Hi Brian, Did you find a solution to your problem? or at least got a working dial-plan for it? I have the same problem again as well and want to know what to do with the dial-plan to off-set the effect at least since Telco says it's not their issue.
Regards, Bruce On Thu, Apr 7, 2011 at 5:53 PM, Brian Henning <bhenn...@pineinst.com> wrote: > Hi, > > Now and then our SIP phones ring with "asterisk" showing as the caller-ID. > Upon picking up the receiver, there is about five seconds of silence and > then the channel is closed (hangup). Can anyone offer some insight? > Here's > relevant snippets from my extensions.conf and Master.csv log: > > This line shows up in Master.csv: > > > "","","1-NOANSWER","inbound","","DAHDI/1-1","SIP/505-00000150","Dial","SIP/5 > 01&SIP/502&SIP/503&SIP/504&SIP/505&SIP/506,10,tTgr","2011-04-07 > 21:37:05","2011-04-07 21:37:16","2011-04-07 > 21:37:21",16,5,"ANSWERED","DOCUMENTATION","1302212225.444","" > > Here's [inbound] from extensions.conf: > [inbound] > exten => s,1,Answer > exten => s,n,Ringing > exten => s,n,Set(CALLERID(num),9${CALLERID(num)}) > exten => s,n,Dial(SIP/504&SIP/506,5,tTgr) > exten => s,n,Goto(1-${DIALSTATUS},1) > exten => 1-ANSWER,1,Hangup > exten => > _1-.,1,Dial(SIP/501&SIP/502&SIP/503&SIP/504&SIP/505&SIP/506,10,tTgr) > exten => _1-.,n,Goto(2-${DIALSTATUS},1) > exten => 2-ANSWER,1,Hangup > exten => _2-.,1,Voicemail(499@default,u) > exten => _2-.,2,Hangup > > The idea is that first 504 and 506 ring, then if neither of them answer, > everyone rings. Works great most of the time. > > I have a hunch that maybe this happens if the inbound caller hangs up while > the first Dial() is ringing, but I would've expected to see the first Dial > (to 504 and 506) show up in the Master.csv log, and it's not there. (The > preceding line of the log is a call from almost an hour earlier). In that > case though I'd expect to see "1-CANCEL" in the log instead. Perhaps if > the > caller happens to hang up right between the two Dial() commands?.. > > As an aside, the Set(CALLERID...) bit doesn't work. The idea was to > prepend > a 9 so that a SIP user could use the "redial" feature of the phone's call > log to return a missed call (automatically including the 9 for outside > line). Unfortunately the 9 does not get prepended. > > Thanks in advance for any and all advice! > ~Brian > > ------------------------------------------------------ > Brian Henning, Software Engineer > > /\ Pine Research Instrumentation > //\\ 5908 Triangle Drive > ///\\\ Raleigh, NC 27617 > ////\\\\ USA > || > || phone: 919.782.8320 > fax: 919.782.8323 > email: bhenn...@pineinst.com > ------------------------------------------------------ > > > > -- > _____________________________________________________________________ > -- 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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