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
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
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
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
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
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.
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
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
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
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
22, 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 <mefhigos...@gmail.com>:
Hi Yves,
Maybe your s
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
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
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
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
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
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
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
On 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 <yves...@gmx.de>:
Hi,
I am pulling my hair for days now
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
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
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
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 ge...@pagestation.com 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
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
--
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
@lists.digium.com
Subject: [asterisk-users] Polycom SoundStation 6000 Dropping Registration [Spam
score:11%]
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
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
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 try to
On Sun, Jan 11, 2015 at 11:19 PM, Michael Englehorn mich...@englehorn.com
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
...@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 Polycom phones
-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@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 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:CENSORED POLYCOM IP:5060 ---
INVITE
On Wed, Sep 17, 2014 at 10:06 PM, Nathan Anderson nath...@fsr.com 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
- 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 this
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, I
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 tnel...@rockbochs.com wrote:
Greetings-
As many of your are Polycom experienced, I was hoping some kind soul
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
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
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
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
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
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
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 setting
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
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
On Wed, May 15, 2013 at 12:10 PM, Ken D'Ambrosio k...@jots.org 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
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
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.com
Hello all,
I need the bootrom.ld file to set up some Polycoms I have
Platform: Model=SoundPoint IP 330
: 316-688-8208
From:Daniel - Asterisk earohua...@gmail.com
To:Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com,
Date:04/12/2013 12:42 PM
Subject:[asterisk-users] Polycom Soundpoint IP 330 provisioning
Sent
...@gmail.com
To:Asterisk Users Mailing List - Non-Commercial Discussion
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.com
...@gmail.com
mailto:earohua...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
mailto:asterisk-users@lists.digium.com,
Date: 04/12/2013 12:42 PM
Subject: [asterisk-users] Polycom Soundpoint IP 330 provisioning
Sent
PM
Subject: [asterisk-users] Polycom Soundpoint IP 330 provisioning
Sent by:
asterisk-users-bounces@lists.**digium.comasterisk-users-boun...@lists.digium.com
mailto:asterisk-users-**boun...@lists.digium.comasterisk-users-boun...@lists.digium.com
Ok, thanks for the info.
-Bryan Anderson
On Thu, Mar 7, 2013 at 6:07 PM, Chad Wallace cwall...@lodgingcompany.comwrote:
On Thu, 7 Mar 2013 17:12:47 -0800
Bryan Anderson shadow...@gmail.com wrote:
Has any one ever worked with placing idle display images onto the
Polycom SPIP331 phones?
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
On Thu, 7 Mar 2013 17:12:47 -0800
Bryan Anderson shadow...@gmail.com 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
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)
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
...@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 No Answer is working fine when enabled. I
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 just
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.com
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
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,
- 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)
updates
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
[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 550 handsets
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
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
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 to dial
On Thu, 14 Jun 2012 00:46:25 +
Klaverstyn, David C david.klavers...@intergraph.com 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
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
On Tue, Jun 12, 2012 at 4:15 PM, Jon Caum jon.c...@xpedeus.com 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
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
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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To: 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
...@lists.digium.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
[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 have this in 1.4 and it
works
] 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 have this in 1.4 and it
works fine:
SIPAddHeader(Alert-Info: intercom
Thanks David. I will check it out.
-Original message-
From: Klaverstyn, David C david.klavers...@intergraph.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Mon, Feb 13, 2012 04:34:30 GMT+00:00
Subject: Re: [asterisk-users] Polycom
-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.com [mailto:asterisk-users-
boun...@lists.digium.com
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
@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 downgrade and make it
work, upgrading kills it again
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
: 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
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 l...@net-wall.com wrote:
Hi,
I just moved many Polycom phones from
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
On Fri, Feb 10, 2012 at 10:30 PM, Mike l...@net-wall.com 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
: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 l...@net-wall.com wrote:
Hi,
I just moved many Polycom phones from firmware v3 to 4.0.1b. Anto-Answer
simply stopped
-Commercial Discussion
Subject: Re: [asterisk-users] Polycom firmware 4.0.1 and paging
On Fri, Feb 10, 2012 at 10:30 PM, Mike l...@net-wall.com 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
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-users-boun
: [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-users-boun
I have a Polycom Soundpoint IP335.
There are no inbound routes set to the phones yet.
However, the phones are getting phantom rings.
What is the legitimacy of these calls?
Is there something I need to block to stop it?
I believe its people trying to hack the phones/phone
...@lists.digium.com] On Behalf Of eherr
Sent: Friday, November 18, 2011 8:34 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] Polycom Phantom Ringing
I have a Polycom Soundpoint IP335.
There are no inbound routes set to the phones yet.
However
.X.Z.Z
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of eherr
Sent: Friday, November 18, 2011 8:34 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] Polycom Phantom Ringing
I have a Polycom
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom Phantom Ringing
Nick.
On Fri, Nov 18, 2011 at 9:38 AM, Danny Nicholas da...@debsinc.com wrote:
If your phones are being hacked you have a firewall problem. Your phones
should only be registering
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Polycom Phantom Ringing
Well this is a remote site.
I am running 1.4.26
I have multiple polycoms that do not experience this.
They are getting dhcp from their local router.
I am wondering if it could
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