[asterisk-users] Really WEIRD: can register but can not call!
David Boyd
dboyd at ignitetrx.com
Mon Aug 25 09:35:28 CDT 2008
You need to reload the configurations, either by reload command or
restart asterisk.
Dave
-----Original Message-----
From: ims.asuser ims.asuser <ims.asuser at gmail.com>
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Cc: dboyd at ignitetrx.com, shariqrazakhan at gmail.com, pavel.jezek at i.cz
Subject: Re: [asterisk-users] Really WEIRD: can register but can not
call!
Date: Mon, 25 Aug 2008 16:17:34 +0200
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)
>
>
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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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