pesb
2004-Mar-23 12:48 UTC
[Asterisk-Users] Asterisk SIP + Grandstream 100 + sip.conf phone HELP
Hi there,
I have been trying with asterisk, plus the h323 module with
Grandstream's bt-100 IP phone. I want the asterisk to work as a SIP-proxy
for
these IP phones.
But, I am having trouble setting the /etc/asterisk/sip.conf file.
This is my file:
#############
;
; 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
;allow=ilbc
;
;register => 1234@mysipprovider.com ; Register with a SIP provider
;register => 2345@mysipprovider.com/1234 ; Register 2345 at sip provider as
1234 here.
;
[243075]
type = friend
context = default
secret = gol
host = dynamic
callerid = fono75 <243075>
[243080]
type = friend
context = default
secret = gol
host = dynamic
callerid = fono80 <243080>
#############
and our SIP phones configuration are the following:
SIP Server: 192.168.0.102
Outbound Proxy: <Empty>
SIP User ID: 243075
Authenticate ID: 243075
Authenticate Password: gol
Name: <Empty>
The IP phones can register to the proxy. But, when I try to dial (ie.:243080),
The SIP-proxy answers with a 404 Not Found and I get a busy tone.
What I am doing wrong here?
Can someone that works with asterisk and these phones send me a sip.conf
sample file, along with the scheme where it is set?
thanks in advance,
Pablo Salinas
Stephen R. Besch
2004-Mar-23 13:22 UTC
[Asterisk-Users] Re: Asterisk SIP + Grandstream 100 + sip.conf phone HELP
--snip--> I am having trouble setting the /etc/asterisk/sip.conf file. > This is my file: >1) Add in the [general] section: disallow=all allow=ulaw allow=alaw allow=any other codec that you want to (or can) support. While some have found that this must be specified for each and every phone, I have found that it works fine specified just once in the general section.> [243075] > type = friend > context = default > secret = gol > host = dynamic > callerid = fono75 <243075> >2) Include dtfmmode=info or inband and match to phone's setting 3) I may have been too tired at the time, but once I tried using long extensions (more than 5 digits) and could not make them work either - same error you are getting. I would limit your extensions to 4 digits and see if it helps. 4) You may also need to add canreinvite=no to each phone definition.> > and our SIP phones configuration are the following: > > SIP Server: 192.168.0.102 > > Outbound Proxy: <Empty> >5) I would set this to be the same as the server if you want to make outbound calls. Hope this helps Stephen R. Besch
willy@yponeinc.com
2004-Mar-25 04:08 UTC
[Asterisk-Users] Re: Asterisk SIP + Grandstream 100 + sip.conf phone HELP
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@mysipprovider.com ; Register with a SIP > provider ;register => 2345@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. > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-usersWilly Wouters ypOne Publishing