[asterisk-users] Linksys/Sipura SPA-3201 FXO/FSA with Asterisk

[Seann Clark]

Yes, the SPA-3201 is set as: (<S0:8028>) on dialplan 8, which is what I 
have the device set to use. My bare bones working dialplan from 
Callweaver works nearly perfectly with Asterisk, and takes all the calls 
and works just as it did in Callweaver (making adjustments for the 
differences in dialplan syntaxes as Callweaver still uses Asterisk 1.2 
syntax). It is just after an hour I can't get calls inbound to Asterisk. 
If I stop Asterisk, and start Callweaver, it can sit for months and 
handle calls no problem, with a like dialplan. SIP users and settings 
aren't changed between the systems either, and my Cisco phones, and the 
other Linksys ATA I have plays well. I am a little stumped on that. I 
will include a SIP dump when I get that back up in test mode (Since it 
is my home telephone system and I need it for work, which I am doing 
right now, I can't afford the downtime right this moment, but tomorrow I 
should have time for this).


Thanks in advance,
Seann Clark

On 4/9/2010 12:08 AM, Jose Flores Galicia wrote:
> 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 <mailto:FloJoSe at gmail.com>>>
> BriefCode && Code Based Training
>
>
> 2010/4/8 Seann Clark <nombrandue at tsukinokage.net 
> <mailto: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
>
>     --
>     _____________________________________________________________________
>     -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>     New to Asterisk? Join us for a live introductory webinar every Thurs:
>     http://www.asterisk.org/hello
>
>     asterisk-users mailing list
>     To UNSUBSCRIBE or update options visit:
>     http://lists.digium.com/mailman/listinfo/asterisk-users
>
>


-------------- next part --------------
A non-text attachment was scrubbed...
Name: smime.p7s
Type: application/pkcs7-signature
Size: 5544 bytes
Desc: S/MIME Cryptographic Signature
Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20100409/41c8d308/attachment.bin