[asterisk-users] False ring problem

Rizwan Hisham rizwanhasham at gmail.com
Wed May 30 08:23:01 MST 2007


Do you mean to say that -- first the carrier sends the msg to us to ring and
then the end user sends the msg to ring?

On 5/30/07, Ricardo Martins <rpoppi77 at gmail.com> wrote:
>
>  It seems that the ring issue is on the CARRIER-OUT signaling. It's
> sending you a SIP-Ring-Message and your asterisk-box is sending it to the
> callee. The second green line ".....is ringing" apears jut because your box
> received a ring signal from the CARRIER-OUT. Got the point?
>
> I don't know what the "left from hold" means but seems to be related to
> the situation when we push the "flash" button on the phone to put "on hold"
> and flash again to put "out of hold". But I'm realy not sure about it.
>
> Rgds, Ricardo Martins
>
>
> Rizwan Hisham escreveu:
>
> Here is my CLI output:
>
> Called 17142545587 at CARRIER-OUT
>     -- SIP/CARRIER-OUT-007d0310 is ringing
>     -- Call on SIP/CARRIER-OUT-007d0310 left from hold
>     -- SIP/CARRIER-007d0310 is making progress passing it to
> SIP/pepsi-00f267e0
> i clearly notice that when the first orange cli msg appears then the
> actual ringing starts. like this tone -- tone -- totone -- tone, and if the
> callee is busy then tone -- tone -- tobeep beep .
>
> does anyone know what this means: -- Call on SIP/CARRIER-OUT-007d0310 left
> from hold
>
> On 5/30/07, Rizwan Hisham < rizwanhasham at gmail.com> wrote:
> >
> > Maybe its a bug in asterisk 1.4.2
> >
> > On 5/30/07, Rizwan Hisham <rizwanhasham at gmail.com > wrote:
> > >
> > > There is no R/r option in my dial application.im only using gM option
> > > here is the dialplan:
> > >
> > > exten=> _1X.,1,NoOp("Dialing Local!!!")
> > > exten=> _1X.,2,Dial(Sip/${EXTEN}@RNKTEL-OUT,,gM(payasyougo^${CDR(accountcode)}^${CDR(userfield)}))
> > >
> > > exten=> _1X.,3,Hangup
> > >
> > >
> > > On 5/30/07, Ricardo Martins < rpoppi77 at gmail.com> wrote:
> > > >
> > > > You should (must!) remove any r/R parameter from your command. If
> > > > you do that, no false ring will be generated anymore...
> > > >
> > > > Att, Ricardo.
> > > >
> > > > Rizwan Hisham escreveu:
> > > >
> > > > Hi all,
> > > > when a user dials any number, asterisk automatically generates
> > > > ringing which caller can hear, and after 2 - 3 rings asterisk detects that
> > > > the called user is busy, then caller hears busy tone. for example user
> > > > hears--- tone--tone--tobeep beep beep ---Can i some how eliminate the false
> > > > ringing at the start so that user hears only beep beep beep if the called
> > > > user is busy. I have used the R and r options in Dial application but they
> > > > dont work.
> > > >
> > > > --
> > > > Rizwan Hisham
> > > > Software Engineer
> > > > AXVOICE Inc.
> > > >
> > > > ------------------------------
> > > >
> > > > _______________________________________________
> > > > --Bandwidth and Colocation provided by Easynews.com --
> > > >
> > > > asterisk-users mailing list
> > > >
> > > >
> > > >
> > > > To UNSUBSCRIBE or update options visit:
> > > >    http://lists.digium.com/mailman/listinfo/asterisk-users
> > > >
> > > >
> > > >
> > > >
> > > >
> > > > _______________________________________________
> > > > --Bandwidth and Colocation provided by Easynews.com --
> > > >
> > > > asterisk-users mailing list
> > > > To UNSUBSCRIBE or update options visit:
> > > >   http://lists.digium.com/mailman/listinfo/asterisk-users
> > > >
> > > >
> > >
> > >
> > > --
> > > Rizwan Hisham
> > > Software Engineer
> > > AXVOICE Inc.
> > >
> >
> >
> >
> > --
> > Rizwan Hisham
> > Software Engineer
> > AXVOICE Inc.
> >
>
>
>
> --
> Rizwan Hisham
> Software Engineer
> AXVOICE Inc.
>
> ------------------------------
>
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
>


-- 
Rizwan Hisham
Software Engineer
AXVOICE Inc.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070530/f3c38513/attachment.htm


More information about the asterisk-users mailing list