[asterisk-biz] $0.0090 US 48 Premium SIP Termination

[Mike Fedyk]

Not only that, but have a PDD timeout.  I have heard of some sip servers 
when they're having trouble will respond with a 100 and then stop 
responding.  This timeout would be useful for dialplan failover in 
situations where the PDD is too long, but shorter than the dial timeout.

Sergey Kuznetsov wrote:
> My fault: Between 100 Trying and 180 Ringing
>
>
> All the Best!
> Sergey.
>
> Sergey Kuznetsov wrote:
>> You have to add some kind of code to measure the time between the ACK 
>> after INVITE and 180 Ringing.
>>
>> That will be the PDD.
>>
>>
>> All the Best!
>> Sergey.
>>
>>
>> Darren Wiebe wrote:
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>>> I had somebody look into measuring PDDs with Asterisk but he didn't 
>>> find
>>> anything without digging into the source code.  We are looking at doing
>>> something with it anyways though.
>>>
>>> Darren Wiebe
>>> darren at aleph-com.net
>>>
>>>
>>> Jean-Michel Hiver wrote:
>>>  
>>>>> Great, thanks for the explanation.
>>>>>
>>>>> Is there anywhere I can find a list of these accepted industry
>>>>> standard terms and their definitions and derivations?  I'd be 
>>>>> happy to
>>>>> start a page on the wiki if someone took the time to explain it a bit
>>>>> to me.
>>>>>       
>>>> I can start with the obvious...
>>>>
>>>> ASR: Answer Seizure Ratio. We've seen that in the thread.
>>>> ACD: (Average Call Duration) or ALOC (Average Length of Call).
>>>> PDD: Post Dialing Delay. Time between the start of a call and the time
>>>> the phone starts ringing.
>>>>
>>>> Can't think of any others right now...
>>>>
>>>> Speaking of PDDs, is there a way of measuring it with *? That would be
>>>> great.
>>>>
>>>> Cheers,
>>>> Jean-Michel.
>>>>
>>>>     
>>>
>>>
>>> - --
>>> Darren Wiebe
>>> darren at aleph-com.net
>>> Aleph Communications
>>> ASTPP - Open Source Voip Billing & Calling Cards
>>> www.aleph-com.net/astpp
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