To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Polycom Phantom Ringing
I have Polycom 501's and they keep a log of all calls. I would expect the
335's to have that capability as well.
-Original Message-
From: asterisk-users-boun
: [asterisk-users] Polycom Phantom Ringing
I have Polycom 501's and they keep a log of all calls. I would expect the
335's to have that capability as well.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf
: [asterisk-users] Polycom Phantom Ringing
They do.
It shows up asterisk on the physical phone.
Nothing in the raw cdr file on the server.
--E
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent
-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dave Fullerton
Sent: Thursday, November 17, 2011 11:27 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] polycom soundpint ip650 question
No, this is all done
On the polycom soundpoint ip 650 six line phone:
Say I have 4 lines on hold, is there way to tell who I put on hold.
I cannot see the caller ID of the other lines, only the last line I placed on
hold.
Thanks,
--E
--
: [asterisk-users] polycom soundpint ip650 question
On the polycom soundpoint ip 650 six line phone:
Say I have 4 lines on hold, is there way to tell who I put on hold.
I cannot see the caller ID of the other lines, only the last line I placed
on hold.
Thanks,
--E
When you perform an attended transfer, the extension of the person transferring
is displayed to the co-worker.
Can I override the caller ID to display the caller's callerID during a blind
transfer?
Thanks,
--E
--
_
--
Of Danny Nicholas
Sent: Wednesday, November 16, 2011 1:49 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] polycom soundpint ip650 question
Core show channels verbose - if you do asterisk -rx cscv from bash
From: asterisk-users-boun
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] Polycom Attended Transfer
When you perform an attended transfer, the extension of the person
transferring is displayed to the co-worker.
Can I override the caller ID to display the caller's callerID during a blind
...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Wednesday, November 16, 2011 1:49 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] polycom soundpint ip650 question
Core show channels verbose - if you do asterisk -rx cscv from bash
From
When you perform an attended transfer, the extension of the person
transferring is displayed to the co-worker.
Can I override the caller ID to display the caller’s callerID during a
blind transfer?
Upgrade to 10.0 – this isn’t available in any of the 1.X flavors
because they had to
Mudgett
Sent: Wednesday, November 16, 2011 2:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom Attended Transfer
When you perform an attended transfer, the extension of the person
transferring is displayed to the co-worker.
Can I override
When you perform an attended transfer, the extension of the person
transferring is displayed to the co-worker.
Can I override the caller ID to display the caller’s callerID
during a
blind transfer?
Upgrade to 10.0 – this isn’t available in any of the 1.X flavors
because they had
, November 16, 2011 2:21 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom Attended Transfer
When you perform an attended transfer, the extension of the person
transferring is displayed to the co-worker.
Can I override the caller ID to display
On Wed, Nov 16, 2011 at 1:49 PM, eherr email.eherr9...@gmail.com wrote:
When you perform an attended transfer, the extension of the person
transferring is displayed to the co-worker.
Can I override the caller ID to display the caller’s callerID during a blind
transfer?
Thanks,
--E
.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ryan Wagoner
Sent: Wednesday, November 16, 2011 3:14 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom Attended Transfer
: [asterisk-users] Polycom Attended Transfer
I understand and agree.
There is one client who prefers having the attended transfer still display
the original caller ID because some users still just hit transfer and
hangup. The boss has a few times got caught saying What when he thought
On Sun, Aug 7, 2011 at 9:32 PM, Mike l...@net-wall.com wrote:
Hi,
[paging]
exten = s,1,Verbose(1,paging)
exten = s,n,Set(TIMEOUT(absolute)=30) ;to prevent call from being stuck
exten = s,n,SIPAddHeader(Alert-Info: Ring Answer)
exten = s,n,Page(SIP/sipphone)
** **
...@lists.digium.com] On Behalf Of Warren Selby
Sent: Monday, August 08, 2011 2:15 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom and auto answer
On Sun, Aug 7, 2011 at 9:32 PM, Mike l...@net-wall.com wrote:
Hi,
[paging]
exten = s,1,Verbose(1
Here's what I have and it works for me in 1.8.5:
in sip.cfg
voIpProt
voIpProt.SIP
voIpProt.SIP.alertInfo
voIpProt.SIP.alertInfo.1.class=ringAutoAnswer
voIpProt.SIP.alertInfo.1.value=Ring Answer
voIpProt.SIP.alertInfo.2.class=autoAnswer
voIpProt.SIP.alertInfo.2.value=Auto Answer /
)?
Mike
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike
Sent: Monday, August 08, 2011 7:18 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Polycom and auto answer
Warren,
Thanks
Hi,
I've been meaning to fix my non-working paging feature here for a while, and
I've just spent the last 5 hours looking at many, many web pages that all
say the same thing. I am using Asterisk 1.6.2.18 and Polycom phones, both
older (501 with latest legacy 3.1.7 firmware) and newer (335 and
2011/7/7 Gord Urquhart gord...@gmail.com
Oliver
Your problem is you have not turned on notifycid=yes in sip.conf. Back
on June 28 in another thread you said
With asterisk 1.6.1.18, I could make this work without setting
notifycid=yes isn sip.conf.
butyes that gets the monitored
Oliver
Your problem is you have not turned on notifycid=yes in sip.conf. Back
on June 28 in another thread you said
With asterisk 1.6.1.18, I could make this work without setting
notifycid=yes isn sip.conf.
butyes that gets the monitored line to blink on an incoming call, but as
you have
Using a Polycom 650 with 3.3.1, I could not have Directed Pickup working.
More precisely, I configured the phone using call and attendant entries
as described in this thread.
Whenever a call comes in, BLF is blinking green.
Pressing the associated key generate generates a general Call Pickup
I am dtmf recognition issues. Out bound calls go though dahdi trunk/sangoma
a400. Dtmf tones are not being recognized. Is there any issues with the latest
polycom firmware?
Sent from my android device.--
_
-- Bandwidth and
Which dtmf method?
I think we use inband here without issue.
I am dtmf recognition issues. Out bound calls go though dahdi
trunk/sangoma a400. Dtmf tones are not being recognized. Is there any
issues with the latest polycom firmware?
Sent from my android device.
--
@lists.digium.com
Sent: Wed, 29 Jun 2011 12:37 PM
Subject: Re: [asterisk-users] Polycom ip320 dtmf issues
Which dtmf method?
I think we use inband here without issue.
I am dtmf recognition issues. Out bound calls go though dahdi trunk/sangoma
a400. Dtmf tones are not being recognized. Is there any
2011/6/20 Gord Urquhart gord...@gmail.com
I missed one important parameter in my setup of BLF for polycom phones (at
least if you want to do one touch directed pickup)
In sip.conf add
notifycid=yes
the notifycid=yes causes asterisk to add a target uri = callID to the XML
of the
I missed one important parameter in my setup of BLF for polycom phones (at
least if you want to do one touch directed pickup)
In sip.conf add
notifycid=yes
the notifycid=yes causes asterisk to add a target uri = callID to the XML
of the SIP notify. Without this target uri the Polycom
From http://www.voip-info.org/wiki/view/Asterisk+presence
Polycom Phones (updated for 3.2.X firmware with asterisk 1.6.1 Jan/2010) With
SIP 3.2.X firmware (available on the Polycom download site) and Asterisk
1.6.1, Polycom phones now support a full featured BLF showing statuses of
Ringing, Inuse
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
Hi,
Anybody know how to set polycom 501 subscription expiry ?
-S
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live
On Thu, May 19, 2011 at 1:24 PM, Ryan Wagoner rswago...@gmail.com wrote:
I updated my phones to the UCS 3.3.1 firmware a few months back. The
scenario is I place a call and receive an incoming call. With 3.3.1
the screen will show call 1/2 and I have to press the down arrow to
see the caller
I updated my phones to the UCS 3.3.1 firmware a few months back. The
scenario is I place a call and receive an incoming call. With 3.3.1
the screen will show call 1/2 and I have to press the down arrow to
see the caller name / number. Has anybody else noticed this with
3.3.1? I had thought with
Currently we have following working page now i want to add custom ring type so
people pay attention. Anybody know about what would be the variable to change
custom ringer
[all-page]
exten = s,1,Set(TIMEOUT(absolute)=15)
exten = s,n,AGI(page.agi)
exten = s,n,SIPAddHeader(Alert-Info: Ring
We're looking to purchase new phones for Asterisk. There are a limited
number of new Polycom 501's on the market, mostly refurbished available.
Can you recommend a replacement phone? What ever model replaced the
501?
-Satish
--
The Polycom 501 has basically been replaced by the Polycom 550.
Thanks,
--Warren Selby, dCAP
On Apr 1, 2011, at 4:25 PM, satish patel satish...@hotmail.com wrote:
We're looking to purchase new phones for Asterisk. There are a limited
number of new Polycom 501's on the market, mostly
You are awesome!!!
--
Sent from my iPhone
On Apr 1, 2011, at 5:40 PM, Warren Selby wcse...@selbytech.com wrote:
The Polycom 501 has basically been replaced by the Polycom 550.
Thanks,
--Warren Selby, dCAP
On Apr 1, 2011, at 4:25 PM, satish patel satish...@hotmail.com
wrote:
We're
2011/2/17 Mike l...@net-wall.com
Hi,
Is there ANY way for me to see the status of the Polycom DND buttons in the
Asterisk hints? I`m using the BLF buttons to see the status of other
people`s lines, and DND should logically be somehow reflected (I don`t care
as much about Polycom showing
2011/3/9 Russell Bryant russ...@digium.com
- Original Message -
I tried to work around this by centralizing DND requests in Asterisk
and sending back a short (You're in DND mode) text to Polycom's
screen (using sipsak for this).
This was rather disappointing as Poycoms redirect
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
Sent: Wednesday, March 09, 2011 12:57 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom Do Not Disturb button and asterisk hints
On Thu, Feb 17, 2011 at 12:02 AM, Patrick Lists
asterisk-l...@puzzled.xs4all.nl wrote:
No parameters were rejected. Maybe my perception of backlight off is
incorrect. When it is off I expect it so be similar to a Cisco 7961. So no
light whatsoever and very hard to read in dim light. Yet in the
List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom IP335
On 02/17/2011 12:17 AM, ERIC HERRON wrote:
To disabled the mwi chirp can be set to silence.
MESSAGE_WAITING se.pat.misc.1.name=message waiting
se.pat.misc.1.inst.1.type=silence ..
This did not work but looking
-Commercial Discussion
Subject: Re: [asterisk-users] Polycom IP335
On 02/17/2011 03:20 AM, Ryan Wagoner wrote:
[snip]
whichsection it is under. My 3.2.x config file worked except for
alert info, ringer, and feature settings, which was outlined
On 02/17/2011 02:28 PM, ERIC HERRON wrote:
I have ind.pattern.
In firmware version 3.3.1? I have only found a reference to ind.pattern
in the Simplify Configuration Improvements Guide which you mentioned
yesterday. Afaict there is no ind.pattern section in the 3.3.0 Admin
Guide and the
On 02/17/2011 02:26 PM, ERIC HERRON wrote:
Yeah it’s the same thing; I think.
I think we have different config files…are you using the split?
Unfortunately I have no idea what the split means. Can you please explain?
Regards,
Patrick
--
On 02/17/2011 02:23 PM, Ryan Wagoner wrote:
[snip]
The color screen must be different or it is a firmware bug. Was it any
different on 3.2.x vs 3.3.x? On my IP550 you can still read the screen
I have not tried firmware 3.2.x. I'll give that a try once I figure out
the old config system.
On 02/17/2011 12:27 PM, Mike wrote:
Hi,
Is there ANY way for me to see the status of the Polycom DND buttons in
the Asterisk hints? I`m using the BLF buttons to see the status of other
people`s lines, and DND should logically be somehow reflected (I don`t
care as much about Polycom showing the
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
On Wed, Feb 16, 2011 at 4:51 PM, ERIC HERRON e...@lanline.com wrote:
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
Subject: Re: [asterisk-users] Polycom IP335
On Wed, Feb 16, 2011 at 4:51 PM, ERIC HERRON e...@lanline.com wrote:
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
On Wed, Feb 16, 2011 at 2:51 PM, ERIC HERRON e...@lanline.com wrote:
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
...@lists.digium.com] On Behalf Of Ryan Wagoner
Sent: Wednesday, February 16, 2011 3:03 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom IP335
On Wed, Feb 16, 2011 at 2:51 PM, ERIC HERRON e...@lanline.com wrote:
I am posting here since you guys
On Wed, Feb 16, 2011 at 3:05 PM, ERIC HERRON e...@lanline.com wrote:
I have it on the 430s.
I think it’s a firmware issue but I am having trouble replicating it on the
430
I could have sworn I had it on one phone before I rebooted it but memory
might be influenced by hopes.
What
: Re: [asterisk-users] Polycom IP335
On Wed, Feb 16, 2011 at 3:05 PM, ERIC HERRON e...@lanline.com wrote:
I have it on the 430s.
I think it's a firmware issue but I am having trouble replicating it on
the
430
I could have sworn I had it on one phone before I rebooted it but memory
On 02/16/2011 09:43 PM, ERIC HERRON wrote:
On IP430s
cat sip.ver
VVX-1500 3.2.2.0481
All others 3.2.2.0477
2345-11402-001.bootrom.ld sip.ld
Phone1.cfg
msg msg.bypassInstantMessage=1
mwi msg.mwi.1.subscribe= msg.mwi.1.callBackMode=contact
msg.mwi.1.callBack=*97 msg.mwi.2.subscribe=
Sip.cfg
On Wed, Feb 16, 2011 at 5:49 PM, Patrick Lists
asterisk-l...@puzzled.xs4all.nl wrote:
I share your pain. I have an IP335 and IP670 here. Have not configured the
IP335 yet but using the latest Admin Guide (3.3.1) did configure the IP670
running the latest bootrom (4.3.0) and firmware (3.3.1).
.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ryan Wagoner
Sent: Wednesday, February 16, 2011 6:10 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom IP335
On Wed, Feb 16, 2011
On 02/17/2011 12:10 AM, Ryan Wagoner wrote:
[snip]
Backlight works fine on a IP550 with 3.3.1 . I have mine set to off
when idle. I like that the 3.3.x series doesn't required the default
sip.cfg and phone1.cfg files. The structure of the XML seems cleaner
and more consistent.
up
On 02/17/2011 12:17 AM, ERIC HERRON wrote:
I haven’t played with the backlights yet.
One annoyance at a time.
Agreed :)
To disabled the mwi chirp can be set to silence.
MESSAGE_WAITING se.pat.misc.1.name=message waiting
se.pat.misc.1.inst.1.type=silence ….
Thanks for the tip Eric. The
On 02/17/2011 12:17 AM, ERIC HERRON wrote:
[snip]
I am trying the different firmwares now to see if it makes any difference.
In the admin guide I just came across: up.oneTouchVoiceMail default 0
If set to 1, the voice mail summary display is bypassed and voice mail
is dialed directly (if
On Wed, Feb 16, 2011 at 8:38 PM, Patrick Lists
asterisk-l...@puzzled.xs4all.nl wrote:
On 02/17/2011 12:10 AM, Ryan Wagoner wrote:
up
up.backlight up.backlight.idleIntensity=0
up.backlight.onIntensity=3
/up.backlight
/up
Here's what I have:
up
up.idleTimeout=10
] On Behalf Of Patrick Lists
Sent: Wednesday, February 16, 2011 9:19 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom IP335
On 02/17/2011 12:17 AM, ERIC HERRON wrote:
[snip]
I am trying the different firmwares now to see if it makes any difference
On 02/17/2011 03:20 AM, Ryan Wagoner wrote:
[snip]
whichsection it is under. My 3.2.x config file worked except for
alert info, ringer, and feature settings, which was outlined in
Simplified_Configuration_Improvements_in_UC_Software3_3_0_TB60519.pdf
Just went through that doc. Interesting
On 02/17/2011 12:17 AM, ERIC HERRON wrote:
To disabled the mwi chirp can be set to silence.
MESSAGE_WAITING se.pat.misc.1.name=message waiting
se.pat.misc.1.inst.1.type=silence ….
This did not work but looking at the example files in the 3.3.1 firmware
the snippet below did work (mind the
Hi,
For future reference, it might be useful to notice (from SIP 3.1 Admin
Manual):
attendant/ attributes are only available to SoundPoint 320/330, 430, 550,
560, 600, 601, 650 and 670 phones only.
For a 3.1.3-enabled 501, has someone been able monitor a third status beyond
Idle, OnCall ones ? I
After someone sent me an email saying his directed pickup did not work. I
realized I forgot to mention that directed pickup needs to be enabled in
extensions.conf i.e. add the following
exten=_*8.,1,SET(GLOBAL(PICKUPMARK)=${EXTEN:2})
exten =
Hi I'm new to this list, so please forgive me off-topic or RTFM-questions.
I have an asterisk/elastix driven phone-environment using Polycom
SoundPoint IP 650 as extensions. When adding just one custom ringtone
(~57KB) in a proper format (ML.wav: RIFF (little-endian) data, WAVE
audio, ITU G.711
All,
I'm using Asterisk 1.6 and using Polycom 500's with SIP firmware
2.1.3. I can not seem to get the Message Waiting Indicator to work
reliably (and in my opinion correctly) with voicemail.
I've got the following in my phone.cfg:
reginfo
msg msg.bypassInstantMessage=1
mwi
Brian C. Huffman wrote:
Does anyone know how to setup this phone to work with asterisk so that
the indicator light comes on when there's a new message and goes off
quickly (less than a minute) after the message is deleted?
My phone.cfg for extension 4221 and the voicemail extension of 4200
On Thu, Jan 20, 2011 at 12:55 PM, Brian C. Huffman
bhuff...@etinternational.com wrote:
Does anyone know how to setup this phone to work with asterisk so that the
indicator light comes on when there's a new message and goes off quickly
(less than a minute) after the message is deleted?
Thanks,
I've got the following in my phone.cfg:
reginfo
msg msg.bypassInstantMessage=1
mwi msg.mwi.1.callBack=*97 msg.mwi.1.callBackMode=contact
msg.mwi.1.subscribe= /mwi
/reginfo
The actual config looks good, but the structure of the XML is off. Here's
what I use (and it works):
phone1
msg
With SIP 3.2.X firmware (available on the Polycom download site) and
Asterisk 1.6.1, Polycom phones now support a full featured BLF showing
statuses of Ringing, Inuse and Online and one touch directed call pickup.
On the asterisk side all that needs to be done is to add a hint to the
extension. On
Is the buddy watch tag activated in your mac-directory.xml file ? bw1/bw
item
lbSebastien/lb
fnSebastien/fn
lnThomas/ln
ct222/ct
sd1/sd
bw1/bw
/item
---
Sebastien Thomas
Amplisys Inc. - Digital Telephony Integration Specialists
T: 514.225.4141 x222 F: 514.225.4162 TF: 1-877-AMPLISYS
Yeah... My directory looks like this:
directory
item_list
item
ln6288/lnfn/fnct6288/ctsd1/sdrt2/rtdc/ad0/adar0/arbw1/bwbb0/bb
/item
item
ln6208/lnfn/fnct6208/ctsd2/sdrt2/rtdc/ad0/adar0/arbw1/bwbb0/bb
/item
item
ln6234/lnfn/fnct6234/ctsd3/sdrt2/rtdc/ad0/adar0/arbw1/bwbb0/bb
/item
item
Ok, that looks good.
We use FreePBX, and I know I had to modify a couple Asterisk files to get the
BLF working ... here are some of my mods but may also be used for FOP2 (I dont
recall which go for BLF and which go FOP2).
vi /etc/asterisk/sip_registrations_custom.conf
allowsubscribe=yes
vi
Thanks! Blf is working now. I forgot I had to set set subscribecontext.
When a phone is ringing, the blf light is solid red and the icon is a
(/) type icon indicating unavailable. I'm also interested in directed
pickup. I set up the following:
call.directedCallPickupString=*6
Would anyone happen to have some examples of polycom configs,
specifically the 650 with sidecar for blf.
I have the asterisk side all configured since I've set up blf with other
types of phones, but I'm missing the polycom side.
I've put together a mac-directory.xml, and the sidecar now
According to the Admin guide EFK is not supported on 501s
This capability applies to the SoundPoint IP 32x/33x, 450, 550, 560, 650,
and
670 desktop phones, the SoundStation IP 5000, 6000, and 7000 conference
phones, and Polycom VVX 1500 business media phones
On Fri, Dec 3, 2010 at 5:02 PM,
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gord Urquhart
Sent: Monday, December 06, 2010 4:44 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom Park by EFK
According
Has anyone gotten one-touch call parking to work on Polycom phones via
the Enhanced Feature Keys feature working? I've looked at various
examples, they appear correct, but the phones (501, 3.1.x firmware)
show the Park button while in a call but this does not actually cause
the call to be parked.
On Fri, Dec 3, 2010 at 8:02 PM, Andrew Joakimsen joakim...@gmail.com wrote:
Has anyone gotten one-touch call parking to work on Polycom phones via
the Enhanced Feature Keys feature working? I've looked at various
examples, they appear correct, but the phones (501, 3.1.x firmware)
show the Park
I've had phones before where, with the phone on-hook, it still implements
the local dialplan. E.g., if I dialed 0 (on-hook), after three seconds,
it would dial the operator, and have the call on speakerphone. Does
Polycom allow this functionality? Clearly, not a necessary feature... but
it
Hi Everyone,
Configuring a Polycom conference bridge IP 5000 to connect to Asterisk. For
some reason I don't see any SIP packets coming in to Asterisk at all. I
don't want to use XML or ftp etc for now and just use the Web Interface to
get it going with basic features. But the Web UI is a bit
Hello,
I have been tearing my hair out on this issue for 2 days, any help
would be appreciated.
We have a normal network and a Cisco SGE2010P switch - a 48 port PoE switch
There are two VLANs, 1(data) 50(VoIP). When Polycoms are connected
to the switch with VLAN 50 hard coded in the config
That switch doesn't seem to support CDP, so the Polycom phone has no way of
figuring out which VLAN to tag itself to automagically. It will grab the
primary VLAN unless you specify otherwise in the phone's setup.
On boot of the phone, go into setup, default password 456, there's an option in
One more thing: Make sure that the port going to your data-DHCP server doesn't
have the voice VLAN set on it. I troubleshot an installation for a few hours
before thinking of this...
Bests,
Seb
On 2010-10-08, at 2:37 AM, Thermal Wetland wrote:
Hello,
I have been tearing my hair out on
On Fri, Oct 8, 2010 at 5:16 AM, Sebastien Thomas li...@amplisys.ca wrote:
One more thing: Make sure that the port going to your data-DHCP server
doesn't have the voice VLAN set on it. I troubleshot an installation for a
few hours before thinking of this...
Interesting, the DHCP server for
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ken D'Ambrosio
Sent: Thursday, October 07, 2010 9:34 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Polycom: full caller ID?
Hi, all. When I
, 2010 7:36 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Polycom dhcp boot
Hi all
I have a few Polycom 331's but after following allot of advice I can't
get them to provision from a dhcp boot server. We have a sonicwall
router in place.
I can press setup and set the FTP boot
Your string and boot-option look good.
In the SonicWall config, its a two step process:
- create the new boot option under DHCP Server menu Advanced button
Add Option
- assign it to your lease scope under DHCP Server menu DHCP Server Lease
Scopes section Edit button Advanced tab DHCP
Use lowercase for ftp:// . That might be the issue but it should be
easy to test. Do your FTP server logs shpw anything?
I will double check but I believe that lower case ftp is being used. I do
have upper case PlcmSpIp:PlcmSpip as the password. I will see if lower case
usernames and
On Sun, Sep 12, 2010 at 8:57 AM, colin mcdermott
colinjamesmcderm...@gmail.com wrote:
Use lowercase for ftp:// . That might be the issue but it should be
easy to test. Do your FTP server logs shpw anything?
I will double check but I believe that lower case ftp is being used. I do
have
Hi all
I have a few Polycom 331's but after following allot of advice I can't
get them to provision from a dhcp boot server. We have a sonicwall
router in place.
I can press setup and set the FTP boot server to my * box. From there
th phones boot fine. But I cannot get them to autoprovision.
I
Use lowercase for ftp:// . That might be the issue but it should be easy to
test. Do your FTP server logs shpw anything?
On Sep 10, 2010, at 5:35 PM, colin mcdermott colinjamesmcderm...@gmail.com
wrote:
Hi all
I have a few Polycom 331's but after following allot of advice I can't
get
Has anyone successfully made this scenario work in 1.4. I found info at
http://www.voip-info.org/wiki/view/Asterisk+presence indicating that this
does not work with 1.4 implementations.
--
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...@lists.digium.com] On Behalf Of David Backeberg
Sent: Monday, August 16, 2010 11:04
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom 331 freezes connecting to FreePBX
On Mon, Aug 16, 2010 at 4:21 PM, Ben Schorr b...@rolandschorr.com wrote
11:04
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom 331 freezes connecting to FreePBX
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
We deployed a single phone handset (Polycom 331) at a remote site. We
have a IPSEC VPN running between the firewall at the remote site and the
firewall at the site where our Asterisk/FreePBX box lives. We have used
a similar configuration for this site before and it worked fine.
We gave the
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