you should issue 'sip show peers' command to see, if your phones are available, put 'qualify=yes' in your sip.conf 'sip show registry' command is usefull to see if your _asterisk_ is registered to some another sip server, eg. voip provider.. PJ
David Boyd wrote: > -----Original Message----- > From: ims.asuser ims.asuser <[EMAIL PROTECTED]> > Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion > <[email protected]> > To: [email protected] > Subject: [asterisk-users] Really WEIRD: can register but can not call! > Date: Mon, 25 Aug 2008 12:26:45 +0200 > > Hi all, > > I have a very weird problem. > > I have 2 users (103 and 105). They are able to register in Asterisk, but > they can not call each other. > > Hereunder is the outcome: > > openwrt3*CLI> > -- Registered SIP '103' at 192.168.3.9 port 6127 expires 3600 > -- Saved useragent "eyeBeam release 3010n stamp 19039" for peer 103 > openwrt3*CLI> > openwrt3*CLI> > -- Registered SIP '105' at 192.168.3.6 port 8377 expires 3600 > -- Saved useragent "eyeBeam release 3010n stamp 19039" for peer 105 > openwrt3*CLI> > openwrt3*CLI> > -- Executing Dial("SIP/105-0ead", "SIP/l03") in new stack > Jan 1 00:19:26 WARNING[498]: chan_sip.c:1407 create_addr: No such host: > l03 > Jan 1 00:19:26 NOTICE[498]: app_dial.c:764 dial_exec: Unable to create > channel > of type 'SIP' > == Everyone is busy/congested at this time > openwrt3*CLI> > openwrt3*CLI> > -- Timeout on SIP/105-0ead > == CDR updated on SIP/105-0ead > -- Executing Goto("SIP/105-0ead", "#|1") in new stack > -- Goto (default,#,1) > -- Executing Playback("SIP/105-0ead", "demo-thanks") in new stack > Jan 1 00:19:36 WARNING[498]: file.c:475 ast_openstream: File > demo-thanks does n > ot exist in any format > Jan 1 00:19:36 WARNING[498]: file.c:787 ast_streamfile: Unable to open > demo-tha > nks (format ulaw): No such file or directory > Jan 1 00:19:36 WARNING[498]: app_playback.c:83 playback_exec: > ast_streamfile fa > iled on SIP/105-0ead for demo-thanks > -- Executing Hangup("SIP/105-0ead", "") in new stack > == Spawn extension (default, #, 2) exited non-zero on 'SIP/105-0ead' > > > The "show sip registry" command shows that no users are registered. > That's really WEIRD! > > > Please see the sip.conf and extension.conf files. > > sip.conf: > > [general] > context=default ; Default context for incoming calls > ;recordhistory=yes ; Record SIP history by default > ; (see sip history / sip no history) > ;realm=mydomain.tld ; Realm for digest authentication > ; defaults to "asterisk" > ; Realms MUST be globally unique > according to RF > ; Set this to your host name or domain > name > port=5060 ; UDP Port to bind to (SIP standard port > is 5060 > bindaddr=x.x.x.x ; IP address to bind to (0.0.0.0 binds to all) > srvlookup=yes ; Enable DNS SRV lookups on outbound > calls > ; Note: Asterisk only uses the first > host > ; in SRV records > ; Disabling DNS SRV lookups disables the > ; ability to place SIP calls based on > domain > ; names to some other SIP users on the > Internet > > [103] ; > ;Turn off silence suppression in X-Lite ("Transmit Silence"=YES)! > type=friend > username=103 ; Authorization User dans X-Lite > secret=1234 > callerid="Philippe" <103> ; nom et numéro affichés dans le X-Lite > appelé l > context=default > host=dynamic > nat=no ; X-Lite is behind a NAT router > canreinvite=no ; Typically set to NO if behind NAT > disallow=all ; désactive tous les codages sauf ceux spécifiés > ci-aprè > allow=gsm ; GSM consumes far less bandwidth than > ulaw > allow=ulaw > allow=alaw > > [105] ; > ;Turn off silence suppression in X-Lite ("Transmit Silence"=YES)! > type=friend > username=105 ; Authorization User dans X-Lite > secret=1234 > callerid="Khalid" <105> ; nom et numéro affichés dans le X-Lite > appelé lor > context=default > host=dynamic > nat=no ; X-Lite is behind a NAT router > canreinvite=no ; Typically set to NO if behind NAT > disallow=all ; désactive tous les codages sauf ceux spécifiés > ci-aprè > allow=ulaw > allow=alaw > > > extension.conf: > > [default] ; context par défaut des utilisateurs SIP répertoriés > dans sip.c > > > exten => 103,1,Dial(SIP/l03) > exten => 105,1,Dial(SIP/l05) > > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2008 - September 22 - 25 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > Your extensions are listed as SIP/l03 and SIP/l05 and should be SIP/103 and > SIP/105. Plus a problem with some recorded files. > > > Regards, > Dave > > > > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2008 - September 22 - 25 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
