It’s an outbound call from the polycom to pstn and the other side (e.g. my
mobile) returns a busy signal.



The ISDN gateway then sometimes sends a 180 Ringing to the polycom and
after that a 486 Busy. If it sends 180 Ringing depends on the destination
that was called. My mobile provider sends an alerting signal before the
busy signal but other providers may not. If I call a provider that doesn’t
send alerting before it refuses the call the gateway does not send 180
ringing to the polycom.



Working scenario:

Polycom sends Invite to Gateway. Gateway returns 180 Ringing and then 486
Busy here.

Not working scenario:

Polycom sends Invite to Gateway. Gateway directly return 486 Busy here. The
user does not see anything on the polycom screen and does not hear a busy
signal. He doesn’t know anything about the reason why the call could not be
established.





_____________________________
Jan Fricke (B.Sc.)

*IANT -
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*Von:* [email protected] [mailto:
[email protected]] *Im Auftrag von *Tony Graziano
*Gesendet:* Mittwoch, 24. Oktober 2012 12:48
*An:* Discussion list for users of sipXecs software
*Betreff:* Re: [sipx-users] Polycom Busy Problem



Why would the phone ring busy? Have you artificially lowered the number of
calls per line? Is VM enabled on the line or not?

# of calls per line and whether there is VM will matter.

On Oct 24, 2012 5:11 AM, "Jan Fricke" <[email protected]> wrote:

Hi,

I’m struggling with a Polycom related problem. This is more Polycom related
than SipX but I think here are a lot of people that know Polycom phones
very well.



In some cases my Polycom does not play the user-busy tone. When calling
pstn using ISDN it depends on the other side.

The case that works:

-          ISDN Alerting arrives -> 180 Ringing to phone

-          ISDN Disconnect user-busy arrives -> 486 Busy here to phone

The case that does not work:

-          ISDN Disconnect user-busy arrives -> 486 Busy here to phone



The polycom seems to play the busy-tone only if there was a 180 RINGING
before the 486 Busy here. Does anybody know if there is an option that I
can set from SipX to manipulate this behavior?



Sincerely



Jan



_____________________________
Jan Fricke (B.Sc.)

*IANT -
APPLIED NGN-TECHNOLOGIES

**Turn-Key VoIP/UC Solutions and More...


*IANT GmbH
Salzdahlumer Straße 46/48
D-38302 Wolfenbüttel

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