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ims.asuser ims.asuser

2008-Aug-25 10:26 UTC

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

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 num?ro affich?s dans le
X-Lite
appel? 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            ; d?sactive tous les codages sauf ceux sp?cifi?s
ci-apr?
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 num?ro affich?s dans le
X-Lite appel?
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            ; d?sactive tous les codages sauf ceux sp?cifi?s
ci-apr?
allow=ulaw
allow=alaw


extension.conf:

[default]       ; context par d?faut des utilisateurs SIP r?pertori?s dans
sip.c


exten => 103,1,Dial(SIP/l03)
exten => 105,1,Dial(SIP/l05)
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Shariq Khan

2008-Aug-25 10:38 UTC

head link

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

My Dear,

You have used 'L03' (alphabet 'l' *EL*) in dial command instead
of '103'.


Shariq


On Mon, Aug 25, 2008 at 3:26 PM, ims.asuser ims.asuser <ims.asuser at
gmail.com> wrote:
> 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 num?ro affich?s
dans le X-Lite
> appel? 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            ; d?sactive tous les codages sauf ceux sp?cifi?s
> ci-apr?
> 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 num?ro affich?s dans
le X-Lite
> appel? 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            ; d?sactive tous les codages sauf ceux sp?cifi?s
> ci-apr?
> allow=ulaw
> allow=alaw
>
>
> extension.conf:
>
> [default]       ; context par d?faut des utilisateurs SIP r?pertori?s 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
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David Boyd

2008-Aug-25 10:40 UTC

head link

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

-----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 num?ro affich?s dans le
X-Lite
appel? 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            ; d?sactive tous les codages sauf ceux sp?cifi?s
ci-apr?
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 num?ro affich?s dans le
X-Lite
appel? 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            ; d?sactive tous les codages sauf ceux sp?cifi?s
ci-apr?
allow=ulaw
allow=alaw

extension.conf:

[default]       ; context par d?faut des utilisateurs SIP r?pertori?s
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

asterisk users - Aug 2008 - Really WEIRD: can register but can not call!

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

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

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