[OpenWrt-Devel] lantiq voip foo

[John Crispin]

Hi,

with the vdsl now working as well as the adsl i would also like to try
to bring the voip/asterisk support up to scratch.

i am a n00b at asterisk and have never used the can_lantiq driver.

any volunteers for this ?

any one successfully used chan_lantiq ?

ideally we could make a sample config file that will allow users to
simply setup the voip of the usual suspects such as sipgate, ekiga,
bt, dt, kpn ...

Having an easy way to set this up would make units such as the a803 or
the 7519 much more attractive.


John

[José Vázquez]

I have only two ARV4518pw but they are enough to make tests, and in a
spanish forum surely there will be a lot of people very pleased to
help you.
First of all i need to learn a bit of asterisk with AA in order to
help you better.

Thanks in advance:

Pepe

Post by John Crispin
Hi,
with the vdsl now working as well as the adsl i would also like to try
to bring the voip/asterisk support up to scratch.
i am a n00b at asterisk and have never used the can_lantiq driver.
any volunteers for this ?
any one successfully used chan_lantiq ?
ideally we could make a sample config file that will allow users to
simply setup the voip of the usual suspects such as sipgate, ekiga,
bt, dt, kpn ...
Having an easy way to set this up would make units such as the a803 or
the 7519 much more attractive.
John
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[Luka Perkov]

Post by John Crispin
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

***@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:***@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

***@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

[Jiří Šlachta]

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

Post by Luka Perkov

Post by John Crispin
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
[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
[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
[default]
[in-USERNAME]
exten => USERNAME,1,Dial(TAPI/1)
exten => USERNAME,n,Hangup
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