[cisco-voip] VOIP deployment

Just Kennie justkennie at gmail.com
Tue Feb 1 01:26:31 EST 2011


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
>>>>
>>>>
>>>
>>
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>>
>
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