Paulo Santos wrote:
> Hello,
>
> Following my first mail about this issue [1], I think I know now what
> the problem is.
>
> When I have both lines being used and a third call comes in, the person
> calling doesn't get a busy tone, he gets something like line unavailable.
>
> I've been debugging mISDN and I think the reason is because asterisk is
> sending the release cause as 0.
>
> P[ 3] --> channel:0 mode:TE cause:0 ocause:0 rad: cad:
>
> The request from the telephone company's switch seems correct, a SETUP
> message (if 08 is Q.931, 05 is SETUP).
>
> 02 ff 03 08 01 04 05 a1 04 03 80 90
> a3 18 01 80 6c 0b 01 83 39 31 36 33
> 39 31 37 34 32 70 03 c1 38 34
>
> I've changed misdn.conf so it sends a release cause as 17 (user busy),
> but I get the same behaviour - cause:0 ocause:0.
>
> Anyone knows how can I force asterisk to send cause 16 or 17 in this
> situation?
>
> Thanks in advance.
>
> Best regards,
> Paulo Santos
>
> misdn.conf: http://pastebin.com/FmgECqkU
> misdn debug: http://pastebin.com/Tg6wPKBD
>
> [1]
> http://www.mail-archive.com/[email protected]/msg244330.html
>
Ok, I've encountered a similar issue on a different installation but
instead of being PTP it's PTMP. Plus, it's a setup with 2 BRI lines with
call forwarding between them - main number of BRI1 forwards to secondary
number of BRI2 when busy/unavailable and vice-versa.
I've called the phone company and confirmed that call waiting is
disabled, yet I get a message in misdn debug saying:
P[ 2] --> Call Waiting on PMP sending RELEASE_COMPLETE
I don't know if this is the actual call waiting feature or if it is just
an information of some kind.
In the misdn debug I get this: http://pastebin.com/D7wv0qqm
The P[ 2] is the port of the BRI line I called in the first place, then
it is forwarded to P[ 1] where I get an error:
P[ 1] Decoding FACILITY failed! (-1)
And the same issue I said in the previews email:
P[ 1] --> channel:0 mode:TE cause:0 ocause:0 rad: cad:
I changed isdn_lib.c and now I'm sending ocause:17 (user busy). I've
done this in the PTP line mentioned in the previews email as well.
For the PTP line it appears to have worked, I have the regular busy
signal. It worked only after the first time I tried to place a 3rd call.
Now the 3rd call doesn't even reach Asterisk, which was what I wanted
from the phone company in the first place.
On the PTMP line it didn't work, I still don't get the busy signal.
Maybe cause 17 isn't the right one? And what can be that "FACILITY"
mentioned in the debug?
Thanks in advance.
Best regards,
Paulo Santos
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
