[Asterisk-Users] Help with sip setup because can't receive
calls!!!!!!
Facundo Ameal
fameal at gmail.com
Tue Jan 31 06:05:19 MST 2006
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
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