Hello Luka and John,

I can do that. There is just one thing that holds me back, I am finishing
my master thesis, so there is not enough time for me to revise that in
near several weeks.

Jiri

Dne 15. 4. 2014 1:00, Luka Perkov napsal(a):
> 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:passw...@sip.foo.bar/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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