Re: [OSL | CCIE_Voice] B-ACD VoiceMail

[kill mill]

The param voice mail will send the call to voice mail after the max call
retry timer has expired. The calling number will be the hunt pilot number
and Unity will play u the fantastic greeting that this mail box does not
exist and transfer u to the AA.

//ankur

On Thu, Jan 29, 2009 at 5:07 AM, kamal yousaf wrote:

> If B-ACD script causes call to be sent to VoiceMail number defined using
> 'param voicemail 5000' , which mailbox is the call routed to ? Is it Pilot
> Point OR HuntGroup Number ?
>
>

Re: [OSL | CCIE_Voice] DTMF digits from PSTN callers are not recognized in the Unity Greeting

[Yung Hung]

Hi Kamal,

Were you able to resolve this problem?



From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of kamal yousaf
Sent: Monday, February 02, 2009 11:04 PM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] DTMF digits from PSTN callers are not recognized in 
the Unity Greeting

Hi,

 When dialing from BR1 into Unity , calls and digit input work .During AAR/SRST 
, callers dialing via PSTN are able to leave voice-mail messages.However, when 
subscriber dials directly into unity during AAR/SRST, unity plays subscriber 
greeting but it doesn't recognize any input dtmf  digit inputs.As per support 
Wiki, call-handlers are working but Unity Subscriber Sign-In fails.Any one else 
had same problem ?

http://supportwiki.cisco.com/ViewWiki/index.php/DTMF_digits_from_PSTN_callers_are_not_recognized_in_the_greeting,_and_Call_Handler_and_callers_are_not_authenticated_on_Cisco_Unity

Re: [OSL | CCIE_Voice] IPCC and MOH Flash scenario

[Sergio Polizer]

Hi Ryan and All,
 
I think the limitation is at the Site B. According to SRND, the router can 
stream only a single audio file from flash and that you can use only a single 
multicast address and port number per router. 
 
If we had other MMoH sorce at Site B It will works.
 
Regards, Sergio.



Date: Thu, 5 Feb 2009 14:51:15 -0600From: ryanstudyvo...@gmail.comto: 
ccie_vo...@onlinestudylist.comsubject: [OSL | CCIE_Voice] IPCC and MOH Flash 
scenarioI got a scenario that I dont think is possible with MOH, but I wanted a 
sanity check.

HQ location and IPCC receives MOH multicast from the CM.  That same MOH file is 
on the flash for SiteB and is used locally so no multicast MOH will go across 
the WAN.  So calls from HQ to SiteB and calls from PSTN you hear MOH just fine 
(MOH source file 1).  

IPCC requires a different MOH file to be played.  So I drop that MOH in the 
folder and use it as multicast and apply it to the CTI Ports so callers can 
hear that new MOH file.  Works great for HQ phones and PSTN callers calling 
into the HQ location.

Since multicast is not allowed across the WAN and MOH is on the flash for the 
IP address of (239.1.1.3, the MOH server is in the HQ DP so G729 IP address to 
the SiteB).  If a caller calls in from SiteB to IPCC and it put into the queue 
you will never hear MOH since it is the second MOH source file that is trying 
to play, which is being announced on a different IP address.  correct?

This scenario is not possible right?

Thanks,
Ryan Trauernicht
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[OSL | CCIE_Voice] IPCC and MOH Flash scenario

[Ryan Trauernicht]

I got a scenario that I dont think is possible with MOH, but I wanted a
sanity check.
HQ location and IPCC receives MOH multicast from the CM.  That same MOH file
is on the flash for SiteB and is used locally so no multicast MOH will go
across the WAN.  So calls from HQ to SiteB and calls from PSTN you hear MOH
just fine (MOH source file 1).

IPCC requires a different MOH file to be played.  So I drop that MOH in the
folder and use it as multicast and apply it to the CTI Ports so callers can
hear that new MOH file.  Works great for HQ phones and PSTN callers calling
into the HQ location.

Since multicast is not allowed across the WAN and MOH is on the flash for
the IP address of (239.1.1.3, the MOH server is in the HQ DP so G729 IP
address to the SiteB).  If a caller calls in from SiteB to IPCC and it put
into the queue you will never hear MOH since it is the second MOH source
file that is trying to play, which is being announced on a different IP
address.  correct?

This scenario is not possible right?

Thanks,
Ryan Trauernicht

Re: [OSL | CCIE_Voice] CCIE Voice Results

[Shadab Abbasi (moabbasi)]

Also it depends, if the couple of days are Holidays, then it will come
on next business day.

 

Regards,

Shadab

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of rob
Sent: Thursday, February 05, 2009 8:56 PM
To: Jose Gregorio Linero (jlinero)
Cc: ccie_voice@onlinestudylist.com; Christian Narvaez
Subject: Re: [OSL | CCIE_Voice] CCIE Voice Results

 

 Same in Brussels. 

Re: [OSL | CCIE_Voice] CCIE Voice Results

[rob]

 Same in Brussels.

Re: [OSL | CCIE_Voice] CCIE Voice Results

[Jose Gregorio Linero (jlinero)]

Hi, same in RTP



From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Christian
Narvaez
Sent: Jueves, Febrero 05, 2009 10:01 AM
To: kamal yousaf; ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] CCIE Voice Results



At least in San Jose the result generally comes between 13:00 and 17:00
of the next business day. 

 

De: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] En nombre de kamal
yousaf
Enviado el: 05 February 2009 11:51
Para: ccie_voice@onlinestudylist.com
Asunto: [OSL | CCIE_Voice] CCIE Voice Results

 

Hi,
  
 Can anyone tell me how long it takes for CCIE voice lab result to come
? I know expected time is b/w 24-48 hours.But since 34+ hours have
already passed since my lab, i am getting nerveless to know what could
be causing delay ? 

Thanks

Re: [OSL | CCIE_Voice] CCIE Voice Results

[Christian Narvaez]

At least in San Jose the result generally comes between 13:00 and 17:00
of the next business day. 

 

De: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] En nombre de kamal
yousaf
Enviado el: 05 February 2009 11:51
Para: ccie_voice@onlinestudylist.com
Asunto: [OSL | CCIE_Voice] CCIE Voice Results

 

Hi,
  
 Can anyone tell me how long it takes for CCIE voice lab result to come
? I know expected time is b/w 24-48 hours.But since 34+ hours have
already passed since my lab, i am getting nerveless to know what could
be causing delay ? 

Thanks

[OSL | CCIE_Voice] CCIE Voice Results

[kamal yousaf]

Hi,

 Can anyone tell me how long it takes for CCIE voice lab result to come ? I
know expected time is b/w 24-48 hours.But since 34+ hours have already
passed since my lab, i am getting nerveless to know what could be causing
delay ?

Thanks

Re: [OSL | CCIE_Voice] SIP Failover commands

[Ryan Trauernicht]

Thank you very much Robert.

On Thu, Feb 5, 2009 at 2:54 AM, Robert Schuknecht wrote:

> Ryan,
>
> some time ago i found an blog article from Vik, where he explained your
> scenario. For SIP failover you need the following:
>
> sip-ua
> retry invite 2
> timers trying 400
>
> You will find Viks article at this link: http://malhi.net/blog/?p=31
>
> HTH
>
> /Robert
>
> >>> Ryan Trauernicht schrieb am Donnerstag, 5.
> Februar
> 2009 um 05:22 in Nachricht 9fa638ba8d82c1ae6ebb60f6e8bb1cf5:
> > For H323 you can have a call up and if CM fails the call stays up.
> > If you have a SIP trunk to an FXS port and that CM fails what commands
> are
> > needed to allow it to failover.  The H323 commands obviously dont work.
> >
> > Thanks,
> > Ryan Trauernicht
>

[OSL | CCIE_Voice] url directory

[omar itani]

hi guys
1-in cme how to enable url directory &ccm  what does it mean
 
2-predot+pefix 0113433132x while we are creating route list into route group 
this digit manipulation where
i should assign it 
 
3-when disable e.164 regiter in cme becoz of gatekeeper config ,how to anable 
again to do another task 
 
 
thanks
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[OSL | CCIE_Voice] Antw: SIP Failover commands

[Robert Schuknecht]

Ryan,

some time ago i found an blog article from Vik, where he explained your 
scenario. For SIP failover you need the following:

sip-ua
retry invite 2
timers trying 400

You will find Viks article at this link: http://malhi.net/blog/?p=31

HTH

/Robert

>>> Ryan Trauernicht schrieb am Donnerstag, 5. Februar
2009 um 05:22 in Nachricht 9fa638ba8d82c1ae6ebb60f6e8bb1cf5:
> For H323 you can have a call up and if CM fails the call stays up.
> If you have a SIP trunk to an FXS port and that CM fails what commands are
> needed to allow it to failover.  The H323 commands obviously dont work.
> 
> Thanks,
> Ryan Trauernicht