[cisco-voip] VOIP deployment

[Nick Matthews]

I could explain most of these concepts, but it would really be better for
you to do some reading on the topic.  These are the building blocks of
voice, and you probably aren't doing yourself any favors by learning them
individually as-needed.  I would go and take a look at the CCVP books.  The
CCNA voice book would go a long way for you to understanding the concepts
here.

-nick

On Tue, Feb 1, 2011 at 1:26 AM, Just Kennie <justkennie at gmail.com> wrote:

> Matthews,
> Thanks a bunch, I am most grateful. I am having a clearer view now, but
> have issues with the following to wrap up.
> - About the DID, are you saying I should remove it from the following
> config, because that the only place I included it.
>
>
> dial-peer voice 1 pots
> destination-pattern 12345
>  direct-inward-dial
>  port 0/0/0
>
> - How do I know the protocol I am using by default, since there is no call
> manager.
>
> - Once I place a call to the PSTN number to be routed to the extension, how
> do I monitor the progress of the call? debug  ???
>
> - How do I add bound ports, voice service voip config, and what do they add
> in this situation.
>
> Thanks as I await your reply.
>
>
> *Cheers,
> Idowu Kehinde
> TEL: 2348024017748
> BB:  30E8E5F9
> *
>
>
>
> On Tue, Feb 1, 2011 at 6:06 AM, Nick Matthews <matthnick at gmail.com> wrote:
>
>> You don't need  direct-inward-dial on FXO ports, only PRI ports.  Other
>> than that, the config looks alright.
>>
>> If you've got other config such as bound ports, voice service voip config,
>> etc it would be helpful to see that.
>>
>> And from that config, you're not using SIP.  'session protocol sip' would
>> have to be added to the voip dial peers.
>>
>> -nick
>>
>>
>> On Mon, Jan 31, 2011 at 11:31 PM, Just Kennie <justkennie at gmail.com>wrote:
>>
>>> I am using SIP.
>>>
>>>
>>> *Cheers,
>>> Idowu Kehinde
>>> TEL: 2348024017748
>>> BB:  30E8E5F9
>>> *
>>>
>>>
>>>
>>> On Tue, Feb 1, 2011 at 5:08 AM, neil dsilva <neild25 at gmail.com> wrote:
>>>
>>>> Which routing protocol you are using ?
>>>>
>>>> Regards,
>>>> Neil Dsilva
>>>>
>>>> On Mon, Jan 31, 2011 at 5:48 PM, Just Kennie <justkennie at gmail.com>wrote:
>>>>
>>>>> Hi guys,
>>>>> I am in need of swift help, I am a cisco network freak, and found my
>>>>> self handling voice in my new company. I have been doing lots of reading on
>>>>> voice, but I now I have a major huddle to scale. I will explain the design
>>>>> in details and concisely.
>>>>>
>>>>>    PSTN(12345) - - - - - -(voice-port 0/0/0) CE1 - - - -  PE1- - - - -
>>>>> - - - PE2 - - - - -CE2(voice-port 0/0/0)- - - -PHONE(1000)
>>>>>
>>>>> That is the design above, Layer 2 and Layer 3 are fine.
>>>>> The requirement is, when external call called 12345, the call should be
>>>>> routed and picked at extension 1000 (PHONE)
>>>>> I have built my config (below), please help me look at it, as I don't
>>>>> have the full equipment to test across.
>>>>>
>>>>> CE1
>>>>> voice-port 0/0/0
>>>>>  connection plar 1000
>>>>>  description connected to 12345
>>>>>
>>>>> dial-peer voice 1 pots
>>>>> destination-pattern 12345
>>>>>  direct-inward-dial
>>>>>  port 0/0/0
>>>>>
>>>>> dial-peer voice 2 voip
>>>>> destination-pattern 1000
>>>>>  session target ipv4:20.0.0.2   (CE2)
>>>>>
>>>>> CE2.........................
>>>>>
>>>>> voice-port 0/0/0
>>>>>  description connected to 1000
>>>>>
>>>>> dial-peer voice 1 pots
>>>>>  destination-pattern 1000
>>>>>  port 0/0/0
>>>>>
>>>>> dial-peer voice 2  voip
>>>>>  destination-pattern 12345
>>>>>  session target ipv4:10.0.0.2  (CE1)
>>>>>
>>>>> Please help out in details, I will understand. Thanks!
>>>>>
>>>>>
>>>>> *Cheers,
>>>>> Idowu Kehinde
>>>>> TEL: 2348024017748
>>>>> BB:  30E8E5F9
>>>>> *
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> cisco-voip mailing list
>>>>> cisco-voip at puck.nether.net
>>>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>>>>
>>>>>
>>>>
>>>
>>> _______________________________________________
>>> cisco-voip mailing list
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>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>>
>>>
>>
>
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