagentsbusy.au
13   83291 Oct 04 2010 11:37:50   en_bacd_disconnect.au
14   63055 Oct 04 2010 11:37:50   en_bacd_enter_dest.au
15   37952 Oct 04 2010 11:37:52   en_bacd_invalidoption.au
16  496521 Oct 04 2010 11:38:04  en_bacd_music_on_hold.au
17  123446 Oct 04 2010 11:38:08  en_bacd_options_menu.au
18   42978 Oct 04 2010 11:38:10   en_bacd_welcome.au

Configuration:
!
application
 service aa flash:app-b-acd-aa-3.0.0.2.tcl
  param number-of-hunt-grps 1
  paramspace english index 0
  param drop-through-option 1
  param handoff-string aa
  paramspace english language en
  param max-time-vm-retry 2
  param aa-pilot 6299
  paramspace english location flash:
  param second-greeting-time 30
  param drop-through-prompt en_bacd_welcome.au
  param call-retry-timer 15
  param max-time-call-retry 700
  param service-name callq
 !
 service callq flash:app-b-acd-3.0.0.2.tcl
  param queue-len 10
  param aa-hunt1 6300
  param queue-manager-debugs 1
  param number-of-hunt-grps 1
 !
dial-peer voice 6299 voip
 service aa
 destination-pattern 6299
 session target ipv4:10.5.21.2
 incoming called-number 6299
 dtmf-relay h245-alphanumeric
 codec g711ulaw
 no vad
!
ephone-hunt 10 longest-idle
 pilot 6300
 list 6201, 6202
 timeout 10, 10
!

Thanks for help.
Tamer,

___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

Re: [OSL | CCIE_Voice] BACD drop through not working on 2821

2010-10-05 Thread Roger Källberg

Your missing some needed parameters, see this config example from the document 
"Cisco Unified CME B-ACD and Tcl Call-Handling Applications". This can be found 
on Cisco support and will be available to you in the lab.


Cisco Unified CME B-ACD with Drop-Through Option: Example
The following example sets parameters for an AA service called aa and a 
call-queue service called callq. The direct-dial number to reach the AA service 
is 800 555-0100. Callers to this number drop through to the ephone hunt group 
that has a pilot number of 5071 after hearing the initial prompt from the file 
en_dt_prompt.au.

ephone-hunt 10 sequential
 pilot 5071
 list 5011, 5012, 5013, 5014, 5015
 timeout 10

application
 service callq tftp://192.168.254.254/user1/CallQ/B-ACD/app-b-acd.tcl
  param queue-manager-debugs 1
  param aa-hunt1 5071
  param number-of-hunt-grps 1
  param queue-len 10

 service aa tftp://192.168.254.254/user1/CallQ/B-ACD/app-b-acd-aa.tcl
  paramspace english location tftp://192.168.254.254/user1/prompts/
  paramspace english index 0
  paramspace english language en
  param aa-pilot 8005550100
  param number-of-hunt-grps 1
  param service-name callq
  param handoff-string aa
  param second-greeting-time 60
  param drop-through-option 1
  param drop-through-prompt _dt_prompt.au
  param call-retry-timer 15
  param max-time-call-retry 700
  param voice-mail 5000
  param max-time-vm-retry 2


Sincerely

Roger Källberg
CCIE #26199 (Voice)
Consultant
Cygate AB

Från: Tamer Ismail [tih...@gmail.com]
Skickat: den 5 oktober 2010 12:26
Till: CCIE_Voice@onlinestudylist.com
Ämne: [OSL | CCIE_Voice] BACD drop through not working on 2821

Hello,
I want to configure BACD drop through on router 2821, when I call the
pilot number, the call is drops.
Can someone check and tell me, may I made something wrong.

Router:
2800 Software (C2800NM-IPVOICEK9-M), c2800nm-ipvoicek9-mz.124-22.YB7.bin

Flash:
10   30421 Oct 04 2010 11:37:44   app-b-acd-3.0.0.2.tcl
11   55599 Oct 04 2010 11:37:46   app-b-acd-aa-3.0.0.2.tcl
12   75650 Oct 04 2010 11:37:48   en_bacd_allagentsbusy.au
13   83291 Oct 04 2010 11:37:50   en_bacd_disconnect.au
14   63055 Oct 04 2010 11:37:50   en_bacd_enter_dest.au
15   37952 Oct 04 2010 11:37:52   en_bacd_invalidoption.au
16  496521 Oct 04 2010 11:38:04  en_bacd_music_on_hold.au
17  123446 Oct 04 2010 11:38:08  en_bacd_options_menu.au
18   42978 Oct 04 2010 11:38:10   en_bacd_welcome.au

Configuration:
!
application
 service aa flash:app-b-acd-aa-3.0.0.2.tcl
  param number-of-hunt-grps 1
  paramspace english index 0
  param drop-through-option 1
  param handoff-string aa
  paramspace english language en
  param max-time-vm-retry 2
  param aa-pilot 6299
  paramspace english location flash:
  param second-greeting-time 30
  param drop-through-prompt en_bacd_welcome.au
  param call-retry-timer 15
  param max-time-call-retry 700
  param service-name callq
 !
 service callq flash:app-b-acd-3.0.0.2.tcl
  param queue-len 10
  param aa-hunt1 6300
  param queue-manager-debugs 1
  param number-of-hunt-grps 1
 !
dial-peer voice 6299 voip
 service aa
 destination-pattern 6299
 session target ipv4:10.5.21.2
 incoming called-number 6299
 dtmf-relay h245-alphanumeric
 codec g711ulaw
 no vad
!
ephone-hunt 10 longest-idle
 pilot 6300
 list 6201, 6202
 timeout 10, 10
!

Thanks for help.
Tamer,___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

Re: [OSL | CCIE_Voice] Vol2 Lab2 5.1 - 4digit GK calls

2010-10-05 Thread Roger Källberg

Your hitting bug CSCsl74701. This is a well known bug that you should be really 
familiar with. There are many posts on the OSL about this and also Matthew 
Barry has an excellent post on his blog about this. See this url, 
http://ciscovoiceguru.com/382/cscsl74701-bug-details/

Sincerely

Roger Källberg
CCIE #26199 (Voice)
Consultant
Cygate AB

Från: David A [david.a...@gmail.com]
Skickat: den 4 oktober 2010 22:43
Till: ccie_voice@onlinestudylist.com
Ämne: [OSL | CCIE_Voice] Vol2 Lab2 5.1 - 4digit GK calls

Hi All,

I am doing the Vol2 Lab2 GK scenario and running into a couple of issues.

issue 1 - Call from CME SCCP phone to CUCM phones is using 128kbps. When I 
check the codec used on the call on both phones it says g729. The gk-tunk is in 
DP GK with region g729 to everyone.

2811-HQ-GW#sh gatekeeper call
Total number of active calls = 1.
 GATEKEEPER CALL INFO
 
LocalCallIDAge(secs)   BW
25-49659   21  128(Kbps) <--- should be 
16kbps as per the requirement
 Endpt(s): Alias E.164Addr
   src EP: SiteC-GW  3003
   CallSignalAddr  Port  RASSignalAddr   Port
   10.10.110.3 1720  10.10.110.3 58555
 Endpt(s): Alias E.164Addr
   dst EP: gk-trunk_21#1002
   CallSignalAddr  Port  RASSignalAddr   Port
   10.137.151.26   1720  10.137.151.26   33447


issue 2 - Call from CUCM to CME SIP phone(g711)  works fine and invokes the 
transcoder and i see a 16kbps GK call. However when I call from CME SIP phone 
to any CUCM phone, CUCM phone rings and I can answer it. However it drops after 
a few seconds and I see no transcoder being used. Here are my configs

Site C -

voice register pool  1
 id mac 0025.4593.0368
 type 7975
 number 1 dn 1
 number 2 dn 2
 template 1
 description 32143002
 codec g711ulaw

!
dial-peer voice 15 voip
 destination-pattern [15]...$
 session target ras
 incoming called-number .
 tech-prefix 1#
 dtmf-relay h245-alphanumeric rtp-nte
!
dial-peer voice 3000 voip
 incoming called-number 3...$
 dtmf-relay h245-alphanumeric
!

Any clues?

Thanks,
DA___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

[OSL | CCIE_Voice] BACD drop through not working on 2821

2010-10-05 Thread Tamer Ismail

Hello,
I want to configure BACD drop through on router 2821, when I call the
pilot number, the call is drops.
Can someone check and tell me, may I made something wrong.

Router:
2800 Software (C2800NM-IPVOICEK9-M), c2800nm-ipvoicek9-mz.124-22.YB7.bin

Flash:
10   30421 Oct 04 2010 11:37:44   app-b-acd-3.0.0.2.tcl
11   55599 Oct 04 2010 11:37:46   app-b-acd-aa-3.0.0.2.tcl
12   75650 Oct 04 2010 11:37:48   en_bacd_allagentsbusy.au
13   83291 Oct 04 2010 11:37:50   en_bacd_disconnect.au
14   63055 Oct 04 2010 11:37:50   en_bacd_enter_dest.au
15   37952 Oct 04 2010 11:37:52   en_bacd_invalidoption.au
16  496521 Oct 04 2010 11:38:04  en_bacd_music_on_hold.au
17  123446 Oct 04 2010 11:38:08  en_bacd_options_menu.au
18   42978 Oct 04 2010 11:38:10   en_bacd_welcome.au

Configuration:
!
application
 service aa flash:app-b-acd-aa-3.0.0.2.tcl
  param number-of-hunt-grps 1
  paramspace english index 0
  param drop-through-option 1
  param handoff-string aa
  paramspace english language en
  param max-time-vm-retry 2
  param aa-pilot 6299
  paramspace english location flash:
  param second-greeting-time 30
  param drop-through-prompt en_bacd_welcome.au
  param call-retry-timer 15
  param max-time-call-retry 700
  param service-name callq
 !
 service callq flash:app-b-acd-3.0.0.2.tcl
  param queue-len 10
  param aa-hunt1 6300
  param queue-manager-debugs 1
  param number-of-hunt-grps 1
 !
dial-peer voice 6299 voip
 service aa
 destination-pattern 6299
 session target ipv4:10.5.21.2
 incoming called-number 6299
 dtmf-relay h245-alphanumeric
 codec g711ulaw
 no vad
!
ephone-hunt 10 longest-idle
 pilot 6300
 list 6201, 6202
 timeout 10, 10
!

Thanks for help.
Tamer,
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

Re: [OSL | CCIE_Voice] Vol2 Lab2 5.1 - 4digit GK calls

2010-10-05 Thread Stutz, Bernhard

For Calls from CME to CUCM you need a MTP and inbound FastStart however
when you have inbound FastStart you will see 128kps on the gatekeeper
call even when you see on the phone g729 being used. That's a limitation
when you use inbound faststart the bandwidth request is hardcoded to
this. When you turn off inbound fast start calls to sip phones won't
work at all.

 

Hth,

Bernhard

 

Von: Stutz, Bernhard 
Gesendet: Dienstag, 5. Oktober 2010 10:14
An: 'CCIE Voice GMAIL'; 'osl osl'
Betreff: AW: [OSL | CCIE_Voice] Vol2 Lab2 5.1 - 4digit GK calls

 

Hi,

 

You are probably hitting the 0 dial-peer. Make sure you have a inbound
dial-peer on the other end.

Have a look which dial-peers you are using:

 

sh call active voice compact

or

sh call active voice brief

 

hth,

Bernhard

 

Von: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] Im Auftrag von CCIE
Voice GMAIL
Gesendet: Montag, 4. Oktober 2010 23:24
An: 'osl osl'
Betreff: Re: [OSL | CCIE_Voice] Vol2 Lab2 5.1 - 4digit GK calls

 

For the first issue, if you add the CME router as an H323 gateway in
CUCM the correct bandwidth will show.  Make sure that the CSS includes
the partition that contains the phones.

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of David A
Sent: Monday, October 04, 2010 1:43 PM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] Vol2 Lab2 5.1 - 4digit GK calls

 

Hi All,

 

I am doing the Vol2 Lab2 GK scenario and running into a couple of
issues.

 

issue 1 - Call from CME SCCP phone to CUCM phones is using 128kbps. When
I check the codec used on the call on both phones it says g729. The
gk-tunk is in DP GK with region g729 to everyone.

 

2811-HQ-GW#sh gatekeeper call
Total number of active calls = 1.
 GATEKEEPER CALL INFO
 
LocalCallIDAge(secs)   BW
25-49659   21  128(Kbps) <--- should
be 16kbps as per the requirement
 Endpt(s): Alias E.164Addr
   src EP: SiteC-GW  3003
   CallSignalAddr  Port  RASSignalAddr   Port
   10.10.110.3 1720  10.10.110.3 58555
 Endpt(s): Alias E.164Addr
   dst EP: gk-trunk_21#1002
   CallSignalAddr  Port  RASSignalAddr   Port
   10.137.151.26   1720  10.137.151.26   33447

 

 

issue 2 - Call from CUCM to CME SIP phone(g711)  works fine and invokes
the transcoder and i see a 16kbps GK call. However when I call from CME
SIP phone to any CUCM phone, CUCM phone rings and I can answer it.
However it drops after a few seconds and I see no transcoder being used.
Here are my configs

 

Site C - 

 

voice register pool  1
 id mac 0025.4593.0368
 type 7975
 number 1 dn 1
 number 2 dn 2
 template 1
 description 32143002
 codec g711ulaw

 

!
dial-peer voice 15 voip
 destination-pattern [15]...$
 session target ras
 incoming called-number .
 tech-prefix 1#
 dtmf-relay h245-alphanumeric rtp-nte
!
dial-peer voice 3000 voip
 incoming called-number 3...$
 dtmf-relay h245-alphanumeric
!

 

Any clues?

 

Thanks,

DA

___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

Re: [OSL | CCIE_Voice] Vol2 Lab2 5.1 - 4digit GK calls

2010-10-05 Thread Stutz, Bernhard

Hi,

 

You are probably hitting the 0 dial-peer. Make sure you have a inbound
dial-peer on the other end.

Have a look which dial-peers you are using:

 

sh call active voice compact

or

sh call active voice brief

 

hth,

Bernhard

 

Von: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] Im Auftrag von CCIE
Voice GMAIL
Gesendet: Montag, 4. Oktober 2010 23:24
An: 'osl osl'
Betreff: Re: [OSL | CCIE_Voice] Vol2 Lab2 5.1 - 4digit GK calls

 

For the first issue, if you add the CME router as an H323 gateway in
CUCM the correct bandwidth will show.  Make sure that the CSS includes
the partition that contains the phones.

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of David A
Sent: Monday, October 04, 2010 1:43 PM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] Vol2 Lab2 5.1 - 4digit GK calls

 

Hi All,

 

I am doing the Vol2 Lab2 GK scenario and running into a couple of
issues.

 

issue 1 - Call from CME SCCP phone to CUCM phones is using 128kbps. When
I check the codec used on the call on both phones it says g729. The
gk-tunk is in DP GK with region g729 to everyone.

 

2811-HQ-GW#sh gatekeeper call
Total number of active calls = 1.
 GATEKEEPER CALL INFO
 
LocalCallIDAge(secs)   BW
25-49659   21  128(Kbps) <--- should
be 16kbps as per the requirement
 Endpt(s): Alias E.164Addr
   src EP: SiteC-GW  3003
   CallSignalAddr  Port  RASSignalAddr   Port
   10.10.110.3 1720  10.10.110.3 58555
 Endpt(s): Alias E.164Addr
   dst EP: gk-trunk_21#1002
   CallSignalAddr  Port  RASSignalAddr   Port
   10.137.151.26   1720  10.137.151.26   33447

 

 

issue 2 - Call from CUCM to CME SIP phone(g711)  works fine and invokes
the transcoder and i see a 16kbps GK call. However when I call from CME
SIP phone to any CUCM phone, CUCM phone rings and I can answer it.
However it drops after a few seconds and I see no transcoder being used.
Here are my configs

 

Site C - 

 

voice register pool  1
 id mac 0025.4593.0368
 type 7975
 number 1 dn 1
 number 2 dn 2
 template 1
 description 32143002
 codec g711ulaw

 

!
dial-peer voice 15 voip
 destination-pattern [15]...$
 session target ras
 incoming called-number .
 tech-prefix 1#
 dtmf-relay h245-alphanumeric rtp-nte
!
dial-peer voice 3000 voip
 incoming called-number 3...$
 dtmf-relay h245-alphanumeric
!

 

Any clues?

 

Thanks,

DA

___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

[OSL | CCIE_Voice] Please help me !!!!!!!!!!

2010-10-05 Thread Akbar Ali

Dear all ,

Can any body help me solving my problem , its not working what to do ,
scenario is as following

primary link ethernet ,

secondary link isdn

static routing

intersting traffic cannot be generated from branch

i need to use isdn interface as soon as ehternet mpls (prmary ) link goes
down.

uff do know what to do.

Regards
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

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[OSL | CCIE_Voice] Vol2 Lab2 5.1 - 4digit GK calls

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Re: [OSL | CCIE_Voice] Vol2 Lab2 5.1 - 4digit GK calls

Re: [OSL | CCIE_Voice] Vol2 Lab2 5.1 - 4digit GK calls

Re: [OSL | CCIE_Voice] Vol2 Lab2 5.1 - 4digit GK calls

[OSL | CCIE_Voice] Order for arranging resources in MRGL

Re: [OSL | CCIE_Voice] Vol2 Lab2 5.1 - 4digit GK calls

Re: [OSL | CCIE_Voice] Vol2 Lab2 5.1 - 4digit GK calls

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Re: [OSL | CCIE_Voice] Vol2 Lab2 5.1 - 4digit GK calls

Re: [OSL | CCIE_Voice] Vol2 Lab2 5.1 - 4digit GK calls

Re: [OSL | CCIE_Voice] Vol2 Lab2 5.1 - 4digit GK calls

Re: [OSL | CCIE_Voice] BACD drop through not working on 2821

Re: [OSL | CCIE_Voice] Vol2 Lab2 5.1 - 4digit GK calls

Re: [OSL | CCIE_Voice] BACD drop through not working on 2821

Re: [OSL | CCIE_Voice] BACD drop through not working on 2821

Re: [OSL | CCIE_Voice] Vol2 Lab2 5.1 - 4digit GK calls

[OSL | CCIE_Voice] BACD drop through not working on 2821

Re: [OSL | CCIE_Voice] Vol2 Lab2 5.1 - 4digit GK calls

Re: [OSL | CCIE_Voice] Vol2 Lab2 5.1 - 4digit GK calls

[OSL | CCIE_Voice] Please help me !!!!!!!!!!

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