Are you using a SIP Softphone or an ATA? 2006/1/31, Facundo Ameal <[EMAIL PROTECTED]>: > 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 <[EMAIL PROTECTED]>: > > 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/[EMAIL PROTECTED]' (case =0)". my configuration files are attached > > below. any help would be greatly appreciated. many thanks in advance. > > ABC > > > > abc def <[EMAIL PROTECTED]> 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:[EMAIL PROTECTED]/2171 > > register => stargate2:[EMAIL PROTECTED]/2172 > > register => stargate3:[EMAIL PROTECTED]/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 > > ---------------------------------------------------------------------------- > > > > > > [EMAIL PROTECTED] wrote: > > > > What error do you get when trying to call the SIP phones? > > > > PaulH > > > > > > ----- Original Message ----- > > From: abc def > > To: asterisk-users@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 > > > > < HR SIZE=1> Do you Yahoo!? > > With a free 1 GB, there's more in store with Yahoo! Mail. > > ________________________________ > > > > _______________________________________________ > > --Bandwidth and Colocation provided by Easynews.com -- > > > > Asterisk-Users mailing list > > To UNSUBSCRIBE or update options visit: > > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > _______________________________________________ > > --Bandwidth and Colocation provided by Easynews.com -- > > > > Asterisk-Users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > > ________________________________ > > Bring words and photos together (easily) with > > PhotoMail - it's free and works with Yahoo! > > Mail._______________________________________________ > > --Bandwidth and Colocation provided by Easynews.com -- > > > > Asterisk-Users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > > > > ________________________________ > > Yahoo! Autos. Looking for a sweet ride? Get pricing, reviews, & more on new > > and used cars. > > > > > > _______________________________________________ > > --Bandwidth and Colocation provided by Easynews.com -- > > > > Asterisk-Users mailing list > > To UNSUBSCRIBE or update options visit: > > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > > > -- > 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. _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users