[asterisk-users] help with Sipura SPA 3000

Francis Augusto Medeiros francis at mgate.com.br
Fri Apr 13 09:42:21 MST 2007


On 11 de abr de 2007, at 21:07, James Harper wrote:

>>> A dialplan of '(S0<:s>)' will get your phone to jump straight into
> the
>>> 's' extension in asterisk as soon as someone picks it up. From
>>> there you
>>> can do something like:
>>
>> It worked perfectly! Thanks!
>
> Just remember that having Asterisk supply the dialtone does add (a
> slight) additional load, rather than it just routing calls between
> endpoints. Not an issue with one or two ATA's though.

i have just one ATA anyway, this is intended to be used solely at  
home... I'll probably give it up in favor of pbxes.org...

>>> [sip_ata_incoming]
>>> exten => s,1,Answer
>>> exten => s,n,DISA(no-password|sip_extension_in)
>>>
>>> so Asterisk will give you dialtone and do the dialplan stuff for
> you.
>>>> From the 'sip_extension_in' context you can make a single '0' or
> '*'
>>> call the PSTN line.
>>
>> On the "sip_extension_in", I entered the following
>>
>> exten => 0,1,NoOp(outgoing call via POTS line to No. ${EXTEN:1})
>> exten => 0,2,Dial(SIP/${EXTEN:1}@LinkSysOut,120)
>> exten => 0,3,Congestion()
>> exten => 0,4,Hangup
>>
>> However, when I press the "0", it does gives me a dialtone, but it
>> doesn't seem to be delivering the tones imediately. I even suspect it
>> isn't my PSTN tone after the 0. Is there something else?
>
> A few things to check:
>
> . ${EXTEN:1} will be empty because the extension can only be '0'.  
> Change
> it to 'SIP/LinkSysOut' instead

Done.

> . I'm not sure but I think that the SPA3000 can either present a  
> 'false'
> dialtone to the SIP call on the PSTN line, take the digits, then send
> them to the PSTN then connect the SIP call to it, or it can give the
> real PSTN dialtone and connect the call immediately. I think the  
> latter
> is what you want but I can't remember the name of the setting. Maybe
> 'one stage dialling'?

Done!!!! It works! I had to disable one stage dialing and setting the  
VOIP DP to none. However, this is giving me one trouble: I use also  
my cellphone (E61) to make calls, and it would be nice to do one  
stage dialing with it. I don't think it's possible to make it one  
stage with the mobile and two stage with the FXS of the Sipura...


Cheers, and thanks a lot!!!


Francis



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