[asterisk-users] Polycom IP301 and Queues

Wed Jul 19 08:49:16 MST 2006

Dean,

Thank you for your help. I have it up and running. As soon as I get some
free time lets chat about what we need going forward. I have some
dollars to move this forward. If I can accommodate additional
requirements, all the better.

Michael

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Dean @
INKnBITs
Sent: Tuesday, July 18, 2006 3:57 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Polycom IP301 and Queues

The setup looks fine, I will run through what I did and the version,
there
might be an easier way.

cd /usr/src
svn checkout
http://svn.digium.com/svn/asterisk/team/bweschke/polycom_acd_functions/
asterisk-poly -r 30432

this will checkout the 30432 release and put in the the asterisk-poly
directory.

cd /usr/src/asterisk-poly

make clean
make  - I found you had to run make (2 or 3 times), it does come up on
the
screen and tells you to re-run. First run I think makes menuconfig,
second.... can't remember.
make mpg123 (if you want mp3 music on hold)
make install

The only problem I can find in this release is the meetme (conference
centre) does not compile, (but ACD does) and in the newer version the
meetme
works but not ACD. So I'm going to have two servers one for ACD on old
software and one for conference on new software. Not great but least it
works.

Hope that helps.

Regards,
Dean.

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com]On Behalf Of Michael
Miller
Sent: 17 July 2006 23:16
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Polycom IP301 and Queues

Thanks for the response and information.

The Asterisk version that I am using is Asterisk
SVN-bweschke-polycom_acd_functions-r37228. I went one revision back
using the following command:

svn checkout -r37228
http://svn.digium.com/svn/asterisk/team/bweschke/polycom_acd_functions
PolycomACD-07172006

 With no results. I am not as familiar with svn as cvs. I am not sure if
the -r option just labels or checks out the requested version. I will do
some reading tonight on svn.

I have install zaptel and libpri from the latest version of trunk.

I am using a Polycom 601 SIP version 1.6.6.0036.

The Polycom <reg> tag includes the following for line button one:

reg.1.displayName="Helpdesk"
reg.1.address="1000"
reg.1.label="Agent"
reg.1.type="private"
reg.1.thirdPartyName=""
reg.1.auth.userId=""
reg.1.auth.password="1000"
reg.1.server.1.address=""
reg.1.server.1.port=""
reg.1.server.1.transport="DNSnaptr"
reg.1.server.2.transport="DNSnaptr"
reg.1.server.1.expires=""
reg.1.server.1.register=""
reg.1.server.1.retryTimeOut=""
reg.1.server.1.retryMaxCount=""
reg.1.server.1.expires.lineSeize=""
reg.1.acd-login-logout="1"
reg.1.acd-agent-available="1"
reg.1.ringType="2"
reg.1.lineKeys="1"
reg.1.callsPerLineKey="2"

I assumed that the property reg.1.auth.userId="" is what you meant by
not putting in a username on the Polycom. I tried it both ways with no
luck.

I set the server addrss in the Polycom sip.cfg file.

The sip.conf entry for the Polycom looks like:

[1000]
type            = friend
secret          = 1000
context         = default
callerid        = "Helpdesk" <1000>
accountcode     = 1000

host            = dynamic
nat             = no
qualify         = 1000
canreinvite     = no

disallow        = all
allow           = ulaw

dtmfmode        = rfc2833

agentlogin	    = yes
agentcbcontext  = default

I also have an agent defined in the agnt.conf as:

agent => 2000,1234,Test Agent

Thanks again for the assistance!

Michael

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Dean @
INKnBITs
Sent: Monday, July 17, 2006 3:42 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom IP301 and Queues

I had the same problems, first of all, what version of asterisk are you
using? If you run the CLI whats the polycom_acd_functions verison 3xxxx.
If
you did a svn checkout http://........polycom_acd_function, then you
most
likely got the newest version. I had trouble with that.

Have you installed and compiled the zaptel/libpri from the trunk?
http://svn.digium.com/svn/zaptel/trunk and
http://svn.digium.com/svn/libpri/trunk ? You need these for the ACD
part.

On the polycom setup, make sure the username field is blank and that set
a
password.

In the Sip.conf, make sure the secret is the same as the polycom, and
that
you do not put a username= or a authname=

I can get you all the release/version numbers to download from the svn
tomorrow when back in work. It would be easier to talk you through it
when
in front of the server, but I'm in the UK and the time differences might
get
in the way!

Regards,
Dean.

----- Original Message -----
From: "Michael Miller" <Michael.Miller at sungardhe.com>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>
Sent: Monday, July 17, 2006 6:56 PM
Subject: RE: [asterisk-users] Polycom IP301 and Queues

I have been unable to get this branch of asterisk to work properly. I
can not get any SIP phone, Polycom or X-Lite, to register with the
server. If, on the same server, I recompile and install Trunk the phones
register properly. In doing this I made no changes to the conf files at
all. I simply recompiled and reinstalled.

Is there a trick to getting the phones to register? I made sure that the
phone SIP config and the agent config did no overlap. The phone will
register if I comment out the secret line.

I have not tried getting the ACD functionality to work at this point in
time...one issue at a time. Although this will be a big leap forward if
it works and I would be willing to put up a bounty to move this forward.

Thanks,

Michael

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Dean @
INKnBITs
Sent: Monday, July 17, 2006 3:38 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Polycom IP301 and Queues

Do you have a soft button on the IP301? I use the 501 and it works fine,
you
do have to use the special asterisk code for it to work correctly. It
lets
me login, logout, make the agent available/unavailable.

You can read about it at http://bugs.digium.com/view.php?id=6119

I found you must also use the trunk version of zaptel and libpri, and
make
sure you use auth on the phones in the config.

Hope thats what you looking for, if so, any problems just ask, its just
taken me 2 weeks to get it working great.

Regards,
Dean.

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com]On Behalf Of Julian
Varanini
Sent: 17 July 2006 00:15
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Polycom IP301 and Queues

Is there any way to use the polycom phones to log into and out of
queues?
So the polycom phone could show their current status in that queue?
logged
in / logged out for example.

Thanks

Julian

----------------------------------------
> Subject: RE: [asterisk-users] PRI dropouts
> From: pdhales at optusnet.com.au
> To: asterisk-users at lists.digium.com
> Date: Sat, 15 Jul 2006 20:47:17 +1000
>
>
> Hmm - I have had 2 bad PRI installs out of about 20, and both times it
> was faulty wiring from the Telco.
> But getting them to fix it can be a real struggle!
>
>
> Paul Hales
> Technical Manager
> www.asteriskit.com.au
>
>
> On Sat, 2006-07-15 at 12:23 +1000, James Sturges wrote:
> > Have had L O T S of trouble like this, the settings zap config files
> > seem to have to e exact, please send email to thinking at 1am.com.au
and
> > I will send config files.
> >
> >
> >
> > Thanks
> >
> >
> >
> > James
> >
> >
> >
> >
> >
______________________________________________________________________
> > From:asterisk-users-bounces at lists.digium.com
> > [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Kevin
> > Withnall
> > Sent: Saturday, 15 July 2006 11:05 AM
> > To: asterisk-users at lists.digium.com
> > Subject: [asterisk-users] PRI dropouts
> >
> >
> >
> >
> > Recently we cut over to using asterisk (trixbox 1.1.1) for our
> > production system.
> >
> >
> >
> > We are using a TE110P digium card (Primary rate) with a Telstra
onramp
> > 10.
> >
> >
> >
> > Sometimes when people call, on their end it doesn't seem to connect.
> > On our end, we get caller id, it passes ok to the sip phone but then
> > no-one is there.
> >
> >
> >
> > Anyone have any similar problems and worked out how to solve it ?
> >
> >
> >
> > Thanks.
> >
> >
> >
> >
> > _______________________________________________
> > --Bandwidth and Colocation provided by Easynews.com --
> >
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> > To UNSUBSCRIBE or update options visit:
> >    http://lists.digium.com/mailman/listinfo/asterisk-users
>
> _______________________________________________
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