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" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> 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 => [EMAIL PROTECTED] ; Register with a SIP provider > ;register => [EMAIL PROTECTED]/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. > > > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
