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

Jose Flores Galicia flojose at gmail.com
Fri Apr 9 12:21:44 CDT 2010


I am just guessing, but sometimes happened to me that the logic on dialplan
does not contain a hungup, so channels on spa3102 continues up even if users
have finished.

On CLI you should put "core show channels", and see if there are channels to
sip/8028

On the  [gw8028] context you send the call to [from-internal] extension 111,
so that extension has to end in a hangup action.

Just guessing.

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


2010/4/9 Seann Clark <nombrandue at tsukinokage.net>

> 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
>>
>>    --
>>    _____________________________________________________________________
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>>
>>
>
>
> --
> _____________________________________________________________________
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