[Asterisk-Users] Dialing out with SIP phone problem

[steve szmidt]

On Wednesday 13 October 2004 07:15 am, James Bean wrote:
> 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'

This message does not affect anything. It's a bug with Snom firmware that it 
sends out a publish even though you tell it not to. Meanwhile no need to 
worry. One of these days Snom will fix it and we won't see the message.

> 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
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-- 

Steve Szmidt

"They that would give up essential liberty for temporary safety 
deserve neither liberty nor safety."
                                Benjamin Franklin