[asterisk-users] Problem with Atxfer for the calling party
Antonio Modesto
modesto at isimples.com.br
Tue Dec 13 13:07:57 CST 2011
On Tue, 2011-12-13 at 16:35 -0200, Roberto Linck wrote:
> Hi Antonio,
>
>
>
> I'd never had used extensions.ael but in extensions.conf, using Macro
> I always set '__TRANSFER_CONTEXT' to the same context of exten and it
> works well.
Thanks, it worked, the MACRO_CONTEXT variable was empty, I've set the
value to my extensions context and it worked fine.
Thanks.
>
>
> 2011/12/13 Antonio Modesto <modesto at isimples.com.br>
>
> Hello everybody,
>
> I found that if i write my macro in the extensions.conf
> (not in ael), the atxfer works well, the problem is that ael
> uses gosub instead of the Macro() application, which doesn't
> change the current context. Does anybody know if i can do
> anything to solve this? I know if i rewrite all my macros in
> the common way, it will work, but that's a lot of coding for
> me.
>
>
>
>
> On Mon, 2011-12-12 at 08:57 -0200, Antonio Modesto wrote:
>
> > Nothing?
> >
> >
> > On Mon, 2011-11-21 at 16:21 -0200, Antonio Modesto wrote:
> >
> > >
> > >
> > >
> > >
> > > Hi There,
> > >
> > > I'm still having this problem, Does somebody know
> > > what can be happening?
> > >
> > >
> > > Regards.
> > >
> > > On Fri, 2011-11-11 at 09:40 -0200, Antonio Modesto wrote:
> > >
> > > > 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
> > > >
> > > >
> > > >
> > > > --
> > > > _____________________________________________________________________
> > > > -- 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
> >
> >
> >
> > --
> > _____________________________________________________________________
> > -- 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
>
>
>
>
>
>
>
> --
> Atenciosamente
>
> ____________________
> Roberto Linck
> robertolinck at gmail.com
> (51) 8140-1372
>
>
> --
> _____________________________________________________________________
> -- 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/20111213/29bbdd55/attachment-0001.htm>
More information about the asterisk-users
mailing list