[asterisk-users] Problem with Atxfer for the calling party
Antonio Modesto
modesto at isimples.com.br
Fri Nov 11 05:40:37 CST 2011
Hello,
The exten is the parameter passed to the macro, which contains the
sip device name. I'll change the name to another less confusing.
* Alexandre, também sou brasileiro hehe, notei que você já escreveu um
livro sobre asterisk, será que você poderia me ajudar com esse problema?
Já tem alguns dias que estou na luta aqui hehe.
On Fri, 2011-11-11 at 08:45 -0200, Alexandre Keller wrote:
> You're using ${exten} inside your macro, you should use ${EXTEN}.
>
> --
> Atenciosamente,
>
> ALEXANDRE KELLER
>
>
> http://twitter.com/alexandrekeller
> http://www.facebook.com/alexandre.keller.BR
>
> "Dinheiro é a consequência de um trabalho bem feito e não o motivo
> para se fazer um bom trabalho."
>
>
> P Antes de imprimir pense em seu compromisso com o Meio Ambiente.
>
>
>
> On 11/11/2011, at 08:38, Antonio Modesto wrote:
>
>
>
> > On Mon, 2011-11-07 at 09:12 -0600, Danny Nicholas wrote:
> >
> > > It can have to do with either the telephones dial plan or the
> > > context in the Asterisk dial plan combined with your features.conf
> > > settings.
> >
> >
> > I noticed that my problem occurs when i use a macro to dial sip
> > devices, my dialplan is like this:
> >
> > - Each sip device has its own context
> > - This context includes the outgoing call contexts that this
> > extension can use for making calls and includes a context called
> > "ramais", which has the dial plan to call another extensions, it
> > uses a macro to do this.
> >
> > Here is the configuration for my extension "modesto" :
> >
> > # sip.conf
> > [modesto](default_extension)
> > username=modesto
> > context=modesto
> > callerid="modesto" <106>
> > callgroup=4
> > pickupgroup=4
> >
> > # Default extension template
> > type=friend
> > dtmfmode=auto
> > host=dynamic
> > disallow=all
> > allow=ulaw
> > allow=alaw
> > deny=0.0.0.0/0.0.0.0
> > permit=192.168.1.0/255.255.255.0
> > canreinvite=yes
> > qualify=no
> > callcounter=yes
> >
> >
> > # context for SIP/modesto
> > context modesto {
> > includes {
> > vivo;
> > tim;
> > oi;
> > claro;
> > vivoddd;
> > timddd;
> > oiddd;
> > claroddd;
> > embratel;
> > embratel2;
> > };
> > includes {
> > ramais;
> > };
> > };
> >
> > # Although the problem is occurring also for others contexts
> > included, i'll show only the "ramais" context, which is used to call
> > local extensions:
> >
> > context ramais {
> > 101 => &dial_sip(suporte1);
> > 102 => &dial_sip(suporte2);
> > 103 => &dial_sip(suporte3);
> > 105 => &dial_sip(suporte05);
> > 106 => &dial_sip(modesto);
> > 107 => &dial_sip(gustavo);
> > 108 => &dial_sip(pauloh);
> > 109 => &dial_sip(fernanda);
> > 111 => &dial_sip(marcos);
> > 112 => &dial_sip(thiago);
> > 115 => &dial_sip(helder);
> > 116 => &dial_sip(atendimento01);
> > 117 => &dial_sip(atendimento03);
> > 118 => &dial_sip(atendimento02);
> > 119 => &dial_sip(marlon);
> > 120 => &dial_sip(suporteemp);
> > 122 => &dial_sip(telemais);
> > 123 => &dial_sip(casagustavo);
> > 127 => &dial_sip(manutencao);
> > 128 => &dial_sip(guilherme);
> > 129 => &dial_sip(marcelo);
> > 130 => &dial_sip(rafael);
> > 132 => &dial_sip(netita2);
> > 133 => &dial_sip(unotel);
> >
> > };
> >
> > If I use the Dial() application instead of this macro, it works
> > well. I noticed that when I use the macro and try to transfer a call
> > (The problem occurs only for the calling party, the called party can
> > do transfers with no problems), asterisk tries to find the extension
> > in the <macro-name> context and of course, there is no dialplan to
> > call the extensions there.
> >
> >
> > Here is the dial_sip macro:
> >
> > macro dial_sip(exten) {
> > Verbose(2,"==> Chamando a MACRO dial_sip - ponto 1
> > macros.ael <==");
> > Verbose(4,"====> Macro dial_sip iniciada.");
> > ChanIsAvail(SIP/${exten});
> > Verbose(2,"==> ${AVAILORIGCHAN}");
> >
> > if ("${AVAILORIGCHAN}" != "")
> > {
> > Verbose(4,"====> SIP/${exten} parece estar
> > disponivel, vou disca-lo agora.");
> > Set(FromExt=${CALLERID(num)});
> > System(/bin/sh /var/spool/asterisk/calllog/log.sh
> > SIP/${FromExt} SIP/${exten} SIP-TO-SIP);
> > Verbose(4,"====> System status: ${SYSTEMSTATUS}");
> > Dial(SIP/${exten},${SIP_DIAL_TIMEOUT},Ttr);
> > Hangup();
> > }
> > else
> > {
> > Verbose(2,"====> SIP/${exten} nao esta
> > disponivel.");
> > Hangup();
> > };
> >
> > NoOp("From ${MACRO_EXTEN} to ${exten});
> > System(${CALLLOGDIR}/log.sh ${exten});
> >
> > return;
> > };
> >
> > Thanks in advance.
> >
> >
> >
> >
> > --
> > _____________________________________________________________________
> > -- 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
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20111111/dfb814a0/attachment-0001.htm>
More information about the asterisk-users
mailing list