thanks.
2008/3/10, Noah Miller <[EMAIL PROTECTED]>:
> Hi Daniel -
>
> > Thank you for guide most things become cleare. No I dont need the dial tone.
> > When I pickup XLITE to dial a number I hear dialtone and after I enter
> > number nothing happens, this behaviar was strange for me, exactly
>
Hi Daniel -
> Thank you for guide most things become cleare. No I dont need the dial tone.
> When I pickup XLITE to dial a number I hear dialtone and after I enter
> number nothing happens, this behaviar was strange for me, exactly
> becase you said I have analog phone in mind :)
The only thin
Thank you for guide most things become cleare. No I dont need the dial tone.
When I pickup XLITE to dial a number I hear dialtone and after I enter
number nothing happens, this behaviar was strange for me, exactly
becase you said I have analog phone in mind :)
2008/3/9, Tzafrir Cohen <[EMAIL PROT
On Sun, Mar 09, 2008 at 10:19:05AM +0400, Daniel Suleyman wrote:
> ok, then I'm not understanding something.
> How I can call with xlite to my Asterisk not sending extension?
> when I want to call I pick up phone, hear ring (piii) and I need
> to type some extension otherwise nothing hapens
On Sunday 09 March 2008 00:19:05 Daniel Suleyman wrote:
> ok, then I'm not understanding something.
> How I can call with xlite to my Asterisk not sending extension?
> when I want to call I pick up phone, hear ring (piii) and I need
> to type some extension otherwise nothing hapens
SIP doe
ok, then I'm not understanding something.
How I can call with xlite to my Asterisk not sending extension?
when I want to call I pick up phone, hear ring (piii) and I need
to type some extension otherwise nothing hapens
2008/3/8, Tilghman Lesher <[EMAIL PROTECTED]>:
> On Saturday 08 March 2
On Saturday 08 March 2008 02:52:39 Daniel Suleyman wrote:
> Here is the log, my extensions is in the default section
> INVITE sip:[EMAIL PROTECTED] SIP/2.0
> [ Context 'default' created by 'pbx_config' ]
> '7007' => 1. Goto(7007-${CNT}|1)
> [pbx_config] '7007-2' =
Here is the log, my extensions is in the default section
*CLI> core set verbose 3
Verbosity is at least 3
*CLI> [Mar 5 15:21:43] NOTICE[15870]: chan_sip.c:13879
handle_request_invite: Call from '7007' to extension '999' rejected
because extension not found.
*CLI> sip set debug
SIP Debugging enabl
On Sat, Mar 08, 2008 at 11:45:14AM +0400, Daniel Suleyman wrote:
> Even if I have s in defult it is not work.
So please provide a trace of that case:
core set verbose 3
And see what happens.
--
Tzafrir Cohen
icq#16849755 jabber:[EMAIL PROTECTED]
+972-50-7952406
Even if I have s in defult it is not work.
2008/3/6, Noah Miller <[EMAIL PROTECTED]>:
> Hi -
>
>
> > Thank you all for answers. As I understand s - i and others is
device specific.
> > I will not need them in my SIP configuration.
>
>
> The "s" extension is not zap-specific.
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