Re: [OSL | CCIE_Voice] CCM to CME calls keeps ringing

[Vik Malhi]

Uncheck it for whenever you invoke an IPIPGW or CUBE. That means a call from
CCM -> BACD and also CCM -> CUE.
-- 
Vik Malhi ­ CCIE #13890, CCSI #31584
Senior Technical Instructor - IPexpert, Inc.

Telephone: +1.810.326.1444
Fax: +1.810.454.0130
Mailto: vma...@ipexpert.com


Join our free online support and peer group communities:
http://www.IPexpert.com/communities
IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand
and Audio Certification Training Tools for the Cisco CCIE R&S Lab, CCIE
Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage
Lab Certifications.








From: Onur Tufekci 
Date: Thu, 30 Apr 2009 19:52:54 -0400
To: Vik Malhi 
Cc: Cliff McGlamry , OSL Group

Subject: Re: [OSL | CCIE_Voice] CCM to CME calls keeps ringing

> WOW That is scary. VIK you are the man. I never had that problem before.
> So do we have to uncheckWait for Far End H.245 Terminal Capability Set for all
> the GK trunks or just IPIPGW?

Re: [OSL | CCIE_Voice] CCM to CME calls keeps ringing

[Onur Tufekci]

Looks like you just need to change it for the IPIPGW.

On Wed, Apr 29, 2009 at 6:42 PM, Onur Tufekci  wrote:

> WOW That is scary. VIK you are the man. I never had that problem
> before. So do we have to uncheckWait for Far End H.245 Terminal Capability
> Set for all the GK trunks or just IPIPGW?
>
>
>
>  On Wed, Apr 29, 2009 at 1:51 PM, Vik Malhi  wrote:
>
>> Also look into the trunk settings- Wait for H245 TCS should not be
>> checked.
>> --
>> Vik Malhi – CCIE #13890, CCSI #31584
>> Senior Technical Instructor - IPexpert, Inc.
>>
>> Telephone: +1.810.326.1444
>> Fax: +1.810.454.0130
>> Mailto: *vma...@ipexpert.com
>>
>> *
>> Join our free online support and peer group communities:
>> *http://www.IPexpert.com/communities<http://www.ipexpert.com/communities>
>> *IPexpert - The Global Leader in Self-Study, Classroom-Based,
>> Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE
>> R&S Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and
>> CCIE Storage Lab Certifications.
>>
>>
>>
>>
>>
>>
>>
>> ------------------
>> *From: *Cliff McGlamry 
>> *Date: *Tue, 28 Apr 2009 23:29:53 -0400
>> *To: *Onur Tufekci , OSL Group <
>> ccie_voice@onlinestudylist.com>
>> *Subject: *Re: [OSL | CCIE_Voice] CCM to CME calls keeps ringing
>>
>> Your issue is probably not on the Gatekeeper.  Your configuration looks
>> okay as far as I can tell.
>>
>> Do a debug dial peer on the CME router and see which dial peer you are
>> coming in on.  My guess is that you're coming in on a dial peer that doesn't
>> have the codec defined correctly, or possibly coming in on the default dial
>> peer (which is always a not so good thing to have happen).
>>
>> I'm betting that the issue is the dial peer you're hitting inbound on CME
>> is either the wrong dial peer, or it's misconfigured.
>>
>>
>>
>> - Original Message -
>>
>> *From:*  Onur  Tufekci <mailto:onurvc...@gmail.com >
>>
>>
>> *To:* ccie_voice@onlinestudylist.com
>>
>> *Sent:* Tuesday, April 28, 2009 11:13  PM
>>
>> *Subject:* [OSL | CCIE_Voice] CCM to CME  calls keeps ringing
>>
>>
>>
>> I have the IPIPGW configured on the HQ Router. Calls from CME to CCM are
>>  successful but other way is not. CCM phone just keeps ringing even after
>>  picking up the call at CME phone.
>>
>>
>>
>> Is this even a valid configuration?
>>
>>
>>
>

Re: [OSL | CCIE_Voice] CCM to CME calls keeps ringing

[Onur Tufekci]

WOW That is scary. VIK you are the man. I never had that problem before.
So do we have to uncheckWait for Far End H.245 Terminal Capability Set for
all the GK trunks or just IPIPGW?



On Wed, Apr 29, 2009 at 1:51 PM, Vik Malhi  wrote:

> Also look into the trunk settings- Wait for H245 TCS should not be checked.
> --
> Vik Malhi – CCIE #13890, CCSI #31584
> Senior Technical Instructor - IPexpert, Inc.
>
> Telephone: +1.810.326.1444
> Fax: +1.810.454.0130
> Mailto: *vma...@ipexpert.com
>
> *
> Join our free online support and peer group communities:
> *http://www.IPexpert.com/communities <http://www.ipexpert.com/communities>
> *IPexpert - The Global Leader in Self-Study, Classroom-Based,
> Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE
> R&S Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and
> CCIE Storage Lab Certifications.
>
>
>
>
>
>
>
> --
> *From: *Cliff McGlamry 
> *Date: *Tue, 28 Apr 2009 23:29:53 -0400
> *To: *Onur Tufekci , OSL Group <
> ccie_voice@onlinestudylist.com>
> *Subject: *Re: [OSL | CCIE_Voice] CCM to CME calls keeps ringing
>
> Your issue is probably not on the Gatekeeper.  Your configuration looks
> okay as far as I can tell.
>
> Do a debug dial peer on the CME router and see which dial peer you are
> coming in on.  My guess is that you're coming in on a dial peer that doesn't
> have the codec defined correctly, or possibly coming in on the default dial
> peer (which is always a not so good thing to have happen).
>
> I'm betting that the issue is the dial peer you're hitting inbound on CME
> is either the wrong dial peer, or it's misconfigured.
>
>
>
> - Original Message -
>
> *From:*  Onur  Tufekci <mailto:onurvc...@gmail.com >
>
>
> *To:* ccie_voice@onlinestudylist.com
>
> *Sent:* Tuesday, April 28, 2009 11:13  PM
>
> *Subject:* [OSL | CCIE_Voice] CCM to CME  calls keeps ringing
>
>
>
> I have the IPIPGW configured on the HQ Router. Calls from CME to CCM are
>  successful but other way is not. CCM phone just keeps ringing even after
>  picking up the call at CME phone.
>
>
>
> Is this even a valid configuration?
>
>
>

Re: [OSL | CCIE_Voice] CCM to CME calls keeps ringing

[Scott ODonnell]

Vik,

Is there any rule of thumb regarding how to configure the trunk via GK for
the lab?
I've seen the trunk configured as an ICT and also as h225-GK-controlled.
My understanding is the ICT "adds" some stuff that CM and CME use.
But given the known layout of the lab, should we always use one way over the
other?

- Scott



On Wed, Apr 29, 2009 at 1:59 PM, Onur Tufekci  wrote:

> Umm! I will check that too. Everything was in default setting when I set
> the trunk up.
>
>
> On Wed, Apr 29, 2009 at 1:51 PM, Vik Malhi  wrote:
>
>> Also look into the trunk settings- Wait for H245 TCS should not be
>> checked.
>> --
>> Vik Malhi – CCIE #13890, CCSI #31584
>> Senior Technical Instructor - IPexpert, Inc.
>>
>> Telephone: +1.810.326.1444
>> Fax: +1.810.454.0130
>> Mailto: *vma...@ipexpert.com
>>
>> *
>> Join our free online support and peer group communities:
>> *http://www.IPexpert.com/communities<http://www.ipexpert.com/communities>
>> *IPexpert - The Global Leader in Self-Study, Classroom-Based,
>> Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE
>> R&S Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and
>> CCIE Storage Lab Certifications.
>>
>>
>>
>>
>>
>>
>>
>> ------------------
>> *From: *Cliff McGlamry 
>> *Date: *Tue, 28 Apr 2009 23:29:53 -0400
>> *To: *Onur Tufekci , OSL Group <
>> ccie_voice@onlinestudylist.com>
>> *Subject: *Re: [OSL | CCIE_Voice] CCM to CME calls keeps ringing
>>
>> Your issue is probably not on the Gatekeeper.  Your configuration looks
>> okay as far as I can tell.
>>
>> Do a debug dial peer on the CME router and see which dial peer you are
>> coming in on.  My guess is that you're coming in on a dial peer that doesn't
>> have the codec defined correctly, or possibly coming in on the default dial
>> peer (which is always a not so good thing to have happen).
>>
>> I'm betting that the issue is the dial peer you're hitting inbound on CME
>> is either the wrong dial peer, or it's misconfigured.
>>
>>
>>
>> - Original Message -
>>
>> *From:*  Onur  Tufekci <mailto:onurvc...@gmail.com >
>>
>>
>> *To:* ccie_voice@onlinestudylist.com
>>
>> *Sent:* Tuesday, April 28, 2009 11:13  PM
>>
>> *Subject:* [OSL | CCIE_Voice] CCM to CME  calls keeps ringing
>>
>>
>>
>> I have the IPIPGW configured on the HQ Router. Calls from CME to CCM are
>>  successful but other way is not. CCM phone just keeps ringing even after
>>  picking up the call at CME phone.
>>
>>
>>
>> Is this even a valid configuration?
>>
>>
>>
>

Re: [OSL | CCIE_Voice] CCM to CME calls keeps ringing

[Onur Tufekci]

Umm! I will check that too. Everything was in default setting when I set the
trunk up.


On Wed, Apr 29, 2009 at 1:51 PM, Vik Malhi  wrote:

> Also look into the trunk settings- Wait for H245 TCS should not be checked.
> --
> Vik Malhi – CCIE #13890, CCSI #31584
> Senior Technical Instructor - IPexpert, Inc.
>
> Telephone: +1.810.326.1444
> Fax: +1.810.454.0130
> Mailto: *vma...@ipexpert.com
>
> *
> Join our free online support and peer group communities:
> *http://www.IPexpert.com/communities <http://www.ipexpert.com/communities>
> *IPexpert - The Global Leader in Self-Study, Classroom-Based,
> Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE
> R&S Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and
> CCIE Storage Lab Certifications.
>
>
>
>
>
>
>
> --
> *From: *Cliff McGlamry 
> *Date: *Tue, 28 Apr 2009 23:29:53 -0400
> *To: *Onur Tufekci , OSL Group <
> ccie_voice@onlinestudylist.com>
> *Subject: *Re: [OSL | CCIE_Voice] CCM to CME calls keeps ringing
>
> Your issue is probably not on the Gatekeeper.  Your configuration looks
> okay as far as I can tell.
>
> Do a debug dial peer on the CME router and see which dial peer you are
> coming in on.  My guess is that you're coming in on a dial peer that doesn't
> have the codec defined correctly, or possibly coming in on the default dial
> peer (which is always a not so good thing to have happen).
>
> I'm betting that the issue is the dial peer you're hitting inbound on CME
> is either the wrong dial peer, or it's misconfigured.
>
>
>
> - Original Message -
>
> *From:*  Onur  Tufekci <mailto:onurvc...@gmail.com >
>
>
> *To:* ccie_voice@onlinestudylist.com
>
> *Sent:* Tuesday, April 28, 2009 11:13  PM
>
> *Subject:* [OSL | CCIE_Voice] CCM to CME  calls keeps ringing
>
>
>
> I have the IPIPGW configured on the HQ Router. Calls from CME to CCM are
>  successful but other way is not. CCM phone just keeps ringing even after
>  picking up the call at CME phone.
>
>
>
> Is this even a valid configuration?
>
>
>

Re: [OSL | CCIE_Voice] CCM to CME calls keeps ringing

[Vik Malhi]

Also look into the trunk settings- Wait for H245 TCS should not be checked.
-- 
Vik Malhi ­ CCIE #13890, CCSI #31584
Senior Technical Instructor - IPexpert, Inc.

Telephone: +1.810.326.1444
Fax: +1.810.454.0130
Mailto: vma...@ipexpert.com


Join our free online support and peer group communities:
http://www.IPexpert.com/communities
IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand
and Audio Certification Training Tools for the Cisco CCIE R&S Lab, CCIE
Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage
Lab Certifications.








From: Cliff McGlamry 
Date: Tue, 28 Apr 2009 23:29:53 -0400
To: Onur Tufekci , OSL Group

Subject: Re: [OSL | CCIE_Voice] CCM to CME calls keeps ringing

Your issue is probably not on the Gatekeeper.  Your configuration looks okay
as far as I can tell.
 
Do a debug dial peer on the CME router and see which dial peer you are
coming in on.  My guess is that you're coming in on a dial peer that doesn't
have the codec defined correctly, or possibly coming in on the default dial
peer (which is always a not so good thing to have happen).
 
I'm betting that the issue is the dial peer you're hitting inbound on CME is
either the wrong dial peer, or it's misconfigured.
 
 
>  
> - Original Message -
>  
> From:  Onur  Tufekci <mailto:onurvc...@gmail.com>
>  
> To: ccie_voice@onlinestudylist.com
>  
> Sent: Tuesday, April 28, 2009 11:13  PM
>  
> Subject: [OSL | CCIE_Voice] CCM to CME  calls keeps ringing
>  
> 
>  
> I have the IPIPGW configured on the HQ Router. Calls from CME to CCM are
> successful but other way is not. CCM phone just keeps ringing even after
> picking up the call at CME phone.
>  
>  
>  
> Is this even a valid configuration?

Re: [OSL | CCIE_Voice] CCM to CME calls keeps ringing

[Onur Tufekci]

Thanks guys! I will check it tonight and let you know.

On Tue, Apr 28, 2009 at 11:29 PM, Cliff McGlamry  wrote:

>  Your issue is probably not on the Gatekeeper.  Your configuration looks
> okay as far as I can tell.
>
> Do a debug dial peer on the CME router and see which dial peer you are
> coming in on.  My guess is that you're coming in on a dial peer that doesn't
> have the codec defined correctly, or possibly coming in on the default dial
> peer (which is always a not so good thing to have happen).
>
> I'm betting that the issue is the dial peer you're hitting inbound on CME
> is either the wrong dial peer, or it's misconfigured.
>
>
>
> - Original Message -
> *From:* Onur Tufekci 
> *To:* ccie_voice@onlinestudylist.com
> *Sent:* Tuesday, April 28, 2009 11:13 PM
> *Subject:* [OSL | CCIE_Voice] CCM to CME calls keeps ringing
>
> I have the IPIPGW configured on the HQ Router. Calls from CME to CCM are
> successful but other way is not. CCM phone just keeps ringing even after
> picking up the call at CME phone.
>
> Is this even a valid configuration?
>
>

Re: [OSL | CCIE_Voice] CCM to CME calls keeps ringing

[Cliff McGlamry]

Your issue is probably not on the Gatekeeper.  Your configuration looks okay as 
far as I can tell.

Do a debug dial peer on the CME router and see which dial peer you are coming 
in on.  My guess is that you're coming in on a dial peer that doesn't have the 
codec defined correctly, or possibly coming in on the default dial peer (which 
is always a not so good thing to have happen).  

I'm betting that the issue is the dial peer you're hitting inbound on CME is 
either the wrong dial peer, or it's misconfigured.


  - Original Message - 
  From: Onur Tufekci 
  To: ccie_voice@onlinestudylist.com 
  Sent: Tuesday, April 28, 2009 11:13 PM
  Subject: [OSL | CCIE_Voice] CCM to CME calls keeps ringing


  I have the IPIPGW configured on the HQ Router. Calls from CME to CCM are 
successful but other way is not. CCM phone just keeps ringing even after 
picking up the call at CME phone.

  Is this even a valid configuration?

[OSL | CCIE_Voice] CCM to CME calls keeps ringing

[Onur Tufekci]

I have the IPIPGW configured on the HQ Router. Calls from CME to CCM are
successful but other way is not. CCM phone just keeps ringing even after
picking up the call at CME phone.

Is this even a valid configuration?


HQ-RTR
Description: Binary data

Re: [OSL | CCIE_Voice] ccm to cme voicemail

[jeffrey liujian]

Hi Cliff and Tech,

Thanks for your reply.

I am not sure if the xcode works properly and I made some show to check it.
The xcodes have been registered to ccm and cme successfully.
On cme site: 
1. "show call active voice compact"-- It shows the g729 to ccm ip phone address 
and g711 to cue ip address.
2. "show dspfarm session"-- It shows the g729 to cme loopback 0 and g711 to cme 
loopback 0.
3: "show voip rtp connections"-- It shows 2 active RTP connections, one from 
cme loopback 0 to ccm ip phone, the other from cme loopback 0 to cue.
On ccm site:
1. There is no xcode involved in the process because IP phone can choose g729 
to cme.
2. If I tick the outbound fast connection and MTP, it shows xcode works from 
the "performance monitor" interface when I made a call.

I guess it may be an IPIPGW h323 to sip promble. 
When I use HDV in cme for xcode, it is a first generation dsp so it can only 
xcode from g729 to g711,but it can not translate from h323 dtmf to sip dtmf.
The second generation dsp has an enhanced MTP funtion which can do xcode and 
translate dtmf together.
Is that right? Someone can help me with this?
Thanks.


Jeffrey











From: Cliff McGlamry 
To: ccie_voice@onlinestudylist.com
Sent: Sunday, April 5, 2009 12:49:10 AM
Subject: Re: [OSL | CCIE_Voice] ccm to cme voicemail

 
Dead air on a connected call means that the media 
stream is not in place.  This can be for several reasons:
 
1.  Transcoder at HQ site (if you're using the 
6608 blade) is not registered, not in the correct MRG, MRGL, or location(this 
is 
a hidden killer), or using the WRONG DEFAULT GATEWAY.  
 
2.  Transcoder at CME site not 
registered.  In addition to configuration of dspfarm profile, three 
sdspfarm commands are required in telephony-service.  Make sure the g729r8 
codec has been set up for transcoding.  Use the show sccp to confirm the 
transcode resource is up, registered and operational.  
 
3.  SCCP is attempting to register to the 
incorrect address.  The sccp ccm command should point at the same ip 
address being used as the source address in the telephony service section of 
your config.
 
4.  Incoming dial-peer not explicitly defined 
correctly causing call to use default dial peer.  Default dial peer cannot 
invoke transcoder.  Use the show voice call status to see which dial peers 
are operational.  This issue is the primary cause of this type problem, 
although anything in this list can cause it.
 
5.  MRGL not correctly assigned to the GK 
Trunk.  
 
 
Cliff
 
 
- Original Message - 
From: jeffrey liujian 
To: Cliff McGlamry ; Voice CCIE ; gughan...@yahoo.in ; rxlm...@gmail.com 
Sent: Saturday, April 04, 2009 8:44  AM
Subject: Re: [OSL | CCIE_Voice] ccm to  cme voicemail

Hi all,

There is some trouble.
ccm --> gk --> cme using  g729, cme --> cue using g711. 
when ccm phone makes a call to cme phone  or cue number:
1.  call can be connected, but no sound at  all.
2.  if I change the code (between ccm and cme) to g711,  everything is fine.
Why does this  happen?

Thanks.

Jeffrey







 From: Cliff McGlamry 
To: Gughan Gug 
Cc: ccie_voice@onlinestudylist.com
Sent: Saturday, March 21, 2009 5:19:05  PM
Subject: Re: [OSL |  CCIE_Voice] ccm to cme voicemail

 
Oh, this one bites lots of folks.
 
In the telephony-service, make sure you  have:  call-forward pattern .T   
 
If you're missing that, it won't  work.
 
Cliff
 
-  Original Message - 
From: Gughan Gug 
To: ccie_voice@onlinestudylist.com 
Sent: Saturday, March 21, 2009 1:46 AM
Subject: [OSL | CCIE_Voice] ccm to cme voicemail

Hi,
CCM phones call to CME phones through gatekeeper . phone from ccm site  can 
call to cme voicemail but if the phone call to cme ip phone and get  redirected 
the calls get disconnected.
 
I have allowed the h323 to sip and sip to h323 coneectiion under voice  service 
voip,transcoder at cme
when a cme phone call the other one and at noan the calls get  redirected 
properly to vociemail.
 
Any help in this regard. anything i am missing here. since ccm phoes  can call 
directly to  cme voice
mail using the  voicemail number no issue, only when it gets  redirectd the 
call get disconnected.
 
Regards
Gug


 Add more friends to your messenger and enjoy! Invite them  now..



  

Re: [OSL | CCIE_Voice] ccm to cme voicemail

[Tech Guy]

I will check the following:

1. Make sure regions on CCM are configured correctly; calls to CME are using 
g729.
2. Configure transcoder in CCM and assign it to the device pool.
3. Configure transcoder on the CME and assign it to the telephone-services.
4. Configure 'cisco voice services' to allow at atleast h323-to-h323; 
h323-to-sip; sip-to-h323; sip-to-sip.

Tech Guy


  - Original Message - 
  From: jeffrey liujian 
  To: Cliff McGlamry ; Voice CCIE ; gughan...@yahoo.in ; rxlm...@gmail.com 
  Sent: Saturday, April 04, 2009 8:44 AM
  Subject: Re: [OSL | CCIE_Voice] ccm to cme voicemail


  Hi all,

  There is some trouble.
  ccm --> gk --> cme using g729, cme --> cue using g711. 
  when ccm phone makes a call to cme phone or cue number:
  1.  call can be connected, but no sound at all.
  2.  if I change the code (between ccm and cme) to g711, everything is fine.
  Why does this happen?

  Thanks.

  Jeffrey






--
  From: Cliff McGlamry 
  To: Gughan Gug 
  Cc: ccie_voice@onlinestudylist.com
  Sent: Saturday, March 21, 2009 5:19:05 PM
  Subject: Re: [OSL | CCIE_Voice] ccm to cme voicemail


  Oh, this one bites lots of folks.

  In the telephony-service, make sure you have:  call-forward pattern .T   

  If you're missing that, it won't work.

  Cliff

- Original Message - 
From: Gughan Gug 
To: ccie_voice@onlinestudylist.com 
Sent: Saturday, March 21, 2009 1:46 AM
    Subject: [OSL | CCIE_Voice] ccm to cme voicemail


Hi,
CCM phones call to CME phones through gatekeeper . phone from ccm site can 
call to cme voicemail but if the phone call to cme ip phone and get redirected 
the calls get disconnected.

I have allowed the h323 to sip and sip to h323 coneectiion under voice 
service voip,transcoder at cme
when a cme phone call the other one and at noan the calls get redirected 
properly to vociemail.

Any help in this regard. anything i am missing here. since ccm phoes can 
call directly to  cme voice
mail using the  voicemail number no issue, only when it gets redirectd the 
call get disconnected.

Regards
Gug




Add more friends to your messenger and enjoy! Invite them now..

Re: [OSL | CCIE_Voice] ccm to cme voicemail

[Cliff McGlamry]

Dead air on a connected call means that the media stream is not in place.  This 
can be for several reasons:

1.  Transcoder at HQ site (if you're using the 6608 blade) is not registered, 
not in the correct MRG, MRGL, or location(this is a hidden killer), or using 
the WRONG DEFAULT GATEWAY.  

2.  Transcoder at CME site not registered.  In addition to configuration of 
dspfarm profile, three sdspfarm commands are required in telephony-service.  
Make sure the g729r8 codec has been set up for transcoding.  Use the show sccp 
to confirm the transcode resource is up, registered and operational.  

3.  SCCP is attempting to register to the incorrect address.  The sccp ccm 
command should point at the same ip address being used as the source address in 
the telephony service section of your config.

4.  Incoming dial-peer not explicitly defined correctly causing call to use 
default dial peer.  Default dial peer cannot invoke transcoder.  Use the show 
voice call status to see which dial peers are operational.  This issue is the 
primary cause of this type problem, although anything in this list can cause it.

5.  MRGL not correctly assigned to the GK Trunk.  


Cliff


  - Original Message - 
  From: jeffrey liujian 
  To: Cliff McGlamry ; Voice CCIE ; gughan...@yahoo.in ; rxlm...@gmail.com 
  Sent: Saturday, April 04, 2009 8:44 AM
  Subject: Re: [OSL | CCIE_Voice] ccm to cme voicemail


  Hi all,

  There is some trouble.
  ccm --> gk --> cme using g729, cme --> cue using g711. 
  when ccm phone makes a call to cme phone or cue number:
  1.  call can be connected, but no sound at all.
  2.  if I change the code (between ccm and cme) to g711, everything is fine.
  Why does this happen?

  Thanks.

  Jeffrey






--
  From: Cliff McGlamry 
  To: Gughan Gug 
  Cc: ccie_voice@onlinestudylist.com
  Sent: Saturday, March 21, 2009 5:19:05 PM
  Subject: Re: [OSL | CCIE_Voice] ccm to cme voicemail


  Oh, this one bites lots of folks.

  In the telephony-service, make sure you have:  call-forward pattern .T   

  If you're missing that, it won't work.

  Cliff

- Original Message - 
From: Gughan Gug 
To: ccie_voice@onlinestudylist.com 
Sent: Saturday, March 21, 2009 1:46 AM
Subject: [OSL | CCIE_Voice] ccm to cme voicemail


Hi,
CCM phones call to CME phones through gatekeeper . phone from ccm site can 
call to cme voicemail but if the phone call to cme ip phone and get redirected 
the calls get disconnected.

I have allowed the h323 to sip and sip to h323 coneectiion under voice 
service voip,transcoder at cme
when a cme phone call the other one and at noan the calls get redirected 
properly to vociemail.

Any help in this regard. anything i am missing here. since ccm phoes can 
call directly to  cme voice
mail using the  voicemail number no issue, only when it gets redirectd the 
call get disconnected.

Regards
Gug




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Re: [OSL | CCIE_Voice] ccm to cme voicemail

[jeffrey liujian]

Hi all,

There is some trouble.
ccm --> gk --> cme using g729, cme --> cue using g711. 
when ccm phone makes a call to cme phone or cue number:
1.  call can be connected, but no sound at all.
2.  if I change the code (between ccm and cme) to g711, everything is fine.
Why does this happen?

Thanks.

Jeffrey







From: Cliff McGlamry 
To: Gughan Gug 
Cc: ccie_voice@onlinestudylist.com
Sent: Saturday, March 21, 2009 5:19:05 PM
Subject: Re: [OSL | CCIE_Voice] ccm to cme voicemail

 
Oh, this one bites lots of folks.
 
In the telephony-service, make sure you have:  
call-forward pattern .T   
 
If you're missing that, it won't work.
 
Cliff
 
- Original Message - 
From: Gughan Gug 
To: ccie_voice@onlinestudylist.com 
Sent: Saturday, March 21, 2009 1:46  AM
Subject: [OSL | CCIE_Voice] ccm to cme  voicemail

Hi,
CCM phones call to CME phones through gatekeeper . phone from ccm site  can 
call to cme voicemail but if the phone call to cme ip phone and get  redirected 
the calls get disconnected.
 
I have allowed the h323 to sip and sip to h323 coneectiion under voice  service 
voip,transcoder at cme
when a cme phone call the other one and at noan the calls get redirected  
properly to vociemail.
 
Any help in this regard. anything i am missing here. since ccm phoes can  call 
directly to  cme voice
mail using the  voicemail number no issue, only when it gets  redirectd the 
call get disconnected.
 
Regards
Gug


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[OSL | CCIE_Voice] ccm to cme voicemail

[Allan Ren]

Hi,

 

Did you add call-forward pattern .T under telephone service?

 

Cheers,

Allan

Re: [OSL | CCIE_Voice] ccm to cme voicemail

[Cliff McGlamry]

Oh, this one bites lots of folks.

In the telephony-service, make sure you have:  call-forward pattern .T   

If you're missing that, it won't work.

Cliff

  - Original Message - 
  From: Gughan Gug 
  To: ccie_voice@onlinestudylist.com 
  Sent: Saturday, March 21, 2009 1:46 AM
  Subject: [OSL | CCIE_Voice] ccm to cme voicemail


  Hi,
  CCM phones call to CME phones through gatekeeper . phone from ccm site can 
call to cme voicemail but if the phone call to cme ip phone and get redirected 
the calls get disconnected.

  I have allowed the h323 to sip and sip to h323 coneectiion under voice 
service voip,transcoder at cme
  when a cme phone call the other one and at noan the calls get redirected 
properly to vociemail.

  Any help in this regard. anything i am missing here. since ccm phoes can call 
directly to  cme voice
  mail using the  voicemail number no issue, only when it gets redirectd the 
call get disconnected.

  Regards
  Gug



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[OSL | CCIE_Voice] ccm to cme voicemail

[Gughan Gug]

Hi,
CCM phones call to CME phones through gatekeeper . phone from ccm site can call 
to cme voicemail but if the phone call to cme ip phone and get redirected the 
calls get disconnected.

I have allowed the h323 to sip and sip to h323 coneectiion under voice service 
voip,transcoder at cme
when a cme phone call the other one and at noan the calls get redirected 
properly to vociemail.

Any help in this regard. anything i am missing here. since ccm phoes can call 
directly to  cme voice
mail using the  voicemail number no issue, only when it gets redirectd the call 
get disconnected.

Regards
Gug



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Re: [OSL | CCIE_Voice] CCM to CME

[Cyrus]

Suresh,


Assume that it's not a CME box, don't make it complicated for yourself by
saying cme should be different .Add it as normal h323 Gw. put ip add of one
of it's interfaceAnd u r there!

It's a very explanation on CCM trunk types

http://www.ipexpert.com/index.cfm/a/p/h323_SIPtrunks



Cyrus

On Tue, Jan 20, 2009 at 6:40 PM, Suresh Solomon wrote:

> Hi All,
>
> Asking a noobie question.
>
> When making a call from CCM to CME ( not mgcp)  you can use the following
> methods.
>
> 1. E1 /T1
> 2. via a Gatekeeper
> 3.Can you create a h.323 trunk to CME directly? ( with no Gatekeeper)
> If yes can anyone provide the config or point to a site that has some
> description?
>
> Is there any other way for this setup?
>
> Any reply is appreciated.
>
> Suresh
>
>
>  --
> *From:* "ccie_voice-requ...@onlinestudylist.com" <
> ccie_voice-requ...@onlinestudylist.com>
> *To:* ccie_voice@onlinestudylist.com
> *Sent:* Tuesday, January 20, 2009 1:40:14 PM
> *Subject:* CCIE_Voice Digest, Vol 35, Issue 175
>
> 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. Re: SRST to VM (Christian Hennrich)
>   2. H.323 GW not showing in RouteGrup (Erwan Erwan)
>   3. Re: H.323 GW not showing in RouteGrup (James Key)
>   4. Re: Routing or another issue? (Cliff McGlamry)
>   5. Re: America numbering plan ,does pstn pass "1" or not ?
>   (Cliff McGlamry)
>
>
> --
>
> Message: 1
> Date: Tue, 20 Jan 2009 00:02:45 +0100
> From: Christian Hennrich 
> Subject: Re: [OSL | CCIE_Voice] SRST to VM
> To: Kevin Porter 
> Cc: ccie_voice@onlinestudylist.com
> Message-ID: <49750695.1080...@intact-is.com>
> Content-Type: text/plain; charset=windows-1252; format=flowed
>
> Hi,
> I found that on the SRST CLI command reference guide:
>
> vm-integration
> To enter voice-mail integration configuration mode and enable voice-mail
> integration with dual tone multifrequency (DTMF) and analog voice-mail
> systems, use the vm-integration command in global configuration mode. To
> disable voice-mail integration, use the no form of this command.
>
> HTH
>
> Kevin Porter schrieb:
> > Good information.  I have not seen that statement before, which probably
> > explains why it doesn?t work with a PRI.  When debugging the PRI you see
> > the router send the correct Called number, immediately followed by the
> > ?* FDN?, which in my case causes a return of ?unallocated/unassigned
> > number? coming from the Callmanager at the Unity site?
> >
> >
> >
> > 
> >
> > *From:* ccie_voice-boun...@onlinestudylist.com
> > [mailto:ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Rogers O.
>
> > OCHIENG
> > *Sent:* Monday, January 19, 2009 2:33 AM
> > *To:* ryanstudyvo...@gmail.com
> > *Cc:* ccie_voice@onlinestudylist.com
> > *Subject:* Re: [OSL | CCIE_Voice] SRST to VM
> >
> >
> >
> > I saw this comment from IPExpert Workbook PG_Lab01 ?VM-Intergration
> > command is normally reserved (in fact Cisco says ONLY supported for use
> > with a FXO port) where we utilize overlap sending of DTMF digits?
> >
> >
> >
> > *From:* ccie_voice-boun...@onlinestudylist.com
> > [mailto:ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Ryan
> > Trauernicht
> > *Sent:* 19 January 2009 08:00
> > *To:* karuna durai
> > *Cc:* ccie_voice@onlinestudylist.com
> > *Subject:* Re: [OSL | CCIE_Voice] SRST to VM
> >
> >
> >
> > Karuna,
> >
> >  I have never been able to get it to work with the vm-integration no
> > matter what I did.  It always went to the opening greeting.
> >
> >
> >
> > Atleast not without changing the dialplan on the PSTN router to except
> > dials that are not NANP (aka 7 digits, 10 digits, and 11 digits).
> >
> >
> >
> > If you ever got it work can you post you config?
> >
> >
> >
> > Thanks,
> >
> > Ryan Trauernicht
> >
> > On Sun, Jan 18, 2009 at 10:55 PM, karuna durai  > > wrote:
> >
> >
> >
> > Hi,
> >
> > Whats your dial plan and are u able to reach HQ site when SRST, you need
> > to configrure , I n you mailbox you need to give alt extn for SRST mode.
> > and below config required at SRST
> >
> > vm-integation
> > pattern trunk busy * FDN
> > pattern trunk no-an * FDN
> >
> >
> >
> > On Mon, Jan 19, 2009 at 2:20 AM, Kevin Porter  > > wrote:
> >
> > What is the solution for VM integration while in SRST mode if you can't
> > use the DID block at

[OSL | CCIE_Voice] CCM to CME

[Suresh Solomon]

Hi All,

Asking a noobie question.

When making a call from CCM to CME ( not mgcp)  you can use the following 
methods.

1. E1 /T1 
2. via a Gatekeeper 
3.Can you create a h.323 trunk to CME directly? ( with no Gatekeeper)
If yes can anyone provide the config or point to a site that has some 
description?

Is there any other way for this setup?

Any reply is appreciated.

Suresh





From: "ccie_voice-requ...@onlinestudylist.com" 

To: ccie_voice@onlinestudylist.com
Sent: Tuesday, January 20, 2009 1:40:14 PM
Subject: CCIE_Voice Digest, Vol 35, Issue 175

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. Re: SRST to VM (Christian Hennrich)
  2. H.323 GW not showing in RouteGrup (Erwan Erwan)
  3. Re: H.323 GW not showing in RouteGrup (James Key)
  4. Re: Routing or another issue? (Cliff McGlamry)
  5. Re: America numbering plan ,    does pstn pass "1" or not ?
      (Cliff McGlamry)


--

Message: 1
Date: Tue, 20 Jan 2009 00:02:45 +0100
From: Christian Hennrich 
Subject: Re: [OSL | CCIE_Voice] SRST to VM
To: Kevin Porter 
Cc: ccie_voice@onlinestudylist.com
Message-ID: <49750695.1080...@intact-is.com>
Content-Type: text/plain; charset=windows-1252; format=flowed

Hi,
I found that on the SRST CLI command reference guide:

vm-integration
To enter voice-mail integration configuration mode and enable voice-mail 
integration with dual tone multifrequency (DTMF) and analog voice-mail 
systems, use the vm-integration command in global configuration mode. To 
disable voice-mail integration, use the no form of this command.

HTH

Kevin Porter schrieb:
> Good information.  I have not seen that statement before, which probably 
> explains why it doesn?t work with a PRI.  When debugging the PRI you see 
> the router send the correct Called number, immediately followed by the 
> ?* FDN?, which in my case causes a return of ?unallocated/unassigned 
> number? coming from the Callmanager at the Unity site?
> 
>  
> 
> 
> 
> *From:* ccie_voice-boun...@onlinestudylist.com 
> [mailto:ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Rogers O. 
> OCHIENG
> *Sent:* Monday, January 19, 2009 2:33 AM
> *To:* ryanstudyvo...@gmail.com
> *Cc:* ccie_voice@onlinestudylist.com
> *Subject:* Re: [OSL | CCIE_Voice] SRST to VM
> 
>  
> 
> I saw this comment from IPExpert Workbook PG_Lab01 ?VM-Intergration 
> command is normally reserved (in fact Cisco says ONLY supported for use 
> with a FXO port) where we utilize overlap sending of DTMF digits?
> 
>  
> 
> *From:* ccie_voice-boun...@onlinestudylist.com 
> [mailto:ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Ryan 
> Trauernicht
> *Sent:* 19 January 2009 08:00
> *To:* karuna durai
> *Cc:* ccie_voice@onlinestudylist.com
> *Subject:* Re: [OSL | CCIE_Voice] SRST to VM
> 
>  
> 
> Karuna,
> 
>  I have never been able to get it to work with the vm-integration no 
> matter what I did.  It always went to the opening greeting.  
> 
>  
> 
> Atleast not without changing the dialplan on the PSTN router to except 
> dials that are not NANP (aka 7 digits, 10 digits, and 11 digits).  
> 
>  
> 
> If you ever got it work can you post you config?
> 
>  
> 
> Thanks,
> 
> Ryan Trauernicht
> 
> On Sun, Jan 18, 2009 at 10:55 PM, karuna durai  > wrote:
> 
> 
> 
> Hi,
> 
> Whats your dial plan and are u able to reach HQ site when SRST, you need 
> to configrure , I n you mailbox you need to give alt extn for SRST mode. 
> and below config required at SRST
> 
> vm-integation
> pattern trunk busy * FDN
> pattern trunk no-an * FDN
> 
>  
> 
> On Mon, Jan 19, 2009 at 2:20 AM, Kevin Porter  > wrote:
> 
> What is the solution for VM integration while in SRST mode if you can't 
> use the DID block at the site where the Voice Mail server is located?
> 
> Thanks,
> 
> Kevin
> 
>  
> 
>  
> 
> 
> __
> This email has been scanned by the MessageLabs Email Security System.
> For more information please visit http://www.messagelabs.com/email
> __
> 
> __
> This email has been scanned by the MessageLabs Email Security System.
> For more information please visit http://www.messagelabs.com/email
> ___

Re: [OSL | CCIE_Voice] CCM to CME/CUE issues

[Devildoc]

Christian,
 
To correct your scenario 1, configure the following dial peer on your CME 
router.
 
dial-peer voice 2 voip
description <>
incoming called-number .
 
For your scenario 2 and 3, pay close attention to dtmf-relay configuration.  
The workbook has a mistake of putting the command "dtmf-relay rtp-nte 
digit-drop h245-alphanumeric" on the outgoing SIP dial-peer to CME.  SIP dial 
peers only support rtp-nte or sip-notify dtmf relay.  If you strip the rtp-nte 
dtmf relay (as indicated by the parameter "rtp-nte digit-drop") on an outbound 
SIP dial peer, and use the h245-alphanumeric dtmt relay, then it won't work.  
Usually, you put "rtp-nte digit-drop" on an incoming SIP dial-peer when you go 
from SIP to H323 network.
 
JD


Date: Fri, 1 Feb 2008 14:17:23 -0800From: [EMAIL PROTECTED]: [EMAIL PROTECTED]: 
[EMAIL PROTECTED]: Re: [OSL | CCIE_Voice] CCM to CME/CUE issues
Scott,
 
For scenario 2: 
Incoming to HQRTR: h245-alphanumeric. 
Outgoing to CME: rtp-nte digit-drop h245-alphanumericIncoming on CME: rtp-nte
Outgoing to CUE: sip-notify
 
Scenario 3: All dial-peers are h245-alphanumber but the one to CUE, which is 
still sip-notify.
 
Thanks for your response!
 
ChristianScott Monasmith <[EMAIL PROTECTED]> wrote:

For Scenario 1: Transcoding codecs between inbound and outbound SIP calls is 
not supported.
For Scenarios 2 & 3: What inbound and outbound DTMF types are you using?
On Feb 1, 2008 3:30 PM, Earnieball78 <[EMAIL PROTECTED]> wrote:

Hello,I am refering to task 4.9 in the workbook. I have a few issues I'd like 
some help straighting out. 
 
Scenario 1: CCM --> H323 G711 to HQGW --> SIP G729 to CME
Calling from CCM, calls works fine, can pick up CME phone and talk. I see 
transcoder on HQRTR being invoked so all is good. Problems start to pile up 
when trying to get to the CUE. 
When call is forwarded to VM, it get fast busy. When calling the VM pilot 
itself it never picks up. It doesn't seem to want to invoke the CME transcoder.
 
Scenario 2: CCM --> H323 G711 to HQGW --> SIP G711 to CME
Now calls go to VM just fine BUT my DTMF is not being recognized.
 
Scenario 3: CCM --> H323 G711 to HQGW --> H323 G729 to CME
Calls work. I can get to VM and I see both the HQ and CME transcoders being 
invoked. 
Problem is DTMF here again. Keep in mind that I change my DTMF settings on the 
dialpeers when going from SIP to H323. 
 
On a side note, when using a gatekeeper, CCM side being G729 to CME, 
transcoder is being invoked on the CME and calls to CUE work just fine, 
including DTMF.
 
So, I'm not quite sure where to go from here. Any suggestions or pointers would 
be greatly appreciated. 
 
Cheers,
Christian



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Re: [OSL | CCIE_Voice] CCM to CME/CUE issues

[Earnieball78]

Scott,
   
  For scenario 2: 
  Incoming to HQRTR: h245-alphanumeric. 
  Outgoing to CME: rtp-nte digit-drop h245-alphanumeric
Incoming on CME: rtp-nte
  Outgoing to CUE: sip-notify
   
  Scenario 3: All dial-peers are h245-alphanumber but the one to CUE, which is 
still sip-notify.
   
  Thanks for your response!
   
  Christian


Scott Monasmith <[EMAIL PROTECTED]> wrote:
For Scenario 1: Transcoding codecs between inbound and outbound SIP calls 
is not supported.

  For Scenarios 2 & 3: What inbound and outbound DTMF types are you using?

  On Feb 1, 2008 3:30 PM, Earnieball78 <[EMAIL PROTECTED]> wrote:
Hello,
I am refering to task 4.9 in the workbook. I have a few issues I'd like some 
help straighting out. 
   
  Scenario 1: CCM --> H323 G711 to HQGW --> SIP G729 to CME
  Calling from CCM, calls works fine, can pick up CME phone and talk. I see 
transcoder on HQRTR being invoked so all is good. Problems start to pile up 
when trying to get to the CUE. 
  When call is forwarded to VM, it get fast busy. When calling the VM pilot 
itself it never picks up. It doesn't seem to want to invoke the CME transcoder.
   
  Scenario 2: CCM --> H323 G711 to HQGW --> SIP G711 to CME
  Now calls go to VM just fine BUT my DTMF is not being recognized.
   
  Scenario 3: CCM --> H323 G711 to HQGW --> H323 G729 to CME
  Calls work. I can get to VM and I see both the HQ and CME transcoders being 
invoked. 
  Problem is DTMF here again. Keep in mind that I change my DTMF settings on 
the dialpeers when going from SIP to H323. 
   
  On a side note, when using a gatekeeper, CCM side being G729 to CME, 
  transcoder is being invoked on the CME and calls to CUE work just fine, 
including DTMF.
   
  So, I'm not quite sure where to go from here. Any suggestions or pointers 
would be greatly appreciated. 
   
  Cheers,
  Christian
  
-
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now.   
  





-- 
"There are only 10 types of people in the world: Those who understand binary, 
and those who don't" 

   
-
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Re: [OSL | CCIE_Voice] CCM to CME/CUE issues

[Scott Monasmith]

For Scenario 1: Transcoding codecs between inbound and outbound SIP calls is
not supported.
For Scenarios 2 & 3: What inbound and outbound DTMF types are you using?
On Feb 1, 2008 3:30 PM, Earnieball78 <[EMAIL PROTECTED]> wrote:

> Hello,
> I am refering to task 4.9 in the workbook. I have a few issues I'd like
> some help straighting out.
>
> Scenario 1: CCM --> H323 G711 to HQGW --> SIP G729 to CME
> Calling from CCM, calls works fine, can pick up CME phone and talk. I see
> transcoder on HQRTR being invoked so all is good. Problems start to pile up
> when trying to get to the CUE.
> When call is forwarded to VM, it get fast busy. When calling the VM pilot
> itself it never picks up. It doesn't seem to want to invoke the CME
> transcoder.
>
> Scenario 2: CCM --> H323 G711 to HQGW --> SIP G711 to CME
> Now calls go to VM just fine BUT my DTMF is not being recognized.
>
> Scenario 3: CCM --> H323 G711 to HQGW --> H323 G729 to CME
> Calls work. I can get to VM and I see both the HQ and CME transcoders
> being invoked.
> Problem is DTMF here again. Keep in mind that I change my DTMF settings on
> the dialpeers when going from SIP to H323.
>
> On a side note, when using a gatekeeper, CCM side being G729 to CME,
> transcoder is being invoked on the CME and calls to CUE work just fine,
> including DTMF.
>
> So, I'm not quite sure where to go from here. Any suggestions or pointers
> would be greatly appreciated.
>
> Cheers,
> Christian
>
> --
> Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it
> now.
>
>


-- 
"There are only 10 types of people in the world: Those who understand
binary, and those who don't"

[OSL | CCIE_Voice] CCM to CME/CUE issues

[Earnieball78]

Hello,
I am refering to task 4.9 in the workbook. I have a few issues I'd like some 
help straighting out. 
   
  Scenario 1: CCM --> H323 G711 to HQGW --> SIP G729 to CME
  Calling from CCM, calls works fine, can pick up CME phone and talk. I see 
transcoder on HQRTR being invoked so all is good. Problems start to pile up 
when trying to get to the CUE. 
  When call is forwarded to VM, it get fast busy. When calling the VM pilot 
itself it never picks up. It doesn't seem to want to invoke the CME transcoder.
   
  Scenario 2: CCM --> H323 G711 to HQGW --> SIP G711 to CME
  Now calls go to VM just fine BUT my DTMF is not being recognized.
   
  Scenario 3: CCM --> H323 G711 to HQGW --> H323 G729 to CME
  Calls work. I can get to VM and I see both the HQ and CME transcoders being 
invoked. 
  Problem is DTMF here again. Keep in mind that I change my DTMF settings on 
the dialpeers when going from SIP to H323. 
   
  On a side note, when using a gatekeeper, CCM side being G729 to CME, 
  transcoder is being invoked on the CME and calls to CUE work just fine, 
including DTMF.
   
  So, I'm not quite sure where to go from here. Any suggestions or pointers 
would be greatly appreciated. 
   
  Cheers,
  Christian

   
-
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