Hi Pasqualotto,
Actually, I've seen your post on Asterisk-Users list yesterday, but I could
not understand back then. Now, I've checked your sip configuration again, I
think you make a mistake in "type" of sip account. I use "friend" not
"peer". I am not sure though.
Following is what I had in my sip.conf file for the FXO port of HT488:
[41]
username=41
type=friend
secret=<put your password here>
host=dynamic
context=<put your context here>
callerid="Outside-line" <41>
dtmfmode=inband
group=1
callgroup=1
pickupgroup=1
Of course, you should configure HT488 FXO sip account accordingly too. You
should make sure that HT488 registers with Asterisk.
Also read again the following thread:
http://lists.digium.com/pipermail/asterisk-users/2005-August/123548.html
Now, when you call 41 from another phone, you should be able to hear the
dial tone. And if you configured HT488 to answer incomming calls to FXO and
where they should be directed to ("Forward to VoIP" box), then you should be
able to call in HT488 FXO and talk to Asterisk after a few rings. (HT488
configuration is also very important, I don't know what settings you have
there.)
I don't have a HT488 these days, so I cannot test your configurations,
sorry.
Soner
----- Original Message -----
From: "Pasqualotto Enrico" <[EMAIL PROTECTED]>
To: <[email protected]>
Sent: Monday, February 27, 2006 9:54 PM
Subject: [Asterisk-Users] Asterisk with HT 488 FXO
Hi, i have a HT 488 and I want using this like an FXO for Asterisk.
I have find some configuration in the list archive & google but my HT with
these config not work.
my sip.conf
[HT-488]
username=400
type=peer
secret=wowowow
qualify=yes
port=5062
nat=no
host=192.168.1.157
fromuser=400
disallow=all
context=from-pstn
allow=g711u
allow=ulaw
allow=alaw
my sip debug:
--------------------------------------------------------------
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bK669516e2;rport
From: "Unknown" <sip:[EMAIL PROTECTED]>;tag=as073738f8
To: <sip:192.168.1.157:5062>;tag=ebc40000a8e20000
Call-ID: [EMAIL PROTECTED]
CSeq: 102 OPTIONS
User-Agent: Grandstream HT488 1.0.2.16
Contact: <sip:[EMAIL PROTECTED]:5062;user=phone>
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Supported: replaces
Content-Length: 0
--- (11 headers 0 lines)---
Destroying call '[EMAIL PROTECTED]'
asterisk1*CLI>
<-- SIP read from 192.168.1.157:5062:
SIP/2.0 481 No Such Call
Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bK669516e2;rport
From: "Unknown" <sip:[EMAIL PROTECTED]>;tag=as073738f8
To: <sip:192.168.1.157:5062>;tag=522400002a6bffff
Call-ID: [EMAIL PROTECTED]
CSeq: 102 OPTIONS
User-Agent: Grandstream HT488 1.0.2.16
Content-Length: 0
--- (8 headers 0 lines)---
Destroying call '[EMAIL PROTECTED]'
REGISTER 12 headers, 0 lines
Reliably Transmitting (no NAT) to 192.168.1.157:5060:
REGISTER sip:192.168.1.157 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bK42700737;rport
From: <sip:[EMAIL PROTECTED]>;tag=as558874a4
To: <sip:[EMAIL PROTECTED]>
Call-ID: [EMAIL PROTECTED]
CSeq: 120 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Expires: 120
Contact: <sip:[EMAIL PROTECTED]>
Event: registration
Content-Length: 0
---
Destroying call '[EMAIL PROTECTED]'
asterisk1*CLI>
<-- SIP read from 192.168.1.157:5060:
SIP/2.0 501 Not Implemented
Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bK42700737;rport
From: <sip:[EMAIL PROTECTED]>;tag=as558874a4
To: <sip:[EMAIL PROTECTED]>;tag=3a7300003fa70000
Call-ID: [EMAIL PROTECTED]
CSeq: 120 REGISTER
User-Agent: Grandstream HT488 1.0.2.16
Content-Length: 0
-------------------------------------------------------
The register string ??
Can anyone help me??
Thanks
--
Pasqualotto Enrico
email: [EMAIL PROTECTED]
web: http://www.pasqualotto.org
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