Re: [OSL | CCIE_Voice] Triggering +

2011-10-27 Thread Adam Frankel (afrankel) 

In 8.5 you can dial a + by holding *.

Adam



*From:* Farkas Péter 
*Sent:* Thu, Oct 27, 2011 11:03:31 AM
*To:* Emanuel Damasceno 
*CC:* ccie_voice@onlinestudylist.com
*Subject:* Re: [OSL | CCIE_Voice] Triggering +

From IP phones you cannot dial the "+" only CUPC and CIPC can support.

Peter

- Original Message -
From: Emanuel Damasceno
Date: Thursday, October 27, 2011 4:41 pm
Subject: [OSL | CCIE_Voice] Triggering +
To: ccie_voice@onlinestudylist.com



 Hello Experts,

  I am on Lab 5A and I am now wondering how I trigger the + on the phones. I
  was told that 7940s don't support + dial, but I don't want to know that, I
  want to know what I need to press so it becomes a +. On my cell phone, I
  press 0 and hold it. Is it the same with Cisco Phones?

  Thanks
  *Antonio Emanuel Damasceno*
  CCNA, CCNA Voice, CCNP Voice, CCIE Voice (written)
  CompTIA Network+
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ccie_voice@onlinestudylist.com

2011-10-01 Thread Adam Frankel (afrankel) 
The only issue you may run into is if your transcoder is picked as a 
Media Termination inadvertently.  For example, if an MTP is required for 
G729-G729 media termination, MRM will still allocate the transcoder, but 
the ORC-ACK will fail on the second leg.  In the same MRG, MRM load 
balances between the two, every other call in this scenario will fail. 
This is until universal transcoders are supported with 7.1(5).


A transcoder is an MTP, but not all MTPs are transcoders.   For this 
reason I put the transcoder in an MRG below the MTP, if an MTP is needed 
it will use the MTP in the first MRG, and if a transcoding resources are 
required MRM will not use the MTP, and use the XCoder in the next MRG.


Depending on requirements this may or may not be necessary.

HTH,

Adam




*From:* Mohd Baqari 
*Sent:* Fri, Sep 30, 2011 1:35:30 PM
*To:* eng_firasoq...@yahoo.com 
*CC:* "ccie_voice@onlinestudylist.com" 
*Subject:* Re: [OSL | CCIE_Voice] MRG&MRGL

No need for 3 mrgs. Create one mrg and put all 3 resources into it. 
Order doesnt matter since those resources are of different types.


Regards,
Mohammed Al Baqari

Sent from my iPhone

On Sep 30, 2011, at 5:14 PM, "eng_firasoq...@yahoo.com 
" > wrote:



Dear All,
 If we have to configure MTP,Xcoder,hd_Conference in the HQ site then 
we will configure 3 MRG MTP_MRG,Xcoder_MRG,conf_MRG after that we will 
create MRGL for hd_hq_MRGL but now what is correct order for MRG 
inside the MRGL

Regards

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Re: [OSL | CCIE_Voice] background image not working..

2011-09-22 Thread Adam Frankel (afrankel) 

What type of phone are you using? For 7965 it is 320x212x16:

R3#more flash:Desktops/320x212x16/List.xml

URL="TFTP:Desktops/320x212x16/NYY1_lrg.png"/>




HTH,

Adam



Original Message--
From: Ray 
Sent: Thu, Sep 22, 2011 11:40:52 Am
To: ccie voice 
CC:
Subject: [OSL | CCIE_Voice] background image not working..
has anyone done background image and it works..  when i go to user 
preferenc>backgroundimage>  it says selection unavailable.  help!
i done know if it is my image.. or what. if anyone has an image that 
works please forward to me.. I created my in windows 7 paint program 
and save it as png file.

br2#
br2#more flash:/Desktops/320x212x12/List.xml




br2#

tftp-server flash:/Desktops/320x212x12/List.xml
tftp-server flash:flash:/Desktops/320x212x12/voicesmall.png
tftp-server flash:flash:/Desktops/320x212x12/TN-voiceslarge.png


br2#sho flash

 0 Nov 17 2009 22:44:36 Desktops/320x212x12
30  131470 Nov 17 2009 22:44:38 Desktops/320x212x12/CampusNight.png
31   80565 Nov 17 2009 22:44:40 Desktops/320x212x12/CiscoFountain.png
328156 Nov 17 2009 22:44:40 Desktops/320x212x12/CiscoLogo.png
33  138278 Nov 17 2009 22:44:40 Desktops/320x212x12/Fountain.png
34 165 Sep 21 2011 15:17:18 Desktops/320x212x12/List.xml
35  109076 Nov 17 2009 22:44:42 Desktops/320x212x12/MorroRock.png
36  108087 Nov 17 2009 22:44:44 
Desktops/320x212x12/NantucketFlowers.png

37   10820 Nov 17 2009 22:44:44 Desktops/320x212x12/TN-CampusNight.png
389657 Nov 17 2009 22:44:46 
Desktops/320x212x12/TN-CiscoFountain.png

392089 Nov 17 2009 22:44:46 Desktops/320x212x12/TN-CiscoLogo.png
407953 Nov 17 2009 22:44:46 Desktops/320x212x12/TN-Fountain.png
417274 Nov 17 2009 22:44:46 Desktops/320x212x12/TN-MorroRock.png
429933 Nov 17 2009 22:44:48 
Desktops/320x212x12/TN-NantucketFlowers.png

434651 Sep 21 2011 15:15:04 Desktops/320x212x12/TN-voiceslarge.png
443961 Sep 21 2011 15:15:54 Desktops/320x212x12/voicesmall.png


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Re: [OSL | CCIE_Voice] how to retrieve list of Active IP Phones ONLY...

2011-06-23 Thread Adam Frankel (afrankel) 


Device Search in RTMT.

HTH,

Adam



Original Message--
From: Rashid Khan 
Sent: Thu, Jun 23, 2011 7:18:14 Am
To: ccie voice 
CC:
Subject: [OSL | CCIE_Voice] how to retrieve list of Active IP Phones 
ONLY...

Dear Team,
I want to know is there any way to findout a list of Currently active 
or registed IP Phones with Call Manager.
Oneway to do this is, write nothing in Text Box and press Find 
button, when I do this I also see non active devices Or the devices 
whose Status is Not found also appearing.

I only want list of phones which are Active or working currently,
Regards
Rashid


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Re: [OSL | CCIE_Voice] sip trunk delay offer

2011-06-22 Thread Adam Frankel (afrankel) 


G729 early offer was introduced in CUCM 7.0, an IOS G.729 capable MTP is 
required. In 8.5 and later CUCM can do early offer with multiple codecs 
by checking the "Early Offer support for voice and video calls" checkbox 
on the SIP Profile.   This was made possible by the addition of the 
StationPortReq,StationPortRes, and StationPortClose message being added 
to SCCPv20.


Regards,

Adam



Original Message--
From: Farkas Péter 
Sent: Wed, Jun 22, 2011 3:00:33 Am
To: Adam Frankel (afrankel) 
CC: wormh...@sch.hu, donny f , 
ccie_voice@onlinestudylist.com

Subject: Re: [OSL | CCIE_Voice] sip trunk delay offer

You are right, I thought early offer when it is the case.

- Original Message -----
From: "Adam Frankel (afrankel)"
Date: Wednesday, June 22, 2011 12:22 am
Subject: Re: [OSL | CCIE_Voice] sip trunk delay offer
To: wormh...@sch.hu, Farkas Péter
Cc: donny f, ccie_voice@onlinestudylist.com


>   This is false.  If region configuration is set to G729 and the calling
>   device supports G729, CUCM can select G729 in its answer, if it is offered.
>
>   Adam
>
>
>   Original Message--
>   From: Farkas Péter
>   Sent: Tue, Jun 21, 2011 3:32:21 Am
>   To: donny f
>   CC: ccie_voice@onlinestudylist.com
>   Subject: Re: [OSL | CCIE_Voice] sip trunk delay offer
>   >  Have you introduce MTP to the call? By default CUCM only capable 
outgoing delay offer based
>  on G.711.
>   >
>   >  Peter
>   >
>   >  - Original Message -
>   >  From: donny f
>   >  Date: Tuesday, June 21, 2011 1:09 am
>   >  Subject: [OSL | CCIE_Voice] sip trunk delay offer
>   >  To: ccie_voice@onlinestudylist.com
>   >
>   >
>   >  >   hi,
>   >  >
>   >  >
>   >  >anybody know why i did not see the G729  in debug ?  it only said 
PCMU in
>   >  >codec
>   >  >
>   >  >v=0
>   >  >o=CiscoSystemsSIP-GW-UserAgent 6036 5234 IN IP4 10.20.100.2
>   >  >s=SIP Call
>   >  >c=IN IP4 10.20.100.2
>   >  >t=0 0
>   >  >m=audio 16522 RTP/AVP 0
>   >  >c=IN IP4 10.20.100.2
>   >  >a=rtpmap:0 PCMU/8000->codec
>   >  >a=ptime:20
>   >  >   ___
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please visit www.ipexpert.com
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>   >  >Are you a CCNP or CCIE and looking for a job? Check out 
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Re: [OSL | CCIE_Voice] sip trunk delay offer

2011-06-21 Thread Adam Frankel (afrankel) 


This is false.  If region configuration is set to G729 and the calling 
device supports G729, CUCM can select G729 in its answer, if it is offered.


Adam


Original Message--
From: Farkas Péter 
Sent: Tue, Jun 21, 2011 3:32:21 Am
To: donny f 
CC: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] sip trunk delay offer

Have you introduce MTP to the call? By default CUCM only capable outgoing delay 
offer based on G.711.

Peter

- Original Message -
From: donny f
Date: Tuesday, June 21, 2011 1:09 am
Subject: [OSL | CCIE_Voice] sip trunk delay offer
To: ccie_voice@onlinestudylist.com


>  hi,
>
>
>   anybody know why i did not see the G729  in debug ?  it only said PCMU in
>   codec
>
>   v=0
>   o=CiscoSystemsSIP-GW-UserAgent 6036 5234 IN IP4 10.20.100.2
>   s=SIP Call
>   c=IN IP4 10.20.100.2
>   t=0 0
>   m=audio 16522 RTP/AVP 0
>   c=IN IP4 10.20.100.2
>   a=rtpmap:0 PCMU/8000->   codec
>   a=ptime:20
>  ___
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visit www.ipexpert.com
>
>   Are you a CCNP or CCIE and looking for a job? Check out 
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Re: [OSL | CCIE_Voice] how to verify class-based traffic shape?

2011-06-15 Thread Adam Frankel (afrankel) 


Do you have "frame-relay trafic-shaping" on the physical serial 
interface? (not the sub-interface)


Adam



Original Message--
From: Greg 
Sent: Wed, Jun 15, 2011 3:11:00 Pm
To: ccie_voice@onlinestudylist.com
CC:
Subject: [OSL | CCIE_Voice] how to verify class-based traffic shape?

Per SRND, I configured FRF.12 as below:
class-map match-any signal
 match  dscp cs3
 match  dscp af31
class-map match-any voice
 match  dscp ef
!
policy-map wan-edge
 class voice
priority percent 33
 class signal
bandwidth percent 5
 class class-default
fair-queue
!
policy-map shape384
 class class-default
shape average 364800 3648 0
  service-policy wan-edge
!
interface Serial0/1/0.102 point-to-point
 bandwidth 384
 frame-relay interface-dlci 102
  class map384
What command we would use to verify the shaping?  I tried command 
"show traffic-shape" but got nothing.  "show policy-map interface" 
didn't give statistics on shaping.

Thanks!


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Re: [OSL | CCIE_Voice] DHCP Debug in CUCM

2011-06-03 Thread Adam Frankel (afrankel) 


  
  
Packet
capture on CUCM CLI:
https://supportforums.cisco.com/docs/DOC-11599

Not really feasible for lab purposes IMO.

You can also see the DHCP activity in the UCM syslog:

 file tail activelog syslog/messages

HTH,
  
  Adam Frankel
Customer Support Engineer
Unified Communications Infrastructure
+1,919-392-7936 9am-5pm EDT (GMT -4)
CCVP, RHCE, CUP Specialist
Onshift Weekend Office Hours: 8am-1:30pm EDT
  
  
Original Message--
From: Miron Kobelski 
Sent: Fri, Jun 03, 2011 11:51:03 Am
To: Friderich Claude 
CC: CCIE VOICE 
Subject: Re: [OSL | CCIE_Voice] DHCP Debug in CUCM
  Much better solution, thanks! :)

On Fri, Jun 3, 2011 at 11:52, Friderich
  Claude 
  wrote:
  

  
Something
very fast and easy ;-)
Debug ip udp
on the hq router or br1 router depending for which
site dhcp pub is asking for …..
 
 
 

  

  

  
  
 
 
  
  
Claude FRIDERICH 
PreSales Voice over IP 
  
  
 

 CCIE Voice #29035
  


  
Office
  
  
+352 31 33
80 – 407
  


  
Mobile 
  
  
+352 621 303 616
  


  

  
  
Fax  
  
  
+352
  31 33 80 412
  


  
Email
  
  
cfrider...@netcore.lu
  

  

 
 

  From: ccie_voice-boun...@onlinestudylist.com
  [mailto:ccie_voice-boun...@onlinestudylist.com]
  On Behalf Of Miron
  Kobelski
  Sent: vendredi 3 juin 2011 08:03
  To: Cristobal Priego
  Cc: CCIE VOICE
  Subject: Re: [OSL | CCIE_Voice] DHCP Debug
  in CUCM


  
 
Utils network capture

  On Jun 3, 2011 1:55 AM,
"Cristobal Priego" 
wrote:
> how do you do the packet capture from the
cucm cli ?
> 
> 2011/6/2 Miron Kobelski 
> 
>> Hi,
>>
>> I was unable to find anything better
then packet capture on CUCM CLI -
it
>> allows you to see if the DHCP messages
hit the CUCM and are sent back,
but
>> not much more ;)
>>
>> best regards
>>
>> On Thu, Jun 2, 2011 at 16:45, John
Anzar 
wrote:
>>
>>>
>>> Hi Guys
>>>
>>> I have a very basic question about
DHCP. most of the time this
works with
>>> out any issue.
>>>
>>> Is there anyway that we can DHCP
debug in CUCM ?? Can we get the
mac biind
>>> with the IP form CUCM ?
>>>
>>> Best

Re: [OSL | CCIE_Voice] [Vol2 Lab3 2.8]LRQ suspension point failed (return code = 0x4009)

2011-05-26 Thread Adam Frankel (afrankel) 

Looks like there is a timegap there, did you get an LRJ from Remote
Gateekeeper? Try adding "debug h225 asn1" to your debugs.

Regards,

Adam 



Original Message--
From: Satoshi Funabashi (sfunabas) 
Sent: Thu, May 26, 2011 9:29:54 Am
To: ccie_voice@onlinestudylist.com
CC:
Subject: [OSL | CCIE_Voice] [Vol2 Lab3 2.8]LRQ suspension point failed
(return code = 0x4009)
>
> Hi,
>
> On Vol2 Lab3 2.8, when I call from HQ Phone2 to +916745738932,
>
> I got following message on HQ Router with "debug gatekeeper main 5"
> and the call failed.
>
> R1#
>
> May 26 12:10:57.681: ////GK/gk_rassrv_arq:
> arqp=0x461B5FE8,crv=0x1, answerCall=0
>
> May 26 12:10:57.681: //0064F4340100/0064F4340100/GK/gk_dns_query: No
> Name servers
>
> May 26 12:10:57.681:
> //0064F4340100/0064F4340100/GK/rassrv_get_addrinfo: (1#916745738932)
> Matched tech-prefix 1#
>
> May 26 12:10:57.685:
> //0064F4340100/0064F4340100/GK/rassrv_get_addrinfo: (1#916745738932)
> Matched zone prefix 91 and remainder 6745738932
>
> May 26 12:10:57.685:
> //0064F4340100/0064F4340100/GK/rassrv_arq_select_viazone: about to
> check the source side, src_zonep=0x48206C2C
>
> May 26 12:10:57.685:
> //0064F4340100/0064F4340100/GK/rassrv_arq_select_viazone: matched zone
> is UCM, and z_invianamelen=0
>
> May 26 12:10:57.685:
> //0064F4340100/0064F4340100/GK/rassrv_arq_select_viazone: about to
> check the destination side, dst_zonep=0x47F90A4C
>
> May 26 12:10:57.685:
> //0064F4340100/0064F4340100/GK/rassrv_arq_select_viazone: matched zone
> is PSTN-WAN, and z_outvianamelen=0
>
> May 26 12:10:57.685:
> //0064F4340100/0064F4340100/GK/rassrv_put_remote_zones_from_zone_list:
> zone PSTN-WAN
>
> *May 26 12:11:03.685:
> //0064F4340100/0064F4340100/GK/gk_rassrv_sep_arq: LRQ suspension point
> failed (return code = 0x4009)*
>
> * *
>
> Does anyone know how to resolve this issue?
>
> Following is the HQ Router gatekeeper configuration.
>
> gatekeeper
>
> zone local UCM ipexpert.com 142.51.1.64
>
> zone remote PSTN-WAN ipexpert.com 142.51.1.63 1719
>
> zone prefix PSTN-WAN 01191*
>
> zone prefix PSTN-WAN 91*
>
> gw-type-prefix 1#* default-technology
>
> no shutdown
>
> !
>
> If you need more information, please let me know.
>
> Thanks and Regards,
>
> Satoshi
>
> 
>
> Satoshi Funabashi
>
> Systems Engineer
>
> Cisco Systems G.K.
>
> Tel:81-3-6434-2824(direct)
>
> 　 81-3-6434-6500(group)
>
> 81-90-4050-1574(mobile)
>
> E-mail: sfuna...@cisco.com
>
> 
>
>
> ___
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> visit www.ipexpert.com
>
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> www.PlatinumPlacement.com
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Re: [OSL | CCIE_Voice] MGCP GW will not register in CM -- Unknown

2011-05-17 Thread Adam Frankel (afrankel) 


Are you saying it is showing up in CUCM as not registered?

From the CUCM side:

1.  Check replication
2.  Restart Cisco RisData Collector on all nodes
3.  Restart CIsco CallManager on all nodes

HTH,

Adam



Original Message--
From: Rrcrumm 
Sent: Tue, May 17, 2011 2:32:58 Am
To: Shrini 
CC: "ccie_voice@onlinestudylist.com" 
Subject: Re: [OSL | CCIE_Voice] MGCP GW will not register in CM -- Unknown
Thanks for the ideas.  Not on the rack but I'll look back at these 
emails if I have a problem


Randall
Sent from my iPhone

On May 16, 2011, at 3:51 PM, Shrini > wrote:



Hi Randall,

I am sure you might have checked but what does "show controller T1" 
show for 0/0/0
Also just try adding isdn-protocol-emulate network under s0/0/0:23, 
there is a possibility PSTN router is configured user side.


-Shrini

On 5/16/2011 2:37 PM, Hough, Earl wrote:
On May 16, 2011 3:48 PM, "Randall Crumm" 
> wrote:

> HI,
>
> Having an issue in Call Manager. My br1 config looks good just 
can't get

> it to register in call manager.
>
>
>
> Any ideas? I've reloaded the router, deleted the GW and recreated 
it. No

> mgcp and mgcp. Etc...
>
>
>
> Here are snipit of the config and CUCM page
> 

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Re: [OSL | CCIE_Voice] Basic question on 911 RG config - Low importance question

2011-04-19 Thread Adam Frankel (afrankel) 


Standard Local Route Group is a more scalable solution for supporting 
multiple sites.


HTH,

Adam


Original Message--
From: Shrini 
Sent: Tue, Apr 19, 2011 2:07:00 Am
To: ccie_voice@onlinestudylist.com 
CC:
Subject: [OSL | CCIE_Voice] Basic question on 911 RG config - Low 
importancequestion
We can configure route list for emergency calling either by assigning 
specific RG or we can configure using Standard local route group 
(which will bind the device to use local GW for emergency calls) IMO 
Standard LRG is best option and recommended.


One binds to DP and other bind to CSS which one is best.

Wanted to know others opinions.


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Re: [OSL | CCIE_Voice] AAR with TEHO

2011-04-17 Thread Adam Frankel (afrankel) 


TEHO can have a backup route group which sends call out the local PSTN 
gateway. Since "Stop Routing on Out of Bandwidth Flag" service parameter 
is set to false by default, it should route to the backup local gateway.


HTH,
Adam

Original Message--
From: voice boy 
Sent: Sun, Apr 17, 2011 12:31:50 Am
To: OSL Questions 
CC:
Subject: [OSL | CCIE_Voice] AAR with TEHO

Hi,

I see if we have + dialing, we can do AAR with only one group

Only the TEHO calls fail to go through WAN


For example,,

We allow 5-calls for HQ-BR1 WAN
and a 6th call: HQ phone calling BR1 PSTN through WAN

and the WAN bw is allready have the allowed 5-calls
How can I let the TEHO call to count from the number of allowed calls  
[I think I should assign location to the gateways] , and if the 
bandwidth is not available and the call will be re-routed , how the 
calling phone know the new called number that the other site gw not 
have EPNM as the phones


Or the call will fail that I think TEHO isn't supported with AAR


Thanks & Best Regards


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Re: [OSL | CCIE_Voice] Hardware conference is stuck in state "TCP_CONN_ERROR"

2011-03-29 Thread Adam Frankel (afrankel) 


Sounds like you are having a TCP Connectivity issue with the CUCM.

Is the Cisco CallManager Service started on all nodes?

Can you telnet to port 2000 on all nodes?

Are the IP addresses correct?

Have you tried removing any QOS config?

Regards,

Adam



Original Message--
From: Rahul Kapor 
Sent: Tue, Mar 29, 2011 11:32:19 Am
To: Rogers Ochieng 
CC: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] Hardware conference is stuck in state   
 "TCP_CONN_ERROR"

Hi All,

I did reset , reload , no sccp , sccp but did not help. DIf f conf 
bridge from diff router can be registered with CUCM so looks like , 
PVDM has gone into hung state.

Is there any way to reset the PVDM ?

th,
Rahul

On Tue, Mar 29, 2011 at 8:30 PM, Rogers Ochieng 
mailto:rogersochi...@gmail.com>> wrote:


Since i think this is CUCM with SRST scenario. Try this
sccp local GigabitEthernet0/0
sccp ccm 14.160.115.200 identifier 1 version 7.0
sccp ccm 14.160.116.40 identifier 2 version 7.0
sccp ccm 14.160.110.22 identifier 3 version 7.0
sccp
sccp ccm group 1
associate ccm 1 priority 1
associate ccm 2 priority 2
associate ccm 3 priority 3
associate profile 1 register BR1-CONF
dspfarm profile 1 conference
 codec g711ulaw
 codec g729r8

 maximum sessions 1
 conference-join custom-cptone join
 conference-leave custom-cptone leave
 associate application SCCP
!
telephony-service
sdspfarm units 1
sdspfarm tag 1 BR1-CONF
conference hardware
ephone-dn 10 octo-line
number B
conference meet-me
no huntstop

On 29 March 2011 14:31, Rahul Kapor mailto:rahul.kapo...@gmail.com>> wrote:

Hi All ,

thanks for replying.
here is my sccp config

sccp local GigabitEthernet0/0
sccp ccm 14.160.115.200 identifier 1 version 7.0
sccp ccm 14.160.116.40 identifier 2 version 7.0
sccp ccm 14.160.110.22 identifier 3 version 7.0
sccp
sccp ccm group 1
 associate ccm 3 priority 1
 associate profile 1 register BR1-CONF

dspfarm profile 1 conference
 codec g711ulaw
 codec g711alaw
 codec g729ar8
 codec g729abr8
 codec g729r8
 codec g729br8
 maximum sessions 1
 conference-join custom-cptone join
 conference-leave custom-cptone leave
 associate application SCCP
!

I am adding Hardware conf in CUCM as BR1-CONF but status is
unknown-unknown.
sh sccp
SCCP Admin State: UP
Gateway Local Interface: GigabitEthernet0/0
IPv4 Address: 14.160.110.35
Port Number: 2000
IP Precedence: 5
User Masked Codec list: None
Call Manager: 14.160.115.200, Port Number: 2000
Priority: N/A, Version: 7.0, Identifier: 1
Trustpoint: N/A
Call Manager: 14.160.116.40, Port Number: 2000
Priority: N/A, Version: 7.0, Identifier: 2
Trustpoint: N/A
Call Manager: 14.160.110.22, Port Number: 2000
Priority: N/A, Version: 7.0, Identifier: 3
Trustpoint: N/A


Conferencing Oper State: ACTIVE_IN_PROGRESS - Cause Code:
TCP_CONN_ERROR
Active Call Manager: NONE
TCP Link Status: CONNECT_PENDING, Profile Identifier: 1
Reported Max Streams: 8, Reported Max OOS Streams: 0
Supported Codec: g711ulaw, Maximum Packetization Period: 30
Supported Codec: g711alaw, Maximum Packetization Period: 30
Supported Codec: g729ar8, Maximum Packetization Period: 60
Supported Codec: g729abr8, Maximum Packetization Period: 60
Supported Codec: g729r8, Maximum Packetization Period: 60
Supported Codec: g729br8, Maximum Packetization Period: 60
Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30
Supported Codec: rfc2833 pass-thru, Maximum Packetization
Period: 30
Supported Codec: inband-dtmf to rfc2833 conversion, Maximum
Packetization Period: 30

reloaded the router again but no help.

thx,
Rahul





On Tue, Mar 29, 2011 at 12:28 PM, Rogers Ochieng
mailto:rogersochi...@gmail.com>> wrote:

Your sccp and telephony service config will help

On 28 March 2011 18:57, Rahul Kapor
mailto:rahul.kapo...@gmail.com>>
wrote:


Hi all ,

I am trying to register my hardware conference to
cucm/cme   but it is stuck in "TCP_CONN_ERROR" error
state.

sh sccp
Conferencing Oper State: ACTIVE_IN_PROGRESS - Cause
Code: TCP_CONN_ERROR
Active Call Manager: NONE
TCP Link Status: CONNECT_PENDING, Profile Identifier: 1

Re: [OSL | CCIE_Voice] A 2nd Corporate Directory

2011-03-29 Thread Adam Frankel (afrankel) 


If you write a custom corporate directory server using the IP Phone 
Services SDK (developer.cisco.com) it would be possible.


Adam



Original Message--
From: Bo Gao 
Sent: Mon, Mar 28, 2011 8:35:25 Pm
To: OSL 
CC:
Subject: [OSL | CCIE_Voice] A 2nd Corporate Directory

Hi all,

Is it possible to set up a Secondary corporate directory as an IP 
Phone service that coexist with the primary corporate directory in 
CUCM 7 or CUCM 8?


Thank you,


Bo


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Re: [OSL | CCIE_Voice] CME phone hears beep-on-hold on PSTN call

2011-03-20 Thread Adam Frankel (afrankel) 


If the PSTN phone is putting the call on hold, its the responsibility of 
the PSTN service provider (or party on the far end) to play the MOH.


Regards,

Adam


Original Message--
From: Michael Luo 
Sent: Sun, Mar 20, 2011 10:34:46 Am
To: ccie_voice@onlinestudylist.com
CC:
Subject: [OSL | CCIE_Voice] CME phone hears beep-on-hold on PSTN call
I was trying to configure a CME router to stream music on hold to its 
e-phones.  The CME router is also the voice gateway to PSTN.


Here are the results:

Scenario #1 Calls between e-phones.  e-phones can hear MOH.

Scenario #2 Calls between e-phone and PSTN phones.  PSTN phones can 
hear MOH.


Scenario #3 Calls between e-phone and PSTN phones.  e-phones hear 
beep-on-hold (beep-beep-beep).


I was trying to troubleshoot Scenario #3.

Here are the configure highlights:

=
ip multicast-routing
!
interface vlan502
 description e-phone vlan
 ip address 142.102.66.254 255.255.255.0
 ip pim dense

interface Loopback0
 ip address 142.1.66.254 255.255.255.255

ccm-manager music-on-hold

telephony-service
 ip source-address 142.102.66.254 port 2000
 moh music-on-hold.au
 multicast moh 239.1.1.1 port 16384 route 142.1.66.254 142.102.66.254
=

Attached are the full router config and debug output.

Any help is appreciated!

Michael


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[OSL | CCIE_Voice] No Transformations applied on Internal Mobile Connect Call

2011-03-14 Thread Adam Frankel (afrankel) 


Just in case you are using a 7.1 version of CUCM for study purposes be 
aware of CSCsx01971.


This defect will cause calling party transformations on route patterns 
and translation patterns to not be applied on Mobile Connect calls.  
This defect is fixed in 7.1(2).


HTH,

--
Adam

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Re: [OSL | CCIE_Voice] CME Hunting - not getting Original Called party number

2011-03-12 Thread Adam Frankel (afrankel) 


Since it's not a CCIE Study question you might want to try the 
cisco-voip mailer (cisco-v...@puck.nether.net) or Cisco Support Forums 
(https://supportforums.cisco.com/community/netpro/collaboration-voice-video/ip-telephony).


Adam



Original Message--
From: Ashar Siddiqui 
Sent: Sat, Mar 12, 2011 10:56:43 Am
To: ccie_voice@onlinestudylist.com 
CC:
Subject: Re: [OSL | CCIE_Voice] CME Hunting - not getting Original 
Calledparty number

Common guys, no one know this answer of this?
Come out of CCIE lab questions for a change (we had enough of them 
already) and see if you can find a way out of this one?


Ash

On 11/03/2011 22:15, Ashar Siddiqui wrote:

Hello,

I would be glad if someone can help in this. I am working on a CME 
hunting setup for a customer where when a call is not answered/busy 
on any phone, it will start ringing a hubt listcustomer has 
reported that although hunting is working perfectly fine but the 
problem is when it starts hunting and a member of that hunt group 
picks up the call, on their phone it does not display who was the 
Original called party instead it shows a name of someone in the hunt 
list.


For example...an external call comes for extension Joe Bloggs 
ext.2878  it doesn't  get answered in 10 secs...it starts ringing 
the pilot 3504 and huntig starts...let say ext 2920 picks up the 
phone...on her phone..she doesn't get that the call was for Joe 
Bloggs instead she get some other name...Is there a possibility of 
showing who was the original called-party?




ephone-dn  1  octo-line
call-forward busy 3504
call-forward noan 3504 timeout 10
!
!
ephone-dn  2  octo-line

  call-forward busy 3504
  call-forward noan 3504 timeout 10

!
!
ephone-dn  3  octo-line
   call-forward busy 3504
  call-forward noan 3504 timeout 10
!
!
ephone-dn  4  octo-line

  call-forward busy 3504
  call-forward noan 3504 timeout 10

!   
!

ephone-hunt 4 peer
  pilot 3504
  list 2934, 2932, 2912, 2913, 2920, 2924, 2918, 2926, 2936, 2935, 2917, 2919
  final 8000
  preference 1
  timeout 12
  hops 15
  no-reg pilot
!



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