[asterisk-users] Callerid Woes

[John Koenig]

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
>
>