>
> Hi
Thankx for reply.i am really new fro libss7.i am using asterisk 1.4.25 and
freepbx 2.4. currently using zaptel 1.4.is this pre request match for
install libss7.if there has any document please send me.
thanx
akalanka
>
>
> ________________________________
> From: aka <akablast at gmail.com>
> To: asterisk-ss7 at lists.digium.com
> Sent: Tuesday, May 26, 2009 12:57:30 PM
> Subject: [asterisk-ss7] send call via define cic range
>
> Hi
>
> i am using chan_ss7 1.1 to connect ss7 & asterisk.in my configuration
> there has a two links(E1 s) to connect asterisk ss7.is there has any way
> to handle outsides calls in link wise.i try to configure asterisk this way
> but no luck for this.(SS7/link1/$OUTNUM)
>
>
>
>
> -------------- next part --------------
> An HTML attachment was scrubbed...
> URL:
> http://lists.digium.com/pipermail/asterisk-ss7/attachments/20090526/b8dc4b40/attachment-0001.htm
>
> ------------------------------
>
> Message: 4
> Date: Tue, 26 May 2009 09:27:42 -0300
> From: Marcelo Pacheco <marcelo at m2j.com.br>
> Subject: Re: [asterisk-ss7] send call via define cic range
> To: asterisk-ss7 at lists.digium.com
> Message-ID: <4A1BE03E.5060009 at m2j.com.br>
> Content-Type: text/plain; charset=ISO-8859-1
>
> Dahdi groups is the feature you want.
> It's well documented in chan_dahdi.conf
>
> Antoine Megalla wrote:
> > Hi,
> >
> > I did not find a way to do that in chan_ss7.
> > However you can use the hunting_policy in the linkset definition to
> > try achieve that in a way
> >
> > you can use
> > hunting_policy => seq_lth
> > to use the first E1 to send calls
> >
> > if you want to use the second E1 then use
> > hunting_policy => seq_htl
> >
> > This will only ensure that the first 30 outbound calls go to the E1 of
> > your choice.
> > Of course you can set a global counter that you increment as the calls
> > are coming in and decrement on hangup and then check to make sure that
> > only 30 calls are sent. This will ensure that only 1 E1 is used for
> > oubound.
> >
> > Exmaple to use the first E1:
> >
> > in ss7.conf use
> > hunting_policy => seq_lth
> > in the linkset definition
> >
> > exten => _X.,1,SetGlobalVar(CALLCOUNTER=${MATH(${CALLCOUNTER}+1,int)})
> > exten => _X.,2,GoToIf($[${CALLCOUNTER} > 30] ? 3:4)
> > exten => _X.,3,Dial(SS7/MSC1/00${EXTEN}|60)
> > exten => _X.,4,Hangup()
> >
> > exten => h,1,SetGlobalVar(CALLCOUNTER=${MATH(${CALLCOUNTER}-1,int)})
> >
> > Regrads,
> >
> > Antoine Megalla.
> >
> > ------------------------------------------------------------------------
> > *From:* aka <akablast at gmail.com>
> > *To:* asterisk-ss7 at lists.digium.com
> > *Sent:* Tuesday, May 26, 2009 12:57:30 PM
> > *Subject:* [asterisk-ss7] send call via define cic range
> >
> > Hi
> >
> > i am using chan_ss7 1.1 to connect ss7 & asterisk.in
> > <http://asterisk.in/> my configuration there has a two links(E1 s) to
> > connect asterisk ss7.is <http://ss7.is/> there has any way to handle
> > outsides calls in link wise.i try to configure asterisk this way but
> > no luck for this.(SS7/link1/$OUTNUM)
> >
> > ------------------------------------------------------------------------
> >
> > _______________________________________________
> > --Bandwidth and Colocation Provided by http://www.api-digital.com--
> >
> > asterisk-ss7 mailing list
> > To UNSUBSCRIBE or update options visit:
> > http://lists.digium.com/mailman/listinfo/asterisk-ss7
>
>
>
>
> ------------------------------
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-ss7/attachments/20090527/07ae14ed/attachment.htm
|