Michael Hamann wrote:
Hi Everybody,
as a relative newby I´m just trying to get a Draytek Vigor Router (2600Vi)
connected to my Asterisk System (CVS-05/31/04). With X-Lite and a Cisco
Phone it is no problem, but the Vigor seems to have some problems with
Asterisk.
The first thing ist when I do a sip show peers on the console I get:
4002/4002172.16.183.37 (D) 255.255.255.255 5060 Unmonitored
4001/4001172.16.183.37 (D) 255.255.255.255 5060 Unmonitored
What does this status unmonitored mean? With my softphone the entry looks
like:
6275/6275172.16.181.49 (D) 255.255.255.255 5060 OK (8 ms)
The next thing is that when I try to call one of the vigors SIP Ports via
X-Lite I see the following message in the debug console:
Jun 18 10:09:54 NOTICE[131081]: chan_sip.c:5150 handle_response: Dunno
anything about a 0 Unkown status code response from SIP/4001-b2fc
No call is signalled to the phone. The other way, my X-Lite rings but the
connection is hung up the moment I accept the call.
The Draytek support says that the Vigor does not support SIP Reinvite and
that I should try to disable it in my PBX system.
So I changed my sip.conf to:
[4001]
type=friend
username=4001
secret=4001
mailbox=2000
canreinvite=no
context=default
host=dynamic
But it still does not work. Does anybody has this combination working and
could send me his config files? Or any other ideas?
best regards from germany
Michael
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I had this working once, now I have a grandstream so it is no longer needed.
It is vital that you get the latest version of the firmware for the
vigor as previous versions do not work with the sip server on the lan
ports only on the other side of the ADSL line.
The reason for this is the sip packets always originated from the ADSL
address instead of the internal address which is the one you want to be
using if you have an internal server.
Next I used a settup a bit like this:
Vigor:
VOIP SETUP SIP Related Functions
SIP:
SIP Port 5060
Registrar asterisk.mydomain.com (or an IP address)
Port1:
Name: p1
Password: (I did not use one)
Expiry Time: 10 mins
VOIP Setuip CODEC/RTP etc:
Codecs:
G.711MU
Packet Size: 20ms
DTMF:
OutBand
Payload Type 101
RTP:
Take the default ports
Asterisk:
Sip.conf:
[general]
port=5060 ; Port to bind to
bindaddr=0.0.0.0; Address to bind to
context=in-sip ; Default for incoming calls
callerid=Call 909090
canreinvite=no
disallow=all
allow=ulaw
allow=alaw
allow=gsm
maxexpirey=1800
defaultexpirey=600
tos=throughput
[p1]
type=friend
host=dynamic
user=p1
;secret=
dtmfmode=rfc2833
[EMAIL PROTECTED]
callerid=p1 3002
qualify=yes
context=home
hope this helps
Chris.
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