[Asterisk-Users] Help with sip setup because can't receive calls!!!!!!

Facundo Ameal fameal at gmail.com
Wed Feb 1 06:38:09 MST 2006 

are you sure your sip phone is registering ok?

2006/2/1, abc def <xisterisk at yahoo.com>:
> Thanks Facundo for instruction but it didn't work. there is nothing new in
> your suggestion compare to my conf files nevertheless I tried it but it
> didn't work. I can make call from my sip phone but can't receive any phone
> call. I am sure some one had had the same problem and solved it.
> as always I appreciate for your suggestion, advice and/or correction to my
> config files.
> if you know how to solve this problem please give me some hint.
>
> thank you
>
> Facundo Ameal <fameal at gmail.com> wrote:
> 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 :
> > Are you using a SIP Softphone or an ATA?
> >
> > 2006/1/31, Facundo Ameal :
> > > 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 :
> > > > 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 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
> & gt; > > 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]
> > > &gt ; 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.
> > > famealgmailcom
> > > Linux User #395088
> > >
> > > FWD: 741664
> > > MSN: asadolamorcillacomar
> > > ICQ: 74005793
> > >
> > >
> > > Open your mind, use open source.
> > >
> >
> >
> > --
> > Facundo Ameal.
> > famealgmailcom
> > Linux User #395088
> >
> > FWD: 741664
> > MSN: asadolamorcillacomar
> > ICQ: 74005793
> >
> >
> > Open your mind, use open source.
> >
>
>
> --
> Facundo Ameal.
> famealgmailcom
> Linux User #395088
>
> FWD: 741664
> MSN: asadolamorcillacomar
> ICQ: 74005793
>
>
> Open your mind, use open source.
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> To UNSUBSCRIBE or update options visit:
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>
>
>
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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.

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