<DIV id=RTEContent> <DIV>I am responding to my own problem because I found the answer finally which it may help others in future.</DIV> <DIV>I just had a break-through after 2 weeks struggling, finally I found the problem.</DIV> <DIV>the problem was in extensions.conf file. I misspelled "include" as "inculde" (finding a misspelled word in a long extensions.conf file is not so easy, trust me) but after checking "show dialplan" and going through it line by line, I found my sip sub-division is not there. </DIV><BR><BR><B><I>abc def <xisterisk@yahoo.com></I></B> wrote: <BLOCKQUOTE class=replbq style="PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #1010ff 2px solid"> <DIV>not sure but this is the output from the pbx:</DIV> <DIV> </DIV> <DIV>>sip show registry<BR>Host
Username Refresh State <BR>local_sip:5060 stargate3 105 Registered <BR>local_sip:5060 stargate2 105 Registered <BR>local_sip:5060 stargate1 105 Registered <BR></DIV> <DIV>from sip phone I can any other phone (cisco with sccp or iax protocol) but
I can't call any other sip phone, or receive phone calls.</DIV> <DIV><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">are you sure your sip phone is registering ok?<BR><BR>2006/2/1, abc def <XISTERISK@YAHOO.COM>:<BR>> Thanks Facundo for instruction but it didn't work. there is nothing new in<BR>> your suggestion compare to my conf files nevertheless I tried it but it<BR>> didn't work. I can make call from my sip phone but can't receive any phone<BR>> call. I am sure some one had had the same problem an d solved it.<BR>> as always I appreciate for your suggestion, advice and/or correction to my<BR>> config files.<BR>> if you know how to solve this problem please give me some hint.<BR>><BR>> thank you<BR>><BR>> Facundo Ameal <FAMEAL@GMAIL.COM>wrote:<BR>> 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 :<BR>> > Are you using a SIP Softphone or an ATA?<BR>> ><BR>> > 2006/1/31, Facundo Ameal :<BR>> > > does it registers well?<BR>> > > although i think you have to add "context=default" to the stargate1<BR>> section.<BR>> ; > ><BR>> > > try that and see what happens.<BR>> > ><BR>> > > 2006/1/31, abc def :<BR>> > > > Hi all, I am resending this message, so far no one has helped me with<BR>> this<BR>> > > > incoming call issue. there is no problem with outbound call but there<BR>> is no<BR>> >
> > inbound call to my sip phone. the only message I get when I call from<BR>> pstn<BR>> > > > is "unable to create local channel for call forward to<BR>> > > > 'Local/sipphone@default' (case =0)". my configuration files are<BR>> attached<BR>> > > > below. any help would be greatly appreciated. many thanks in advance.<BR>> > > > ABC<BR>> > > ><BR>> > > > abc def wrote:<BR>> > > ><BR>> > > > there is no error message coming up on the pbx for in-bound calls<BR>> (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>>
-------------------------------------------------------------------------<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>> & gt; > > exten => 2172,1,Dial(SIP/stargate2,20)<BR>><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>> -------------------------------------------------------------------------<BR>> > > > sip.conf<BR>> > > ><BR>> -------------------------------------------------------------------------<BR>> > > > [general]<BR>> > > > context=default ; Default context for incoming calls<BR>&g t; > > > ; 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<BR>> 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>><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>> > &g t; > 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>> > > ><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<BR>> calls<BR>> >
> ><BR>> > > ><BR>> > > ><BR>> > > > Hi all,<BR>> > > > I read many posts on asterisk mail site and been trying many<BR>> 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<BR>> fine.<BR>> > > > however, I have been struggeling to configure my sip phones (polycom<BR>> 601,<BR>> > > > Aastra 480i and cisco 9760) to work with aster isk. I can call out from<BR>> sip<BR>> > > > phones to anywhere else but not receive phone calls. I can see the<BR>> phones on<BR>> > > > "sip show registry" and "sip show peers" but no track phone calls for<BR>> 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>> > > ><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>> > > ><BR>> http://lists.digium.com/mailman/listinfo/asterisk-users<BR>> > > ><BR>> > > ><BR>> > > ><BR>> > > > ________________________________<BR>> > > > Bring words and photos to gether (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>> > > ><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<BR>> 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>> & gt; > ><BR>> http://lists.digium.com/mailman/listinfo/asterisk-users<BR>> > > ><BR>> > > ><BR>> > > ><BR>> > ><BR>> > ><BR>> > > --<BR>> > > Facundo Ameal.<BR>> > > famealgmailcom<BR>> > > Linux User #395088<BR>> > ><BR>> > > FWD: 741664<BR>> > > MSN: asadolamorcillacomar<BR>> > > ICQ: 74005793<BR>> > ><BR>> > ><BR>> > > Open your mind, use open
source.<BR>> > ><BR>> ><BR>> ><BR>> > --<BR>> > Facundo Ameal.<BR>> > famealgmailcom<BR>> > Linux User #395088<BR>> ><BR>> > FWD: 741664<BR>> > MSN: asadolamorcillacomar<BR>> > ICQ: 74005793<BR>> ><BR>> ><BR>> > Open your mind, use open source.<BR>> ><BR>><BR>><BR>> --<BR>> Facundo Ameal.<BR>> famealgmailcom<BR>> Linux User #395088<BR>><BR>> FWD: 741664<BR>> MSN: asadolamorcillacomar<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>><BR>><BR>><BR>><BR>> ________________________________<BR>><BR>> What are the most popular cars? Find out at
Yahoo! Autos<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>--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> <div> <HR SIZE=1> <A href="http://us.rd.yahoo.com/evt=38381/%20ylc=X3oDMTEzcGlrdGY5BF9TAzk3MTA3MDc2BHNlYwNtYWlsdGFncwRzbGsDMWF1dG9z/*http://autos.yahoo.com/index.html">Yahoo!
Autos</A>. Looking for a sweet ride? Get pricing, reviews, & more on new and used cars._______________________________________________<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><BR></DIV><p>
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