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 <<floj...@gmail.com>> BriefCode && Code Based Training 2010/4/9 Seann Clark <nombran...@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 >> <<floj...@gmail.com <mailto:floj...@gmail.com>>> >> BriefCode && Code Based Training >> >> >> 2010/4/8 Seann Clark <nombran...@tsukinokage.net <mailto: >> nombran...@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 >> >> >> > > > -- > _____________________________________________________________________ > -- 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 >
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