[Asterisk-Users] Asterisk + GrandStream SIP phones

[Sergio Serrano]

Try to add a qualify=XXXX to sip.conf, and try to exec a sip show peers.
In spite of phones appears like register, if you use NAT, your firewall
can cut communication. Try the next:


Just after phone register call to it, and then wait for a minutes and
try to call again. Could you call first time but not in second one? It
is due to your firewall. Try to configure wuth next config:

[1004]
......
.....
qualify=XXXX
.......
......

In you grandstream configuration try to put time to expire register  1
minute and then try to do the previous test.


I'm sorry for my english, but I hope this let you call.

Regards,

srsergio



-----Mensaje original-----
De: asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com] En nombre de pesb
Enviado el: lunes, 29 de marzo de 2004 20:26
Para: asterisk-users at lists.digium.com
Asunto: [Asterisk-Users] Asterisk + GrandStream SIP phones


-This is my 'sip.conf' file:

;*************************************************************
;
; SIP Configuration for Asterisk
;
[general]
port = 5060                     ; Port to bind to
bindaddr = 0.0.0.0              ; Address to bind to
context = default               ; Default for incoming calls
tos=184
maxexpirey=3600         ; Max length of incoming registration we allow
defaultexpirey=120              ; Default length of incoming/outoing 
registration
disallow=all                    ; Disallow all codecs
allow=ulaw                      ; Allow codecs in order of preference
allow=alaw


[1004]
type=friend
username=1004
secret=
reinvite=no
canreinvite=no
host=dynamic
dtmfmode=inband
mailbox=1004
nat=1
disallow=all
allow=ulaw
allow=alaw

[1005]
type=friend
username=1005
secret=
reinvite=no
canreinvite=no
host=dynamic
dtmfmode=inband
mailbox=1005
nat=1
disallow=all
allow=ulaw
allow=alaw

;*******************************************


-And this is the basic seting of my two GrandStream SIP phones:

***************[1005]****************
IP Address:192.168.0.105
Subnet Mask:255.255.255.0
SIP Server: 192.168.0.103
Outbound Proxy:<empty>
SIP User ID:1005
Authenticate ID:1005
Authenticate Password:123
Name:1005

Preferred Vocoder:
choice 1: PCMU
choice 2: PCMA
choice 3: G723
choice 4: G729
choice 5: G726-32
choice 6: G728

G723 rate:  6.3kbps
Silence Suppression:No
Send DTMF:in-audio

***************[1004]****************
IP Address:192.168.0.104
Subnet Mask:255.255.255.0
SIP Server: 192.168.0.103
Outbound Proxy:<empty>
SIP User ID:1004
Authenticate ID:1004
Authenticate Password:123
Name:1004

Preferred Vocoder:
choice 1: PCMU
choice 2: PCMA
choice 3: G723
choice 4: G729
choice 5: G726-32
choice 6: G728

G723 rate:  6.3kbps
Silence Suppression:No
Send DTMF:in-audio

******************************

I have 2 SIP GrandStream phones, both phones are correctly registered to
the 
Asterisk server. But, when I try to make a call from registered phone
'1005' 
to registered phone '1004', dialing 1004, Asterisk responds with the
'Status: 
404 Not Found' message.
How do I have to dial? What else do I need to set?
Find attached my traffic captured on ethereal.