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

[willy at yponeinc.com]

Well ...
For starters, in your sip.conf you have
dtmfmode=rfc2833
but your phone setup gives
send_dtmf=in-audio
In your post (below) you also left out
authenticate_password=gol
but that may be an oversight?
BTW: My GS setup uses dtmfmode=info (in my sip.conf for each
phone)
and send_dtmf=SIP_IPNFO in the phone config
Cheers, Willy

----- Original Message Follows -----
> 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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Willy Wouters
ypOne Publishing