[asterisk-users] Really WEIRD: can register but can not call!

[Pavel Jezek]

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)
>
>
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>
>
> 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
>
>
>
>
> _______________________________________________
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