Thanks Tony,
unfortunatly I cant reproduce the problem anymore. Really confusing.
Upgraded Polycom to 3.2.7. The problem is gone. Downgraded to 3.2.6. Its
still gone. Maybe there was something wrong in the polycom that vanished
just because it restarted when updating to 3.2.7.Ill have to wait until
murphy is back. L
I dont think this is a Patton problem because we are working with Patton
Gateways for a couple of years without any problem.
_____________________________
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*Von:* sipx-users-***@list.sipfoundry.org [mailto:
sipx-users-***@list.sipfoundry.org] *Im Auftrag von *Tony Graziano
*Gesendet:* Mittwoch, 24. Oktober 2012 14:00
*An:* Discussion list for users of sipXecs software
*Betreff:* Re: [sipx-users] Polycom Busy Problem
I would use 3.27, it's what we use here.
The only thing of note with the 4960 gateway is this: your channel hung
order (i.e. if the provider is sending you calls from channel 23 descending
and you are also sending the calls descending), it would create momentary
errors where people can't dial out, which is true for any PRI or ISDN
service. It's an instant error and the way to solve that is to change your
order where you send the calls to the gateway.
The patton has an amazing logging and debug capability too, using that to
further troubleshoot might provide enough detail as to what is going on,
including ISDN messages being sent/received from the carrier in the event
of a failure.
On Wed, Oct 24, 2012 at 7:49 AM, Jan Fricke <***@iant.de> wrote:
The gateway is a Patton Smartnode 4960. I think it is not the bottleneck.
In my test only two of the thirty channels were in use.
I compared the 486 messages in both cases. Aside the call-id, from and
to-tag etc. they are exactly the same. The only difference is the 180
ringing before that exists or not.
Just now I found the release notes of the 3.2.7 firmware of polycom. We are
currently using 3.2.6. The release notes contain something interesting
regarding this:
http://downloads.polycom.com/voice/voip/relnotes/SIP_Software_Release_Notes_3_2_7.pdf
*7**7**0**3**8 *Added support to generate ring back after a SIP 183 message,
followed by a SIP 180 message.
Thats not exactly the problem I see but maybe its worth a try.
Is there any reason against 3.2.7?
_____________________________
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*IANT -
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*Von:* sipx-users-***@list.sipfoundry.org [mailto:
sipx-users-***@list.sipfoundry.org] *Im Auftrag von *Tony Graziano
*Gesendet:* Mittwoch, 24. Oktober 2012 13:38
*An:* Discussion list for users of sipXecs software
*Betreff:* Re: [sipx-users] Polycom Busy Problem
I understand the call flow better now. What type of gateway it is "i think"
has a lot to do with it.
There is a difference between the called user is actually busy and when the
gateway has all of its channels full and cannot process the call. Will you
reveal what type of ISDN gateway you are using?
In the non-working scenario is the ISDN "full" (meaning does it have
channels available)? Do you have a call trace of a failed scenario?
What might be meaningful is any text message in the header as to why the
call failee (i.e. sip header is 486 but the reason code is "user busy"
which in this case might be coming from the gateway itself").
On Wed, Oct 24, 2012 at 7:14 AM, Jan Fricke <***@iant.de> wrote:
Its 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 doesnt
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 doesnt know anything about the reason why the call could not be
established.
_____________________________
Jan Fricke (B.Sc.)
*IANT -
APPLIED NGN-TECHNOLOGIES
**Turn-Key VoIP/UC Solutions and More... *
Fon: +49 (5331) 6794 0
Fax: +49 (5331) 6794 499
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IANT is eZuce <http://www.ezuce.com/> Elite Partner for EMEA
IANT is Member of GROUPLINK <http://www.grouplink.de/>
*Von:* sipx-users-***@list.sipfoundry.org [mailto:
sipx-users-***@list.sipfoundry.org] *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" <***@iant.de> wrote:
Hi,
Im 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
_____________________________
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