[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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