[asterisk-users] Really WEIRD: can register but can not call!
ims.asuser ims.asuser
ims.asuser at gmail.com
Mon Aug 25 09:17:34 CDT 2008
That's right, I used a 'l' instead of '1'! Thank you.
I've made the modification on extension.conf (there's nothing to change on
the sip.conf) but the call can not go through...is there another file to
modify?
The new outcome is:
-- Executing Dial("SIP/105-6298", "SIP/l03") in new stack
Jan 1 00:54:38 WARNING[606]: chan_sip.c:1407 create_addr: No such host: l03
Jan 1 00:54:38 NOTICE[606]: app_dial.c:764 dial_exec: Unable to create
channel
of type 'SIP'
== Everyone is busy/congested at this time
-- Saved useragent "eyeBeam release 3010n stamp 19039" for peer 103
-- Timeout on SIP/105-6298
== CDR updated on SIP/105-6298
-- Executing Goto("SIP/105-6298", "#|1") in new stack
-- Goto (default,#,1)
-- Executing Playback("SIP/105-6298", "demo-thanks") in new stack
Jan 1 00:54:48 WARNING[606]: file.c:475 ast_openstream: File demo-thanks
does n
ot exist in any format
Jan 1 00:54:48 WARNING[606]: file.c:787 ast_streamfile: Unable to open
demo-tha
nks (format ulaw): No such file or directory
Jan 1 00:54:48 WARNING[606]: app_playback.c:83 playback_exec:
ast_streamfile fa
iled on SIP/105-6298 for demo-thanks
-- Executing Hangup("SIP/105-6298", "") in new stack
== Spawn extension (default, #, 2) exited non-zero on 'SIP/105-6298'
-- Saved useragent "eyeBeam release 3010n stamp 19039" for peer 105
extension.conf
[default]
exten => 103,1,Dial(SIP/103)
exten => 105,1,Dial(SIP/105)
Thank you all!
Khaldon
2008/8/25 Pavel Jezek <pavel.jezek at i.cz>
> 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 <ims.asuser at gmail.com>
> > Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
> > <asterisk-users at lists.digium.com>
> > To: asterisk-users at lists.digium.com
> > 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
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> > 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
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