[asterisk-users] *#&%! Polycom...

[Ken D'Ambrosio]

I used to do lots of Asterisk, but got "an offer I couldn't refuse," and
went SysAdmin.  Well, now I'm trying to bring Asterisk in-house, and want
to set up a test system.  One thing I'd really like to get my hands on is
recent firmware, etc., for SoundPoint IP 430's.  Freedomphones.net, my old
source, seems to have been kaput about as long as I've been a sysadmin;
are there any other sources out there?  (And, yeah, if anyone wants to
e-mail them to me directly, I won't say no.)

Thanks much,

-Ken


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[asterisk-users] Polycom

[Forrest Beck]

I know this doesn't belong on this list but...  I am looking to see if
anyone is using Polycom and knows of a web based software for
creating/managing the cfg files for polycom phones.  I see that the
AsteriskNow will add provisioning support for Polycom phones.  Since
it is still in beta, I was just looking to see if there was anything
else out there.

Thanks!

--
***
Forrest Beck
IAXTEL: 17002871718
[EMAIL PROTECTED]
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[asterisk-users] Polycom 650

[Brent Torrenga]

List,

Question about the Polycom 650: when dialing the digits for a phone number,
and an incoming call comes in, does the phone prevent you from completing
your outgoing call until the phone stops ringing, like a Cisco 79X0 does?

--Brent




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[asterisk-users] Polycom Speakerphone

[Eric Jacksch]

Hello all,

We're using a lot of the linksys phones, and while user feedback is
generally positive, the speakerphone leaves a bit to be desired.

For those of you using the polycom desk phones, how do you find the built-in
speakerphone?

Thanks,
Eric



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[asterisk-users] Polycom Paging

[Michael Munger]

Does anyone have a link to a tutorial on how to do paging with Polycom
phones?

 

I am also looking for a tutorial on how to use the programmable buttons
on the Polycom to do speed dial, line presence (buddy watch) etc...

 

Yours,

Michael Munger

404-438-2128

[EMAIL PROTECTED]  

 

Attachment encrypted? click here
 .

 

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[asterisk-users] Polycom VLAN

[Jeremy Mann]

Just curious, if I have my Polycom IP 550 phone VLAN tag 30, will the packets I 
send from my PC(on the PC port of the phone) have the same VLAN tag?  THe PC is 
sending untagged packets.

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[asterisk-users] Polycom MyStat

[Douglas Garstang]

Has anyone ever gotten the Polycom Status feature, accessible via the 'MyStat' 
soft-key to work? When you change the status in this way, the phone does not 
send any communication to Asterisk, and it seems to have no effect in incoming 
calls. So... what's it for?

Doug


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[asterisk-users] Polycom Firmware

[Dovid B]

Hi Guys,
A while back (several months ago) I was having issues with wmy Polycom's and 
Asterisk. I was told to use a specific set of firmware and sip version. I am 
unable to find that email. Anyone know which ones work well with Asterisk ? (I 
believe it was 2.x )

Thanks,

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Re: [asterisk-users] *#&%! Polycom...

[Matt Watson]

On May 23, 2008 05:27:49 pm Ken D'Ambrosio wrote:
> I used to do lots of Asterisk, but got "an offer I couldn't refuse," and
> went SysAdmin.  Well, now I'm trying to bring Asterisk in-house, and want
> to set up a test system.  One thing I'd really like to get my hands on is
> recent firmware, etc., for SoundPoint IP 430's.  Freedomphones.net, my old
> source, seems to have been kaput about as long as I've been a sysadmin;
> are there any other sources out there?  (And, yeah, if anyone wants to
> e-mail them to me directly, I won't say no.)
>

My source was google, and I came across this almost right away: 
http://yum.trixbox.org/centos/5/RPMS/repodata/repoview/firmware-polycom-0-3.0.1-2.html

presumably their RPM includes firmware for all of the polycom's

I don't use polycom's and never actually downloaded the RPM... but it seems to 
me thats what you are looking for.

You should also be able to contact whomever you bought your Polycom's from to 
obtain the most recent versions.

-- 
Matt
http://www.mattgwatson.ca

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Re: [asterisk-users] *#&%! Polycom...

[Tilghman Lesher]

On Friday 23 May 2008 16:27:49 Ken D'Ambrosio wrote:
> I used to do lots of Asterisk, but got "an offer I couldn't refuse," and
> went SysAdmin.  Well, now I'm trying to bring Asterisk in-house, and want
> to set up a test system.  One thing I'd really like to get my hands on is
> recent firmware, etc., for SoundPoint IP 430's.  Freedomphones.net, my old
> source, seems to have been kaput about as long as I've been a sysadmin;
> are there any other sources out there?  (And, yeah, if anyone wants to
> e-mail them to me directly, I won't say no.)

Polycom no longer requires a reseller agreement to obtain firmware directly
from them:
http://www.polycom.com/usa/en/support/voice/soundpoint_ip/soundpoint_ip430.html
http://downloads.polycom.com/voice/voip/sp_ss_sip/spip_ssip_2_2_2_release_sig.zip

This isn't the latest firmware, but it's the latest that most people seem to
be running at this time.  I think people are still waiting for version 3.0 to
stabilize.

-- 
Tilghman

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[asterisk-users] Polycom 3.1.0RevB

[Andrew Joakimsen]

Could someone please tell me where to download Polycom 3.1.0RevB?
Polycom.com is not possible. Thanks.

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[asterisk-users] Polycom MWI.

[Ken D'Ambrosio]

Hey, all.  I'm all over MWI, but I gotta say that I think the Polycoms go
a bit over the top.  The blinking LED is enough for me; how do I disable
the stuttered dialtone and the audible warble?  I've looked through the
config files, but there are a HELL of a lot of options, and I haven't been
able to find those particular ones yet.

Thanks!

-Ken


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[asterisk-users] Polycom IP321?

[Matt Darnell]

A client of mine asked about a Polycom IP321..anyone else heard about it?

-Matt

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[asterisk-users] Polycom Digitmap

[Justin Phelps]

I'm working on replacing a SoundPoint 600 with a 650. I need to merge 
these two sets of digitmaps in the polycom sip.cfg file, because the 650 
locks up when I try to use the digitmap from the 600. I've included the 
default digitmap from a 3.1.3 RevB polycom release.

I'd like to merge these two digitmaps, but I don't want to reintroduce 
the lockup issue I was having with the 600 digitmap on the 650.

I've included my understanding of each definition below as well. I 
understand that the T at the end of certain definitions set a timeout, 
which is defined right after the digitmap. The Default timeout has a 
series of six 3's seperated by a |. The 600 definition has a single 3. 
What is the proper way to define these timeouts as well?

It also seems some of the definitions in the 600 group are redundant. 
Can you think of any reason as to why these shouldn't be combined?

Default:
[2-9]11|0T|011xxx.T|[0-1][2-9]x|[2-9]x|[2-9]xxxT

600:
[2-9]11|[*]xx|891xxx|[1-7]xxx|8[0-46-8]xx|8500|851xxx|9,1[2-9]x|9,xxxT

600
[2-9]11Information/Emergency (411 911 etc)
[*]xx  Service Numbers (*69)
891xxx 891 then 3 digits
[1-7]xxx   4 Digit Extension (1801, 1101, 5674)
8[0-46-8]xx8 then 0-4 or 6-8 then 2 digits
8500   8500
851xxx 851 then 3 digits
9,1[2-9]x  9, then 1, then [2-9] then 9 digits
9,xxxT 9, then 7 digit number

Default
[2-9]11  Information/Emergency (411 911 etc)
0Operator
011xxx.T 011 then three digits, than anything
[0-1][2-9]x  0-1 then 2-9 then 9 digits
[2-9]x   2-9 then 9 digits
[2-9]xxxT2-9 then 3 digits
-- 
Justin Phelps
www.onitato.com
850.866.6864

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[asterisk-users] Polycom Phones

[Robert Grignon]

Sorry if this is off topic

I have a "loud talker" in our call center and was asked if I can make
his voice louder to make him talk softer :-)

Does anyone know if you can do that with Polycom 430's

I found voice.gain.tx.headset but wasn't sure if that will make his
voice louder to the calling party or to himself...

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[asterisk-users] Polycom Questions

[»Steven Ringwald«]

Any Polycom gurus out there? If so, I have a few config file questions.

First off, does anyone have the daylight savings time rules written for 
this Sunday's big change?


Secondly, if there any way in the config file to tell the phone not to 
display the number of missed calls? I don't mind it keeping the missed 
calls list, I just don't want that running count.


Lastly, I am trying to get the dialplan to work, but have had no luck so 
far. I have tried defining it in the sip.cfg and/or the phone1.cfg, but 
have had no luck getting the phone to latch onto the numbers, and 
immediately dial. I am running with the 2.0.1 firmware, if that matters.


from sip.cfg:

  dialplan.removeEndOfDial="1">
 dialplan.digitmap="9[2-9]xx[2-9]xx|91[2-9]xx[2-9]xx" 
dialplan.digitmap.timeOut="3"/>

 
dialplan.routing.server.1.port="5060"/>
dialplan.routing.emergency.1.server.1="1"/>

 
  

from phone1.cfg:

dialplan.1.removeEndOfDial="1" dialplan.2.impossibleMatchHandling="0" 
dialplan.2.removeEndOfDial="1" dialplan.3.impossibleMatchHandling="0" 
dialplan.3.removeEndOfDial="1" dialplan.4.impossibleMatchHandling="0" 
dialplan.4.removeEndOfDial="1" dialplan.5.impossibleMatchHandling="0" 
dialplan.5.removeEndOfDial="1" dialplan.6.impossibleMatchHandling="0" 
dialplan.6.removeEndOfDial="1">
 dialplan.1.digitmap="9[2-9]xx[2-9]xx|91[2-9]xx[2-9]xx" 
dialplan.1.digitmap.timeOut="3" dialplan.2.digitmap="" 
dialplan.2.digitmap.timeOut="" dialplan.3.digitmap="" 
dialplan.3.digitmap.timeOut="" dialplan.4.digitmap="" 
dialplan.4.digitmap.timeOut="" dialplan.5.digitmap="" 
dialplan.5.digitmap.timeOut="" dialplan.6.digitmap="" 
dialplan.6.digitmap.timeOut=""/>

 
dialplan.1.routing.server.1.port="5060" 
dialplan.2.routing.server.1.address="" 
dialplan.2.routing.server.1.port="" 
dialplan.3.routing.server.1.address="" 
dialplan.3.routing.server.1.port="" 
dialplan.4.routing.server.1.address="" 
dialplan.4.routing.server.1.port="" 
dialplan.5.routing.server.1.address="" 
dialplan.5.routing.server.1.port="" 
dialplan.6.routing.server.1.address="" dialplan.6.routing.server.1.port=""/>
dialplan.1.routing.emergency.1.server.1="" 
dialplan.2.routing.emergency.1.value="" 
dialplan.2.routing.emergency.1.server.1="" 
dialplan.3.routing.emergency.1.value="" 
dialplan.3.routing.emergency.1.server.1="" 
dialplan.4.routing.emergency.1.value="" 
dialplan.4.routing.emergency.1.server.1="" 
dialplan.5.routing.emergency.1.value="" 
dialplan.5.routing.emergency.1.server.1="" 
dialplan.6.routing.emergency.1.value="" 
dialplan.6.routing.emergency.1.server.1=""/>

 
  



Thanks in advance!
Steve

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[asterisk-users] Polycom Power

[Mike Hammett]

I have a 501 with traditional power and a 301 with PoE. I rightfully assumed
that the traditional power from the 501 would work on the 301.

How do I get the PoE to work? Do I use the Polycom PoE cable in addition to
whatever PoE injection method I use? I have a Cisco PoE injector that works
on my Cisco AP350 and my 7960. No combination of this injector, the Polycom
cable, and the phone result in success.

I have 18v PoE injectors that I use for other things, but I hear that
802.3af is 48v, therefore probably wouldn't work.

How do I use Polycom PoE?

 

 

 

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Re: [asterisk-users] Polycom

[Andrew Joakimsen]

VGPS is a PHP/MySQL based provisioning system intended to generate
vendor-specific configuration files for Voice-over-IP (VoIP) devices
via a generic HTTP API.

Good luck...

http://sourceforge.net/projects/vgps

On 4/4/07, Forrest Beck <[EMAIL PROTECTED]> wrote:

I know this doesn't belong on this list but...  I am looking to see if
anyone is using Polycom and knows of a web based software for
creating/managing the cfg files for polycom phones.  I see that the
AsteriskNow will add provisioning support for Polycom phones.  Since
it is still in beta, I was just looking to see if there was anything
else out there.

Thanks!

--
***
Forrest Beck
IAXTEL: 17002871718
[EMAIL PROTECTED]
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[asterisk-users] polycom repair

[James Andrewartha]

Hi all,

Has anyone had any experience getting Polycom phones repaired? The screen on
one of our IP600s got smashed, and I'm wondering if it's worth the effort to
get it repaired, or if it'd just be cheaper to buy a new phone.

Thanks,

-- 
James Andrewartha
Systems Administrator
Data Analysis Australia Pty Ltd
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[asterisk-users] Polycom Phones

[Wiley Siler]

 

Can anyone tell me which config file tells the phone what file to load
as bootrom.ld?

Or is this hardcoded in the phone?  I just got a IP501 but I have a
bunch of IP500s...

Will the bootrom (2.6.2) work OK with both the IP500 and 501?

 

Thanks!

 

Wiley E. Siler
Director of Information Technology
4110 N. Scottsdale Rd. Ste 110
Scottsdale, Arizona 85251
(480) 296.0260
(866) 320.2083 ext. 1003
mailto:[EMAIL PROTECTED]
www.education2020.com   

 

 

 

Helping students on a mission. Graduation and beyond.

 

<>
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[asterisk-users] Polycom 650

[Klaverstyn, David C]

All,

 

I have a Polycom 650 phone, when turned on displays "Checking
application".

 

Can any give me some information as to what is wrong?  I have copied the
CFG files from a 601 phone to work with this 650. 

 

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[Asterisk-Users] polycom phones

[Richard]

Hi,

Has anyone experienced any problem with polycom phones not sending voice?
Occasionally some phones didn't send voice, a few times phone didn't play
the incoming voice. It didn't happen often on the IP 500 I bought last
November. But it happened quite a lot on the IP 600 I got recently.

Thanks,
Richard


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[Asterisk-Users] Polycom Images

[Manjit Riat]

Couple of months ago someone had posted to the list that polycom won't
support/provide to customers using polycom phones with asterisk. Is it still
the same way? If not then where do you get the images from coz I am looking
to pick up a polycom.



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[Asterisk-Users] Polycom IP4000

[Nathan]

I'm trying to get a few Polycom IP4000s working with asterisk, the
incoming calls from inside and outside of the network work fine with it,
but when I try calling out it just kicks me over to a busy siginal.
Anyone have any ideas on what causes this or how to fix it?

Thanks, 
Nathan


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[Asterisk-Users] Polycom configuration

[Chris Mason]

How do you configure your Polycom phones? Is it enough to configure one line
appearance? Or is there a way to configure a roll over?



Chris Mason
US Number: (646)722-0001 US Fax (815)301-9759
Skype: netconcepts
 

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[Asterisk-Users] Polycom Echo

[John Bittner]

Hi,

Just install 6 new polycoms at a customer and all of them have a major echo
issue. Have asterisk connected to the PSTN via digium 4 port fxo card in a
P4 running fedora.

I have tweaked zapata ran ztmonitor... just as a test I attached a cisco
7960 the cisco has no echo problems.

Another issue I came up on is when I enable echo training I no longer can
hear any inbound voice.

Does anyone know if there is a setting in the polycom config that will cause
this. 

The echo on the phones happen only on my side and only when I speak. The
polycoms play back my voice but delayed.

Any help would be appreciated.

Thanks

John Bittner
Simlab.net

 

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[Asterisk-Users] Polycom IP400

[Sean Cook]

I know that this phone will not do sip as it has limited memory, however
has anyone been successful getting it to work with mgcp/h323?

I mistakenly bought one of these thinking it was a 500 (description
noted the 500 documentation).  Anyway... just hoping I didn't blow $150
and that I can at least get it to work

sean

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[Asterisk-Users] Polycom Buddies

[Nihal]

I've got two Polycom 500's that I'm playing with, and I want view the status of 
either phone, (busy/on the phone/etc.) from the other.

I've got this cute little  'Buddies' button, and I can add contacts to that. 
But the status doesnt actually update.

Do I need to setup realtime for asterisk? Can anyone point me to some 
documentation or give me some hints?

Thanks,
Nihal
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[Asterisk-Users] Polycom IP500

[Tim Jackson]

Earlier tonight I moved our * box from an old Compaq w/ 3 X100P clone
cards to a 1U IBM server with a TDM04B card. I finally got the card
working in the server, but I'm having issues with these Polycom IP500s
now. Using the exact same config from the old server I'm getting weird
errors. Dial a number on the phone and it gives you dialtone but no user
interaction (if that makes sense) then after about 35-40 seconds it
displays "Line used remotely" and hangs up. Inbound calls ring, but you
can't answer them, registration seems to be ok, but I'm at a loss.

sip.conf:
[101]
type=friend
callerid="Tim Jackson - Home" <101>
secret=itsasekret
username=101
host=dynamic
dtmfmode=rfc2833
nat=yes
canreinvite=no
context=default
allow=all

OR

[101]
type=peer
callerid="Tim Jackson" <101>
secret=itsasekret
host=dynamic
dtmfmode=rfc2833
nat=yes
mailbox=101
canreinvite=no
context=default
disallow=all
allow=ulaw


app log from phone:

0106005724|res  |4|00|[ResFinderC]: Failed to download file BigLogo.bmp,
errno 0xdd.
0106005724|net  |*|00|Initial log entry. Current logging level 4
0106005724|key  |*|00|Initial log entry. Current logging level 4
0106005724|ssps |*|00|Application, comp. 1: Label=PolyDSP Orion Mem2
FS1, Version=1.1.2.0002 17-May-04 15:28
0106005724|ssps |*|00|Application, comp. 1: P/N=3150-11580-112.
0106005724|pps  |*|00|Initial log entry. Current logging level 4
0106005724|sip  |*|00|Initial log entry. Current logging level 4
0106005724|cfg  |4|00|Edit|Parse error 4 with local cfg
/ffs0/local/0004f2010524-phone_cfg.zzz
0106005724|app1 |*|00|Initial log entry. Current logging level 4
0106005724|cfg  |4|00|Edit|Parse error 4 with local cfg
/ffs0/local/0004f2010524-phone_cfg.zzz
0106005724|cfg  |5|00|Edit|Simple|error 0x0 when locking (param get)
0106005726|cfg  |5|00|Edit|Simple|error 0x0 when locking (param get)
0106005726|so   |*|00|[SoNcasC]: App-Ctx (Tim Jackson) [0-101]
0106005727|cfg  |5|00|Edit|Simple|error 0x0 when locking (setting
lcl.ml.lang="")
0106005727|cfg  |5|00|Edit|Simple|error 0x0 when locking (param get)
0106005727|slog |*|00|Initial log entry. Current logging level 4
0106005927|cfg  |4|00|Edit|Parse error 4 with local cfg
/ffs0/local/0004f2010524-phone_cfg.zzz


debug sip peer 101

Sip read:
REGISTER sip:192.9.200.9:5060 SIP/2.0
Via: SIP/2.0/UDP 192.9.202.2:5060;branch=z9hG4bK7ad588653F72B1C6
From: "Tim Jackson" ;tag=36767043-B9FDB2DA
To: 
CSeq: 1 REGISTER
Call-ID: [EMAIL PROTECTED]
Contact: ;methods="INVITE, ACK, BYE, CANCEL,
OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.3.4
Max-Forwards: 70
Expires: 3600
Content-Length: 0


11 headers, 0 lines
Using latest request as basis request
Sending to 192.9.202.2 : 5060 (non-NAT)
Transmitting (NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
192.9.202.2:5060;branch=z9hG4bK7ad588653F72B1C6;received=192.9.202.2;rpo
rt=5060
From: "Tim Jackson" ;tag=36767043-B9FDB2DA
To: ;tag=as024fe72d
Call-ID: [EMAIL PROTECTED]
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: 
Content-Length: 0


 to 192.9.202.2:5060
Transmitting (NAT):
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
192.9.202.2:5060;branch=z9hG4bK7ad588653F72B1C6;received=192.9.202.2;rpo
rt=5060
From: "Tim Jackson" ;tag=36767043-B9FDB2DA
To: ;tag=as024fe72d
Call-ID: [EMAIL PROTECTED]
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: 
WWW-Authenticate: Digest realm="angelinacounty.net", nonce="243b35d1"
Content-Length: 0


 to 192.9.202.2:5060
Scheduling destruction of call '[EMAIL PROTECTED]'
in 15000 ms
asterisk*CLI>

Sip read:
REGISTER sip:192.9.200.9:5060 SIP/2.0
Via: SIP/2.0/UDP 192.9.202.2:5060;branch=z9hG4bKf79496d429FB42CD
From: "Tim Jackson" ;tag=36767043-B9FDB2DA
To: 
CSeq: 2 REGISTER
Call-ID: [EMAIL PROTECTED]
Contact: ;methods="INVITE, ACK, BYE, CANCEL,
OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.3.4
Authorization: Digest username="101", realm="angelinacounty.net",
nonce="243b35d1", uri="sip:192.9.200.9:5060",
response="11f3478d812d35993018150f29fb5e81", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


12 headers, 0 lines
Using latest request as basis request
Sending to 192.9.202.2 : 5060 (NAT)
Transmitting (NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
192.9.202.2:5060;branch=z9hG4bKf79496d429FB42CD;received=192.9.202.2;rpo
rt=5060
From: "Tim Jackson" ;tag=36767043-B9FDB2DA
To: ;tag=as024fe72d
Call-ID: [EMAIL PROTECTED]
CSeq: 2 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: 
Content-Length: 0


 to 192.9.202.2:5060
Transmitting (NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.9.202.2:5060;branch=z9hG4bKf79496d429FB42CD;received=192.9.202.2;rpo
rt=5060
From: "Tim Jackson" ;tag=36767043-B9FDB2DA
To: ;tag=as024fe72d
Call-ID: [EMAIL PROTECTED]
CSeq: 2 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, R

[Asterisk-Users] Polycom registration

[Roger]

I have a PolyCom Soundpoint 500 sip phone.  I'm tring to get the phone 
registered on an asterisk box but am having no luck.  I get the 
following errors  192.168.22.196 being the phone and 22.254 being the 
asterisk box..

Apr 23 11:41:33 NOTICE[1133742896]: chan_sip.c:5623 handle_request: 
Registration from '"110" ' failed for 
'192.168.22.196'
Apr 23 11:42:05 NOTICE[1133742896]: chan_sip.c:5623 handle_request: 
Registration from '"110" ' failed for 
'192.168.22.196'
Apr 23 11:42:05 NOTICE[1133742896]: chan_sip.c:5623 handle_request: 
Registration from '"110" ' failed for 
'192.168.22.196'
Apr 23 11:42:37 NOTICE[1133742896]: chan_sip.c:5623 handle_request: 
Registration from '"110" ' failed for 
'192.168.22.196'

Attempting to dial out from the Polycom Phones gives a fast busy..  
Below I've included my sip.conf file - I'm wanting to set phone as x110.

[110]   
type=friend
username=110
secret=test
host=dynamic
context=home
callgroup=1
pickupgroup=1
canreinvite=yes
dtmfmode=rfc2833
;dtmfmode=inband
;[EMAIL PROTECTED]; put in for voicemail notification
callerid="Polycom" <110> ; put in for internal caller id only

I've reset the phone to factory defaults and started from scratch but 
still - no dice when it comes to registering this puppy.  I used the web 
interface to specify the username/password but still nothing. 

Any ideas or docs I could look at to get this Polycom phone setup?

--
Rock River Internet  Roger Grunkemeyer
202 W. State St, 8th Floor[EMAIL PROTECTED]
Rockford, IL 61101   815-968-9888 x101
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[Asterisk-Users] polycom dialplan

[Roger]

I recently had a bear of a time getting a Polycom Soundpoint 500IP up 
and registered..  Now that its registered I ran into a problem w/ the 
dialplan.

Needing to dial x101 I'd dial 10 - then get a fast buzy..  Also making a 
local call - dialing 95551212- would give me a fast busy after the 7th 
digit - so 9555121..  Same w/ LD calls...

This dialplan really got me down as I didn't find any docs on this.  
Look toward the web based interface was no help either.  To fix the 
problem I eventually did the following...

On the ftp server I replaced the dialplan settings in the sip.cfg from

   
   
   

to this..

   
   
   

This standard dialplan and files for the ftp server was grabbed off this 
page

http://www.freedomphones.net/polycom/files/

Hopefully this message will help someone down the road.

--
Rock River Internet  Roger Grunkemeyer
202 W. State St, 8th Floor[EMAIL PROTECTED]
Rockford, IL 61101   815-968-9888 x102
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[Asterisk-Users] polycom ip300

[harry gaillac]

hi all,

anybody could help me with polycom ip300
bootrom-2.6.1 sip 1.4.1 
conf files store on ftp server

SUBSCRIBE/NOTIFY and MESSAGE method failed.
either chan_sip is bad or i made mistakes but 
SUBSCRIBE/NOTIFY and MESSAGE method work with many sip
servers.

REGISTER is ok but asterisk send 407 error for
SUBSCRIBE ?!

I do hope a user could help me.
Harry







Découvrez le nouveau Yahoo! Mail : 250 Mo d'espace de stockage pour vos mails ! 
Créez votre Yahoo! Mail sur http://fr.mail.yahoo.com/
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[Asterisk-Users] Polycom IP501

[Noah Miller]

Hi All -

I noticed that the Polycom IP501's are now shipping.  Has anyone  
gotten one yet, and if so, what's different about the phone?  Any UI  
improvements, or is it just better hardware?


Thanks,
Noah
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[Asterisk-Users] Polycom 500...

[Carlos Chavez]

 I am having a strange problem with a couple of Polycom IP 500 phones.  I
know this is not related to Asterisk, but maybe someone here had the same 
problem.

 I configured my phones following the documentation at voip-info.org and
they are working very well.  The only problem I have is that when I dial an
extension like 1100 the phone changes that to 0110 and obviously the call
fails.  I have to dial slowly to get the 1100.  Does this have anything to do
with the dialplan?

--
Carlos Chavez
Director de Tecnología
Telecomunicaciones Abiertas de México S.A. de C.V.
Tel: +52-55-91169161 Ext 2001

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[Asterisk-Users] Polycom DTMF

[David Gomillion]

Problem:
   Polycom SoundPoint IP phones (running SIP) ceases to send DTMF that
Asterisk can detect and use.  It worked in 1.0.5, but has not worked
since.  This has been verified on SoundPoint IP 300's and SoundPoint IP
600's.

Workaround:
   It used to be that for DTMF to work, I had to set the mode in
sip.conf to "inband".  Without making any configuration changes on the
phones, I changed the DTMF mode to "rfc2833".  The DTMF is recognized.
No reboot to the phone is necessary, and remember that you can reload
the sip configuration with a reload in Asterisk, meaning your PBX
doesn't have to be restarted either.

Discussion:
   This is probably not the "right" way to fix this, as Polycom's
configurations, by default, will "encode DTMF in the active RTP stream".
There may have been a change in the sip channel's code that is causing
this.  Others on the list have indicated that they worked around the
problem by reverting the version of the sip app to an older version.
   As the new code usually fixes other problems, the solution of
reverting seemed to be counter-productive, so I tried other DTMF
signalling modes.  Thankfully, the stock Polycom configs will work with
Asterisk's sip.conf "rfc2833" DTMF mode, at least as of
CVS-v1-0-03/23/05-21:40:48.  When I get more time, or if someone else
has the time, an examination of what changed to cause this could enable
us to fix the heart of the matter.
   Other users on the Asterisk list (see thread "*-1.0.7 DTFM => Not
working" from 03/23/2005) have reported other UAs not working.
Therefore, there may be a bigger problem with the fundamental issue at
hand: when do we change DTMF in channels, to ensure compliance with
standards, as well as compatibility with older UAs.  

Hope this helps someone.

Sincerely,
David Gomillion

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[Asterisk-Users] Polycom Phones

[Peter Osborne]

Hi,

I have Asterisk running with some Polycom IP 300's. I've noticed that the 
default behaviour for these phones is to increment the "missed calls" 
indicator whenever you fail to pick up the phone, this is fine if a call is 
destined to my extension but often I have calls ringing multiple phones. This 
means that the missed call indicator is increased on all the phones that 
didn't answer the call.

Now, it seems there is a setting in the Polycom to only increment the 
indicator when the SIP server tells it to, I have enabled this option but I 
can't figure out how to get Asterisk to send a missed call indicator. 

Does anyone have some experience with this?

Thanks,
Pete
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[Asterisk-Users] Polycom Problems

[Tim Jackson]

We have Polycom IP500’s, and just starting recently
(after the broadvoice patch I might add) after about 1-2 days these phones
ring, and answer, but we get no audio on the phones. The caller can hear us,
but we cannot hear the caller. Its happened 4-5 times
and is only intermittent. No errors on the console, using g.711u. Any ideas?

 

Tim Jackson



Network Engineer





Angelina County, Texas





(936)639-4827x101 office





(936)414-6723 mobile



 






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[Asterisk-Users] Polycom IP500

[Chris Cherry]

Hey All,

First Time Writing.

I'm trying to set up my IP500 phones to register SIP with *. I input all
the (I assume) correct data in to the fields on the Web Interface. And I
get no notification that the phone is even attempting to register, no
failed messages etc. I have read that the Web interface is crap and the
XML config files is the way to go. Does anyone have a basic config file
that doesn’t change any defaults? I couldn't seem to find one. 

Extra Info:
Server is 192.168.0.3
Phone name/ext I want to be 301

Thanks,
Chris Cherry

-- 
No virus found in this outgoing message.
Checked by AVG Anti-Virus.
Version: 7.0.289 / Virus Database: 265.4.4 - Release Date: 11/30/2004
 

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[Asterisk-Users] Polycom IP500

[Chris]

Does anyone have a location to download the latest Polycom firmware etc?
Other than the extranet site, because I am not a reseller, there fore I
have no login.

[minirant]
And shouldn't end users be granted access to this kind of thing anyway?
Geeze
[/minirant]

Thanks,
Chris Cherry

-- 
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Version: 7.0.289 / Virus Database: 265.4.5 - Release Date: 12/3/2004
 

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[Asterisk-Users] Polycom Microbrowser

[David Gomillion]

Title: Message



I have just spent 
the morning playing around with a Polycom IP600's microbrowser.  Everything 
is working pretty well.  In answer to the question of what type of XML it 
runs, it appears to be more or less XHTML-compliant.  I have created a 
basic set of web pages allowing users to clock in and out against our MySQL 
timeclock system, running a PHP back-end.  It's running like a 
champ.
 
Please refer to the 
Administrator's guide, pages 106, 107, and 53 to get it set up 
correctly.
 
When creating web 
applications, be very careful about syntax.  Here's a few gotcha's that I 
have found:
 
WRONG:


 
RIGHT:


 
Notice the little 
slash on the end.  This is very XML-based, so you MUST have perfectly 
balanced tags.  Most browsers are intelligent enough to ignore little 
errors like these, but the polycom browser is very touchy.
 

Forget about 
 and  tags, they work if you remember the  and 
 counterparts, but don't seem to make much of a difference.  
Same with .
 
Other things I 
have noticed: selects (i.e. drop-down menus) don't seem to work, using a 
 gave me problems, and tables look like crap, so just use 
paragraphs.  One thing to note is that the title of the page will appear at 
the top, centered, so give your pages good titles.
 
One important 
security note, I have not yet found a way to clear or limit the history.  
So, if I press the back button 3 or 4 times, I tend to get re-clocked-out and 
other fun stuff.  I'm using server sessions, but it seems that the phone 
will keep those alive between times you press the "services" button.  If 
anybody finds a setting to fix this, let me know.  For now, I have set some 
session variables to allow an authentication to only be used one time before 
resetting it.  My next try will be setting 30-second sessions for PHP, but 
I really want the sessions to reset when the user switches away from the 
microbrowser.
 
Thanks,
David
 
 
 
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[Asterisk-Users] Polycom IP500

[Paul Hales]

The Polycom phones have an instant messaging function - any idea what is
required to make it work?

PaulH
Adairs

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[Asterisk-Users] Polycom Echo

[Matthew Marlowe]

Lately I have been experiencing a lot of echo from my Polycom phones.
Only I hear the echo and it's not on every call.  I've researched it
via google and the forums and every echo problem usually relates when
it's using a Zap card and not an IAX provider.

Can anyone give me some advice or where to look to help solve this
echo problem?  This never occurs on any of our other phones, Ciscos,
Grandstreams, Sipuras, etc.. Only on the polycoms.

Any help would be greatly appreciated.

Thanks in advance.

-- 
MBM
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[Asterisk-Users] Polycom phones

[Sudhir Kumar]

I have a couple of Polycom phones, bootrom 2.5.0, SIP 1.3.1.0056. Works
great with Asterisk when I power on the phone. However, after some time,
say an hour, I cannot receive calls on this phone. On Asterisk, when I
do "database show", it does show the phone in there, but it cannot reach
the phone. Is there a way to keep alive the connection with Polycom?

Any suggestions? 

Thanks,
-- sudhir

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[Asterisk-Users] Polycom Echo

[Anton Krall]

Guys.

I have about 20 Polycom 301, some 501 and some 600 and I really like the
phones, but I have a question and maybe somebody else has seen this. Seems
sometimes when people talk a bit loud, Polycom phones have a tiny bit of
echo, can this be controled with some kind of gain or AGC or something on
the xml files?

Has anybody seen this type of echo?

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[Asterisk-Users] Polycom 501

[MBIT Technologies]

Hi Guys

 

Just a quick question regarding on the 501, has
anyone been able to configure the transfer button and messaging buttons to work
with asterisk?

 

Can you share a configuration to do this?

 

Thanks in advance.






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[Asterisk-Users] Polycom configs?

[Michael Graves]

I have a number of Polycom phones to setup with my * server. For my
initial few phones I hand wrote configs. Does anyone here who uses
Polycom phones have some form of management utility for automating
their setup?

Michael

--
Michael Graves   [EMAIL PROTECTED]
Sr. Product Specialist  www.pixelpower.com
Pixel Power Inc. [EMAIL PROTECTED]

o713-861-4005
o800-905-6412
c713-201-1262



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[Asterisk-Users] Polycom Phones

[Kenneth Shaw]

Slightly off topic as this doesn't pertain directly to Asterisk, but
with the Polycom 500/501 phones, does anyone know how to correctly put a
custom logo for the idle screen on the device? I've read the Admin Guide
through and through and the information there is not enough to implement
it.

Any information you may have is much appreciated. Thanks!

-- 
Kenneth Shaw
Director of Technology
ExpiTrans, Inc.
2428 Newport Blvd #8
Costa Mesa, CA 92627
tel: 949 278 7288
fax: 866 494 5043
[EMAIL PROTECTED]

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[Asterisk-Users] Polycom Phones

[Chris Gamble]

Just got in a bunch of polycom phones for use on my shiny new asterisk box, but 
found 2 small issues I was wandering if someone could help me with.

First, though the phones support 2 call appearances, if I am on a call, the 
second call does not ring through -- it goes to voicemail instead of letting me 
put the first on hold to talk to the second. Is there a way to fix this?

The second is: a lot of my phones will not ring for internal extensions. They 
show up on the screen as a call ringing in, but the phone itself wont ring. 
About 50% however do ring. What could cause this?

As usual, thank you all for your kind  support in getting this far!
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[Asterisk-Users] polycom backlight?

[Dean Collins]

Do any of the polycoms have a backlight capability?

 

I have a Polycom 501 and during the day the numbers are easy
to read but I work in the evenings a lot and because of the placement/lighting
etc the numbers are impossible to read. 

 

 

Does the 601 (or 701 if such a thing exists have
backlighting?)

 

 

Cheers,

Dean

 






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[Asterisk-Users] Polycom Software

[Darrick Hartman]

Just wanted to let everyone know that Polycom now has firmware and 
bootrom software available for download on their public site.  It is one 
version back from what they currently supply to their authorized 
resellers, but at least it is available.


SIP Software version 1.5.3 and BootROM 3.1.0 are available (as well as 
older versions).


Darrick

--
Darrick Hartman
DJH Solutions, LLC
http://www.djhsolutions.com

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[Asterisk-Users] Polycom FW

[Bill Michaelson]

Anyone know how to obtain firmware and starter .cfg files for Polycom 
phones?  Despite registering at the Polycom web site, I can't locate 
this stuff.



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[Asterisk-Users] polycom dialplan

[Jonathan k. Creasy]

I have been having some trouble understanding the Polycom dialplan. I
would like the phone to put a 9 on the front of a 7 digit string such
that [2-9]XX becomes [9][2-9]XX. 

What do I need to do? 

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[asterisk-users] Polycom MyStat

[Douglas Garstang]

Has anyone ever gotten the Polycom MyStat soft-key to do anything?

Setting the status to something like 'Away', does not generate any outgoing SIP 
traffic from the phone. Calling into the phone either from a watched buddy, or 
other number, acts as if the status was never changed. A call to Polycom 
yielded no results. 

Thanks,
Doug.
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[asterisk-users] Polycom Firmware

[Forum Expansive]

What is the latest polycom firmware and where can I
get it?

 

Steve

 

 






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[Asterisk-Users] Polycom MWI

[Wilson Pickett]

Hi,

I have lookedaround and don't see this anywhere. Is there a way to
tell the ip500 to not make the aural MWI blips?
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[Asterisk-Users] polycom software

[Bartosz Jozwiak]

Dear users,

It might be slightly off topic.
I own couple 500 and 600  Polycom SoundPoint IP phones and
need to download new software for them.
The phones has been purchased from voipsupply.com

Is there a way to download such a software without becoming certified 
reseller?


Thanks,
Bartosz 


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[Asterisk-Users] Polycom files

[Jerry Jones]

Has anyone been able to get Polycom phones to use a different  
directory when it creates mac-phone.cfg and mac-directory.xml instead  
of placing them in the root directory. similar to specifying a log  
directory which does work.

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[Asterisk-Users] Polycom Echo

[Kevin Ragsdale]

Hello,

We just experienced a problem that we though might be useful to anyone
using Polycom phones.  We are installing a new system at one of our
remote offices and were experienced a ton of echo on our side while the
remote side was on speakerphone.  It turns out that the desk surface was
causing the echo - when the phone was lifted off the desk, the echo
disappeared.  We also were able to put a mousepad under the phone to
eliminate the echo as well.  Since the microphone is on the bottom of
the phone, there must be some kind of sound reflection going on.

Just a heads up - this is the first time we've seen this happen.

Kevin
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[Asterisk-Users] Polycom 601

[Dmitry Rudman [NSCGI]]

Can anyone please 
provide sip.ld for polycom ip 601 phone?
Thanks in 
advance.
 
Dmitry RudmanNetwork Systems 
Consulting Group, Inc.1778 Tahoe Circle DriveWheeling, IL 
60090Phone (847) 942-5003   Fax (847) 229-0585[EMAIL PROTECTED]
 

NSCGI is one source Total Solution Provider offering Computer Consulting Services to businesses throughout Chicagoland area.

The information transmitted is intended only for the person or entity to which it is addressed and may contain confidential and/or privileged material. Any review, retransmission, dissemination or other use of, or taking of any action in reliance upon, this information by persons or entities other than the intended recipient is prohibited. If you received this in error, please contact the sender immediately for further instruction.

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[Asterisk-Users] Polycom 501

[Curt Shaffer]

Does anyone out there have a sample config they can share
for the Polycom 501? Is it possible to do “sub configs” like you
can with the Aastra 9133i? It could be just me but the boot configs seem a bit
cryptic compared to the aastra. Also do any of you have any comparisons between
these and the Aastra 9133i?

 

Thanks

 

Curt






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[Asterisk-Users] polycom ftp

[hgaillac-sip]

Anydody need some access to polycom ftp server ?
Harry



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[Asterisk-Users] Polycom Files

[hgaillac-sip]

Hello,

If somebody need the latest Polycom Files contact me
or look at ftp://nxs.yi.org

Harry



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[Asterisk-Users] Polycom Configuration

[Sean Cook]

I have been playing around with sipX for a couple of days now, and while
I don't really like it (just feels wierd), I do really like the
management interface for provisioning phones.  I was wondering if anyone
had considered ripping this out of sipX or porting it to a simple php
interface or something?

If not I think it would be of interest to create a phone provisioning
tool for asterisk (although not directly in asterisk...).


Takers?  Thoughts?  proverbial bugger off?

Regards,

Sean
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[Asterisk-Users] Polycom subscriptions

[Douglas Garstang]

Somewhat off topic.

We upgraded a Polycom phone from SIP v1.6.3 to v1.6.6
The phone will no longer send SIP subscription messages for buddies to 
Asterisk. I have broken the directory file down to make it as simple as 
possible.
Here is what it contains.
 





Presley
Elvis
2944093
1
3

0
0
1
0



 
I ran a network trace of all traffic to and from the phone on boot. Here is the 
output of that...
 
  1   0.00 219.187.128.95 -> 219.187.142.203 SIP Request: REGISTER 
sip:ua1.ipt.twoeighty.com
  2   0.000101 219.187.142.203 -> 219.187.128.95 SIP Status: 100 Trying(1 
bindings)
  3   0.000148 219.187.142.203 -> 219.187.128.95 SIP Status: 401 Unauthorized   
 (1 bindings)
  4   0.211291 219.187.128.95 -> 219.187.142.203 SIP Request: REGISTER 
sip:ua1.ipt.twoeighty.com
  5   0.211432 219.187.142.203 -> 219.187.128.95 SIP Status: 100 Trying(1 
bindings)
  6   0.237595 219.187.142.203 -> 219.187.128.95 SIP Status: 200 OK(1 
bindings)
  7   3.987556 219.187.142.203 -> 219.187.128.95 SIP Request: NOTIFY sip:[EMAIL 
PROTECTED] (text/plain)
  8   4.087355 219.187.128.95 -> 219.187.142.203 SIP Status: 200 OK
 
The phone is not sending a SIP subscription message for the watched buddy in 
the directory.
This worked previously in SIP software version 1.6.3. 
We completely upgraded the sip and phone1 xml files to the ones supplied with 
sip software 1.6.6
It obviously isn't an Asterisk problem.

Doug.
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[Asterisk-Users] Polycom Queues

[Douglas Garstang]

Has anyone integrated Asterisk Queues with Polycom phones?

What I'd like to do is display the agent status next to their appearance. I 
don't see much discussion about this.
This is not the same thing as setting 1 against the appearance in the 
phone directory.

Thanks
Doug.

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[Asterisk-Users] Polycom TOS

[Jonathan k. Creasy]

Does anyone know the format for the TOS element in the Polycom
config?

 

-Jonathan

 

Jonathan Creasy
Network Engineer

BluegrassNet Development

www.bgnd.com www.bluegrass.net

o. 502-589-4638

c. 502-889-5567

h. 502-541-0566

 






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[Asterisk-Users] Polycom VLANs

[Rob Terhaar]

So has anyone had any experience working with the polycom 501 or 301 and vlans? We run dell managed switches here, so we don't have the luxury of running CDP to force the VOIP vlan. I haven't been able to get the polycom phones to talk on a manually set vlan. I have some junky sipura phones that work fine-(get dhcp, register to asterisk etc) when i manually set them to vlan4.
Any advice you guys have would be greatly appreciated!
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[Asterisk-Users] Polycom Microbrowser

[hgaillac-sip]

Hello,

I read the polycom microbrowser post here
http://www.voip-info.org/wiki/index.php?page=Polycom+Microbrowser

Can we access a webmail application like horde/imp or
others (which ones) to read and listen  voicemails ,
send e-mails, ... ?

Regards
Harry








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[Asterisk-Users] Polycom MWI

[Kerry Garrison]

I have tried 
everything from voip-info and I still cant get the Polycom 501/601 to display 
the MWI indicator light. Everything else works just fine. I am using FreePBX set 
to users and devices mode. Here is the MWI section of the phonexxx.cfg 
file:
 

 
msg.mwi.1.subscribe="" msg.mwi.1.callBackMode="contact" 
msg.mwi.1.callBack="*97" msg.mwi.2.subscribe="" 
msg.mwi.2.callBackMode="disabled" msg.mwi.2.callBack="" 
msg.mwi.3.subscribe="" msg.mwi.3.callBackMode="disabled" 
msg.mwi.3.callBack="" msg.mwi.4.subscribe="" 
msg.mwi.4.callBackMode="disabled" msg.mwi.4.callBack="" 
msg.mwi.5.subscribe="" msg.mwi.5.callBackMode="disabled" 
msg.mwi.5.callBack="" msg.mwi.6.subscribe="" 
msg.mwi.6.callBackMode="disabled" msg.mwi.6.callBack=""/> 

 
   

 
i have also 
tried
 
msg.mwi.1.callBackMode="register"
 
 Kerry GarrisonDirector of 
Technical ServicesTech Data Pros - Orange County's Mobile IT Service 
Provider(949) 502-7819 x200 - [EMAIL PROTECTED]http://www.techdatapros.com 

 
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[Asterisk-Users] Polycom Delay

[Kevin Smith]

Hey everyone,

Hopefully someone can point me in the right direction for this. 
Currently we have two offices, all using Polycom 601 Revsion E I think. 
All have the same configurations and firmware versions.


The differences:
Office A: public IP address.
Office B: NAT (router has a static IP)

Office A: Same state as the asterisk server (Michigan)
Office B: Wisconsin

Office A: T1 network to the colo where the asterisk server is located
Office B: Wireless connection (2 tower hops I think) (our wireless 
connection, we are a small ISP) to our backbone to the colo


Okay, so calls going to and from office A have no problems at all. 
Office B is having a bit of a delay (about 5 seconds before the CLI 
shows the call is even started). The odd part is, it only happens when 
they are making an outbound call. Incoming calls go directly to them 
without any problems. Both offices for external calls use our PRI we 
have installed and all interal are SIP. I think also internal calls are 
having the same problem, but that I haven't had a 100% sure answer if it 
is or isn't, but I know for sure the PRI calls are.


My question is, does it sound like the phone is causing the problem, or 
the network being NAT, wireless connection, or both having more to do 
with the problem. While I know it isn't an answer you can say, hey this 
is the solution, I would like any input or experience that anyone has 
had with a problem like this.


Thanks
Kevin
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[asterisk-users] Polycom MyStat

[Douglas Garstang]

Has anyone ever gotten the Polycom MyStat soft-key to do anything?

Setting the status to something like 'Away', does not generate any outgoing SIP 
traffic from the phone. Calling into the phone either from a watched buddy, or 
other number, acts as if the status was never changed. A call to Polycom 
yielded no results. 

Thanks,
Doug.
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[asterisk-users] Polycom HDVoice

[Forrest Beck]

Has anyone used the Polycom HDvoice phone yet?  I am curious if it
uses a different codec.  Does it actually sound any better?
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[asterisk-users] Polycom IP650

[Dean Collins]

Does anyone have an actual delivery date on the new Polycom
HD IP650’s?

 

I’m getting sick of not having a backlit screen and
thinking of upgrading.

 

 

 

 

Cheers,

 

Dean

 

 






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[asterisk-users] Polycom BLF

[Jeff LaCoursiere]

Struggling with an IP650 and 7 IP335s this morning.  I have the following 
hints defined (courtesy of FreePBX 2.9):


extensions_additional.conf:exten => 300,hint,SIP/300
extensions_additional.conf:exten => 301,hint,SIP/301
extensions_additional.conf:exten => 302,hint,SIP/302
extensions_additional.conf:exten => 303,hint,SIP/303
extensions_additional.conf:exten => 304,hint,SIP/304
extensions_additional.conf:exten => 305,hint,SIP/305
extensions_additional.conf:exten => 307,hint,SIP/307
extensions_additional.conf:exten => 308,hint,SIP/308
extensions_additional.conf:exten => 322,hint,SIP/322
extensions_additional.conf:exten => 350,hint,SIP/350
extensions_additional.conf:exten => 400,hint,SIP/400

The Polycoms are all pulling an XML directory via FTP where each extension 
has "" (Buddy Watch) set to 1:



Mehra
Ray
301
101
1


This all actually works fine, and from the reception phone (the 650) I can 
see the status of all the extensions, and if I dig into some menus on the 
335 I can see status as well.  So I would expect that "core show hints" 
would show '8' for all extensions, but it doesn't:


artha*CLI> core show hints

-= Registered Asterisk Dial Plan Hints =-
300@ext-local   : SIP/300 
State:IdleWatchers  7
301@ext-local   : SIP/301 
State:IdleWatchers  8
302@ext-local   : SIP/302 
State:IdleWatchers  8
303@ext-local   : SIP/303 
State:IdleWatchers  8
304@ext-local   : SIP/304 
State:InUse   Watchers  8
305@ext-local   : SIP/305 
State:IdleWatchers  7
307@ext-local   : SIP/307 
State:IdleWatchers  1
308@ext-local   : SIP/308 
State:IdleWatchers  7
350@ext-local   : SIP/350 
State:IdleWatchers  1
400@ext-local   : SIP/400 
State:InUse   Watchers  7


- 11 hints registered


Something seems broken here.  And the 650 seems to "lose" its hint for a 
phone once in a while, and report it as unreachable, even though it can 
easily make and receive calls from it.


Am I tilting at windmills?  Is this really unstable or has someone made it 
work solidly?


Thanks!

--

Jeff LaCoursiere
SunFone
340-715-7600 x222
j...@sunfone.com


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[asterisk-users] Polycom IP335

[ERIC HERRON]

I am posting here since you guys are my last hope.

 

I am trying to configure a Polycom Soundpoint IP 335 with MWI.

Is there any way to eliminate the scrolling messages and Msgs softkey?

I am trying to get it where it's just the light that indicates the new
messages.

I don't know if Asterisk has to send a different notification or what have
you.

Thanks,

--Eric

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[Asterisk-Users] Polycom IP Phones

[Kevin Thompson]

Does any one have any experience setting up asterisk with polycom IP 
phones? All i have been able to figure out about them is that they connect 
to an FTP site on "boot". I tried going to the site to see what files are 
there but it seems they deny directory browsing.

Any one have any clues as to how i could configure my polycom IP phones?

thanks

Kevin

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[Asterisk-Users] Polycom Soundpoint IP600

[Roman Pelikh]

Does anyone have the Admin password for the phone
in order to change configuration

Roman
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[Asterisk-Users] Polycom phones update

[mattf]

Hello,

We have updated the Wiki page for Polycom phones:
http://www.voip-info.org/tiki-index.php?page=Polycom+Phones

We posted several configuration specs as well as a link to an admin guide
for the phone.

We also posted a link on there to two firmware versions for download.

The official Asterisk-Polycom support website should be up and live sometime
in January.

If anyone has anything to add or would like to be on our list of experts for
Asterisk-Polycom when it goes live, please let me know.

Enjoy,

MATT---
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[Asterisk-Users] Polycom Sip Registration

[Brent Franks]

Hello,

 

Has anyone on the list been able to successfully setup a Polycom Soundpoint 500 IP
phone?  I am getting failed
registrations, and the Polycom documentation is not
very precise.  Their web interface isn’t
helping much either.

 

Thanks in advance,

 

Brent

[Asterisk-Users] Polycom Soundpoint IP400

[Elijah Hobbs]

I am trying to use a Polycom Soundpoint IP400 with my asterisk setup, 
and have been unable to find the proper firmware and application files 
to make it work.  The phone can access the FTP server, and downloads the 
IP500/600 configuration files, but claims that the sip.ld file is larger 
than its file system.

I can think of two possible ways to make this work.

One is to find the IP400 firmware and configure the phone to use H.323.  
I have had no success with finding the firmware for this phone on the 
internet, and Polycom tech. support has been rather elusive, so if 
anyone could point me to the files I would be much abliged. 

The second method is to upgrade the file system on the IP400 and see if 
it will run the SIP application.  This would be prefferable, because 
this phone is intended to be a low cost proof of concept, and we will go 
to Sip phones once the concept has been proven to the right people.  I 
imagine that this would involve replaceing a non-volatile memory chip 
inside the phone with an equivalent chip of higher capacity.  The 
possibility of doing this depends heavily on the design of the phone 
(whether the memory size is hard coded into the OS, whether the memory 
is external to the processor, what type of processor is used, etc.)

There is also a possibility that I am barking up the wrong tree.  If I 
am, I would appreciate it if someone could guide me to the right tree.

Thanks in advance.

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[asterisk-users] Polycom 320 Issue

[Klaverstyn, David C]

Hi All,

 

I'm not sure if this is related directly to asterisk or not but on my
Polycom 320 when I try to dial a number smaller than 4 digits I get an
error on the phone saying "Enter more digits".

 

The dial plan section is listed below.

 

   

  

  

 

 

  

   

 

Any help would be appreciated. 

 

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[asterisk-users] Polycom Key Assignment

[Tim Nelson]

Hello! Is it possible to assign any of the soft keys on the Polycom IP series 
handsets to a specific function in the feature menu? I'd like to assign one of 
the keys below the LCD to function as a Do Not Disturb button but I have not 
been able to find a helpful guide or proper documentation that is 
understandable for the task. Any help is greatly appreciated. Thank you!

Tim Nelson
Systems/Network Support
Rockbochs Inc.


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Re: [asterisk-users] Polycom 650

[Scott Plante]

No, you can keep dialing and make your call if you wish, or you can 
answer the call.

-- 
Scott Plante, CTO
Insight Systems, Inc.
(+1) 404 873 0058 x104
[EMAIL PROTECTED]
http://zyross.com 



Brent Torrenga wrote:
> List,
>
> Question about the Polycom 650: when dialing the digits for a phone number,
> and an incoming call comes in, does the phone prevent you from completing
> your outgoing call until the phone stops ringing, like a Cisco 79X0 does?
>
> --Brent
>
>
>
>
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[asterisk-users] polycom auto answer

[Jerry Geis]

I was trying to get my polycom phone to auto answer.
I added this to the dialplan. Used a different phone to call "22"
and the phone rang it did not auto answer.

Did I miss something?

exten => 22,1,SipAddHeader(Call-Info:=\;answer-after=0)
exten => 22,n,SipAddHeader(Alert-Info: Ring Answer)
exten => 22,n,Set(__SIPADDHEADER=Call-Info:\;answer-after=0)
exten => 22,n,Set(__ALERT_INFO=Ring Answer)
exten => 22,n,Set(__SIP_URI_OPTIONS=intercom=true)
exten => 22,n,Dial(SIP/404)

Jerry

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[asterisk-users] Polycom phone reboots

[Steven C. Blair]

   We are using Asterisk and SER  with Polycom 550 phones running SIP version 
2.2.2.0084. The phones register to SER. If an AOR appears on more than one 
phone when a call arrives for that AOR one, some or all of the Polycom phones 
reboot. I can't seem to find the source of this problem. Has anyone else 
encountered this problem?

Thanks,Steve


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[asterisk-users] Polycom Advanced Features

[Andreas van dem Helge]

Anyone have shared lines (sla.conf) working with Polycom phones? Also,
has anyone figured out if its possible to do 1 button call park with
the softkeys?

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[asterisk-users] Polycom IP300 Language

[Jose Flores Galicia]

Hi all.

Does anyone know how to change default language on polycom SoundPoint IP300?

Thanks

-- 
Jose Flores Galicia
<<[EMAIL PROTECTED]>>
BriefCode && Code Based Training
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[asterisk-users] Polycom 601 + Headset

[Dovid B]

Hi List,
I am using a Plantronics CS50 head set with my Polycom 601. I use the button on 
it to pick up calls. Is there any way to have the phone set up that if I pick 
up with the button on the headset that it sends the call to the headset and 
that I don't have to press the headset button on the phone every time ?

Thanks.

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[asterisk-users] Polycom Provisioning Tool

[Michael Munger]

Not sure if one exists, but someone had asked me for this a while ago.
Here it is! My Polycom Provisioning Tool. Notice the version is 0.0.1.
Just a concept program (but it works well).

 

I am open for suggestions, feature additions, and bug fixes. Email me
with any requests. I want to improve this to make it really useful for
the community, so let me know what you think.

 

http://www.wintrisk.com/ppt.html

 

 

Michael Munger, dCAP

High Powered Help, Inc

[EMAIL PROTECTED]  

404-438-2128 x 101

 

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[asterisk-users] Polycom Park Button

[Kelly Opal]

Hi
I have a Polycom 501 phone. I set the park feature to 1 in sip.cfg and
the button shows up just fine. However when you press it it does
nothing. I have the t and T in the dial string. Is there some trick to
getting it to work with asterisk 1.4.

Thanks

Kelly


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Re: [asterisk-users] Polycom Speakerphone

[John Millican]

On Monday November 12 2007 9:38 am, Eric Jacksch wrote:
> Hello all,
>
> We're using a lot of the linksys phones, and while user feedback is
> generally positive, the speakerphone leaves a bit to be desired.
>
> For those of you using the polycom desk phones, how do you find the
> built-in speakerphone?
>
> Thanks,
> Eric

I have found the polcom speaker phone to be very good on the 320's, 330's, and 
the 501's.  Clear clean voice even in relatively noisy areas.
JohnM


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Re: [asterisk-users] Polycom Speakerphone

[Doug Lytle]

Eric Jacksch wrote:
> For those of you using the polycom desk phones, how do you find the built-in
> speakerphone?
>
>   

Excellent!


Doug


-- 
 
Ben Franklin quote:

"Those who would give up Essential Liberty to purchase a little Temporary 
Safety, deserve neither Liberty nor Safety."



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Re: [asterisk-users] Polycom Speakerphone

[Michael Graves]

On Mon, 12 Nov 2007 09:38:57 -0500, Eric Jacksch wrote:

>Hello all,
>
>We're using a lot of the linksys phones, and while user feedback is
>generally positive, the speakerphone leaves a bit to be desired.
>
>For those of you using the polycom desk phones, how do you find the built-in
>speakerphone?
>
>Thanks,
>Eric

Actually, IMHO, the Polycom speakerphone is the standard by which all
other should be judged. I have a number of 500, 600 and 430 models in
service and they're all very good.

Even the little CS100 USB speakerphone device has been excellent.

Michael

--
Michael Graves
mgravesmstvp.com
o713-861-4005
c713-201-1262
sip:[EMAIL PROTECTED]
skype mjgraves
fwd 54245



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Re: [asterisk-users] Polycom Speakerphone

[Alan Lord]

Michael Graves wrote:
> On Mon, 12 Nov 2007 09:38:57 -0500, Eric Jacksch wrote:
> 
>> Hello all,
>>
>> We're using a lot of the linksys phones, and while user feedback is
>> generally positive, the speakerphone leaves a bit to be desired.
>>
>> For those of you using the polycom desk phones, how do you find the built-in
>> speakerphone?
>>

I am using the Polycom Communicator C100S on Ubuntu Linux. And despite 
most of the echo cancellation and noise reduction technology being only 
available in their Windows XP driver, the sound quality is excellent. I 
sometimes need to plug in a headset, but the desktop mic(s) work 
brilliantly.

HTH

Alan


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Re: [asterisk-users] Polycom Speakerphone

[Doug]

At 08:38 11/12/2007, Eric Jacksch wrote:
 >Hello all,
 >
 >We're using a lot of the linksys phones, and while user feedback is
 >generally positive, the speakerphone leaves a bit to be desired.
 >
 >For those of you using the polycom desk phones, how do you find the built-in
 >speakerphone?
 >
 >Thanks,
 >Eric

Excellent speakerphone.  Extremely cumbersome to
configure.


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Re: [asterisk-users] Polycom Speakerphone

[John Millican]

On Monday November 12 2007 1:50 pm, Doug wrote:
> At 08:38 11/12/2007, Eric Jacksch wrote:
>  >Hello all,
>  >
>  >We're using a lot of the linksys phones, and while user feedback is
>  >generally positive, the speakerphone leaves a bit to be desired.
>  >
>  >For those of you using the polycom desk phones, how do you find the
>  > built-in speakerphone?
>  >
>  >Thanks,
>  >Eric
>
> Excellent speakerphone.  Extremely cumbersome to
> configure.
>
I do not understand how you can say that the Polycoms are  "Extremely 
cumbersome to configure".   I find them rather nice.  Once you have one 
working config it is very easy to copy that config over to the mac address 
files for the other phones that you have and only change the per phone bits.  
Set up a site wide sip.cfg and then use phone-(macaddress).cfg files for the 
individual settings for each phone. real nice when you have more than a 
couple phones to configure.
It is not my intention to start any war here just giving my 2 cents worth.
JohnM



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Re: [asterisk-users] Polycom Speakerphone

[David Gomillion]

Doug wrote:
> At 08:38 11/12/2007, Eric Jacksch wrote:
>  >Hello all,
>  >
>  >We're using a lot of the linksys phones, and while user feedback is
>  >generally positive, the speakerphone leaves a bit to be desired.
>  >
>  >For those of you using the polycom desk phones, how do you find the built-in
>  >speakerphone?
>  >
>  >Thanks,
>  >Eric
>
> Excellent speakerphone.  Extremely cumbersome to
> configure.
>   
I agree about the speakerphone, and disagree with the claim about 
configuration. The XML is extremely to generate through scripts, and 
once the framework is built, I find it to be far simpler to manage the 
deployment than other IP phones.

Of course, YMMV.

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Re: [asterisk-users] Polycom Speakerphone

[Doug]

At 13:05 11/12/2007, John Millican, wrote:
 >> Excellent speakerphone.  Extremely cumbersome to
 >> configure.
 >>
 >I do not understand how you can say that the Polycoms are  "Extremely
 >cumbersome to configure".   I find them rather nice.  Once you have one
 >working config it is very easy to copy that config over to the mac address
 >files for the other phones that you have and only change the per 
phone bits.
 >Set up a site wide sip.cfg and then use phone-(macaddress).cfg files for the
 >individual settings for each phone. real nice when you have more than a
 >couple phones to configure.
 >It is not my intention to start any war here just giving my 2 cents worth.
 >JohnM


At 13:13 11/12/2007, David Gomillion wrote:
 >> Excellent speakerphone.  Extremely cumbersome to
 >> configure.
 >>
 >I agree about the speakerphone, and disagree with the claim about
 >configuration. The XML is extremely to generate through scripts, and
 >once the framework is built, I find it to be far simpler to manage the
 >deployment than other IP phones.
 >
 >Of course, YMMV.



I agree that once the .cfg files are working, duplicating
them to use on other phones if fairly straightforward.

That having been said, getting the .cfg files hammered
into a usable form is quite tedious.

Also, getting the Polycoms to accept the new configs
frequently involve defaulting the phone, or resetting
the "local configuration".

Upgrading firmware on older phones may require many
steps by upgrading through intermediate versions.

Compared to an analog ATA, Polycoms are about 10 times
more difficult and time consuming to get running well.
If you haven't had to deal with these problems, count
yourself very lucky.





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Re: [asterisk-users] Polycom Speakerphone

[F6HQZ]

Hi,

Excellent ! For me, Polycom have the best audio.
Just behind, I like also Aastra.

Best Regards,
Francois


-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] la part de Eric Jacksch
Envoyé : lundi 12 novembre 2007 15:39
A : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : [asterisk-users] Polycom Speakerphone


Hello all,

We're using a lot of the linksys phones, and while user feedback is
generally positive, the speakerphone leaves a bit to be desired.

For those of you using the polycom desk phones, how do you find the built-in
speakerphone?

Thanks,
Eric



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No virus found in this incoming message.
Checked by AVG Free Edition.
Version: 7.5.503 / Virus Database: 269.15.30/1125 - Release Date: 11/11/2007
21:50


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[asterisk-users] Polycom call drops

[YT Lim]

Call transfer drops in the following scenario:

1) incoming call to a Polycom via Asterisk
2) call answered
3) tranfer button pressed
4) talk to intended B-party for about 5-10 seconds
5) incoming call drops

This happens every time. Has anyone encountered the
same problem? Would appreciate any suggestion.

Asterisk version: 1.4; Polycom phone: IP301

/Why Tea


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