SIP: No translator path ?!?

Asterisk Asterisk Support
1

Two Asterisk in same lan (for testing). One on 10.0.0.138 and another on 10.0.0.6. Both of them have TDM11B cards installed. On Zap1 both of them have pluged regular old phones.
I have defined that they connect to each other over SIP. When I call from first (138) another * over SIP i get this error on Asterisk 2 (6).

No translator path exist for channel type Zap (native 68 ) to 256
Unable to create channel od type ‘Zap’

And when I try to call from 2’nd * (6) first * (138), on 2’nd * (6) i get this error message.

No path to transkate from Zap/1-1 (68 ) to SIP/jure-d53c(256)
Had to drop call because I couldn’t make Zap/1-1 SIP/jure-d53c

All works perfectly with IAX2. So, it can’t be hardware error.

On 2’nd * my conf files looks like this.

sip.conf
[general]
port=5060
bindaddr=0.0.0.0
context=sip
srvlookup=yes
dtmfmode=rfc2833
disallow=all
allow=g729
allow=gsm
allow=alaw
allow=ulaw

[jure]
type=friend
host=10.0.0.138
; END sip.conf

extensions.conf
[default]
exten => _097.,1,Dial(SIP/jure/${EXTEN:3},30,tr)

[sip]
exten => 1001,1,Dial(Zap/1,30,tr)

[iax]
exten => 1001,1,Dial(Zap/1,30,tr)
; END extensions.conf

2

For anybody else who has this problem, here is the solution.

I have pluged one more ethernet card in 2’nd Asterisk (6). Because of that I couldn’t use Digium’s g729 codec anymore. And they didn’t try to use any other codec instead of first one defined (g729)

Any way I have remuved line
allow=g729
and now it works.

Tomislav

3

You can register the codec up to two times without any problems. Then you have to email sales @ digium and explain your situation and ask them to reset your registration before you can register it again. It’s a good idea to email them after you’ve registered it the second time in case you need to do it again later.