[SOLVED] PRI signaling - Make user hear telco messages

Asterisk Asterisk Support
1

We have a PRI and whenever something goes wrong at the other side (like “number has changed”, “number out of service”, etc.) asterisk returns the DIALSTATUS and hangs up. is there a way to make asterisk actually play those original messages from the telco side?

Our hardware is a Digium te102p

Thanks
wonderworld

2

Hi

try priindication=inband

or use the Dialstatus and play the tones yourself

Ian

3

Hey.

priindication=inband is the default behaviour of asterisk. Just for testing i tried to set it to priindication=outofband. Neither setting changed the asterisk behaviour.

I know that i can read DIALSTATUS and play the appropiate message bymyself, but there is a problem. There are more possible causes of a failed call than DIALSTAUSes. For example:

You get a CHANUNAVAIL if a number doesn’t exist at all.
You get a CHANUNAVAIL also if you call a cellphone that has been switched off.

We really need to get the “real” messages from the telco-side.

Is there a way?

Thanks
wonder

4

try adding

progressinband=yes to the sip.conf

Ian

5

There is also a variable that reflects the more detailed termination cause. As I’d have to look it up, I’ll leave you to do that.

6

voip-info.org/wiki/index.php … angupcause

Ian

7

Ok, I tried progrssinband=yes and it didn’t change the way asterisk behaves.

I am wondering if it’s possible at all to make this work.
Could it be a problem at the configuration of the line at telco-side?

I re-checked HANGUPCAUSE and got an idea what messages to record bymyself, but then I’d still need the external anouncements for changed numbers

“The number has changed. The new number is 0123456789”

Another thing i found was:

;inbanddisconnect=yes
; Allow inband audio (progress) when a call is RELEASEd by the far end of a PRI

I am not sure if that option could do what i want. The problem is, that it isn’t available in our quiet old Asterisk setup from the Debian Stable Repos (1.2.13). I don’t want to setup a new system with the latest Asterisk without being sure, that it would solve my problems, because everything else is working perfectly for us.

Thanks for any help

wonder

8

Hi

Could you post you zapata.conf so we can see what you have setup at the moment.

Its hard to guess what you have or havent in it.

Ian

9

Sure…

[code]
[channels]
language=de

context=versatel-in

switchtype=euroisdn
pridialplan=unknown
prilocaldialplan=national
signalling=pri_cpe

priindication=inband

usecallerid=yes
hidecallerid=no
callwaiting=no
usecallingpres=yes

callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes

echocancel=yes

immediate=no
overlapdial=yes
callerid=asreceived
resetinterval=60

internationalprefix = 00
nationalprefix = 0
localprefix = our local prefix
privateprefix = our private prefix
unknownprefix =

group=1
channel=>1-15,17-31

[/code][/code]

10

I gave up and moved to Asterisk 1.6.0.3 last night.

The inbanddisconnect=yes setting in chan_dahdi.conf made it finaly work. I still don’t get busy tones, but I can generate them bymyself.

Thanks for all the help in this thread. I am still unsure if it was possible at all with my older Asterisk version but at least it works now.

Thanks
wonder