[Asterisk-Users] Re: Asterisk SIP + Grandstream 100 + sip.conf phone HELP

[pesb]

Dear Chris,
My firmware version is 1.0.4.39, how can I make the upgrade? 
where (url site) can I get the firmware?

thanks again,
                   Pablo S.

On Thursday 25 March 2004 06:32, Chris Stenton wrote:
> What version of the Phone firmware are you running ? I had the same problem
> until I upgrade to
> 1.0.4.54
>
>
>
> Chris
>
> ----- Original Message -----
> From: "pesb" <pesb at conexion.com.py>
> To: <asterisk-users at lists.digium.com>
> Sent: Wednesday, March 24, 2004 9:41 PM
> Subject: Re: [Asterisk-Users] Re: Asterisk SIP + Grandstream 100 + sip.conf
> phone HELP
>
> > Hi there,
> > I am still trying to make the asterisk SIP proxy server work with my
> > Grandstream 100 IP phones.
> > I tried Stephen advice and it did not work. I stil got the 404 error
>
> message.
>
> > So, rigth now, I am trying the following configuration(sip.conf):
> >
> > ###########################
> > ;
> > ; SIP Configuration for Asterisk
> > ;
> > [general]
> > port = 5060   ; Port to bind to
> > bindaddr = 0.0.0.0  ; Address to bind to
> > ;externip = 200.201.202.203 ; Address that we're going to put in SIP
>
> messages
>
> > if we're behind a NAT
> > ;localnet = 192.168.0.0         ; Internal NETWORK address
> > ;localmask = 255.255.255.0      ; Internal netmask
> > context = default  ; Default for incoming calls
> > ;srvlookup = yes  ; Enable SRV lookups on outbound calls
> > ;pedantic = yes   ; Enable slow, pedantic checking for Pingtel
> > ;tos=lowdelay
> > ;tos=184
> > ;maxexpirey=3600  ; Max length of incoming registration we allow
> > ;defaultexpirey=120  ; Default length of incoming/outoing registration
> > ;notifymimetype=text/plain ; Allow overriding of mime type in NOTIFY
> > ;videosupport=yes  ; Turn on support for SIP video
> > ;disallow=all   ; Disallow all codecs
> > ;allow=ulaw   ; Allow codecs in order of preference
> > dtmfmode=rfc2833
> > disallow=all
> > allow=ulaw
> > allow=alaw
> > ;allow=ilbc
> >
> > ;register => 1234 at mysipprovider.com ; Register with a SIP provider
> > ;register => 2345 at mysipprovider.com/1234 ; Register 2345 at sip provider
>
> as
>
> > 1234 here.
> > ;
> > ;[snomsip]
> > ;type=friend
> > ;secret=blah
> > ;host=dynamic
> > ;dtmfmode=inband  ; Choices are inband, rfc2833, or info
> > ;defaultip=192.168.0.59
> > ;mailbox=1234,2345  ; Mailbox for message waiting indicator
> > ;restrictcid=yes  ; To have the callerid restriced -> sent as ANI
> >
> > ;[pingtel]
> > ;type=friend
> > ;username=pingtel
> > ;secret=blah
> > ;host=dynamic
> > ;qualify=1000   ; Consider it down if it's 1 second to reply
> > ;callgroup=1,3-4
> > ;pickupgroup=1,3-4
> > ;defaultip=192.168.0.60
> >
> > ;[cisco]
> > ;type=friend
> > ;username=cisco
> > ;secret=blah
> > ;nat=yes   ; This phone may be natted
> > ;host=dynamic
> > ;canreinvite=no   ; Cisco poops on reinvite sometimes
> > ;qualify=200   ; Qualify peer is no more than 200ms away
> > ;defaultip=192.168.0.4
> >
> > ;[cisco1]
> > ;type=friend
> > ;username=cisco1
> > ;fromuser=markster  ; Specify user to put in "from" instead of callerid
> > ;secret=blah
> > ;host=dynamic
> > ;defaultip=192.168.0.4
> > ;amaflags=default  ; Choices are default, omit, billing, documentation
> > ;accountcode=markster  ; Users may be associated with an accountcode tp
>
> ease
>
> > billing
> >
> >
> > [1001]
> > type = friend
> > context = default
> > secret = gol
> > host = dynamic
> > callerid = "STREAM-1001" <1001>
> > ;dtfmmode=inband
> > canreinvite=no
> > defaultip=192.168.0.105
> >
> >
> > [1002]
> > type = friend
> > context = default
> > secret = gol
> > host = dynamic
> > callerid = "STREAM-1002" <1002>
> > ;dtfmmode=inband
> > canreinvite=no
> > defaultip=192.168.0.104
> > ##############################
> >
> > This is the configuration of my SIP-phones:
> >
> >
> > ipaddr=192.168.0.105
> > sipserver=192.168.0.102
> > sipserver_port=5060
> > outboundproxy=null
> > outboundproxy_port=null
> > userid=1001
> > authenticateid=1001
> > codec1=PCMU
> > codec2=PCMA
> > codec3=G723
> > codec4=G729
> > codec5=null
> > codec6=null
> > silence_supporession=no
> > voice_frames_per_tx=2
> > ipqos=48
> > vlantag=0
> > registration_expiration=10
> > local_sip_port=5060
> > local_rtp_port=5004
> > use_random_rtp_port=no
> > send_dtmf=in-audio
> > dtmf_payload_type=101
> > time_zone=GMT-0
> >
> > ipaddr=192.168.0.104
> > sipserver=192.168.0.102
> > sipserver_port=5060
> > outboundproxy=null
> > outboundproxy_port=null
> > userid=1004
> > authenticateid=1004
> > codec1=PCMU
> > codec2=PCMA
> > codec3=G723
> > codec4=G729
> > codec5=null
> > codec6=null
> > silence_supporession=no
> > voice_frames_per_tx=2
> > ipqos=48
> > vlantag=0
> > registration_expiration=10
> > local_sip_port=5060
> > local_rtp_port=5004
> > use_random_rtp_port=no
> > send_dtmf=in-audio
> > dtmf_payload_type=101
> > time_zone=GMT-0
> >
> >
> > What's wrong here??
> >
> > When I try to dial from one phone to the other, I get 404 error message.
> >
> > Please, somebody help me.
> >
> >
> > _______________________________________________
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