[Asterisk-Users] Draytek Vigor 2600Vi as SIP client on Asterisk

[Chris Lee]

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:
> 
> 4002/4002        172.16.183.37   (D)  255.255.255.255  5060     Unmonitored
> 4001/4001        172.16.183.37   (D)  255.255.255.255  5060     Unmonitored
> 
> What does this status unmonitored mean? With my softphone the entry looks
> like:
> 
> 6275/6275        172.16.181.49   (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
mailbox=3002 at home
callerid="p1" <3002>
qualify=yes
context=home

hope this helps

Chris.