[asterisk-users] DID is not working (call is not routing)

[R.R Libera]

Hello Chandra,

What about Teliax´s service? Is it recommended? How´s their call quality?
Thanks in advance...




On 10/10/06, Crazy Boy <crazymoonboy at yahoo.com> wrote:
>
> Hi William,
>
> My DID is working and am receiving calls. The problem is with Teliax
> settings from their end. Thank you for spending your valuable time for me.
>
> Regards,
> Chandra.
>
> *William Piper <william.piper at gmail.com>* wrote:
>
> Your server seems to be doing exactly what you are telling it to do:
>
>  -- Executing Playback("SIP/216.89.79.2-09e1d020", "ss-noservice") in new
> stack
>  -- Playing 'ss-noservice' (language 'en')
>
> Read the extensions.conf directions on the wiki site:
>
> http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20extensions.conf
>
> bp
>
>
> On 10/8/06, Crazy Boy <crazymoonboy at yahoo.com> wrote:
> >
> > Hi,
> >
> > I have created SIP extenstions and created Teliax Trunk using IAX2. I am
> > making outgoing calls to USA successfully.
> >
> > When I am making a call to my DID number from outside, its telling that "The
> > number you have dialed is not inservice". Here I am giving the output
> > from Asterisk server console:
> >
> > *CLI>
> >     -- IAX2/teliax-2 answered SIP/350-09e3b540
> >     -- Executing GotoIf("SIP/216.89.79.2 -09e1d020", "0?from-trunk||1")
> > in new stack
> >     -- Executing Set("SIP/216.89.79.2-09e1d020", "TIMEOUT(absolute)=15")
> > in new stack
> >     -- Channel will hangup at 2006-10-06 11:27:55 UTC.
> >     -- Executing Answer("SIP/216.89.79.2-09e1d020", "") in new stack
> >     -- Executing Wait("SIP/216.89.79.2-09e1d020", "2") in new stack
> >     -- Executing Playback("SIP/216.89.79.2-09e1d020", "ss-noservice") in
> > new stack
> >     -- Playing 'ss-noservice' (language 'en')
> >     -- Executing Congestion("SIP/216.89.79.2-09e1d020", "") in new stack
> >
> >   == Spawn extension (from-sip-external, s, 6) exited non-zero on
> > 'SIP/216.89.79.2-09e1d020'
> >     -- Executing NoOp("SIP/216.89.79.2-09e1d020", "Hangup") in new stack
> >     -- Executing Set("SIP/216.89.79.2-09e1d020", "DID=s") in new stack
> >     -- Executing Goto("SIP/216.89.79.2-09e1d020", "s|1") in new stack
> >     -- Goto (from-sip-external,s,1)
> >     -- Executing GotoIf("SIP/216.89.79.2-09e1d020", "0?from-trunk|s|1")
> > in new stack
> >     -- Executing Set("SIP/216.89.79.2-09e1d020", "TIMEOUT(absolute)=15")
> > in new stack
> >     -- Channel will hangup at 2006-10-06 11:28:04 UTC.
> >     -- Executing Answer("SIP/216.89.79.2-09e1d020", "") in new stack
> >   == Spawn extension (from-sip-external, s, 3) exited non-zero on
> > 'SIP/216.89.79.2-09e1d020'
> >
> > When I am calling from outside phone, call is coming to my server and is
> > not routing. I am making calls to USA and between SIP extensions
> > successfully.  Please tell me the solution. Looking forward to your
> > response. Thank you.
> >
> > Regards,
> > Chandra.
> >  ------------------------------
> > Talk is cheap. Use Yahoo! Messenger to make PC-to-Phone calls. Great
> > rates starting at 1¢/min.
> > <http://us.rd.yahoo.com/mail_us/taglines/postman7/*http://us.rd.yahoo.com/evt=39666/*http://messenger.yahoo.com>
> >
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