[Asterisk-Users] Dialing out with SIP phone problem
James Bean
james at hdcs.com.au
Wed Oct 13 04:15:53 MST 2004
I am trying to setup a SNOM 190 with my asterisk box but having a few
problems....
When a call comes in it connects and rings and I can talk no problems...
If I try to call out with the phone I get...
NOTICE[-165364816]: chan_sip.c:7561 handle_request: Unknown SIP command
'PUBLISH' from '192.168.69.250'
I know dialing out works correctly from my analog phone plugged into my
TDM400P but the sip phone doesn't seem to dial properly?
I updated the latest firmware on the snom190...
The configuration on the SNOM190 is pretty standard with just Line 1
configured for asterisk with the correct password etc, I get the
-- Saved useragent "snom190-3.54" for peer snom-james
And
[2]24/12/2001 11:00:09: Registered at registrar as
snom-james at 192.168.69.1
So the phone and asterisk sync and talk ok.
------------------------------------
/etc/asterisk/sip.conf
[general]
port = 5060
bindaddr = 192.168.69.1
context = sip
disallow = gsm
allow = alaw
disallow = ulaw
srvlookup=no
[snom-james]
type=friend
secret=<password removed>
host=dynamic
callerid="James" <690>
defaultip=192.168.69.250
dtmfmode=rfc2833
mailbox=900
[bt-karen]
type=friend
secret=<password removed>
host=dynamic
callerid="Karen" <691>
defaultip=192.168.69.251
dtmfmode=rfc2833
mailbox=901
/etc/asterisk/extension.conf
[pstn]
exten => s,1,Wait(2)
exten => s,2,NoOp(Comment Only: Call from ${CALLERIDNUM}) ; Just put a
comment in the CLI for info.
exten => s,3,Dial(SIP/snom-james,45,t) ;Dial the group=1 zap card mod
above
exten => s,4,Hangup
;exten => s,5,VoiceMail(u100) ;Whatever box you want.
[internal]
exten => i,1,Playback(invalid)
exten => i,2,Hangup
exten => t,1,Hangup
exten => 099,1,Echo ;simple echo test when you dial 099 on your
phone
include => outgoing
include => voip
include => sip
[outgoing]
exten => _9X.,1,Dial(Zap/g1/${EXTEN:1})
exten => _9X.,2,Congestion()
exten => _9X.,3,Hangup
[voip]
exten => _1XX,1,Dial(OH323/${EXTEN}@192.168.254.250/${CALLERIDNUM})
; 1xx extension to Salisbury
exten => _2XX,1,Dial(OH323/${EXTEN}@192.168.20.250/${CALLERIDNUM})
; 2xx extension to Marcoola
exten => 610,1,Dial(OH323/${EXTEN}@192.168.30.250/${CALLERIDNUM}) ; 610
to Jindalee
exten => 620,1,Dial(OH323/${EXTEN}@192.168.40.250/${CALLERIDNUM}) ; 620
to Batteryhill
;exten => _54XXXXXX,1,Dial(OH323/${EXTEN}@192.168.20.250) ; 54 to
Marcoola
;exten => _0754XXXXXX,1,Dial(OH323/${EXTEN}@192.168.20.250) ; 54 to
Marcoola
[sip]
exten => 690,1,Dial(SIP/snom-james,30,tr)
exten => 690,2,voicemail2,u900
exten => 690,102,voicemail2,b900
exten => 691,1,Dial(SIP/bt-karen,30,tr)
exten => 691,2,voicemail2,u901
exten => 691,102,voicemail,b901
---------------------------------------------------------
Although something strange, on bootup asterisk console displays
WARNING[-165811280]: chan_sip.c:681 retrans_pkt: Maximum retries
exceeded on call 3d1b58ba507ed7ab2eb141f241b71efb at 192.168.69.1 for seqno
102 (Non-critical Request)
Any help would be very much appreciated.
James
More information about the asterisk-users
mailing list