[asterisk-users] SPA3102 registration problem

Mandeep Singh Bhabha mandeep.bhabha at dream-world.spb.ru
Thu Feb 28 05:47:28 CST 2008


hello everyone 
	what i did to configure SPA3102 is 
---------------------sip.conf-----------------

;spa-fxs
[108]
type=friend
host=dynamic
context=sipphones
secret=VerySecretPass
mailbox=108
dtmfmode=rfc2833
;dtmfmode=inband
disallow=all
allow=alaw

;spa-fxo-in
[118]
type=friend
host=dynamic
context=home
secret=secret2
dtmfmode=rfc2833
disallow=all
allow=alaw
insecure=very

;spa-fxo-out
[pstn-spa3k]
type=peer
host=XX.XX.XX.XX ; put your ip address here (i had internet address here)
port=5061
secret=testingasterisk
username=asterisk
fromuser=asterisk
dtmfmode=rfc2833
context=home
insecure=very

---------------------sip.conf-----------------
	don't for get to create contexts in extension.conf
-----------------------spa3102----------------
################Line 1################
SIP SETTINGS:
	SIP Port:5060
Proxy and Registration:
	Proxy: YY.YY.YY.YY ; its ip address of your asterisk server
	Register: YES	
Subscriber Information:
	Display Name:spafxs
	User ID:108
	Use Auth ID: NO
VoIP Fallback To PSTN :
	Auto PSTN Fallback:YES
Dial Plan:
	Dial
Plan:([2-79]11<:@gw0>|xx.|*xx.|**xx.|<#,:>xx.<:@gw0>|<#,:>*xx<:@gw0>)
	Enable IP Dialing:NO

################PSTN LINE################
SIP SETTINGS:
        SIP Port:5061
Proxy and Registration:
        Proxy: YY.YY.YY.YY ; its ip address of your asterisk server
        Register: NO
Subscriber Information:
        Display Name:CallerIDforWorld
        User ID:118
        Use Auth ID: NO
Dial Plans:
	Dial Plan 1:(S0<:YY.YY.YY.YY>); its ip address of your asterisk
VoIP-To-PSTN Gateway Setup:
	VoIP-To-PSTN Gateway Enable:YES
	VoIP Caller Auth Method:None
	Line 1 VoIP Caller DP:None
	VoIP Caller Default DP:None
	Line 1 Fallback DP:None
VoIP Users and Passwords (HTTP Authentication):
	VoIP User 1 Auth ID:asterisk
	VoIP User 1 DP:1
PSTN-To-VoIP Gateway Setup :	
	PSTN-To-VoIP Gateway Enable:Yes
	PSTN Caller Auth Method:None
	PSTN Ring Thru Line 1:No
	PSTN CID For VoIP CID:Yes
	PSTN Caller Default DP:1 
	Line 1 Signal Hook Flash To PSTN:Disabled


-----------------------spa3102----------------
	I think this is enough for starting. These settings are working 
perfectly for my needs. (testing)

On Wed, Feb 27, 2008 at 09:14:50PM +0100, Jaap Winius wrote:
> Quoting Tim Johnson <tjapml at cometonovascotia.ca>:
> 
> > I see you put a password line in your sip.conf, but I do not see a
> > username line. Also, you might want to check the port #'s for both the
> > Line 1 and PSTN line. I use 5060 and 5061, respectively.  Hopefully
> > this either helps, or puts you on the right track.
> 
> The username is 8000, so I don't believe it's necessary to mention it.  
> As for the ports, I'm using them in the same way you suggest. Yet it  
> refuses to work.
> 
> My first attempt involved copying my SPA3000's working configuration  
> to the SPA3102. That didn't work. So, I reset the device and applies a  
> configuration generated by Voxilla's wizard, which worked for me with  
> the SPA3000. Not that this has lead to any real differences, but it's  
> still not working.
> 
> There must be something else different about the SPA3102. I did see a  
> problem with it mentioned somewhere in which it's connection with the  
> local Asterisk server would fail (I think temporarily) when changes to  
> the state of its Internet connection occurred (obviously not an issue  
> with the SPA3000). I hope this has nothing to do with my problem.
> 
> Thanks anyway!
> 
> Cheers,
> 
> Jaap
> 
> 
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-- 
С Уважением,
Мандип Сингх Бхабха
email: mandeep.bhabha at dream-world.spb.ru



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