[asterisk-users] Problem with Atxfer for the calling party

[Antonio Modesto]

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.


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