[asterisk-users] Callerid Woes
Sherwood McGowan
sherwood.mcgowan at gmail.com
Tue Jul 29 18:02:52 CDT 2008
John Koenig wrote:
> Thanks for the tip about sip set debug <peer>. I was able to capture
> some information about the call in progress, but I am confused as to
> what I see. When I pick up my sip phone I dial
> *811<area_code><prefix><number>, and the first invite I see is going to
> 1<area_code><prefix><number>@<my_asterisk_ip>. Shouldn't the *81 be
> included in the request?
>
> Is it possible that the linksys pap2 that I am using is removing the *81
> prior to placing the invite request?
>
> John
>
> John Millican wrote:
>
>> John Koenig wrote:
>>
>>
>>> I tried all of the suggestions, and still the callerid remains intact.
>>> I guess at this point I am starting to wonder what bit of logic is being
>>> run when I dial *8111XXXXXXXXXX...
>>>
>>> Is there a way I can trace how a call is being processed within
>>> asterisk? Or even see what I am sending to my VoIP terminating node?
>>>
>>> John
>>>
>>> John Millican wrote:
>>>
>>>
>>>> Doug Lytle wrote:
>>>>
>>>>
>>>>
>>>>> John Koenig wrote:
>>>>>
>>>>>
>>>>>
>>>>>> exten=s,1,set(CALLERID(all)= null)
>>>>>> exten=s,n,Dial(${ARG1})
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>> Just a guess.
>>>>>
>>>>> exten => s,1,Set(CALLERID(all)= null <0>)
>>>>> exten => s,n,SetCallerPres(prohib)
>>>>> exten => s,n,Dial(${ARG1})
>>>>>
>>>>>
>>>>> Doug
>>>>>
>>>>>
>>>>>
>>>>>
>>>> I believe you need to use:
>>>> exten => s,1,Set(CALLERID(all)=)
>>>> To set an empty callerId
>>>>
>>>>
>>>>
>>>>
>> typing:
>> sip set debug peer <peer_name>
>> at the CLI will give you a bunch of information as to what is going on
>> with that peer
>>
>>
>>
>
>
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The Linksys is taking *81 as a local spree code, causing it to be
stripped. The problem you're then getting is probably that Asterisk is
using _it's_ caller id information for your peer
--
Sherwood McGowan
VoIP / Telecom Solutions
sherwood.mcgowan at gmail.com
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