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
