system toolbox » voip

Firmware 5.1.7 lacking UK FXS impedance setting - what to use instead?

http://forum.voxilla.com/linksys-sipura-voip-support-forum/firmware-5-1-7-lacking-uk-fxs-impedance-setting-use-instead-26939.html

FXS Port Impedence: 600 <North America>
270+750||150nF <most of Europe>
220+820||120nF <Australia, New Zealand>
220+820||115nF <Austria, Bulgaria, South Africa>

na osnovu toga postavio

admin/regional:

• impendance: 270+750||150nf
• caller id method "EDSI DMTF" (ranije bilo bellcore N.America China)
• more echo suppression = Yes (bilo no)

• stavio silence treshold: "high" (bilo medium .. ali nisam siguran :()

• callerid: bellcore

jer sam primjetio da nemam broja pozivaoca nakon ovih promjena

AstRecipes » Installing the Linksys PAP2-NA port adapter

This recipe will help installing a Linksys PAP2 port adapter to work with your Asterisk box.

In this example we will imagine that the PAP2 will be on a local network with your Asterisk box and the Asterisk box will take care of all communication with the rest of the world. We imagine not to have any bandwidth problem between Asterisk and the PAP2.
We also imagine that the PAP2 will have a DHCP assigned address, while the Asterisk server will have a static IP address.

The Linksys does not start in DHCP client mode, so you have to activate it manually.

star Connect an analog phone to the line "phone1"
star Enter ** on the keyboard (4 stars)
star Key in the sequence 101#1#1 to turn DHCP client on
star Key in 110# to have the unit say its IP address

Open your browser and go tho the said IP address. As a default there is no password to access the unit.

star Select "Admin login" and set "Advanced view"
star Go to "Line 1"

Set the following parameters:
Line enable: yes
Proxy: 10.10.3.5
Subscriber:
Display name e User Id: 712
Password: ***
Use Auth ID: no
Preferred codec: G711a

Set it the same way for line 2, using a different account (713 in our case). Note that the SIP ports will be different; that's okay.

Then go to:
star Regional -> Hook Flash timer min: .05 (needed for European phones so that the flash hook works)
star Regional -> Interdigit Long Timer: 5 (seconds before the unit starts dialing after the last entered digit)

Reboot your unit.

Notice that the line LEDs will stay ON when the line is registered and be flashing while the line is in use.

Enter the following parameters in your /etc/asterisk/sip.conf to configure both lines:

[712]
type=friend
secret=***
callerid="My Linksys PAP2-NA p1" <712>
host=dynamic
canreinvite=no
disallow=all
allow=alaw
allow=ulaw
context=sip
dtmfmode=rfc2833
outgoinglimit=1
;incominglimit=1
[713]
type=friend
secret=***
callerid="My Linksys PAP2-NA p2" <713>
host=dynamic
canreinvite=no
disallow=all
allow=alaw
allow=ulaw
context=sip
dtmfmode=rfc2833
outgoinglimit=1
;incominglimit=1

http://forums.whirlpool.net.au/forum-replies-archive.cfm/482283.html

In the PAP2, you can set the DTMF volume and length.
Mine is:

DTMF Playback Level: -16
DTMF Playback Length: 0.1

You might like to try a longer louder signal such as:

DTMF Playback Level: -10
DTMF Playback Length: 0.2

Googling it, most people seem to suggest DTMF Method = ATV. My PAP2 provisioned originally by nehos.net is set to this also.

Edit #1: In Asterisk, "dtmfmode=rfc2833" is the default so unless you specifically set it to something else you probably dont have to touch your asterisk config files. sip.conf is located in /etc/asterisk/sip.conf

Edit #2: AVT stands for "Audio/Visual Transport" and is the name of the working group that drafted the RFC2833 standard for DTMF. So I believe AVT and RFC2833 are the same thing in so far as VoIP settings are concerned.