[asterisk-users] Problem with Atxfer for the calling party
Alexandre Keller
alexandrekeller at gmail.com
Fri Nov 11 04:45:46 CST 2011
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
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20111111/5dec140e/attachment.htm>
More information about the asterisk-users
mailing list