[Asterisk-Dev] delayed sip answer

lconroy lconroy at insensate.co.uk
Sun Sep 25 11:01:56 MST 2005


Hi Folks,
... or, to phrase it another way, the "SIP FXO" is badly broken, and  
no one is likely to want to change any code that would break the  
existing chan_SIP for everyone else, which you already surmise is  
what would be required to accommodate the broken unit.

Of course, if you wanted to do such an abomination in your  
installation, then chan_sip.c awaits. Rely on early media? - Gak!

Trust me on this, just say no - I would be surprised if the -users  
list gave any better answer.

atb,  Lawrence
On 25 Sep 2005, at 18:47, Jens Kübler wrote:

> Am Sonntag 25 September 2005 19:37 schrieb Mark Willis:
>
>> Suppose I had a SIP FXO that sent the 200 OK immediately after  
>> dialling,
>> and no way to do call progress. But I need to not send answer
>> supervision to my carriers until the call is really answered.  
>> Short of
>> detecting "hello, hello" the best way to handle it seems to be to  
>> delay
>> the answer supervision for about 20 seconds and rely on early media
>> should the call be answered before then.
>>
>> So, is there a way with asterisk to delay sending the answer back  
>> to my
>> carrier? i.e, the 200 OK comes in from the FXO but I "queue" it  
>> for 20
>> seconds before sending it onward. Any ideas?
>>
>> Mark
>>
>>
> This is a user question as it seems you want to use asterisk.
> So please post to asterisk-users unless you want something that  
> involves the
> "art of creating code" :-)
>
> Jens
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