Good Morning,
I have an asterisk18-1.8.7.1 running on a FreeBSD 8.2-STABLE, and it
is working well so far, i'm just having some problems with atxfer.
I have written this macro to dial sip extensions:
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;
};
It is working, but the calling party is not able to transfer the calls
because asterisk doesn't wait all the digits be typed, it tries to
transfer the call when the first digit is pressed (We use 3 digits
extensions):
[Oct 31 09:04:01] WARNING[2926]: features.c:2315 builtin_atxfer:
Extension '1' does not exist in context 'dial_sip'
== Spawn extension (dial_sip, ~~s~~, 11) exited non-zero on
'SIP/modesto-0000000d'
[Oct 31 09:04:03] WARNING[2926]: features.c:2319 builtin_atxfer: No
digits dialed for atxfer.
Does anyone have suggestions?
Regards.
--
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