...@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
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
. What
version of firmware and SIP?
From: Barry D. Hassler barry.hass...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, February 20, 2009 8:41:33 AM
Subject: [asterisk-users
for about a
year.
Mike
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Barry D.
Hassler
Sent: Tuesday, February 24, 2009 15:38
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom Phones
PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Polycom Phones start to break up after being
up a LONG time
That`s an old version, I've had plenty of issues (nothing like what you
describe) since 2.2.0. Try going to the latest and greatest (SIP
...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Tuesday, February 24, 2009 15:58
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Polycom Phones start to break up after being
up a LONG time
This is probably out
, 2009 15:58
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Polycom Phones start to break up after being
up a LONG time
This is probably out of date, but my Polycom documentation recommends
keeping the boot rom at less than 2.0 for use with Asterisk
Has anyone else encountered this? I have a fairly large installation (~50
phones, almost all Polycom 501's and a handful of 601's. We're running into
a number of phones on which the outbound voice (Polycom phone user doesn't
hear any problems, but the other end does) is breaking up occasionally --
, February 20, 2009 8:41:33 AM
Subject: [asterisk-users] Polycom Phones start to break up after being up a
LONG time
Has anyone else encountered this? I have a fairly large installation (~50
phones, almost all Polycom 501's and a handful of 601's. We're running into a
number of phones on which
?
From: Barry D. Hassler barry.hass...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, February 20, 2009 8:41:33 AM
Subject: [asterisk-users] Polycom Phones start to break up after being up a
LONG time
Has
-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Asterisk
Asterisk
Sent: Friday, February 20, 2009 11:25 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: nt_jnew...@yahoo.com
Subject: Re: [asterisk-users] Polycom Phones start to break up
Nicholas
Sent: Friday, February 20, 2009 12:45 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Polycom Phones start to break up after
beingupa LONG time
This is just a hack, but why don't you schedule a sip notify
polycom-restart during lunch hour
: [asterisk-users] Polycom Phones start to break up after beingup
a LONG time
That's interesting - I haven't noticed this with any of my installs. What
version of firmware and SIP?
_
From: Barry D. Hassler barry.hass...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial
Jeff LaCoursiere wrote:
On Fri, 20 Feb 2009, Danny Nicholas wrote:
This is just a hack, but why don't you schedule a sip notify
polycom-restart during lunch hour? You could run it from a cron job
using this line for each phone:
Asterisk -rx sip notify polycom-check-cfg 100 replacing 100
Sent: Friday, February 20, 2009 12:44 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom Phones start to break up after beingup
a LONG time
Jeff LaCoursiere wrote:
On Fri, 20 Feb 2009, Danny Nicholas wrote:
This is just a hack, but why don't you
Just in case anyone is having DNS SRV timeouts with their Polycom
phones, the following Polycom KB article should help:
http://knowledgebase.polycom.com/kb/search.do?cmd=displayKCdocType=kcexternalId=12856sliceId=SAL_PUBLIC_1_2dialogID=7620671stateId=1
We have set
Hello,
A few days ago I've posted two questions about Polycom phones: How to access
corporate phone directory from the phone and how to use a conference server
with it. After I got zero responses I tried openning a support call in
Polycom's site. Here are the replies I got from them:
-
I want to enable on hold reminder function on polycom 430 phones. I have
enabled it in sip.cfg
using this setting
hold
localReminder call.hold.localReminder.enabled=1
call.hold.localReminder.period=60
call.hold.localReminder.startDelay=90/
/hold
But still if the call is on
Can anyone tell me which config file tells the phone what file to load
as bootrom.ld?
Or is this hardcoded in the phone? I just got a IP501 but I have a
bunch of IP500s...
Will the bootrom (2.6.2) work OK with both the IP500 and 501?
Thanks!
Wiley E. Siler
Director of Information
Hi Wiley -
Can anyone tell me which config file tells the phone what file to load as
bootrom.ld?
Or is this hardcoded in the phone?
Yup, it's hardcoded. I believe this is the way it works: If there's a
bootrom.ld on your configuration server, and it is newer than the one
on the phone, the
On Mon, 2006-01-23 at 16:26 -0500, Bill Gibbs wrote:
I know the Polycoms work with NAT, but you have to specify the public
IP.
No you don't, at least, I never have, and it works perfectly for me
every time I have a client who regularly moves their polycom 501
from home - work and back
I know the Polycoms work with NAT, but you have to specify
the public IP.
Is there anyway for it to discover the external IP
automatically?
I like the phones (been playing with a 301) but for some of our
clients who have a dynamic IP (and no hope of getting a static ie cable or
Where do you have to set the public IP? We use dhcp Poly behind
firewalls daily. Just set nat=yes in sip.conf
On Jan 23, 2006, at 3:26 PM, Bill Gibbs wrote:
I know the Polycoms work with NAT, but you have to specify the
public IP.
Is there anyway for it to discover the external IP
I have PolyCom phones in one office working perfectly, but in another
office with a new subnet, new server, new everything, the time does not
work. Everything else about the phones seems fine, but the time. If you
look at the internal webpage in the phone, it shows clock. Our
server, which is
Cc:
Subject: [Asterisk-Users] PolyCom phones with blinking clock and wrong
time
I have PolyCom phones in one office working perfectly, but in another
office with a new subnet, new server, new everything, the time does not
work. Everything
:01 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Polycom Phones
Uhmm. Well, he may just be using the typical asterisk configuration of
just editing the .conf files rather than using AMP or [EMAIL PROTECTED]
For the first issue, that kind of sounds like
Just got in a bunch of polycom phones for use on my shiny new asterisk box, but
found 2 small issues I was wandering if someone could help me with.
First, though the phones support 2 call appearances, if I am on a call, the
second call does not ring through -- it goes to voicemail instead of
Chris Gamble wrote:
Just got in a bunch of polycom phones for use on my shiny new
asterisk box, but found 2 small issues I was wandering if someone
could help me with.
Are you using AMP or Asterisk @ Home?
First, though the phones support 2 call appearances, if I am on a
call, the second
Uhmm. Well, he may just be using the typical asterisk configuration of
just editing the .conf files rather than using AMP or [EMAIL PROTECTED]
For the first issue, that kind of sounds like a problem with the
polycom configuration. I don't have my pdf of the polycom config in it
with me right now,
You are attempting to put the second registration on the backup
server for the first registration. Move farthur down the file and
setit up on the second line. Works great. My IP600 on my desk has 3
different registrations right now to different servers. Great debug
tool.
On Aug 2, 2005,
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Polycom phones w/ two lines on different
servers
I do it all the time, the phone on my desk has four server
registrations.
Don't use OVERRIDE or web configurations, do it this way:
?xml version=1.0 standalone=yes
I use the default. Try this.
cd /home/PlcmSpIP
cat log/YOURMAC-boot.log
se what the log file says, also do the same with the YOURMAC-app.log
--
Chris Mason
NetConcepts
(264) 497-5670 Fax: (264) 497-8463
Int: (305) 704-7249 Fax: (815)301-9759
Cell: 264-235-5670
Yahoo IM: [EMAIL PROTECTED]
List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Polycom phones w/ two lines on different
servers
I use the default. Try this.
cd /home/PlcmSpIP
cat log/YOURMAC-boot.log
se what the log file says, also do the same with the YOURMAC-app.log
--
Chris Mason
NetConcepts
(264) 497-5670
Of Adam Robins
Sent: Thursday, August 04, 2005 2:41 PM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: RE: [Asterisk-Users] Polycom phones w/ two lines on different
servers
This is in the -app.log file:
0804194926|sip |4|00|Registration failed User
settings?
Thanks,
Adam
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris
Mason (Lists)
Sent: Tuesday, August 02, 2005 3:44 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Polycom phones w/ two lines
:55 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Polycom phones w/ two lines on different servers
My Polycom 300 is registered on two different servers on two different subnets.
It was failing the same way for me as well because we had server
Slightly off topic as this doesn't pertain directly to Asterisk, but
with the Polycom 500/501 phones, does anyone know how to correctly put a
custom logo for the idle screen on the device? I've read the Admin Guide
through and through and the information there is not enough to implement
it.
Any
Information for configuring idle images on the polycom phones is now
available on the wiki at
http://www.voip-info.org/tiki-index.php?page=Polycom+Phones
Regards,
Derek
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
Slightly off topic as this doesn't pertain directly to
Asterisk, but with the Polycom 500/501 phones, does anyone
know how to correctly put a custom logo for the idle screen
on the device? I've read the Admin Guide through and through
and the information there is not enough to implement
Much Appreciated! Thanks!
On Wed, 2005-08-03 at 15:15 -0600, dbruce wrote:
Information for configuring idle images on the polycom phones is now
available on the wiki at
http://www.voip-info.org/tiki-index.php?page=Polycom+Phones
Regards,
Derek
Hi all -
This isn't really directly Asterisk related, but has anyone successfully
set up a Polycom phone to register two lines on two different Asterisk
boxes? I can get the first line to register, but the second one does not.
I can still place calls from that second line, which indicates to me
[EMAIL PROTECTED] wrote:
Hi all -
This isn't really directly Asterisk related, but has anyone successfully
set up a Polycom phone to register two lines on two different Asterisk
boxes? I can get the first line to register, but the second one does not.
I can still place calls from that second
I do it all the time, the phone on my desk has four server registrations.
Don't use OVERRIDE or web configurations, do it this way:
?xml version=1.0 standalone=yes?
PHONE_CONFIG
phone1
preferences voice.codecPref.G711Mu=1 voice.codecPref.G711A=
voice.codecPref.G729AB=2 /
cfg
appearances per CO key, rather than the default (I
think it was 8?)
Worked like a champ!
Chris Coulthurst
[EMAIL PROTECTED]
- Original Message -
From: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Tuesday, August 02, 2005 9:05 AM
Subject: [Asterisk-Users] Polycom phones w
Has anybody tried to use a Polycom phone (I have 500s and 600s) with a
netmask shorter than /24? (A network bigger than 255.255.255.0). We've
run out of IPs in our initial /24 network, and I'd like to expand it to
255.255.248.0.
When I set it to 255.255.248.0 I can ping the phone while the
9:25 AM
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: [Asterisk-Users] Polycom Phones shorter than /24 netmasks
|
|Has anybody tried to use a Polycom phone (I have 500s and 600s) with a
|netmask shorter than /24? (A network bigger than 255.255.255.0). We've
|run out of IPs
, 2005 12:25 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Polycom Phones shorter than /24 netmasks
Has anybody tried to use a Polycom phone (I have 500s and 600s) with a
netmask shorter than /24? (A network bigger than 255.255.255.0). We've
run out of IPs
]
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Charlie Watts
Sent: Tuesday, June 07, 2005 9:25 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Polycom Phones shorter than /24 netmasks
Has
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of C F
Sent: Monday, May 30, 2005 11:02 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Polycom phones, UNREACHABLE
I now have the problem solved (or so I think) even
I now have the problem solved (or so I think) even without the qualify
in sip.conf.
It happens to be that the problem was just that the firewall would not
allow the packets back in after a specific time. All I had to do was
create trigger rules on the firewall to allow the packets back in
based on
?
- Original Message -
From: Michael George [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Friday, May 27, 2005 11:26 PM
Subject: [Asterisk-Users] Polycom phones, UNREACHABLE
I'm having some trouble with Polycom Soundpoint phones. I have had good
luck
deploying them
Michael George wrote:
On Sat, May 28, 2005 at 11:10:30AM -0400, Steve Totaro wrote:
qualify = yes is what is causing the messages. You can assign a value
rather than yes. like 1000 or something or you can remove the qualify
statement alltogether. The message is just a warning. Eliminating
I'm having the same issues with the polycom phones, as well as with
Sipura ata's. I am also using on another natted network a sipura ata,
that I changed the settings on the sipura that might help, and it did
help, I havn't had an unreachable message since.
I'm not sure if on the second network the
Actually, it looks like I'm getting this problem on all my phones. When I was
testing my phones, most worked pretty well with an occasional complaint from
the Polycom.
I've moved them now to a different location and the ISP must have different
NAT translation going on that make it more difficult
: Michael George [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Friday, May 27, 2005 11:26 PM
Subject: [Asterisk-Users] Polycom phones, UNREACHABLE
I'm having some trouble with Polycom Soundpoint phones. I have had good
luck
deploying them on a local network, but now I've tried
I'm having some trouble with Polycom Soundpoint phones. I have had good luck
deploying them on a local network, but now I've tried putting some in place
which access their * server across the network.
The * server is on a public IP and the polycoms are behind a NAT on a cable
modem broadband
This this may sound ridiculous, but we've had problems with this when the
users did not plug the handset cord in completely. 8 out of our 12 employees
made the mistake, as the plug on the IPX00's appears to be all the way in
when it is actually not.
Probably not the cause of your problems,
Hi,
Has anyone experienced any problem with polycom phones not sending voice?
Occasionally some phones didn't send voice, a few times phone didn't play
the incoming voice. It didn't happen often on the IP 500 I bought last
November. But it happened quite a lot on the IP 600 I got recently.
Jason Brown wrote:
| Anyone have experiece with polycom phones?
|
| I am experiencing a really weird problem. In an office where I have
| the following extensions:
| On the Polycom phones, when I want to dial from extension
100 to any
| extension 120 or above, or dial out, it dials
Anyone have experiece with polycom phones?
I am experiencing a really weird problem. In an
office where I have the following extensions:
100
101
102
103
104
110
111
120
130
140
141
150
200
On the Polycom phones, when I want to dial from
extension 100 to any extension 120 or above, or dial
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Jason Brown wrote:
| Anyone have experiece with polycom phones?
|
| I am experiencing a really weird problem. In an office where I have the
| following extensions:
| On the Polycom phones, when I want to dial from extension 100 to any
| extension 120
We have bought PBXware GUI from Bicom systems and configured extensions
with Polycom Phones as UAs.
The Polycom Phones can dial out and make calls but I cannot make
extension to extension calling.
Googling did not help much.
As you may be aware PBXware is a closed source software GUI from Bicom
Never used pbxware, but the context the sip phones dial out using specified in sip.conf needs to include the dialplan context of the phones in extensions.conf.
On Thu, 2005-03-10 at 15:08, Kanuri, Seshu (Company IT) wrote:
We have bought PBXware GUI from Bicom systems and configured
Never used pbxware, but the context the sip
phones dial out using
specified
in sip.conf needs to include the dialplan context of
the phones in extensions.conf.
PBXware does not
use the Conf files, but it uses its own database and encryption schemes to
configure users from a database and dial
Kanuri, Seshu (Company IT) wrote:
Never used pbxware, but the context the sip phones dial out using
specified in sip.conf needs to include the dialplan context of
the phones in extensions.conf.
As mentioned above check the context for phones... :)
PBXware does not use the Conf files, but
] On Behalf Of Kanuri, Seshu
(Company IT)
Sent: Thursday, March 10, 2005 4:09 PM
To: Asterisk-Users@lists.digium.com
Subject: [Asterisk-Users] Polycom phones do not talk to each other andcannot
answer when we pickup
We have bought PBXware GUI from Bicom systems and configured extensions
with Polycom
[EMAIL PROTECTED] wrote:
You need to call PBXware it should not have anything to do with the
phone. If a phone registers there gui should put it in the correct
context. If it does not then get your money back due to there non
standard setup
Ariel.. for your information Seshu was told BEFORE
I have a couple of Polycom phones, bootrom 2.5.0, SIP 1.3.1.0056. Works
great with Asterisk when I power on the phone. However, after some time,
say an hour, I cannot receive calls on this phone. On Asterisk, when I
do database show, it does show the phone in there, but it cannot reach
the
In almost all my calls now, I am getting beeps and loud and soft parts
of a conversation. It is getting very irritating. Has anyone had this
happen? How do I get rid of it?
Thanks
Sean Garland, MCP+I, A+
Siskiyou Technology Consultants
205 N. Mt. Shasta Blvd. Suite 100
Mt. Shasta, CA 96067
Sean Garland wrote:
In almost all my calls now, I am getting beeps and loud and soft parts
of a conversation. It is getting very irritating. Has anyone had this
happen? How do I get rid of it?
Set relaxdtmf=no in /etc/asterisk/zapata.conf.
___
On Thu, 2003-12-18 at 17:09, Juan J. Sierralta P. wrote:
Does anybody know if Polycoms has a three finger salute as Cisco 79XX
does ? I really hate to unplug ethernet cable since you have to release
the stand first.
I respond myself, hold down: Volume+, Volume-, Hold and
Hello,
We have updated the Wiki page for Polycom phones:
http://www.voip-info.org/tiki-index.php?page=Polycom+Phones
We posted several configuration specs as well as a link to an admin guide
for the phone.
We also posted a link on there to two firmware versions for download.
The official
On Thu, 2003-12-18 at 13:30, mattf wrote:
Hello,
We have updated the Wiki page for Polycom phones:
http://www.voip-info.org/tiki-index.php?page=Polycom+Phones
We posted several configuration specs as well as a link to an admin guide
for the phone.
We also posted a link on there to two
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