<DIV>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. </DIV> <DIV>as always I appreciate for your suggestion, advice and/or correction to my config files.</DIV> <DIV>if you know how to solve this problem please give me some hint.</DIV> <DIV> </DIV> <DIV>thank you<BR><BR><B><I>Facundo Ameal <fameal@gmail.com></I></B> wrote:</DIV> <BLOCKQUOTE class=replbq style="PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #1010ff 2px solid">i've tested it with this config files and i worked:<BR><BR>extensions.conf<BR><BR>exten => 55,1,Dial(SIP/2271,20)<BR><BR><BR>sip.conf<BR><BR>[2271]<BR>type=friend<BR>host=dynamic<BR>secret=sip<BR>allow=all<BR>qualify=200<BR>nat=no<BR><BR><BR>Instead of 2271 you can put whatever you want.<BR><BR>good
luck.<BR><BR><BR><BR>2006/1/31, Facundo Ameal <FAMEAL@GMAIL.COM>:<BR>> Are you using a SIP Softphone or an ATA?<BR>><BR>> 2006/1/31, Facundo Ameal <FAMEAL@GMAIL.COM>:<BR>> > does it registers well?<BR>> > although i think you have to add "context=default" to the stargate1 section.<BR>> ><BR>> > try that and see what happens.<BR>> ><BR>> > 2006/1/31, abc def <XISTERISK@YAHOO.COM>:<BR>> > > Hi all, I am resending this message, so far no one has helped me with this<BR>> > > incoming call issue. there is no problem with outbound call but there is no<BR>> > > inbound call to my sip phone. the only message I get when I call from pstn<BR>> > > is "unable to create local channel for call forward to<BR>> > > 'Local/sipphone@default' (case =0)". my configuration files are attached<BR>> > > below. any help would be greatly appreciated. many thanks in advance.<BR>> > > ABC<BR>> >
><BR>> > > abc def <XISTERISK@YAHOO.COM>wrote:<BR>> > ><BR>> > > there is no error message coming up on the pbx for in-bound calls (there is<BR>> > > only debugging messages for outbound calls).<BR>> > ><BR>> > > thanks in advance for any hint or suggestion.<BR>> > > Ama<BR>> > ><BR>> > > I just post my configuration file here for sip phone:<BR>> > > extensions.conf<BR>> > > -------------------------------------------------------------------------<BR>> > > [globals]<BR>> > > [default]<BR>> > > include => incoming<BR>> > > include => outgoing<BR>> > > include => iax<BR>> > > inculde => sip<BR>> > > include => sccp<BR>> > > [sip]<BR>> > > exten => 2171,1,Dial(SIP/stargate1,20)<BR>> > > ;exten => 2171,1,Dial(SIP/2171,20)<BR>> > > exten => 2171,2,Hangup<BR>> >
> exten => 2172,1,Dial(SIP/stargate2,20)<BR>> > > ;exten => 2172,1,Dial(SIP/2172,20)<BR>> > > exten => 2172,2,Hangup<BR>> > > exten => 2173,1,Dial(SIP/stargate3,20)<BR>> > > ;exten => 2173,1,Dial(SIP/2173,20)<BR>> > > exten => 2173,2,Hangup<BR>> > > [sccp]<BR>> > > [skinny]<BR>> > > [incoming]<BR>> > > exten => ; _214943[5-9]6,1,Dial(SIP/stargate3)<BR>> > > exten => _214943[5-9]6,2,Hangup<BR>> > > [outgoing]<BR>> > > exten => _XXXXXXXX,1,Dial(Zap/g1/${EXTEN})<BR>> > > exten => _XXXXXXXX,2,Hangup<BR>> > > -------------------------------------------------------------------------<BR>> > > sip.conf<BR>> > > -------------------------------------------------------------------------<BR>> > > [general]<BR>> > > context=default ; Default context for incoming calls<BR>> > > ; Set this to your
host name or domain name<BR>> > > bindport=5060 ; UDP Port to bind to (SIP standard port is<BR>> > > 5060)<BR>> > > bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to<BR>> > > all)<BR>> > > srvlookup=yes ; Enable DNS SRV lookups on outbound calls<BR>> > ><BR>> > > register => stargate1:1stargate@local_sip/2171<BR>> > > register => stargate2:2stargate@local_sip/2172<BR>> > > register => stargate3:3stargate@local_sip/2173<BR>> > > ;---------------------------------------------- NAT SUPPORT<BR>> > > ------------------------<BR>> > > nat=no ; Global NAT settings (Affects all peers and<BR>> > > users)<BR>> > ><BR>> > ><BR>> > > [local_sip]<BR>> > > type=friend<BR>> > > host=10.47.200.136<BR>> > > context=default<BR>> > > [stargate1] ;cisco 9760<BR>> > > ;[2171]<BR>> > >
type=friend<BR>> > > host=dynamic ;10.47.200.140 ;dynamic<BR>> > > defaultip=10.47.200.140<BR>> > > username=stargate1<BR>> > > secret=xxx<BR>> > > callerid="21495071" <2171><BR>> > > allow=all<BR>> > > qualify=200<BR>> > > nat=no<BR>> > > defaultip=10.47.200.140<BR>> > ><BR>> > > [stargate2] ;Polycom 601<BR>> > > ;[2172]<BR>> > > type=friend<BR>> > > host=dynamic ;10.47.200.141 ;dynamic<BR>> > > defaultip=10.47.200.141<BR>> > > username=xxx<BR>> > > secret=2stargate<BR>> > > callerid="21495072" <2172><BR>> > > allow=all<BR>> > > qualify=200<BR>> > > nat=no<BR>> > > defaultip=10.47.200.141<BR>> > > [stargate3] ;Aastra 480i<BR>> > > ;[2173]<BR>> > > type=friend<BR>> > > host=dynamic ;10.47.200.137 ;dynamic<BR>> > >
defaultip=10.47.200.137<BR>> > > username=stargate3<BR>> > > callerid="starg ate3" <2173><BR>> > > secret=xxx<BR>> > > allow=all<BR>> > > qualify=200<BR>> > > nat=no<BR>> > > defaultip=10.47.200.137<BR>> > > ----------------------------------------------------------------------------<BR>> > ><BR>> > ><BR>> > > pdhales@optusnet.com.au wrote:<BR>> > ><BR>> > > What error do you get when trying to call the SIP phones?<BR>> > ><BR>> > > PaulH<BR>> > ><BR>> > ><BR>> > > ----- Original Message -----<BR>> > > From: abc def<BR>> > > To: asterisk-users@lists.digium.com<BR>> > > Sent: Wednesday, January 25, 2006 11:58 PM<BR>> > > Subject: [Asterisk-Users] Help with sip setup because can't receive calls<BR>> > ><BR>> > ><BR>> > ><BR>> > > Hi all,<BR>>
> > I read many posts on asterisk mail site and been trying many different<BR>> > > things but still I can't get my sip phones to work with asterisk.<BR>> > > I have a full blown-up voip netwok with two asterisk servers connected<BR>> > > to pstn network with iax phones and cisco sccp phones which all work fine.<BR>> > > however, I have been struggeling to configure my sip phones (polycom 601,<BR>> > > Aastra 480i and cisco 9760) to work with asterisk. I can call out from sip<BR>> > > phones to anywhere else but not receive phone calls. I can see the phones on<BR>> > > "sip show registry" and "sip show peers" but no track phone calls for sip.<BR>> > ><BR>> > > can you please shed some light on me how to go about solving this<BR>> > > problem?<BR>> > ><BR>> > > thank you and best regards,<BR>> > > Ama<BR>> > ><BR>> > > < HR SIZE=1> Do you
Yahoo!?<BR>> > > With a free 1 GB, there's more in store with Yahoo! Mail.<BR>> > > ________________________________<BR>> > ><BR>> > > _______________________________________________<BR>> > > --Bandwidth and Colocation provided by Easynews.com --<BR>> > ><BR>> > > Asterisk-Users mailing list<BR>> > > To UNSUBSCRIBE or update options visit:<BR>> > ><BR>> > > http://lists.digium.com/mailman/listinfo/asterisk-users<BR>> > > _______________________________________________<BR>> > > --Bandwidth and Colocation provided by Easynews.com --<BR>> > ><BR>> > > Asterisk-Users mailing list<BR>> > > To UNSUBSCRIBE or update options visit:<BR>> > > http://lists.digium.com/mailman/listinfo/asterisk-users<BR>> > ><BR>> > ><BR>> > ><BR>> > > ________________________________<BR>> > > Bring words and photos together
(easily) with<BR>> > > PhotoMail - it's free and works with Yahoo!<BR>> > > Mail._______________________________________________<BR>> > > --Bandwidth and Colocation provided by Easynews.com --<BR>> > ><BR>> > > Asterisk-Users mailing list<BR>> > > To UNSUBSCRIBE or update options visit:<BR>> > > http://lists.digium.com/mailman/listinfo/asterisk-users<BR>> > ><BR>> > ><BR>> > ><BR>> > ><BR>> > > ________________________________<BR>> > > Yahoo! Autos. Looking for a sweet ride? Get pricing, reviews, & more on new<BR>> > > and used cars.<BR>> > ><BR>> > ><BR>> > > _______________________________________________<BR>> > > --Bandwidth and Colocation provided by Easynews.com --<BR>> > ><BR>> > > Asterisk-Users mailing list<BR>> > > To UNSUBSCRIBE or update options visit:<BR>> > ><BR>> >
> http://lists.digium.com/mailman/listinfo/asterisk-users<BR>> > ><BR>> > ><BR>> > ><BR>> ><BR>> ><BR>> > --<BR>> > Facundo Ameal.<BR>> > fameal<AT>gmail<DOT>com<BR>> > Linux User #395088<BR>> ><BR>> > FWD: 741664<BR>> > MSN: asado<AT>lamorcilla<DOT>com<DOT>ar<BR>> > ICQ: 74005793<BR>> ><BR>> ><BR>> > Open your mind, use open source.<BR>> ><BR>><BR>><BR>> --<BR>> Facundo Ameal.<BR>> fameal<AT>gmail<DOT>com<BR>> Linux User #395088<BR>><BR>> FWD: 741664<BR>> MSN: asado<AT>lamorcilla<DOT>com<DOT>ar<BR>> ICQ: 74005793<BR>><BR>><BR>> Open your mind, use open source.<BR>><BR><BR><BR>--<BR>Facundo Ameal.<BR>fameal<AT>gmail<DOT>com<BR>Linux User #395088<BR><BR>FWD: 741664<BR>MSN: asado<AT>lamorcilla<DOT>com<DOT>ar<BR>ICQ: 74005793<BR><BR><BR>Open your mind, use open
source.<BR>_______________________________________________<BR>--Bandwidth and Colocation provided by Easynews.com --<BR><BR>Asterisk-Users mailing list<BR>To UNSUBSCRIBE or update options visit:<BR>http://lists.digium.com/mailman/listinfo/asterisk-users<BR></BLOCKQUOTE> <DIV><BR></DIV><p>
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