asterisk users - Apr 2010 - Linksys/Sipura SPA-3201 FXO/FSA with Asterisk

All,


    I am looking at a little support on this, as I haven't found it on 
google yet. I have had this work on Callweaver, but am moving to 
Asterisk for a variety of reasons. My dial plans, and everything else 
transferred perfectly, though I am not sure they are 'correct' for 
Asterisk 1.6.1, with simple things like SIP users outlined in the 
sip.conf file, not in the users file, and my dialplan syntaxes don't 
appear to be liked by the asterisk-gui program (not a big deal, was just 
something shiny to look at for me, to try to figure out a way to get 
this going).

    What my problem is with Asterisk is my SPA-3201 is my primary voice 
gateway, as I do not own any Digium hardware, and currently do not have 
a SIP provider outside of my PBX at home. When I restart Asterisk, 
everything works perfectly. I let Asterisk sit for an hour or so, and it 
stops allowing calls to be routed into the assigned extension. I do see 
stuff from the communications, at the time the call lands on the 
Asterisk server:

  == Using SIP RTP CoS mark 5
  == Using SIP VRTP CoS mark 6

The logic is that the SPA is registered as an extension on my system, 
and incoming calls are routed into the system VIA that extension. The 
dialplan that the SPA connects to is:


[gw8028]
        exten => 8028,1,Answer
        exten => 8028,n,Set(CallerNum=${CALLERID(num)})
        exten => 8028,n,Set(CallerName=${CALLERID(name)})
        exten => 8028,n,Set(CDR(accountcode)="8203")
        exten => 8028,n,Set(CDR(UserField)="POTS")
        exten => 8028,n,Goto(from-internal,111,1)
        exten => 8028,n,Hangup


the 'from-internal' is my current call filtering/processing subsystem.

The outbound side of this works just fine though, as well as my ATA's 
and Cisco 7960's are able to make and receive calls when this is 
happening. I can include any additional details if requested, as I don't 
know exactly what would be helpful to others with this. The SPA itself 
hasn't been changed in seven months, and is stable with Callweaver.



Thanks in Advance,
Seann Clark
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Hi.

On the Spa 3102 is set as Dialplan <s0:8028> on PSTN line tab, since other
way the incoming call will try to be routed to a non set extension on
[gw8028] context

Best Regards
Jose Flores Galicia
<<FloJoSe at gmail.com>>
BriefCode && Code Based Training


2010/4/8 Seann Clark <nombrandue at tsukinokage.net>
> All,
>
>
>   I am looking at a little support on this, as I haven't found it on
google
> yet. I have had this work on Callweaver, but am moving to Asterisk for a
> variety of reasons. My dial plans, and everything else transferred
> perfectly, though I am not sure they are 'correct' for Asterisk
1.6.1, with
> simple things like SIP users outlined in the sip.conf file, not in the
users
> file, and my dialplan syntaxes don't appear to be liked by the
asterisk-gui
> program (not a big deal, was just something shiny to look at for me, to try
> to figure out a way to get this going).
>
>   What my problem is with Asterisk is my SPA-3201 is my primary voice
> gateway, as I do not own any Digium hardware, and currently do not have a
> SIP provider outside of my PBX at home. When I restart Asterisk, everything
> works perfectly. I let Asterisk sit for an hour or so, and it stops
allowing
> calls to be routed into the assigned extension. I do see stuff from the
> communications, at the time the call lands on the Asterisk server:
>
>  == Using SIP RTP CoS mark 5
>  == Using SIP VRTP CoS mark 6
>
> The logic is that the SPA is registered as an extension on my system, and
> incoming calls are routed into the system VIA that extension. The dialplan
> that the SPA connects to is:
>
>
> [gw8028]
>       exten => 8028,1,Answer
>       exten => 8028,n,Set(CallerNum=${CALLERID(num)})
>       exten => 8028,n,Set(CallerName=${CALLERID(name)})
>       exten => 8028,n,Set(CDR(accountcode)="8203")
>       exten => 8028,n,Set(CDR(UserField)="POTS")
>       exten => 8028,n,Goto(from-internal,111,1)
>       exten => 8028,n,Hangup
>
>
> the 'from-internal' is my current call filtering/processing
subsystem.
>
> The outbound side of this works just fine though, as well as my ATA's
and
> Cisco 7960's are able to make and receive calls when this is happening.
I
> can include any additional details if requested, as I don't know
exactly
> what would be helpful to others with this. The SPA itself hasn't been
> changed in seven months, and is stable with Callweaver.
>
>
>
> Thanks in Advance,
> Seann Clark
>
> --
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