[asterisk-users] PRI failover to SIP trunk
Dave Fullerton
dfullertasterisk at shorelinecontainer.com
Fri Jul 10 11:04:18 CDT 2009
Steve Totaro wrote:
> On Fri, Jul 10, 2009 at 11:40 AM, Dave Fullerton <
> dfullertasterisk at shorelinecontainer.com> wrote:
>
>> Steve Totaro wrote:
>>> On Fri, Jul 10, 2009 at 10:45 AM, Dave Fullerton <
>>> dfullertasterisk at shorelinecontainer.com> wrote:
>>>
>>>> Tzafrir Cohen wrote:
>>>>> On Thu, Jul 09, 2009 at 05:31:18PM -0400, Steve Totaro wrote:
>>>>>
>>>>> You have a small typo:
>>>>>
>>>>>> exten => _.,1,Dial(Zap,g1,${EXTEN})
>>>>>> exten => _.,2,Dial(SIP,Provider,${EXTEN})
>>>>> exten => _.,1,Dial(Zap/g1/${EXTEN})
>>>>> exten => _.,2,Dial(SIP/Provider/${EXTEN})
>>>>>
>>>>> ('/' instead of ',')
>>>>>
>>>> While this will work, be aware that there are circumstances where you
>>>> may end up calling the number twice, once through each provider. One
>>>> example is if the number you dial is busy, that progress will be passed
>>>> via the PRI to asterisk and the dialplan will continue to the next
>>>> priority. In this case, dialing the number again through the SIP
>>>> provider. To avoid this you will need to use some dialplan logic and
>>>> check the result of the DIALSTATUS variable. See this page for examples:
>>>>
>>>> http://www.voip-info.org/wiki/view/Asterisk+variable+DIALSTATUS
>>>>
>>>> -Dave
>>>>
>>>>
>>> Good point.
>>>
>>> I was unaware that "busy" back from a TDM circuit would progress in the
>>> dialplan rather than going to the h exten.
>>>
>>> What other cases are there like that?
>> It is my understanding (through trial and error, reading, etc) that any
>> Dial command that does not result in an answered state will continue in
>> the dialplan after a timeout (if specified) or some sort of progress is
>> received. If the called channel results in an answer then dialplan
>> processing stops as soon as one party hangs up (unless the g option is
>> specified).
>>
>> This works on any channels that can pass progress (SIP/IAX and Zap/DAHDI
>> PRI and BRI). Zap/DAHDI Analog channels are considered answered as soon
>> as the dial is complete so you won't be able to use this trick under
>> normal circumstances.
>>
>> -Dave
>>
>>
> True I guess except that if the call fails as the OP posted, because the PRI
> is down, it should work then right?
I believe so, I haven't tried it. I imagine DIALSTATUS would be either
CHANUNAVAIL or CONGESTION.
>
> Another thing. For outbound calls, I do not have a timeout. So the user
> hangs up when they are ready, or when the other side hangs up or gets
> congestion, which amounts to the h exten, or am I not correct.
I can't answer to the use of the h exten, I've never used it.
> Why have a timeout on outbound dialing (unless you are a dialer app?) It is
> not like voicemail where you want it to ring for so many seconds and then
> roll to VM.
You usually wouldn't use a timeout for outbound PSTN calls. I only
mentioned it to try to be as complete as possible.
-Dave
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