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. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070530/ac78db36/attachment.htm
Do you have 'r' in ur dial command ? 'r' makes asterisk produce ring . On 30/05/07, Rizwan Hisham <rizwanhasham@gmail.com> wrote:> 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 > >
Rizwan Hisham wrote:> Hi all, > when a user dials any number, asterisk automatically generates ringing > which caller can hear, and after 2 - 3 rings asteriskShow use that section of your dial plan. Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety."
<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN"> <html> <head> <meta content="text/html;charset=ISO-8859-1" http-equiv="Content-Type"> </head> <body bgcolor="#ffffff" text="#000000"> You should (must!) remove any r/R parameter from your command. If you do that, no false ring will be generated anymore...<br> <br> Att, Ricardo.<br> <br> Rizwan Hisham escreveu: <blockquote cite="mid:4809880c0705300303r75c3f039k6934823b55d48c6b@mail.gmail.com" type="cite">Hi all,<br> 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. <br clear="all"> <br> -- <br> Rizwan Hisham<br> Software Engineer<br> AXVOICE Inc. <pre wrap=""> <hr size="4" width="90%"> _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: <a class="moz-txt-link-freetext" href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users</a> </pre> </blockquote> <br> </body> </html>
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)}^${CDR(userfield)}))
exten=> _1X.,3,Hangup
On 5/30/07, Ricardo Martins <rpoppi77@gmail.com>
wrote:>
> You should (must!) remove any r/R parameter from your command. If you do
> that, no false ring will be generated anymore...
>
> Att, Ricardo.
>
> Rizwan Hisham escreveu:
>
> Hi all,
> when a user dials any number, asterisk automatically generates ringing
> which caller can hear, and after 2 - 3 rings asterisk detects that the
> called user is busy, then caller hears busy tone. for example user hears---
> tone--tone--tobeep beep beep ---Can i some how eliminate the false ringing
> at the start so that user hears only beep beep beep if the called user is
> busy. I have used the R and r options in Dial application but they dont
> work.
>
> --
> Rizwan Hisham
> Software Engineer
> AXVOICE Inc.
>
> ------------------------------
>
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
--
Rizwan Hisham
Software Engineer
AXVOICE Inc.
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20070530/1329de08/attachment.htm
Maybe its a bug in asterisk 1.4.2 On 5/30/07, Rizwan Hisham <rizwanhasham@gmail.com> wrote:> > There is no R/r option in my dial application.im only using gM option > here is the dialplan: > > exten=> _1X.,1,NoOp("Dialing Local!!!") > exten=> _1X.,2,Dial(Sip/${EXTEN}@RNKTEL-OUT,,gM(payasyougo^${CDR(accountcode)}^${CDR(userfield)})) > > exten=> _1X.,3,Hangup > > > On 5/30/07, Ricardo Martins < rpoppi77@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. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070530/4130e709/attachment.htm
Here is my CLI output:
Called 17142545587@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@gmail.com>
wrote:>
> Maybe its a bug in asterisk 1.4.2
>
> On 5/30/07, Rizwan Hisham <rizwanhasham@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@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.
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20070530/a61f18f0/attachment.htm
Rizwan Hisham wrote:> 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.Remove the "r" option from Dial. I assume you have the following: SIP Phone <-> Asterisk w/FXO card <-> POTS line If you are using AMP or any other GUI for Asterisk, then my advice is not valid, since those GUIs take over everything, hide the important stuff, and add options to Dial that you never see.
Nobody is using r option anywhere in my dialplan, thats 4 sure. And im also not using any PSTN line to connect to outside world. my system is based on voip only. SIP PHONE<---------->ASTERISK<---------->CARRIER-OUT No GUI is involved. i edit the conf files myself. On 5/31/07, Eric ManxPower Wieling <eric@fnords.org> wrote:> > Rizwan Hisham wrote: > > 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. > > Remove the "r" option from Dial. > > I assume you have the following: > > SIP Phone <-> Asterisk w/FXO card <-> POTS line > > If you are using AMP or any other GUI for Asterisk, then my advice is > not valid, since those GUIs take over everything, hide the important > stuff, and add options to Dial that you never see. > _______________________________________________ > --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/20070531/314e3a51/attachment.htm
Rizwan Hisham wrote:> Nobody is using r option anywhere in my dialplan, thats 4 sure. And im also > not using any PSTN line to connect to outside world. my system is based on > voip only. > > SIP PHONE<---------->ASTERISK<---------->CARRIER-OUTMy only idea is that the carrier might be using the "r" option. If they are then you should switch carriers. Also callprogress=yes might cause the problem you are experiencing, but I doubt this is the case.
Well, you r right. This was the carrier`s fault. Its been removed on our request and now we r okay. thanx to all. On 5/31/07, Eric ManxPower Wieling <eric@fnords.org> wrote:> > Rizwan Hisham wrote: > > Nobody is using r option anywhere in my dialplan, thats 4 sure. And im > also > > not using any PSTN line to connect to outside world. my system is based > on > > voip only. > > > > SIP PHONE<---------->ASTERISK<---------->CARRIER-OUT > > My only idea is that the carrier might be using the "r" option. If they > are then you should switch carriers. > > Also callprogress=yes might cause the problem you are experiencing, but > I doubt this is the case. > _______________________________________________ > --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/20070601/37ff9cee/attachment.htm
In my opinion, any carrier that adds "r" to a Dial line without a VERY, VERY good reason is not a carrier that I want to use. Using "r" is a classic newbie problem. It indicates a serious lack of understanding about Asterisk. Rizwan Hisham wrote:> Well, you r right. This was the carrier`s fault. Its been removed on our > request and now we r okay. thanx to all. > > On 5/31/07, Eric ManxPower Wieling <eric@fnords.org> wrote: >> >> Rizwan Hisham wrote: >> > Nobody is using r option anywhere in my dialplan, thats 4 sure. And im >> also >> > not using any PSTN line to connect to outside world. my system is based >> on >> > voip only. >> > >> > SIP PHONE<---------->ASTERISK<---------->CARRIER-OUT >> >> My only idea is that the carrier might be using the "r" option. If they >> are then you should switch carriers. >> >> Also callprogress=yes might cause the problem you are experiencing, but >> I doubt this is the case.
I agree with Eric. The situation gets worse when you comes to know that some bad carriers uses the "-r" statement to lead the user to think that its call is already ringing when it is, in fact, still looking for a circuit/network to connect.... Well, in any of those cases, the solution is simple: Stay far from them! Rgds, Ricardo Martins. Eric "ManxPower" Wieling escreveu:> In my opinion, any carrier that adds "r" to a Dial line without a > VERY, VERY good reason is not a carrier that I want to use. Using "r" > is a classic newbie problem. It indicates a serious lack of > understanding about Asterisk. > > Rizwan Hisham wrote: >> Well, you r right. This was the carrier`s fault. Its been removed on our >> request and now we r okay. thanx to all. >> >> On 5/31/07, Eric ManxPower Wieling <eric@fnords.org> wrote: >>> >>> Rizwan Hisham wrote: >>> > Nobody is using r option anywhere in my dialplan, thats 4 sure. >>> And im >>> also >>> > not using any PSTN line to connect to outside world. my system is >>> based >>> on >>> > voip only. >>> > >>> > SIP PHONE<---------->ASTERISK<---------->CARRIER-OUT >>> >>> My only idea is that the carrier might be using the "r" option. If >>> they >>> are then you should switch carriers. >>> >>> Also callprogress=yes might cause the problem you are experiencing, but >>> I doubt this is the case. > _______________________________________________ > --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 >