[Asterisk-Users] Help with sip setup because can't receive
calls!!!!!!
Facundo Ameal
fameal at gmail.com
Tue Jan 31 11:15:52 MST 2006
i've tested it with this config files and i worked:
extensions.conf
exten => 55,1,Dial(SIP/2271,20)
sip.conf
[2271]
type=friend
host=dynamic
secret=sip
allow=all
qualify=200
nat=no
Instead of 2271 you can put whatever you want.
good luck.
2006/1/31, Facundo Ameal <fameal at gmail.com>:
> Are you using a SIP Softphone or an ATA?
>
> 2006/1/31, Facundo Ameal <fameal at gmail.com>:
> > does it registers well?
> > although i think you have to add "context=default" to the stargate1 section.
> >
> > try that and see what happens.
> >
> > 2006/1/31, abc def <xisterisk at yahoo.com>:
> > > Hi all, I am resending this message, so far no one has helped me with this
> > > incoming call issue. there is no problem with outbound call but there is no
> > > inbound call to my sip phone. the only message I get when I call from pstn
> > > is "unable to create local channel for call forward to
> > > 'Local/sipphone at default' (case =0)". my configuration files are attached
> > > below. any help would be greatly appreciated. many thanks in advance.
> > > ABC
> > >
> > > abc def <xisterisk at yahoo.com> wrote:
> > >
> > > there is no error message coming up on the pbx for in-bound calls (there is
> > > only debugging messages for outbound calls).
> > >
> > > thanks in advance for any hint or suggestion.
> > > Ama
> > >
> > > I just post my configuration file here for sip phone:
> > > extensions.conf
> > > -------------------------------------------------------------------------
> > > [globals]
> > > [default]
> > > include => incoming
> > > include => outgoing
> > > include => iax
> > > inculde => sip
> > > include => sccp
> > > [sip]
> > > exten => 2171,1,Dial(SIP/stargate1,20)
> > > ;exten => 2171,1,Dial(SIP/2171,20)
> > > exten => 2171,2,Hangup
> > > exten => 2172,1,Dial(SIP/stargate2,20)
> > > ;exten => 2172,1,Dial(SIP/2172,20)
> > > exten => 2172,2,Hangup
> > > exten => 2173,1,Dial(SIP/stargate3,20)
> > > ;exten => 2173,1,Dial(SIP/2173,20)
> > > exten => 2173,2,Hangup
> > > [sccp]
> > > [skinny]
> > > [incoming]
> > > exten => ; _214943[5-9]6,1,Dial(SIP/stargate3)
> > > exten => _214943[5-9]6,2,Hangup
> > > [outgoing]
> > > exten => _XXXXXXXX,1,Dial(Zap/g1/${EXTEN})
> > > exten => _XXXXXXXX,2,Hangup
> > > -------------------------------------------------------------------------
> > > sip.conf
> > > -------------------------------------------------------------------------
> > > [general]
> > > context=default ; Default context for incoming calls
> > > ; Set this to your host name or domain name
> > > bindport=5060 ; UDP Port to bind to (SIP standard port is
> > > 5060)
> > > bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to
> > > all)
> > > srvlookup=yes ; Enable DNS SRV lookups on outbound calls
> > >
> > > register => stargate1:1stargate at local_sip/2171
> > > register => stargate2:2stargate at local_sip/2172
> > > register => stargate3:3stargate at local_sip/2173
> > > ;---------------------------------------------- NAT SUPPORT
> > > ------------------------
> > > nat=no ; Global NAT settings (Affects all peers and
> > > users)
> > >
> > >
> > > [local_sip]
> > > type=friend
> > > host=10.47.200.136
> > > context=default
> > > [stargate1] ;cisco 9760
> > > ;[2171]
> > > type=friend
> > > host=dynamic ;10.47.200.140 ;dynamic
> > > defaultip=10.47.200.140
> > > username=stargate1
> > > secret=xxx
> > > callerid="21495071" <2171>
> > > allow=all
> > > qualify=200
> > > nat=no
> > > defaultip=10.47.200.140
> > >
> > > [stargate2] ;Polycom 601
> > > ;[2172]
> > > type=friend
> > > host=dynamic ;10.47.200.141 ;dynamic
> > > defaultip=10.47.200.141
> > > username=xxx
> > > secret=2stargate
> > > callerid="21495072" <2172>
> > > allow=all
> > > qualify=200
> > > nat=no
> > > defaultip=10.47.200.141
> > > [stargate3] ;Aastra 480i
> > > ;[2173]
> > > type=friend
> > > host=dynamic ;10.47.200.137 ;dynamic
> > > defaultip=10.47.200.137
> > > username=stargate3
> > > callerid="starg ate3" <2173>
> > > secret=xxx
> > > allow=all
> > > qualify=200
> > > nat=no
> > > defaultip=10.47.200.137
> > > ----------------------------------------------------------------------------
> > >
> > >
> > > pdhales at optusnet.com.au wrote:
> > >
> > > What error do you get when trying to call the SIP phones?
> > >
> > > PaulH
> > >
> > >
> > > ----- Original Message -----
> > > From: abc def
> > > To: asterisk-users at lists.digium.com
> > > Sent: Wednesday, January 25, 2006 11:58 PM
> > > Subject: [Asterisk-Users] Help with sip setup because can't receive calls
> > >
> > >
> > >
> > > Hi all,
> > > I read many posts on asterisk mail site and been trying many different
> > > things but still I can't get my sip phones to work with asterisk.
> > > I have a full blown-up voip netwok with two asterisk servers connected
> > > to pstn network with iax phones and cisco sccp phones which all work fine.
> > > however, I have been struggeling to configure my sip phones (polycom 601,
> > > Aastra 480i and cisco 9760) to work with asterisk. I can call out from sip
> > > phones to anywhere else but not receive phone calls. I can see the phones on
> > > "sip show registry" and "sip show peers" but no track phone calls for sip.
> > >
> > > can you please shed some light on me how to go about solving this
> > > problem?
> > >
> > > thank you and best regards,
> > > Ama
> > >
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> >
> >
> > --
> > Facundo Ameal.
> > fameal<at>gmail<dot>com
> > Linux User #395088
> >
> > FWD: 741664
> > MSN: asado<at>lamorcilla<dot>com<dot>ar
> > ICQ: 74005793
> >
> >
> > Open your mind, use open source.
> >
>
>
> --
> Facundo Ameal.
> fameal<at>gmail<dot>com
> Linux User #395088
>
> FWD: 741664
> MSN: asado<at>lamorcilla<dot>com<dot>ar
> ICQ: 74005793
>
>
> Open your mind, use open source.
>
--
Facundo Ameal.
fameal<at>gmail<dot>com
Linux User #395088
FWD: 741664
MSN: asado<at>lamorcilla<dot>com<dot>ar
ICQ: 74005793
Open your mind, use open source.
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