[Asterisk-Users] configuration to Grandstream via tftp

[David J Carter]

Hans,

Attached is the config file I send to my Grandstream.

Change IP address & Phone ID to suite.


Dave

-----Original Message-----
From: asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com]On Behalf Of Hans-Henrik
Andresen
Sent: 19 January 2004 08:43
To: asterisk-users at lists.digium.com
Subject: [Asterisk-Users] configuration to Grandstream via tftp


Hi,

Anyone know how to set up tftp server for grandstream.

I gues it should be somethink like

<tftpserver-dir>
     <mac-address>
          firmware.bin
          config.txt

Is this correct ?

And how should the config-file look like. ?

I had search sipphone.com but did'nt find anything.

/HHA

_________________________________________________________________
Rethink your business approach for the new year with the helpful tips here.
http://special.msn.com/bcentral/prep04.armx

_______________________________________________
Asterisk-Users mailing list
Asterisk-Users at lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
-------------- next part --------------
# SIP Configuration File, Plug'n'Dial APS v1.1 
mac=000b820c2371 
sipserver=proxy01.sipphone.com 
sipserver_port=5060 
outboundproxy=null 
outboundproxy_port=null 
userid=8003 
authenticateid=8003 
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 
stun=stun01.sipphone.com 
stun_port=3478 
tftp_server=192.168.x.x
tftp_server_port=69 
send_dtmf=in-audio 
dtmf_payload_type=101 
ntp_server=ntp01.sipphone.com 
time_zone=GMT-0