On Thu, Apr 10, 2014 at 09:23:36AM +0200, John Crispin wrote:
> any volunteers for this ?
> 
> any one successfully used chan_lantiq ?

I have but in "production" I am still running owsip that we did before. 

One of the example configs I have is below. But somebody who knows
asterisk should revise that before packaging... Also, one might need to
edit [1] in order to get asterisk working properly on the device.

If we can get somehow working TAPI on ar9 and/or vr9 and not just danube
I will rebase the asterisk_channel_lantiq on top of newer asterisk
version. AFAIK additional kernel patching is needed in order to get TAPI
working there.

Luka

[1] 
http://git.nanl.de/?p=asterisk_channel_lantiq.git;a=blob;f=src/configs/lantiq.conf.sample

root@OpenWrt:/# cat /etc/asterisk/sip.conf
[general]
allowguest=no
match_auth_username=yes
allowoverlap=no
udpbindaddr=0.0.0.0
tcpenable=no
tcpbindaddr=0.0.0.0
srvlookup=no
defaultexpiry=1200
;preferred_codec_only=yes
disallow=all
allow=alaw
allow=ulaw
relaxdtmf=yes
videosupport=no
alwaysauthreject = yes
use_q850_reason = yes
allowsubscribe = yes
notifyringing = yes
notifyhold = yes
notifycid = yes
callcounter = yes
nat=no
directmedia=no
register => USERNAME:[email protected]/USERNAME

[out-USERNAME]
type=peer
context=in-USERNAME
dtmfmode=inband
host=sip.foo.bar
insecure=invite,port
remotesecret=PASSWORD
defaultuser=USERNAME
qualify=yes
nat=no
directmedia=no
transport=udp
vad=no

root@OpenWrt:/# cat /etc/asterisk/extensions.conf
[default]
exten => _X.,1,Dial(SIP/${EXTEN:0}@out-USERNAME,30)

[in-USERNAME]
exten => USERNAME,1,Dial(TAPI/1)
exten => USERNAME,n,Hangup
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