[asterisk-users] Two lines for outgoing calls

[Jonathan GF]

Dominik,

apart from the good responses, please get rid of the 't' in the options of
dial or you will be allowing the called party to transfer the call while you
are paying.

Regards,

Jonathan GF



On Dec 26, 2007 3:32 PM, Dominik Zalewski <dzalewski at open-craft.com> wrote:

> Dear All,
>
> I'm using Asterisk 1.4.16.2 with Zaptel 1.4.7 on Debian with kernel
> 2.6.18.
>
> I have two analog lines Zap/1 and Zap/2 as group 1 in zapata.conf. I'm
> using below context for dialing out.
>
> [outbound-local]
> exten => _9XXX,1,Dial(Zap/g1/${EXTEN:1},30,tTr)
> exten => _9XXXXX,1,Dial(Zap/g1/${EXTEN:1},30,tTr)
> exten => _9ZXXXXXX,1,Dial(Zap/g1/${EXTEN:1},30,tTr)
> exten => _9ZXXXXXXX,1,Dial(Zap/g1/${EXTEN:1},30,tTr)
> exten => _90900XXXXX,1,Dial(Zap/g1/${EXTEN:1},30m,tTr)
>
> When Zap/1 is busy and I try to call, it will use Zap/2 which is fine
> but there is something wrong cause I hear one ring and then a weird
> sound like a noise or something and then hangup. I have to reload zaptel
> modules and then everything works fine for a while.
>
>   -- Executing [9150 at from-internal:1] Dial("SIP/200-08221590",
> "Zap/g1/150|30|tTr") in new stack
>    -- Called g1/150
>    -- Zap/2-1 answered SIP/200-08221590
>    -- Hungup 'Zap/2-1'
>
> I even thought that second fxo module is broken so I changed it. No
> results.
>
> Any ideas?
>
> Thank you in advance,
>
> Dominik
>
>
> _______________________________________________
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> 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/20080110/26dbdf71/attachment.htm