On Mon, Aug 16, 2010 at 4:21 PM, Ben Schorr b...@rolandschorr.com wrote:
We gave the phone a static IP address and pointed it to the configuration
server on the remote end that has the CFG files for it. The phone starts
up, downloads SIP and the “new application” and otherwise seems to be
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David
Backeberg
Subject: Re: [asterisk-users] Polycom 331 freezes connecting to FreePBX
snip
I have a suggestion...
Put back the 'old application', and determine whether the 'new
http://downloads.polycom.com/voice/voip/sip_sw_releases_matrix.html
Howdy, all. What's the difference between split and combined
firmware, which can be seen at the above link? I've googled to no avail,
I'm afraid.
Thanks!
-Ken
--
This message has been scanned for viruses and
dangerous
On Mon, Jun 21, 2010 at 12:10 PM, Ken D'Ambrosio k...@jots.org wrote:
http://downloads.polycom.com/voice/voip/sip_sw_releases_matrix.htmlHowdy,
all. What's the difference between split and combined
firmware, which can be seen at the above link? I've googled to no avail,
I'm afraid.
The
:37 -0400 (EDT)
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Polycom firmware: split vs. combined
http://downloads.polycom.com/voice/voip/sip_sw_releases_matrix.html
Howdy, all. What's the difference between split and combined
firmware, which can be seen at the above link
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ken
D'Ambrosio
Sent: Monday, June 21, 2010 1:11 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Polycom firmware: split vs. combined
http://downloads.polycom.com/voice/voip/sip_sw_releases_matrix.html
Howdy, all. What's
On Mon, Jun 21, 2010 at 10:19 AM, Warren Selby wcse...@selbytech.com wrote:
On Mon, Jun 21, 2010 at 12:10 PM, Ken D'Ambrosio k...@jots.org wrote:
Howdy, all. What's the difference between split and combined
firmware, which can be seen at the above link? I've googled to no avail,
I'm afraid.
I was just wondering if anyone is having the same problem will Polycom 330 ip
phone. The phone looses the network and when you reboot the phone it can no
longer find the DHCP server. I put an address in manually, but the phone is
still not able to connect to the network. I replaced the phone
-users@lists.digium.com
Sent: Fri, March 19, 2010 12:00:14 PM
Subject: Re: [asterisk-users] Polycom not updating the directory list
The mac-addr-directory.cfg permission is 777 with a symoblic link pointing
to -directory.xml with permission of 644.
I would manually edit
This works for me using DNSMasq:
dhcp-host=00:04:f2:*:*:*,net:polycom # creates a 'polycom' group for all
equipment with MAC prefix of 0004f2
dhcp-range=net:polycom,192.168.1.151,192.168.1.180 # dhcp range for
'polycom' group
dhcp-option=net:polycom,66,http://pbxserver/gui/phoneprov; # polycom
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Mike Diehl
Sent: Thursday, March 18, 2010 1:39 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom DHCP
I know for a fact that you can provision a Polycom via ftp.
I've included
much of my dhcpd.conf file below. Pick out what you need.
Let me know if
I can confirm that using option 66 will work with FTP (and HTTP, for
that matter) with newer BootROM versions. I don't know the exact
version
@lists.digium.com
Sent: Wed, March 17, 2010 11:05:09 PM
Subject: Re: [asterisk-users] Polycom not updating the directory list
The very obvious thing to check is the permission of the
mac-addr-directory.cfg.
From: asterisk-users-boun...@lists.digium.com
anyone?
From: hin lee hi...@yahoo.com
To: Asterisk Users asterisk-users@lists.digium.com
Sent: Fri, March 12, 2010 10:08:53 AM
Subject: Polycom not updating the directory list
Hi,
I have a strange problem with all of our Polycom 550 650 phones. I am
Subject: Re: [asterisk-users] Polycom not updating the directory list
anyone?
From: hin lee hi...@yahoo.com
To: Asterisk Users asterisk-users@lists.digium.com
Sent: Fri, March 12, 2010 10:08:53 AM
Subject: Polycom not updating the directory list
Hi,
I have
- Original Message -
From: Lee, John (Sydney) john@compuware.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, March 17, 2010 9:50 PM
Subject: Re: [asterisk-users] Polycom DHCP Option 66 FTP provisioning
I'll see
On Thursday 18 March 2010 11:24:18 am Karl Fife wrote:
- Original Message -
From: Lee, John (Sydney) john@compuware.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, March 17, 2010 9:50 PM
Subject: Re: [asterisk-users
Is anyone successfully using DHCP option 66 to specify an FTP [sic]
provisioning of Polycom Sounpoint phones instead of TFTP? I know option 66
is typically used TFTP booting, but the Polycom doc doesn't appear to
specify that option 66 implies TFTP instead of FTP (since you explicitly
call
On Wednesday 17 March 2010 5:29:36 pm Karl Fife wrote:
Is anyone successfully using DHCP option 66 to specify an FTP [sic]
provisioning of Polycom Sounpoint phones instead of TFTP? I know option 66
is typically used TFTP booting, but the Polycom doc doesn't appear to
specify that option 66
Is anyone successfully using DHCP option 66 to specify an FTP [sic]
provisioning of Polycom Sounpoint phones instead of TFTP? I know option 66
is typically used TFTP booting, but the Polycom doc doesn't appear to
specify that option 66 implies TFTP instead of FTP (since you explicitly
call
Fife
Sent: Thursday, 18 March 2010 10:30 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Polycom DHCP Option 66 FTP provisioning
Is anyone successfully using DHCP option 66 to specify an FTP [sic]
provisioning of Polycom Sounpoint phones instead of TFTP
@compuware.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, March 17, 2010 8:48 PM
Subject: Re: [asterisk-users] Polycom DHCP Option 66 FTP provisioning
Yes, this is still one of the unsolved mysteries I wanted to find out
about
...@lists.digium.com] On Behalf Of Karl Fife
Sent: Thursday, 18 March 2010 1:27 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom DHCP Option 66 FTP provisioning
Now that I know that I'm not the only person (i.e. it's less likely
that I
just made
Hi,
I have a strange problem with all of our Polycom 550 650 phones. I am
running a TFTP server on my Asterisk server and option 66 Boot Host pointing to
Asterisk on my DHCP server. The auto-provisioning is working because the
phones are registering correctly with their extension. If I
On 19 February 2010 15:28, Steve Davies davies...@gmail.com wrote:
[snip]
I just upgraded to the new bootblock and 3.2.2 firmware, and these
phones will now talk video to other devices. Nothing in the changelogs
indicates why, but there is a definite jump up from the previous
release of this
On 18 February 2010 00:14, Michael Graves mgra...@mstvp.com wrote:
On Wed, 17 Feb 2010 17:12:01 +, Steve Davies wrote:
On 17 February 2010 16:56, asterisk aster...@nbsvoice.com wrote:
Can anyone tell if asterisk and Polycom VVX1500 work with video yet?
If so what version? Is there a
Can anyone tell if asterisk and Polycom VVX1500 work with video yet?
If so what version? Is there a patch?
Thank you!
Doug
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing
On 17 February 2010 16:56, asterisk aster...@nbsvoice.com wrote:
Can anyone tell if asterisk and Polycom VVX1500 work with video yet?
If so what version? Is there a patch?
Thank you!
Doug
According to my experimentation, Polycom VVX1500 phones work with
all versions of Asterisk as far
-Commercial Discussion
Subject: Re: [asterisk-users] Polycom VVX1500 video working yet?
On 17 February 2010 16:56, asterisk aster...@nbsvoice.com wrote:
Can anyone tell if asterisk and Polycom VVX1500 work with video yet?
If so what version? Is there a patch?
Thank you!
Doug
According
Sent: Wednesday, February 17, 2010 12:23 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom VVX1500 video working yet?
We use Asterisk 1.6.2 with the latest Polycom firmwares on the VVXs with no
problem.
They key is the new bootblock polycom released
On Wed, 17 Feb 2010 17:12:01 +, Steve Davies wrote:
On 17 February 2010 16:56, asterisk aster...@nbsvoice.com wrote:
Can anyone tell if asterisk and Polycom VVX1500 work with video yet?
If so what version? Is there a patch?
Thank you!
Doug
According to my experimentation, Polycom
DND works flawlessly, but whenever using BLF I can only tell that a line is
either in use (on a call) or not. I cannot tell a phone is on DND, or on
hold for that matter. Would be extremely useful.
Would be willing to pay for this developpement if it can be done as long as
the feature makes
] On Behalf Of Stuart McQuade
Sent: Wednesday, January 27, 2010 7:50
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom phone DND state
Hi,
At my previous company we ran 1.4.x.x (underneath DiVitas.com software) and our
Polycom IP 550 would use
Hi Mike,
What version of spip are you using ?
Jimmy
-Original Message-From: l...@virtutel.caSent: Thu, 04 Feb 2010 20:46:07 -0500To: asterisk-users@lists.digium.comSubject: Re: [asterisk-users] Polycom phone DND state
Sorry it took awhile to answer.
DND works flawlessly
: [asterisk-users] Polycom phone DND state
Hi Mike,
What version of spip are you using ?
Jimmy
-Original Message-
From: l...@virtutel.ca
Sent: Thu, 04 Feb 2010 20:46:07 -0500
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Polycom phone DND state
Sorry it took
: Friday, 5 February 2010 12:46 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Polycom phone DND state
Sorry it took awhile to answer.
DND works flawlessly, but whenever using BLF I can only tell that a line
is either in use (on a call) or not. I
On Thu, Feb 4, 2010 at 9:43 PM, Mike l...@virtutel.ca wrote:
*From:* l...@virtutel.ca
*Sent:* Thu, 04 Feb 2010 20:46:07 -0500
*To:* asterisk-users@lists.digium.com
*Subject:* Re: [asterisk-users] Polycom phone DND state
Sorry it took awhile to answer.
DND works flawlessly, but whenever
Hello list,
anyone have a manual for the webGUI of the above phone ? Just bought a
Polycom Soundpoint IP300 and on the site of Polycom I see a user manual
and a administrator's manual but none of these 2 guides explain the
fields in the webGUI.
Trying to understand the difference between the
From: Lee, John (Sydney) john@compuware.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wed, 27 January, 2010 8:02:14
Subject: Re: [asterisk-users] Polycom phone DND state
I am using 1.4.21.2 and DND is definitely
On Fri, Jan 22, 2010 at 7:50 AM, Mike l...@virtutel.ca wrote:
I know having Asterisk aware of Polycom Do No Disturb state wasn't working
before (1.4), but is this working in any recent version? Is there any
custom way of doing this?
Our Asterisk servers (1.2 and 1.4) get SIP response 603
Discussion'
Subject: [asterisk-users] Polycom phone DND state
Hi,
I know having Asterisk aware of Polycom Do No Disturb state wasn't
working before (1.4), but is this working in any recent version? Is
there any custom way of doing this?
Regards,
Mike
Hi,
I know having Asterisk aware of Polycom Do No Disturb state wasn't working
before (1.4), but is this working in any recent version? Is there any
custom way of doing this?
Regards,
Mike
--
_
-- Bandwidth
Does anyone know if you can use the Polycom Norstar Clarity
speakerphones with Asterisk?
Model number is 2501-03308-001 'C'
Is it a SIP handset or analog style unit (or worse proprietary).
Cheers,
Dean
--
On 20 Jan 2010, at 14:39, Dean Collins wrote:
Is it a SIP handset or analog style unit (or worse proprietary).
I'd say analogue.
W--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users
Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom Soundstation Conferencing Unit
On 20 Jan 2010, at 14:39, Dean Collins wrote:
Is it a SIP handset or analog style unit (or worse proprietary).
I'd say analogue.
W
] On Behalf Of Danny
Nicholas
Sent: Wednesday, January 20, 2010 10:16 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Polycom Soundstation Conferencing Unit
According to what I see on Ebay, it is an Analogue handset. You would
have to hook it to an FXO
Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Polycom Soundstation Conferencing Unit
On 20 Jan 2010, at 14:39, Dean Collins wrote:
Is it a SIP handset or analog style unit (or worse proprietary).
I'd say analogue.
W
: [asterisk-users] Polycom Soundstation Conferencing Unit
FXS port is correct answer.
FXO ports are for PSTN or PBX lines
FXS SUPPLIES battery and ringing, and receives DTMF ( or pulse ) dialing
FXO receives battery and supplies DTMF ( or pulse ) dialing
Danny Nicholas wrote:
According to what I
Sent: Tuesday, January 12, 2010 9:56 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Polycom Mute Problem
Hey Yall
I have an interesting situation. When a person is on the phone (Polycom 501)
and another call hits the phone the phone mutes on the users side not the
person outside
: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michael
Sent: Tuesday, January 12, 2010 9:56 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Polycom Mute Problem
Hey Yall
I have an interesting situation. When a person
List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Polycom Mute Problem
Upgrade the phone. I ran into the same issue a year or so ago. There was
some setting that was screwed up in the config file and upgrading to the
newest version at the time fixed it. It was something like
Discussion'
Subject: Re: [asterisk-users] Polycom Mute Problem
By upgrade the phone I assume you mean upgrade the bios, not purchase a
newer phone?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Peder
Sent: Wednesday
Hey Yall
I have an interesting situation. When a person is on the phone (Polycom 501)
and another call hits the phone the phone mutes on the users side not the
person outside the asterisk system. It stays mute until the call goes to
voicemail. Its like the beep we should get from the call waiting
If anybody who want to earn a quick $50 via paypal and can help me on
setting up a polycom ip7000 to work with asterisk please email sam __ tam AT
hotmail DOT com
Do not email me through my gmail account.
Sam
___
-- Bandwidth and Colocation Provided
On Saturday 28 November 2009 06:48:01 pm Darrick Hartman wrote:
I'd configure the phone using the XML files, but take a look at the
method that Karl Fife has documented here:
http://www.kfife.com/voip/
Minimal changes are made to files. The base config files are never
touched which makes
I have one client that is telling me that their Polycom 500's format the
file system every time they reboot, and also that they are unable to make
changes locally on the phone itself, only via the config files. If the
config file is not available when they try to boot the phone, then they
receive
Hi Warren -
I have one client that is telling me that their Polycom 500's format the
file system every time they reboot, and also that they are unable to make
changes locally on the phone itself, only via the config files. If the
config file is not available when they try to boot the phone,
On Saturday 28 November 2009 06:48:01 pm Darrick Hartman wrote:
Mike Diehl wrote:
On Saturday 28 November 2009 04:48:13 pm Darrick Hartman wrote:
Mike Diehl wrote:
On Saturday 28 November 2009 06:54:16 am Kevin P. Fleming wrote:
Mike Diehl wrote:
On Friday 27 November 2009 11:09:02 am
On Saturday 28 November 2009 05:05:47 pm Kevin P. Fleming wrote:
Darrick Hartman wrote:
The phone is a Polycom 501; it's been discontinued. I am working on a
testing/migration plan to move to the latest Asterisk 1.6.x, but I'm
hesitant to upgrade a system that doesn't currently work right.
On Friday 27 November 2009 11:09:02 am Noah Miller wrote:
Hi Mike -
I've got a Polycom 501 that's been working with Asterisk for some time.
However, I don't seem to be able to put a call on hold and get it back.
It goes on hold just fine. But when I press the resume button, nothing
Mike Diehl wrote:
On Friday 27 November 2009 11:09:02 am Noah Miller wrote:
Hi Mike -
I've got a Polycom 501 that's been working with Asterisk for some time.
However, I don't seem to be able to put a call on hold and get it back.
It goes on hold just fine. But when I press the resume
On Saturday 28 November 2009 06:54:16 am Kevin P. Fleming wrote:
Mike Diehl wrote:
On Friday 27 November 2009 11:09:02 am Noah Miller wrote:
Hi Mike -
I've got a Polycom 501 that's been working with Asterisk for some time.
However, I don't seem to be able to put a call on hold and get
Mike Diehl wrote:
On Saturday 28 November 2009 06:54:16 am Kevin P. Fleming wrote:
Mike Diehl wrote:
On Friday 27 November 2009 11:09:02 am Noah Miller wrote:
Hi Mike -
I've got a Polycom 501 that's been working with Asterisk for some time.
However, I don't seem to be able to put a call on
Darrick Hartman wrote:
The phone is a Polycom 501; it's been discontinued. I am working on a
testing/migration plan to move to the latest Asterisk 1.6.x, but I'm
hesitant
to upgrade a system that doesn't currently work right.
There's no particular reason that you need to move to 1.6.x,
During a call, I get the animated arrows. When I put a
call on hold, I get the flashing phone with the handset upside down. When
I
try to retreive the call, I get the animated arrow for a second,
This is normal and expected behavior so far, at least on my Polycom 500,
asterisk 1.4.27.
On Saturday 28 November 2009 04:48:13 pm Darrick Hartman wrote:
Mike Diehl wrote:
The phone is a Polycom 501; it's been discontinued. I am working on a
testing/migration plan to move to the latest Asterisk 1.6.x, but I'm
hesitant to upgrade a system that doesn't currently work right.
On Saturday 28 November 2009 04:48:13 pm Darrick Hartman wrote:
Mike Diehl wrote:
On Saturday 28 November 2009 06:54:16 am Kevin P. Fleming wrote:
Mike Diehl wrote:
On Friday 27 November 2009 11:09:02 am Noah Miller wrote:
Hi Mike -
I've got a Polycom 501 that's been working with
Mike Diehl wrote:
On Saturday 28 November 2009 04:48:13 pm Darrick Hartman wrote:
Mike Diehl wrote:
On Saturday 28 November 2009 06:54:16 am Kevin P. Fleming wrote:
Mike Diehl wrote:
On Friday 27 November 2009 11:09:02 am Noah Miller wrote:
Hi Mike -
I've got a Polycom 501 that's been
Hi Mike -
I've got a Polycom 501 that's been working with Asterisk for some time.
However, I don't seem to be able to put a call on hold and get it back. It
goes on hold just fine. But when I press the resume button, nothing
happends.
Anyone seen this befor? Any ideas on where to start
Hi all,
I've got a Polycom 501 that's been working with Asterisk for some time.
However, I don't seem to be able to put a call on hold and get it back. It
goes on hold just fine. But when I press the resume button, nothing
happends.
Anyone seen this befor? Any ideas on where to start to
That will increase the gain on the tranmission side of the phone. That's
exactly what you need.
Vinícius Fontes
www.asteriskforum.com.br - Informações e discussão sobre Asterisk e telefonia IP
- Robert Grignon rgrig...@fleetone.com escreveu:
Sorry if this is off topic
I have
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
Hi,
Sorry about posted a protected link, I forgot we'd closed the site to
spammers since we don't use it anymore. The useful content was
re-posted in our list.
---
URI
Hi,
I have been trying a (really simple) push application for the Polycom
microbrowser, using a Polycom 650 with 3.2 firmware.
I can't do anything, I always get Push message cannot be displayed back
from the Polycom phone, and all I am sending is the Polycom example :
PolycomIPPhone
Mike wrote:
Hi,
I have been trying a (really simple) push application for the Polycom
microbrowser, using a Polycom 650 with 3.2 firmware.
I can't do anything, I always get Push message cannot be displayed back
from the Polycom phone, and all I am sending is the Polycom example
Fullerton
Sent: Thursday, September 24, 2009 1:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom push application for microbrowser
Mike wrote:
Hi,
I have been trying a (really simple) push application for the Polycom
microbrowser, using
-Commercial Discussion
Subject: Re: [asterisk-users] Polycom push application for microbrowser
Mike wrote:
Hi,
I have been trying a (really simple) push application for the Polycom
microbrowser, using a Polycom 650 with 3.2 firmware.
I can't do anything, I always get Push message cannot
Hi,
Take a look at this:
http://food4wine.ning.com/forum/topics/submit-an-application-for
Way down the page Dave VG submitted some scripts that hold the answers.
We also did a Polycom App conference at the VUC, but I can't find the
link right now.
/r
On Thu, 24 Sep 2009, randulo wrote:
Take a look at this:
http://food4wine.ning.com/forum/topics/submit-an-application-for
Grrr.
Have to have a Ning ID and you have to be invited.
--
Thanks in advance,
-
Steve Edwards
- Non-Commercial
Discussion
Subject: Re: [asterisk-users] Polycom push application for microbrowser
On Thu, 24 Sep 2009, randulo wrote:
Take a look at this:
http://food4wine.ning.com/forum/topics/submit-an-application-for
Grrr.
Have to have a Ning ID and you have to be invited.
--
Thanks
14:15
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom push application for microbrowser
Mike wrote:
Hi,
I have been trying a (really simple) push application for the Polycom
microbrowser, using a Polycom 650 with 3.2 firmware
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Thursday, September 24, 2009 14:20
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Polycom push
Michael Graves wrote:
On Tue, 14 Jul 2009 22:58:14 + (UTC), Jeff LaCoursiere wrote:
Jeff, yeah i saw the posts, i followed Bob Pierce config and had no
luck, BUT it just started to work, i changed AP's, seems like theres
something wrong with Ubiquiti NanoStation2 WMM implementation, i
On Tue, 14 Jul 2009 22:58:14 + (UTC), Jeff LaCoursiere wrote:
Jeff, yeah i saw the posts, i followed Bob Pierce config and had no
luck, BUT it just started to work, i changed AP's, seems like theres
something wrong with Ubiquiti NanoStation2 WMM implementation, i used a
Linksys WRT54G2
Has anyone played with this phone? i cant seem to get it to work
properly, i manged to get it registered and can make calls from it, but
i havent been able to make it receive calls. Weird thing its that if you
make a call from it and while you are on that call you dial its number
does calls go
Search the archives - we had a big discussion about this phone about six
months ago. If you make it work and want another one I will give you
special price!.
j
On Tue, 14 Jul 2009, Cesar Gonzalez wrote:
Has anyone played with this phone? i cant seem to get it to work
properly, i manged to
Jeff LaCoursiere wrote:
Search the archives - we had a big discussion about this phone about six
months ago. If you make it work and want another one I will give you
special price!.
j
Jeff, yeah i saw the posts, i followed Bob Pierce config and had no
luck, BUT it just started to
On Tue, 14 Jul 2009, Cesar Gonzalez wrote:
Jeff LaCoursiere wrote:
Search the archives - we had a big discussion about this phone about six
months ago. If you make it work and want another one I will give you
special price!.
j
Jeff, yeah i saw the posts, i followed Bob Pierce config
Does anybody know of a way to tell the Polycom phones to stop trying to
download their config? We have some setup for tftp and some for ftp and if
they cannot reach the server, they just keep rebooting over and over and
over and never stop. I would think it should try once or twice and stop,
but
Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] Polycom Stop Downloading Config
Does anybody know of a way to tell the Polycom phones to stop trying to
download their config? We have some setup for tftp and some for ftp and if
they cannot reach the server, they just keep rebooting
-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Wednesday, June 17, 2009 10:46 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Polycom Stop Downloading Config
Touch
Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Polycom Stop Downloading Config
But It still needs to hit the server to see that at some point. I just want
it to stop pulling config totally, unless I tell it to. It is web based, so
I would think there should be some way
On 2009-06-14 at 00:19, Jeff LaCoursiere (j...@jeff.net) wrote:
[hft0]
type=friend
username=hft0
secret=mysecret
context=outtrunk-office
host=192.168.200.99
Change the above to host=dynamic
I just did this and did a 'reload'.
reg.1.server.1.address=jtsd05
Can the phone
On Mon, 15 Jun 2009, Jim Gottlieb wrote:
On 2009-06-14 at 00:19, Jeff LaCoursiere (j...@jeff.net) wrote:
[hft0]
type=friend
username=hft0
secret=mysecret
context=outtrunk-office
host=192.168.200.99
Change the above to host=dynamic
I just did this and did a 'reload'.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom registration errors
On Mon, 15 Jun 2009, Jim Gottlieb wrote:
On 2009-06-14 at 00:19, Jeff LaCoursiere (j...@jeff.net) wrote:
[hft0]
type=friend
username=hft0
secret=mysecret
context=outtrunk-office
On 2009-06-15 at 19:12, Jeff LaCoursiere (j...@jeff.net) wrote:
chan_sip.c: Registration from 'sip:6193644...@jtsd05.nccom.com' failed
for '192.168.200.99' - Username/auth name mismatch
I am a bit confused as to the names and addresses involved here. Which
name/address is the server,
Jim Gottlieb wrote:
I'm evaluating using Polycom phones for our call center and I've set
up my first phone (a SoundPoint 560) to give it a try.
The phone is working and can successfully place and receive calls.
But every minute, there's an error in the log file:
chan_sip.c:
On 2009-06-15 at 17:04, Dave Fullerton
(dfullertaster...@shorelinecontainer.com) wrote:
Try changing reg.1.address to hft0. My hunch is asterisk is looking at
the from of 6193644...@jtsd05 and going huh? I don't know a
6193644...@jtsd05.
That makes sense and it fixed it. Thanks!
I'm evaluating using Polycom phones for our call center and I've set
up my first phone (a SoundPoint 560) to give it a try.
The phone is working and can successfully place and receive calls.
But every minute, there's an error in the log file:
chan_sip.c: Registration from
On Sat, 13 Jun 2009, Jim Gottlieb wrote:
I'm evaluating using Polycom phones for our call center and I've set
up my first phone (a SoundPoint 560) to give it a try.
The phone is working and can successfully place and receive calls.
But every minute, there's an error in the log file:
201 - 300 of 2635 matches
Mail list logo
