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

[ims.asuser ims.asuser]

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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