[Asterisk-Users] Help with sip setup because can't receive
calls!!!!!!
Facundo Ameal
fameal at gmail.com
Tue Jan 31 06:07:35 MST 2006
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.
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