Title: Re: [asterisk-users] POlycom phone not ringing behind firewall (401 permission denied)
Hello Jerry,
Tuesday, June 30, 2020, 5:23:15 AM, you wrote:
I have polycom phones setup in an office connected to a cloud asterisk server. The polycom phones can call out just fine - audio just
Hi All,
I have polycom phones setup in an office connected to a cloud asterisk
server.
The polycom phones can call out just fine - audio just fine.
However a call coming into the cloud asterisk answers fine - get the
autoattendant, enter the extension and the polycom does not ring. The CLI
shows t
Does polycom support "normal" multicast from asterisk as the source?
I'm getting the impression that it only supports its OWN phone to phone
multicast or something.
Thanks,
Jerry
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On 2019-12-19 06:10, Antony Stone wrote:
> On Thursday 19 December 2019 at 14:04:36, Jerry Geis wrote:
>
>> I presume it would just be sending a SIP message - no need to get anything
>> back. Just want to pop a message on the phone.
I think there are some Polycoms that support RFC 3428 SIP MESSAG
On Thursday 19 December 2019 at 14:04:36, Jerry Geis wrote:
> I presume it would just be sending a SIP message - no need to get anything
> back. Just want to pop a message on the phone.
Yes, but *what* message do you need to send?
How does a Polycom do this?
Without knowing that, I don't think
I presume it would just be sending a SIP message - no need to get anything
back. Just want to pop a message on the phone.
Thanks,
Jerry
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Check out the ne
On Wednesday 18 December 2019 at 22:32:24, Jerry Geis wrote:
> Hi all,
>
> I want to send a text message to a polycom phone.
> I know how to create a call file - but that will "call" the phone and
> nothing happens till the phone is answered.
>
> How do I create a call file that will "send" a te
Hi all,
I want to send a text message to a polycom phone.
I know how to create a call file - but that will "call" the phone and
nothing happens till the phone is answered.
How do I create a call file that will "send" a text message over SIP to the
polycom phone?
So the phone does not have to answ
This is done via the custom extension state or hints. Basically you
create a custom hint for 444 and monitor that on your phone like any
other extension. You then enable or disable the hint in the same
dialplan for 444 and 555.
https://wiki.asterisk.org/wiki/display/AST/Extension+State+an
Hi All. I have an interesting scenario. We use the Polycom VXX phones and
have an auto-attendant on our Asterisk system. The receptionist can turn
the auto-attendant off and on as she would like (she dials 444 to enable
and 555 to disable). However, I’d like to have one of the BLFs on her
Polycom l
Solved it!
Turns out UCS Polycoms are quite picky about blank callerids, to the
extant they ignore those packets completely.
My global "callerid=" in sip.conf was intentionally blank. In ten
years, in never caused a problem.
By setting to 0, the Polycoms that didn't respond to SIP OPTIONS (nor
the
I always set it to no, but set the registration time to 60 seconds,
and that has always worked for me.
On Wed, 23 Aug 2017 17:23:38 -0400,
Gary Reuter wrote:
>
> Hello,
> We've had dozens of Polycom 3.x firmware phones deployed and working
> great for years.
> Now I've finally been charged with t
Hello,
We've had dozens of Polycom 3.x firmware phones deployed and working
great for years.
Now I've finally been charged with the long-overdue task of figuring
out why newer Polycom devices with 4.x firmware register fine but do
not respond to SIP OPTIONS request and therefore always become
UNREA
2016 6:31 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom SoundStation IP 6000 does not register
Do you have any LLDP or CDP enabled anywhere ?
2016-12-21 19:50 GMT+01:00 Victor Villarreal :
Hi Yves,
Maybe your switch put your Polycom ins
Do you have any LLDP or CDP enabled anywhere ?
2016-12-21 19:50 GMT+01:00 Victor Villarreal :
> Hi Yves,
>
> Maybe your switch put your Polycom inside a Voice VLAN, based on the MAC
> of the phone. Maybe with the snom this not happen because your switch don't
> see the MAC of the Snom as a "suppe
Hi Yves,
Maybe your switch put your Polycom inside a Voice VLAN, based on the MAC of
the phone. Maybe with the snom this not happen because your switch don't
see the MAC of the Snom as a "supperted IP Phone".
2016-12-21 13:59 GMT-03:00 Yves :
> sorry... typo
> the problematic phone has the 1
sorry... typo
the problematic phone has the 192.168.0.13
the asterisk has 192.168.1.211
when i connect a snom phone on the cable that was in the soundstation
6000 before and configure the
phone to use 192.168.0.13 it does register on the asterisk via 5060/UDP...
it would be helpful if some
On Wed, Dec 21, 2016 at 7:50 AM, Yves wrote:
> Hi Mark,
>
> yes, you are right... these are different VLANs
> I configured the other phone to use the same IP (192.168.1.13)... and it
> worked flawlessly... on the SAME Networkcable in the same plug...
> so it must have something to do with the poly
Hi Mark,
yes, you are right... these are different VLANs
I configured the other phone to use the same IP (192.168.1.13)... and it
worked flawlessly... on the SAME Networkcable in the same plug...
so it must have something to do with the polycom phone config...
remember... when I use tcp the pho
Yves,
Didn't you say that
AsteriskServer: 192.168.1.211
SIP-user: 165
?
On 12/21/2016 4:24 AM, Yves wrote:
. It is sure for 100% that there is no firewall or something else
mangeling
in between... another Hardphone works as expected using the same
Netzworkcable on the same Networkplug with
Hi,
I do not have a switch to mirror the traffic... I am only remotely
connected to the office, where all is set up.
I have full control over asterisk and the phone and I tcpdumped the
traffic coming from the phone.
The weird thing is... if I configure the SIP-Server Setting to use TCP
on Port
On 12/19/2016 10:26 AM, Yves wrote:
There are no SIP Packets arriving at my asterisk at all... and it has
nothing to do with a firewall or similar...
I can ping the phone from the asterisk,
If both of these items are true, then I'd look at the phone
configurations. Does the provisioning f
Behalf Of Olivier
Sent: Monday, December 19, 2016 9:51 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom SoundStation IP 6000 does not register
2016-12-19 16:26 GMT+01:00 Yves :
Hi,
I am pulling my hair for days now...
I can´t get
2016-12-19 16:26 GMT+01:00 Yves :
> Hi,
>
> I am pulling my hair for days now...
>
> I can´t get a PolyCom SoundStation IP 6000 (Conferencephone) to register
> with my Asterisk.
>
> There are no SIP Packets arriving at my asterisk at all... and it has
> nothing to do with a firewall or similar...
Hi,
I am pulling my hair for days now...
I can´t get a PolyCom SoundStation IP 6000 (Conferencephone) to register
with my Asterisk.
There are no SIP Packets arriving at my asterisk at all... and it has
nothing to do with a firewall or similar...
Simple Question:
Does anybody have a running
It sounds like you have problems with your firewall. Your 401 replies don't
reach the phones.
On Thursday, October 08, 2015 02:50:24 PM Jerry Geis wrote:
> Do polycom phones not LIKE using something other than port 5060 ???
>
> I have five of them behind a firewall and my asterisk server is remo
Do polycom phones not LIKE using something other than port 5060 ???
I have five of them behind a firewall and my asterisk server is remote.
Other devices are registering just fine, just not my polycom phones.
Today I notices that ONE registered, but it grabbed port 5060.
1004/1004
On Thu, 27 Aug 2015 16:17:38 -0400
Jerry Geis wrote:
> I have a polycom phone behind a firewall.
> The phone registers - but I only hear half channel audio.
What version of Asterisk?
Which half can you hear?
After a recent update I had a problem with one way audio. Maybe you
are having the same
I have a polycom phone behind a firewall.
The phone registers - but I only hear half channel audio.
I have tried nat=yes, nat=force_rport,comedia and
nat=autio_force_rport,auto_comedia (reloading asterisk every time).
made no difference.
How might I get full audio path?
Thanks,
Jerry
--
_
asterisk-users-boun...@lists.digium.com wrote on 01/23/2015 10:38:27 AM:
> Would you be willing to send the configuration from asterisk for this?
It is pretty bog standard, but sure:
[111]
callerid="Conf Rm" <111>
secret=x
type=friend
host=dynamic
call-limit=5
context=xx
qualify=
> We run a variety of 5000, 6000, and 7000 series Soundstations running
> Asterisk 11.6.0 and the phones are at 4.0.3.7562. We do not see these
> registration issues.
Would you be willing to send the configuration from asterisk for this?
This message may be private and confidential. If you have
ailing List - Non-Commercial Discussion
mailto:asterisk-users@lists.digium.com>>
Date: Friday, 23 January 2015 16:24
To: "asterisk-users@lists.digium.com<mailto:asterisk-users@lists.digium.com>"
mailto:asterisk-users@lists.digium.com>>
Subject: [asterisk-users]
asterisk-users-boun...@lists.digium.com wrote on 01/23/2015 10:24:24 AM:
> Hello,
>
> I'm having a problem with a few Polycom SoundStation 6000s.
> Everything works fine, but they drop registration to asterisk after
> about maybe 30 minutes – the phone does not re-try to register and
> if you t
Hello,
I'm having a problem with a few Polycom SoundStation 6000s. Everything works
fine, but they drop registration to asterisk after about maybe 30 minutes - the
phone does not re-try to register and if you try to dial out on the phone it
says "URI Dialing is Disabled"
Has anyone else had th
e just way too cool.
> Thanks;
> John
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michael
> Englehorn
> Sent: Monday, January 12, 2015 12:19 AM
> To: asterisk-users@list
risk-users-boun...@lists.digium.com] On Behalf Of Michael
Englehorn
Sent: Monday, January 12, 2015 12:19 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Polycom instant messages
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Is it possible to use the instant messaging feature of Po
On Sun, Jan 11, 2015 at 11:19 PM, Michael Englehorn
wrote:
> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
>
> Is it possible to use the instant messaging feature of Polycom phones in
> Asterisk? At the moment I'm seeing this in the SIP messaging when I try
> to send one from a Polycom 450.
>
>
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Is it possible to use the instant messaging feature of Polycom phones in
Asterisk? At the moment I'm seeing this in the SIP messaging when I try
to send one from a Polycom 450.
<--- SIP read from UDP::5060 --->
INVITE sip:0100@:5060;user=phone SIP/2.0
On Thursday, September 18, 2014 10:31 AM, John Kiniston wrote:
> There is one product that I know of that is Compatible with Polycom
> paging. The Algo 8180 Audio Alerter. [snip]
>
> You can call it via SIP from asterisk and it can multicast in the special
> Polycom format to your phones.
Wow,
- Original Message -
> Tim,
> I THINK but I'm not sure that you can do this with the Polycom
> multicast page function. Have you attempted this yet?
> Thanks
> david
Given the odd nature of multicast paging with Polycom, I was hoping to avoid
such a setup. My recollection is having thi
On Wed, Sep 17, 2014 at 10:06 PM, Nathan Anderson wrote:
> BUT Polycom handsets cannot be configured to just listen to RTP being
> multicasted to a particular multicast IP like many other IP phones
> can...the signalling for Polycom multicast paging and PTT functionality is
> completely proprieta
Yes, I am pretty sure that if a Polycom unit is set DND and you initiate a
multicast page from another Polycom handset on a page or PTT channel that the
DND handset is subscribed to (like the emergency channel), then you will hear
audio on that handset.
BUT Polycom handsets cannot be configured
Tim,
I THINK but I'm not sure that you can do this with the Polycom multicast
page function. Have you attempted this yet?
Thanks
david
On Tue, Sep 16, 2014 at 10:07 PM, Tim Nelson wrote:
> Greetings-
>
> As many of your are Polycom "experienced", I was hoping some kind soul
> could provide dir
Greetings-
As many of your are Polycom "experienced", I was hoping some kind soul could
provide direction on a specific issue.
On a system running Asterisk 11.11.0 (and latest FreePBX), I'm finding an
instance where, using intercom/paging functionality of FreePBX, I need to
override an end u
I've just deployed several VVX 600's with the Color Expansion Module.
And I'm having a minor issue with them.
Intermittently when a call comes into a ring group the user is
presented with the call pickup option associated with a BLF entry. Not
the normal answer/reject option.
I've explicitly disa
Hi,
I'm configuring a brand new polycom SSIP 7000.
To my surprise, when this telephone boots up, my DHCP server receives a
request that Wireshark classifies a BootP request from which I can't find
any Vendor identification.
The trouble is my DHCP server uses option vendor-class-identifier to ser
Hello;
I have asterisk Asterisk 1.8.23.0-vici and Polycom 331 and I am able to
register from local area network and not able to register from outside the
office. Also from outside the office, I am able to register via PhonerLite
softphone and not able to register via Zoiper softphone.
So from
>> 1) The red light and the beep: How I can let the Phone only have the red
>> light without the beep sound that keep hearing it periodically and it is
>> bothering? Because I tried from the Polycom web based >>settings but nothing
>> related to this .. Maybe, it is settings need to be from the
Hello;
I am using vicidial which is using asterisk 1.8, mean while when the extension
has voicemail, I always see the red light on the Polycom and hear the beep
sound (toot toot) in period time. Also, I can see at the LCD an option to
select it for accessing the voicemail but I am facing the f
On Wed, May 15, 2013 at 12:10 PM, Ken D'Ambrosio wrote:
> Hey, all. I've got an office set up with Asterisk, and forwarding's got a
> bit of a glitch:
> When they forward, they listen for the remote phone to ring, then hang up.
> If the remote phone doesn't connect, it goes to the original phon
> Hey, all. I've got an office set up with Asterisk, and forwarding's
> got
> a bit of a glitch:
> When they forward, they listen for the remote phone to ring, then
> hang
> up. If the remote phone doesn't connect, it goes to the original
> phone's VM. Is this Polycom's "fault," or Asterisk's?
Hey, all. I've got an office set up with Asterisk, and forwarding's got
a bit of a glitch:
When they forward, they listen for the remote phone to ring, then hang
up. If the remote phone doesn't connect, it goes to the original
phone's VM. Is this Polycom's "fault," or Asterisk's? I've been
us/support/voice/soundpoint_ip/soundpoint_ip330_320.html_>
>>
>>
>> Kevin Larsen - Systems Analyst - Pioneer Balloon - Ph: 316-688-8208
>>
>>
>>
>> From: Daniel - Asterisk > <mailto:earohua...@gmail.com>>
>>
>>
..@gmail.com>>
To: Asterisk Users Mailing List - Non-Commercial Discussion
mailto:asterisk-users@lists.digium.com>>,
Date: 04/12/2013 12:42 PM
Subject: [asterisk-users] Polycom Soundpoint IP 330 provisioning
Sent by: asterisk-users-boun...@lists.digium.co
t; Kevin Larsen - Systems Analyst - Pioneer Balloon - Ph: 316-688-8208
>>
>>
>>
>> From:Daniel - Asterisk
>> To:Asterisk Users Mailing List - Non-Commercial Discussion <
>> asterisk-users@lists.digium.com>,
>> Date:04/
; Kevin Larsen - Systems Analyst - Pioneer Balloon - Ph: 316-688-8208
>
>
>
> From:Daniel - Asterisk
> To:Asterisk Users Mailing List - Non-Commercial Discussion <
> asterisk-users@lists.digium.com>,
> Date: 04/12/2013 12:42 PM
> Subject:
:42 PM
Subject:[asterisk-users] Polycom Soundpoint IP 330 provisioning
Sent by:asterisk-users-boun...@lists.digium.com
Hello all,
I need the bootrom.ld file to set up some Polycoms I have
Platform: Model=SoundPoint IP 330, Assembly=2345-12200-001 Rev=A
I've publiched
Hello all,
I need the bootrom.ld file to set up some Polycoms I have
Platform: Model=SoundPoint IP 330, Assembly=2345-12200-001 Rev=A
I've publiched on my FTP files downloaded from
http://support.polycom.com/PolycomService/support/us/support/voice/soundpoint_ip/soundpoint_ip330_320.html
(3.2.3
Ok, thanks for the info.
-Bryan Anderson
On Thu, Mar 7, 2013 at 6:07 PM, Chad Wallace wrote:
> On Thu, 7 Mar 2013 17:12:47 -0800
> Bryan Anderson wrote:
>
> > Has any one ever worked with placing idle display images onto the
> > Polycom SPIP331 phones? I have got it working but when the image
On Thu, 7 Mar 2013 17:12:47 -0800
Bryan Anderson wrote:
> Has any one ever worked with placing idle display images onto the
> Polycom SPIP331 phones? I have got it working but when the image is
> displayed the clock is moved to the top of the screen. That is
> great but it scrolls between the
Has any one ever worked with placing idle display images onto the Polycom
SPIP331 phones? I have got it working but when the image is displayed the
clock is moved to the top of the screen. That is great but it scrolls
between the clock and the registered extension(s) . Has anyone figured out
a
Justin,
I haven't seen it on that model, but I did have a case awhile back where
it happened to me with a different conference phone. Pretty much the same
symptoms you had. Even more fun it was remote so I couldn't get my hands
on it.
I tracked mine down to being an incorrect firmware for that
I have a Polycom IP6000 conference phone, along with a lot of Polycom IP550
units. I've been updating all the 650s to Polycom's from 3.2.3 to the 4.0.3
software release, by hodling 468*and having them pull the update.
It's been fine with the 650s, but the IP6000 (held 68* for that one) keeps
m
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of motty cruz
Sent: Wednesday, December 12, 2012 5:14 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom phones and ring no answer/302 Moved
Temporarily
I have Polycom IP550. The "Forward"
I have Polycom IP550. The "Forward" "No Answer" is working fine when
enabled. I was looking at the sip.cfg but don't know exactly what to look
for, can you give me a hint to where would i find that option?
Thanks,
On Wed, Dec 12, 2012 at 1:48 PM, Justin Sherrill <
justin.sherr...@americanrocksalt
I have several Polycom IP550 phones running UC 4.0.3, connected to Asterisk 1.8.
Setting forwarding for "Always" works as expected; the phone issues a 302
Moved Temporarily, and Asterisk shifts the call to the new location.
Setting forwarding to "No Answer" means a 302 never gets issued. It ju
On 12-09-06 10:46 AM, Chris Nighswonger wrote:
I have some Polycom 351 on Asterisk 10. On the same box as * I have a tftp
server running to handle configs, etc. The Polycom phones have no problem
grabbing config foo from the tftp server as well as writing log files back
to the server. However, wh
I have some Polycom 351 on Asterisk 10. On the same box as * I have a tftp
server running to handle configs, etc. The Polycom phones have no problem
grabbing config foo from the tftp server as well as writing log files back
to the server. However, when I use the web-if on a phone to set a custom
ri
- Original Message -
> On 07/26/2012 03:32 PM, Danny Nicholas wrote:
> > Question 1 - I think asterisk only supports a limited set of
> > statuses
>
> Asterisk does not *receive* presence updates from Polycom phones (or
> really, non-Digium phones) at all. Instead, the presence (status)
>
On 07/26/2012 04:28 PM, Tim Nelson wrote:
Greetings-
I've got a handful of Polycom IP 550 handsets connected to an Asterisk 1.8.12.0
system. Everything is running smoothly with few problems. However, I have an
issue that maybe someone could shed light on...
Many of the phones have 'buddy watc
On 07/26/2012 03:32 PM, Danny Nicholas wrote:
Question 1 - I think asterisk only supports a limited set of statuses
Asterisk does not *receive* presence updates from Polycom phones (or
really, non-Digium phones) at all. Instead, the presence (status)
updates you are seeing appear on your phon
m.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tim Nelson
Sent: Thursday, July 26, 2012 3:28 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Polycom Presence with Asterisk 1.8.12.0
Greetings-
I've got a handful of Polycom IP 55
Greetings-
I've got a handful of Polycom IP 550 handsets connected to an Asterisk 1.8.12.0
system. Everything is running smoothly with few problems. However, I have an
issue that maybe someone could shed light on...
Many of the phones have 'buddy watch' enabled for the other phones, basically
On Thu, 14 Jun 2012 00:46:25 +
"Klaverstyn, David C" wrote:
> I have a Polycom Handset on a front door and I'd like the phone to
> dial a number as soon as the handset is lifted without having to
> press and buttons or enter any numbers. I know how to do this on a
> Linksys but I can't find
David
C
Sent: Wednesday, June 13, 2012 8:46 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
(asterisk-users@lists.digium.com)
Subject: [asterisk-users] Polycom, Dial Specific Number on Handset Pickup
Hi All,
I have a Polycom Handset on a front door and I'd like the phone
On Tue, Jun 12, 2012 at 4:15 PM, Jon Caum wrote:
> Hello,
>
> I have an issue I remember seeing a while ago and forgot to investigate
> further. Now it is turning into an issue and will need to be resolved. A
> customer has Polycom 335 phones (and a couple Soundstation 6000s), and when
> an exten
Hi All,
I have a Polycom Handset on a front door and I'd like the phone to dial a
number as soon as the handset is lifted without having to press and buttons or
enter any numbers. I know how to do this on a Linksys but I can't find out how
to do it on a Polycom.
I would be greatly appreciate
Hello,
I have an issue I remember seeing a while ago and forgot to investigate
further. Now it is turning into an issue and will need to be resolved. A
customer has Polycom 335 phones (and a couple Soundstation 6000s), and when an
extension is calling out, the screen on the 335 shows the compan
: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Polycom CX3000 IP with Asterisk?
I have a customer that has a CX3000 IP that was designed for MS Lync.
Anyone know if these can run as standard SIP so we can use it with Asterisk?
Thanks
Bryant
I have a customer that has a CX3000 IP that was designed for MS Lync.
Anyone know if these can run as standard SIP so we can use it with
Asterisk?
Thanks
Bryant
--
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Discussion
Subject: Re: [asterisk-users] Polycom firmware 4.0.1 and paging
It appears you need the "info=" if the string you are using is enclosed in
angle brackets.
Alert-Info: fooworks
Alert-Info: does not work
Alert-Info:info= works
On Wed, Feb 15, 2012 at 2:09 PM, M
to:asterisk-users-
> > boun...@lists.digium.com] On Behalf Of Dave Fullerton
> > Sent: Wednesday, February 15, 2012 10:20 AM
> > To: asterisk-users@lists.digium.com
> > Subject: Re: [asterisk-users] Polycom firmware 4.0.1 and paging
> >
> > Which version of ast
um.com [mailto:asterisk-users-
> boun...@lists.digium.com] On Behalf Of Dave Fullerton
> Sent: Wednesday, February 15, 2012 10:20 AM
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] Polycom firmware 4.0.1 and paging
>
> Which version of asterisk are you using? I just
um.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Mike
Sent: Monday, February 13, 2012 10:17 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Polycom firmware 4.0.1 and paging
Thanks Dave, it at least gives me hope that my effor
ling List - Non-Commercial Discussion'
> Subject: Re: [asterisk-users] Polycom firmware 4.0.1 and paging
>
> Thanks Dave, it at least gives me hope that my efforts aren`t wasted.
>
> Mike
>
> > -Original Message-
> > From: asterisk-users-boun...@lists.digium
Thanks David. I will check it out.
-Original message-
From: "Klaverstyn, David C"
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Mon, Feb 13, 2012 04:34:30 GMT+00:00
Subject: Re: [asterisk-users] Polycom IP331 Configuration
This may help you
asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] Polycom firmware 4.0.1 and paging
>
> On 02/10/2012 05:30 PM, Mike wrote:
> > Hi,
> >
> > I just moved many Polycom phones from firmware v3 to 4.0.1b.
> > Anto-Answer simply stopped functioning. I can do
On 02/10/2012 05:30 PM, Mike wrote:
Hi,
I just moved many Polycom phones from firmware v3 to 4.0.1b. Anto-Answer
simply stopped functioning. I can downgrade and make it work, upgrading
kills it again. There obviously is a difference in how the newer
firmware is treating this auto answer sip head
isk-users@lists.digium.com
Subject: [asterisk-users] Polycom IP331 Configuration
I hope this doesn't already exist, but I couldn't find anything to help. I am
installing a brand new Asterisk server, and want to use the Polycom IP331
phones. Does anyone have any steps on how to config
-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike
Sent: Sunday, February 12, 2012 3:34 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Polycom firmware 4.0.1 and paging
Tha
f Of C F
Sent: Sunday, February 12, 2012 12:53 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom firmware 4.0.1 and paging
Are you using the same cfg files?
If yes I would try rewriting them from scratch using the blank cfg files
that come
I hope this doesn't already exist, but I couldn't find anything to help. I am
installing a brand new Asterisk server, and want to use the Polycom IP331
phones. Does anyone have any steps on how to configure these? I have
softphones working just fine, but for some reason I can't find a clear s
Are you using the same cfg files?
If yes I would try rewriting them from scratch using the blank cfg files
that come with new firmware. I have seen wiered things by using olde cfg
files
On Friday, February 10, 2012, Mike wrote:
> Hi,
>
>
>
> I just moved many Polycom phones from firmware v3 to
: Friday, February 10, 2012 7:39 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom firmware 4.0.1 and paging
Mike. Yes sip.ld is the firmware.
I wanted to jump in because i saw you had the phantom ringing problem as well.
I am running 3.3.1 and
Mailing List - Non-Commercial Discussion'
> Subject: Re: [asterisk-users] Polycom firmware 4.0.1 and paging
>
>
>
> Did the 4.0.1b update overwrite sip.ld on these phones? If I recall
> correctly you have to tweak that file to make auto-answer work correctly.
>
>
rcial Discussion'
Subject: Re: [asterisk-users] Polycom firmware 4.0.1 and paging
Did the 4.0.1b update overwrite sip.ld on these phones? If I recall
correctly you have to tweak that file to make auto-answer work correctly.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-
2012 5:37 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom firmware 4.0.1 and paging
On Fri, Feb 10, 2012 at 10:30 PM, Mike wrote:
Hi,
I just moved many Polycom phones from firmware v3 to 4.0.1b. Anto-Answer
simply stopped functioning
:37 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom firmware 4.0.1 and paging
On Fri, Feb 10, 2012 at 10:30 PM, Mike wrote:
Hi,
I just moved many Polycom phones from firmware v3 to 4.0.1b. Anto-Answer
simply stopped functioning. I
On Fri, Feb 10, 2012 at 10:30 PM, Mike wrote:
> Hi,
>
> ** **
>
> I just moved many Polycom phones from firmware v3 to 4.0.1b. Anto-Answer
> simply stopped functioning. I can downgrade and make it work, upgrading
> kills it again. There obviously is a difference in how the newer firmware
> i
Hi,
I just moved many Polycom phones from firmware v3 to 4.0.1b. Anto-Answer
simply stopped functioning. I can downgrade and make it work, upgrading
kills it again. There obviously is a difference in how the newer firmware is
treating this auto answer sip header.
Can anybody tell me if the
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