[asterisk-users] Issue with asterisk

Silver Thorne zoraxus at gmail.com
Tue Nov 2 13:24:17 CDT 2010


Steve;

You are so right - it was the end of the day, I was tired and pissy.

Let me try this again:

Version:
ns211156*CLI> core show version
Asterisk 1.4.31 built by root @ some_server.foo.net on a x86_64 running 
Linux on 2010-06-10 14:32:34 UTC

Name and version of endpoints involved:

Sip Settings:

Global Settings:
----------------
   SIP Port:               5060
   Bindaddress:            0.0.0.0
   Videosupport:           No
   AutoCreatePeer:         No
   Allow unknown access:   Yes
   Allow subscriptions:    Yes
   Allow overlap dialing:  Yes
   Promsic. redir:         No
   SIP domain support:     No
   Call to non-local dom.: Yes
   URI user is phone no:   No
   Our auth realm          asterisk
   Realm. auth:            No
   Always auth rejects:    No
   Call limit peers only:  No
   Direct RTP setup:       No
   User Agent:             Asterisk PBX
   MWI checking interval:  10 secs
   Reg. context:           (not set)
   Caller ID:              asterisk
   From: Domain:
   Record SIP history:     Off
   Call Events:            Off
   IP ToS SIP:             none
   IP ToS RTP audio:       none
   IP ToS RTP video:       none
   T38 fax pt UDPTL:       No
   RFC2833 Compensation:   No
   SIP realtime:           Disabled

Global Signalling Settings:
---------------------------
   Codecs:                 0x8000e (gsm|ulaw|alaw|h263)
   Codec Order:            none
   T1 minimum:             100
   No premature media:     No
   Relax DTMF:             No
   Compact SIP headers:    No
   RTP Keepalive:          0 (Disabled)
   RTP Timeout:            0 (Disabled)
   RTP Hold Timeout:       0 (Disabled)
   MWI NOTIFY mime type:   application/simple-message-summary
   DNS SRV lookup:         Yes
   Pedantic SIP support:   No
   Reg. min duration       60 secs
   Reg. max duration:      3600 secs
   Reg. default duration:  120 secs
   Outbound reg. timeout:  20 secs
   Outbound reg. attempts: 0
   Notify ringing state:   Yes
   Notify hold state:      No
   SIP Transfer mode:      open
   Max Call Bitrate:       384 kbps
   Auto-Framing:           No

Default Settings:
-----------------
   Context:                default
   Nat:                    RFC3581
   DTMF:                   rfc2833
   Qualify:                0
   Use ClientCode:         No
   Progress inband:        Never
   Language:               (Defaults to English)
   MOH Interpret:          default
   MOH Suggest:
   Voice Mail Extension:   asterisk

----
Parsing /etc/asterisk/extconfig.conf

sip show peer

  * Name       : 155
   Secret       :<Set>
   MD5Secret    :<Not set>
   Context      : extern
   Language     : en
   AMA flags    : Unknown
   Transfer mode: open
   MaxCallBR    : 384 kbps
   CallingPres  : Presentation Allowed, Not Screened
   Call limit   : 0
   Callgroup    :
   Pickupgroup  :
   Callerid     : "Glen's Sysadmin Test Line"<200111222>
   ACL          : No
   Codec Order  : (none)
   Auto-Framing:  No

sip.conf


[general]
;port = 5060
;bindaddr=0.0.0.0
;srvlookup=yes
;context=extern
;nat=yes
;localnet=192.168.0.0/255.255.0.0
;allowguest=no

[Axialys]
type=peer
host=sip-proxy.xxx.xxx.net
fromuser=USERID_1
secret=password-1
qualify=yes
context=extern
quality=yes
dtmfmode=rfc2833
disallow=all
allow=gsm
allow=ulaw
nat=yes
insecure=port,invite

[Axialys2]
type=peer
host=sip-proxy.xxx.xxx.net
host=dynamic
fromuser=userid_1
secret=password_1
qualify=yes
context=extern
quality=yes
dtmfmode=rfc2833
disallow=all
allow=gsm
allow=ulaw
nat=yes
insecure=port,invite

[GlenAxialys3]
type=peer
host=sip-proxy.xxx.xxx.net
fromuser=userid_1
secret=password_1
qualify=yes
context=extern
quality=yes
dtmfmode=rfc2833
disallow=all
allow=gsm
allow=ulaw
nat=yes
insecure=port,invite



[Nov  2 17:10:04] NOTICE[13804] chan_sip.c: Call from '6839' to 
extension '33173793697' rejected because extension not found.
[Nov  2 17:10:06] NOTICE[13804] chan_sip.c: Call from '6839' to 
extension '33173793697' rejected because extension not found.
[Nov  2 17:10:07] NOTICE[13804] chan_sip.c: Call from '6839' to 
extension '33173793697' rejected because extension not found.
[Nov  2 17:10:09] NOTICE[13804] chan_sip.c: Call from '6839' to 
extension '33173793697' rejected because extension not found.
[Nov  2 17:10:09] NOTICE[13804] chan_sip.c: Call from '6839' to 
extension '33173793697' rejected because extension not found.
[Nov  2 17:10:17] NOTICE[13804] chan_sip.c: Call from '6839' to 
extension '33173793697' rejected because extension not found.
[Nov  2 17:10:24] NOTICE[13804] chan_sip.c: Call from '6839' to 
extension '33173793697' rejected because extension not found.
[Nov  2 17:10:24] NOTICE[13804] chan_sip.c: Call from '6839' to 
extension '33173793697' rejected because extension not found.
[Nov  2 17:10:31] NOTICE[13804] chan_sip.c: Call from '6839' to 
extension '33173793697' rejected because extension not found.
[Nov  2 17:10:31] NOTICE[13804] chan_sip.c: Call from '6839' to 
extension '33173793697' rejected because extension not found.
[Nov  2 17:10:35] NOTICE[13804] chan_sip.c: Call from '6839' to 
extension '33173793697' rejected because extension not found.

So, when I call the 33173793697 number, the above entry is what I see in 
the log.

Glen

On 11/1/2010 17:32, Steve Edwards wrote:
> On Mon, 1 Nov 2010, Silver Thorne wrote:
>
>> >  Anyone see this before:
>> >
>> >  [Nov 1 19:55:49] WARNING[30497] chan_sip.c: username mismatch, have
>> >  <6839>, digest has<3169>
> You may have better luck with a more descriptive subject. Lots of users
> have an issue or two with Asterisk.
>
> Some details will also help. Like:
>
> ) Version of Asterisk.

> ) Name and version of the endpoints involved.
>
> ) Relevant sections of sip.conf as well as the console output from 'sip
> show settings,' 'sip show user<username>,' and 'sip show peer
> <peername>.' (I'm a 1.2 Luddite.)
>
> ) Console output of 'sip debug ip<address>' illustrating the 'issue.'
>
> Don't forget to 'sanitize' any IP addresses, usernames, and passwords that
> you consider valuable. (Actually, it would be better to redo your
> configuration with 'throw-away' credentials (like username1 and password1)
> for the duration of your issue -- less chance of exposing something or
> mistyping an important detail.)
>



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