Yes, perhaps a script would always be better than hand-touching these
files, and getting an XML editor only really makes it easier on the
eyes.
On the same subject, I have noticed that Snom and Linksys phones do
not support FTP provisioning - only TFTP and HTTP. With TFTP being an
insecure
I am confused how TFTP is less secure than HTTP. TFTP does not allow any
browsing, ever. Neither technologies will allow the device to
authenticate before downloading a configuration file, and both are
easily secured by only permitting connections from specific hosts.
Robert McNaught wrote:
On Thu, May 15, 2008 at 10:08 PM, Robert McNaught
[EMAIL PROTECTED] wrote:
The way I understood it is that TFTP does not allow you to set a
username and password in a URL
like tftp://username:[EMAIL PROTECTED] is not possible
when setting option 66
Is it not possible to require a username
On Thu, 15 May 2008 10:23:14 -0700, Robert McNaught wrote:
Yes, perhaps a script would always be better than hand-touching these
files, and getting an XML editor only really makes it easier on the
eyes.
On the same subject, I have noticed that Snom and Linksys phones do
not support FTP
Limiting to HTTP would be OK if every customer had a static IP - if
you have small offices, then they maybe on DSL without static IP,
which makes that difficult - you could of course force your users to
have static IPs.
Robert
On Thu, May 15, 2008 at 1:45 PM, Atis Lezdins [EMAIL PROTECTED]
.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Robert
McNaught
Sent: May 15, 2008 6:41 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom XML Files / asterisk
Limiting to HTTP would be OK if every customer had
Sorry to be a pest, but does anyone have any ideas on this? I've
opened a bug, but I was hoping someone else on the list has
encountered this issue before.
Thanks,
Jason
On May 9, 2008, at 12:36 PM, Jason Dixon wrote:
We have a remote office that's having problems with their Polycom.
Sorry to be a pest, but does anyone have any ideas on this? I've
opened a bug, but I was hoping someone else on the list has
encountered this issue before.
Jason
Does the Polycom have the Buddy List turned on? We had an IP601 that
would reboot (or lock up) about 60% of the time when IT
We have a remote office that's having problems with their Polycom.
Sometime after they start a conference, the audio will halt and the
Polycom will become unresponsive. The only recourse is to kill the
Polycom meetme. Symptoms include a flood of RTP packets from the
Asterisk server to
Jason Dixon schrieb:
We have a remote office that's having problems with their Polycom.
Sometime after they start a conference, the audio will halt and the
Polycom will become unresponsive. The only recourse is to kill the
Polycom meetme. Symptoms include a flood of RTP packets from
On May 9, 2008, at 12:27 PM, Philipp Kempgen wrote:
Jason Dixon schrieb:
We have a remote office that's having problems with their Polycom.
Sometime after they start a conference, the audio will halt and the
Polycom will become unresponsive. The only recourse is to kill the
Polycom meetme.
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?
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing
Anyone use the LDAP feature yet on the polycom phones? If so how well
does it work? How are you using it in your environment?
http://polycom.com/usa/en/products/voice/desktop/soundpoint_ip/applicati
ons/corporate_directory_access.html
Roy Anciso
Director of Technology
Manistee Intermediate
We've just upgraded to asterisk 1.4 and we have changed the way we
handle our calls a bit. This seems to be giving us a bit of an issue.
We now allow the phones to reinvite. In the rtp.conf file, i've set the
range from 1-2. However, when the phones begin talking to one
another, they
On Fri, Apr 18, 2008 at 1:17 PM, Rob Schall [EMAIL PROTECTED] wrote:
If not, is there a way to configure this range on the phones? (They are
polycom 501s).
You're right, how could the phone read your asterisk rtp.conf setup?
Download the SIP Administrator Guide for the 501 from the Polycom
I havent tried it. I have quite a few polycoms and didnt even know
polycom had this feature! :)
This is obviously a separate peice of software that must be purchased
and installed on the phones. Looks amazing though- any idea on
pricing?.
On Fri, 2008-04-18 at 14:53 -0400, Anciso, Roy wrote:
: Friday, April 18, 2008 6:21 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom LDAP Corporate Directory
I havent tried it. I have quite a few polycoms and didnt even know
polycom had this feature! :)
This is obviously a separate peice
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of faraz
Sent: Friday, April 18, 2008 6:21 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom LDAP Corporate Directory
I havent tried it. I have
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
Hi all,
I have a polycom 501 phone that I rebooted today. It stopped working...
Normally the screen shows New call, Forward and that is all...
Now the screen shows New call, Forward, MyStat, Buddies.
It no longer accepts incoming calls nor can I make outgoing calls.
I have reloaded factory
try rebotting the phone. at boot up. hold down 4,8,6,* - that should reset
the local config for the phone
On Tue, Apr 15, 2008 at 5:58 PM, Jerry Geis [EMAIL PROTECTED] wrote:
Hi all,
I have a polycom 501 phone that I rebooted today. It stopped working...
Normally the screen shows New call,
try rebotting the phone. at boot up. hold down 4,8,6,* - that should reset
the local config for the phone
On Tue, Apr 15, 2008 at 5:58 PM, Jerry Geis geisj at pagestation.com
http://lists.digium.com/mailman/listinfo/asterisk-users wrote:
/ Hi all,
//
// I have a polycom 501 phone that I
You mentioned that you removed sip.cfg. Do you have the file with the
way it was before you made any changes?
If so, try placing it back where it's supposed to be, then reboot the
server, then reboot the phone. (p.s. I don't have an *, I don't know
anything about your problem! this is just
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 =
At 15:06 4/14/2008, Jerry Geis wrote:
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 =
Jerry,
Did you enable Ring Answer in the phone?
Look at your sip.cfg file for:
alertInfo voIpProt.SIP.alertInfo.1.value=Ring Answer
voIpProt.SIP.alertInfo.1.class=4/
and
ringType se.rt.enabled=1
se.rt.modification.enabled=1
DEFAULT
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
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
___
--
Hi James -
The VLAN used by the phone can be configured in several ways:
1. Hard-code it on the phone. Not recommended if you have lots of phones.
2. Auto-discovery using CDP. Requires Cisco or older HP switches.
3. Auto-discovery using DHCP. Disabled by default in SIP 2.1.x.
We use
The 330/550/650 phones have a built-in 2-port switch that speaks
802.1q. Usual use of this is to send two VLANs down the wire. The
phone is configured to use one, and the phone transparently passes the
other to the phone's PC port. On Cisco, this would be a trunk port
with two VLANs, one for the
Hi, all. I see that the Polycom SoundPoint IP 330 supports VLAN... but I
don't quite see how that works. Do you point a non-VLAN'd segment at it
(akin to when you uplink a VLAN_enabled switch), and have the phone
implement the VLAN? Or...? *puzzled*
Thanks much,
-Ken
On Tue, Mar 11, 2008 at 04:48:19PM -0400, Ken D'Ambrosio wrote:
Hi, all. I see that the Polycom SoundPoint IP 330 supports VLAN... but I
don't quite see how that works. Do you point a non-VLAN'd segment at it
(akin to when you uplink a VLAN_enabled switch), and have the phone
implement the
Without portfast, you're looking at about 30
seconds for STP to negotiate whenever the port bounces, during which
time higher layer protocols are unavailable. This may interfere with
CDP and DHCP, if you're using those.
I am using DHCP and I could briefly recall my PC hanging for a short
while
.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of James Sneeringer
Sent: Saturday, 1 March 2008 9:57 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom IP600 + PC share same
Anyone have any experience tying the Polycom VSX 7000e Asterisk
together? It says it supports standards based SIP servers but thought
I'd see if anyone had real world experience.
Thanks,
Ken
___
-- Bandwidth and Colocation Provided by
.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of James Sneeringer
Sent: Saturday, 1 March 2008 9:57 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom IP600 + PC share same switch
port
withVLAN
Discussion
Subject: Re: [asterisk-users] Polycom IP600 + PC share same switch port with
VLAN
You can paste and copy
innterface FastEthernet2/0/1 switchport access vlan 20 switchport mode
access switchport voice vlan 120 srr-queue bandwidth share 10 10 60 20
srr-queue bandwidth shape 10 0
Have you set the VLAN tag on the phone?
CP
Lee, John (Sydney) wrote:
Hi all,
I have been googling and testing without any luck and would appreciate
any guidance from anyone.
A port has already been configured on the CISCO switch with the
following:
interface FastEthernet2/0/1
Hi all,
I have been googling and testing without any luck and would appreciate
any guidance from anyone.
A port has already been configured on the CISCO switch with the
following:
interface FastEthernet2/0/1
description VOIP VLAN 100
switchport access vlan 100
switchport mode access
duplex full
portfast
- Original Message -
From: Lee, John (Sydney)
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Polycom IP600 + PC share same switch port
with VLAN
Date: Fri, 29 Feb 2008 16:39:24 +1100
Hi all,
I have been googling and testing without any luck and would
.
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bob G
Sent: Friday, 29 February 2008 5:02 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom IP600 + PC share same switch port with
VLAN
You can paste and copy
innterface
I know how to remap a key on a polycom 301 and 501
But does anyone know of a list of mapping keys?
For example, the Do Not Disturb on a 301 is #23. I got that one by just
guessing though.
Thanks,
Rob
___
-- Bandwidth and Colocation Provided by
That can be found in the monstrous admin guide for the phone, seemly in
Section 3.1.7 in my ancient version 1.5.0 document. It shows me that on
the 501, that button is 9 instead of 23.
http://www.polycom.com/usa/en/support/voice/soundpoint_ip/soundpoint_ip301.html
There's a link to the
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
I'm assuming you're talking about the 320/330s, 'cause the bigger phones
all have a DND key.
Yes, it's possible but don't do it. Those functions of those soft keys
are context-specific and they are used as navigation keys in some
contexts. I did exactly what you propose, and found that I
Yes you can, but it is not easy. First off you will need the
Administration guide from polycoms website. Check in the support section
under phones. You will have to set up a provisioning server and the
like. Also check voip-info.org. If I remember correctly that is where I
read about how to do
]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, February 20, 2008 3:30:01 PM (GMT-0600) America/Chicago
Subject: Re: [asterisk-users] Polycom Key Assignment
I'm assuming you're talking about the 320/330s, 'cause the bigger phones
all have
List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, February 20, 2008 3:30:01 PM (GMT-0600) America/Chicago
Subject: Re: [asterisk-users] Polycom Key Assignment
I'm assuming you're talking about the 320/330s, 'cause the bigger phones
all have a DND key.
Yes
Is there a way to configure the buttons on the phone that are normally
reserved for line registrations so that I can do a one-button pickup of
a parked call complete with Presence?
The goal is to have a couple of the line registration buttons show me
who is on park orbits 701 and 702 so that I
Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Polycom BLF / Speed Dial
Is there a way to configure the buttons on the phone that are normally
reserved for line registrations so that I can do a one-button pickup of
a parked call complete with Presence?
The goal is to have
-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, February 6, 2008 5:05:20 PM (GMT-0600) America/Chicago
Subject: Re: [asterisk-users] Polycom BLF / Speed Dial
I figured it out. Thanks anyway!
Yours,
Michael Munger, dCAP
404-438-2128
[EMAIL PROTECTED]
Attachment encrypted? click
06, 2008 1:43 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Polycom BLF / Speed Dial
Is there a way to configure the buttons on the phone that are normally
reserved for line registrations so that I can do a one-button pickup of
a parked call complete
Check your sip.cfg for the line:
feature.1.name=presence feature.1.enabled=1
I would imagine that you have enabled=0
That was it!
Thanks - Thermal
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing
Hello,
On our Polycom phones we can not activate the Buddy Watch feature.
When you add or edit a contact, the list ends at Auto Divert.I know it
is the end of the list b/c the down arrow on the right side of the screen
disappears when I get to Auto Divert.
When I add bw1/bw manually to the
On Feb 1, 2008 3:53 PM, Thermal Wetland [EMAIL PROTECTED] wrote:
Hello,
On our Polycom phones we can not activate the Buddy Watch feature.
When you add or edit a contact, the list ends at Auto Divert.I know
it is the end of the list b/c the down arrow on the right side of the screen
On 1/22/08, Steve Johnson [EMAIL PROTECTED] wrote:
Hi list,
There are many Polycom experts on this list -- hopefully someone has a
solution.
With *several* versions of Asterisk 1.4.x, doing a reload of Asterisk
causes the Polycom 601 phones to start dumping these messages to the
CLI.
I am using Polycom's SIP 2.2.0047 (the current release) and am seeing
this. It seems to occur less often with extensions reload rather
than just reload, but it would be nice to fix this.
Tx.
On Jan 22, 2008 8:30 AM, Steve Davies [EMAIL PROTECTED] wrote:
On 1/22/08, Steve Johnson [EMAIL
I have just retested and agree that this error eventually does clear
itself. However, in this test it took about 35 minutes and each
Polycom phone produced between 1000 and 1300 error message lines at 1
to 0 second intervals (which I captured to the debug log). Once one
phone starts flagging an
Yes, this prompt will shows up on SIP 2.2.2 as well.
I never had any issues with this though, it will clear up after next
registration of phone.
I just downloaded SIP 3.0 and have not got a chance to check and see if it
happens with this firmware as well.
On Jan 22, 2008 2:53 PM, Steve Johnson
Hi list,
There are many Polycom experts on this list -- hopefully someone has a solution.
With *several* versions of Asterisk 1.4.x, doing a reload of Asterisk
causes the Polycom 601 phones to start dumping these messages to the
CLI.
-- Incoming call: Got SIP response 500 Internal Server
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.
dialplan dialplan.impossibleMatchHandling=0
Sorry everyone. There was an error in the dial plan in Asterisk.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Klaverstyn, David C
Sent: Monday, 21 January 2008 12:05 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Polycom 320 Issue
Kevin Kiely wrote:
Great suggestion, thanks. The boot failed with the mac-phone.cfg removed. I
re-touched the file and followed your suggestion.
Any way of removing the call forwarding feature via the xml configs?
Kevin Kiely wrote:
I have a remote user on a Polycom IP Phone who has
Kevin Kiely wrote:
I have a remote user on a Polycom IP Phone who has set call forwarding
by accident and is away from the phone. Does anyone know of a way to
remotely un-forward the phone? I tried to reboot the phone but that
didn’t work and removing the mac-phone.cfg caused problems
I have a remote user on a Polycom IP Phone who has set call forwarding by
accident and is away from the phone. Does anyone know of a way to remotely
un-forward the phone? I tried to reboot the phone but that didn't work and
removing the mac-phone.cfg caused problems
Great suggestion, thanks. The boot failed with the mac-phone.cfg removed. I
re-touched the file and followed your suggestion.
Any way of removing the call forwarding feature via the xml configs?
Kevin Kiely wrote:
I have a remote user on a Polycom IP Phone who has set call forwarding
by
When setting a forward on the phone, the phone will upload to your ftp
server a modified macaddr-phone.cfg XML file that (amongst other
locally made changes) contains an OVERRIDE statement similar to this:
OVERRIDE reg.1.fwdContact= reg.1.fwdStatus=1 ... /
Change the .fwdStatus attribute to
Subject: Re: [asterisk-users] Polycom Remotely Cancel Call Forward
When setting a forward on the phone, the phone will upload to your ftp
server a modified macaddr-phone.cfg XML file that (amongst other
locally made changes) contains an OVERRIDE statement similar to this:
OVERRIDE reg.1.fwdContact
I misread then. Even though your original message said you wanted to
un-forward a phone. That can be done with the recipe BJ and I
outlined.
I am not aware of any way to disable the forward function, i.e.
prevent a user from forwarding in the first place.
]
[mailto:[EMAIL PROTECTED] On Behalf Of Kevin
Kiely
Sent: Thursday, January 17, 2008 17:48
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Polycom Remotely Cancel Call Forward
I guess I was interested in Disabling the forwarding feature completely
via
21, 2007 2:16 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: [EMAIL PROTECTED]
Subject: Re: [asterisk-users] Polycom 330 beep on new VM
This is pretty easy to suppress using the configuration files. Check:
http://www.voip-info.org/wiki/view/Polycom+SoundPoint+IP+MWI+audio
What zap driver are you using? ztdummy?
PaulH
On Wed, 2008-01-09 at 15:41 -0500, Mike Coakley wrote:
I'm setting up a new Asterisk system on a Dell server and I'm getting
fuzzy voice between the Polycom IP SoundStation 550 and the Asterisk
server. I've checked all of my codec settings
Doug,
That bug ID was a dead ringer. The workarounds in the bug worked
perfectly. BTW I'm on a openSuSE 10.3 system with gcc 2.4.1.
Thanks for the pointer.
Mike
On Jan 9, 2008, at 8:30 PM, Doug Lytle wrote:
Mike Coakley wrote:
I'm setting up a new Asterisk system on a Dell server and I'm
Mike Coakley wrote:
I'm setting up a new Asterisk system on a Dell server and I'm getting
fuzzy voice between the Polycom IP SoundStation 550 and the Asterisk
Probably related to this bug:
http://bugs.digium.com/view.php?id=11243
Doug
--
Ben Franklin quote:
Those who would give up
Try 'ip4000_1' instead of '207' for your address.
CP
Kevin DeGraaf wrote:
I am trying to configure a Polycom IP4000 for use with Asterisk 1.4.5 on
a flat local network.
I followed the provisioning guides that I found on the Web, and I have
the phone downloading bootrom.ld, sip.ld, and a
I brick'd two of my polycom phones trying to get the shoretel phones
working with *...
does anyone have the equipment to unbrick them?
there is a jtag serial cable that is needed along with the knowledge of
embedded systems..
that is all I currently know.
polycom wants $180 and 30 days to fix
).
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of dave cantera
Sent: Saturday, 5 January 2008 1:10 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Polycom IP phones that are brick'd
I brick'd
I am trying to configure a Polycom IP4000 for use with Asterisk 1.4.5 on
a flat local network.
I followed the provisioning guides that I found on the Web, and I have
the phone downloading bootrom.ld, sip.ld, and a bunch of configuration
files. This all works properly.
However, I receive the
This is the whole point behind using VLAN on the phone. Tagged VLAN
for your phone with QoS configured accordingly on your switch and
untagged VLAN for your PC, both on the same wire. This way you can
always guarantee enough bandwidth for your VoIP packets.
Thanks,
Wojtek
On 2-Jan-08, at
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.
This e-mail, facsimile, or letter and any files or attachments transmitted with
it contains information that
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Polycom VLAN
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.
This e-mail, facsimile, or letter
On Wed, 2 Jan 2008, Jeremy Mann wrote:
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.
According to this --
I need the digit map to call China. Example number:
011-86-10-6887-
011-International (obvious)
86 is country code (China)
10 is city code (Beijing)
Last 8 digits are the number.
I tried using 011xxx.T but it always asks me to enter more digits. Tried
some variations as well,
On Dec 31, 2007, at 11:36 AM, Michael Munger wrote:
I need the digit map to call China. Example number:
011-86-10-6887-
011-International (obvious)
86 is country code (China)
10 is city code (Beijing)
Last 8 digits are the number.
I tried using 011xxx.T but it always asks
Jerry Jones wrote:
Yours should work if you wait long enough for t to timeout.
I think your digit map needs a T on the end of it if you want to allow
timeouts for that match.
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
]
[mailto:[EMAIL PROTECTED] On Behalf Of Jerry
Jones
Sent: Monday, December 31, 2007 12:53 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom Digit Map
On Dec 31, 2007, at 11:36 AM, Michael Munger wrote:
I need the digit map to call China
Michael Munger wrote:
only connects me to a dial tone and says Enter More Digits.
It actually says this?
I would say then it's not the phone, but your phone system's
programming. The Polycoms don't verbally say anything, at least not the
ones I deal with.
Doug
--
Ben Franklin
Doug Lytle wrote:
Michael Munger wrote:
only connects me to a dial tone and says Enter More Digits.
It actually says this?
I would say then it's not the phone, but your phone system's
programming. The Polycoms don't verbally say anything, at least not the
ones I deal
Mojo with Horan Company, LLC wrote:
So try: 011XXT in your digit map, meaning 011 plus at least six
digits, consider it good
Err duh, that's ten X's not six :) To account for the Tajikistan
example plus a little bit of local number.
Really, it's dead simple to just do it like
At 14:27 12/31/2007, Mojo with Horan Company, LLC wrote:
Mojo with Horan Company, LLC wrote:
So try: 011XXT in your digit map, meaning 011 plus at least six
digits, consider it good
Err duh, that's ten X's not six :) To account for the Tajikistan
example plus a little bit of
Doug wrote:
At 14:27 12/31/2007, Mojo with Horan Company, LLC wrote:
Mojo with Horan Company, LLC wrote:
So try: 011XXT in your digit map, meaning 011 plus at least six
digits, consider it good
Err duh, that's ten X's not six :) To account for the Tajikistan
example plus a
Hi,
I have a Polycom 330 that emits a beep every 30s or so when there is a
message waiting. Is there a way to disable that? It is pretty annoying.
Regards,
Ugo
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
This is pretty easy to suppress using the configuration files. Check:
http://www.voip-info.org/wiki/view/Polycom+SoundPoint+IP+MWI+audio
On Dec 21, 2007 11:55 AM, Ugo Bellavance [EMAIL PROTECTED] wrote:
Hi,
I have a Polycom 330 that emits a beep every 30s or so when there is a
Steve Johnson wrote:
This is pretty easy to suppress using the configuration files. Check:
http://www.voip-info.org/wiki/view/Polycom+SoundPoint+IP+MWI+audio
Looks easy once you have the config file provisioning in place, but it
looks overkill and a lot of work to set this up for the only
Ugo Bellavance wrote:
Steve Johnson wrote:
This is pretty easy to suppress using the configuration files. Check:
http://www.voip-info.org/wiki/view/Polycom+SoundPoint+IP+MWI+audio
Looks easy once you have the config file provisioning in place, but it
looks overkill and a lot of
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] mailto:[EMAIL
-2128
[EMAIL PROTECTED]
Attachment encrypted? click here.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alex
Balashov
Sent: Wednesday, December 12, 2007 12:31 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users
Michael,
There is a tutorial about this on the voip-wiki. Search for it on Google.
Basically, it involves provisioning the Polycom phones to auto-answer in
a certain situation, and then sending an additional SIP header field to
provide that situation.
Cheers,
-- Alex
On Wed, 12 Dec 2007,
Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alex
Balashov
Sent: Wednesday, December 12, 2007 12:31 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom Paging
Michael,
There is a tutorial about this on the voip-wiki
PROTECTED] On Behalf Of Alex
Balashov
Sent: Wednesday, December 12, 2007 12:31 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom Paging
Michael,
There is a tutorial about this on the voip-wiki. Search for it on
Google.
Basically, it involves
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