Search the archives - we had a big discussion about this phone about six
months ago. If you make it work and want another one "I will give you
special price!".
j
On Tue, 14 Jul 2009, Cesar Gonzalez wrote:
> Has anyone played with this phone? i cant seem to get it to work
> properly, i manged
Has anyone played with this phone? i cant seem to get it to work
properly, i manged to get it registered and can make calls from it, but
i havent been able to make it receive calls. Weird thing its that if you
make a call from it and while you are on that call you dial its number
does calls go
'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Polycom Stop Downloading Config
But It still needs to hit the server to see that at some point. I just want
it to stop pulling config totally, unless I tell it to. It is web based, so
I would think th
ssage-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Wednesday, June 17, 2009 10:46 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Polycom Stop Downloading Co
Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] Polycom Stop Downloading Config
Does anybody know of a way to tell the Polycom phones to stop trying to
download their config? We have some setup for tftp and some for ftp and if
they cannot reach the server, they just keep
Does anybody know of a way to tell the Polycom phones to stop trying to
download their config? We have some setup for tftp and some for ftp and if
they cannot reach the server, they just keep rebooting over and over and
over and never stop. I would think it should try once or twice and stop,
but
On 2009-06-15 at 17:04, Dave Fullerton
(dfullertaster...@shorelinecontainer.com) wrote:
> Try changing reg.1.address to "hft0". My hunch is asterisk is looking at
> the from of "6193644...@jtsd05" and going "huh? I don't know a
> 6193644...@jtsd05".
That makes sense and it fixed it. Thanks!
Jim Gottlieb wrote:
> I'm evaluating using Polycom phones for our call center and I've set
> up my first phone (a SoundPoint 560) to give it a try.
>
> The phone is working and can successfully place and receive calls.
> But every minute, there's an error in the log file:
>
> chan_sip.c: Reg
On 2009-06-15 at 19:12, Jeff LaCoursiere (j...@jeff.net) wrote:
> > chan_sip.c: Registration from '' failed
> > for '192.168.200.99' - Username/auth name mismatch
>
> I am a bit confused as to the names and addresses involved here. Which
> name/address is the server, and which is the phone?
T
: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom registration errors
On Mon, 15 Jun 2009, Jim Gottlieb wrote:
> On 2009-06-14 at 00:19, Jeff LaCoursiere (j...@jeff.net) wrote:
>
>>> [hft0]
>>> type=friend
>>> username=hft
On Mon, 15 Jun 2009, Jim Gottlieb wrote:
> On 2009-06-14 at 00:19, Jeff LaCoursiere (j...@jeff.net) wrote:
>
>>> [hft0]
>>> type=friend
>>> username=hft0
>>> secret=mysecret
>>> context=outtrunk-office
>>> host=192.168.200.99
>>
>> Change the above to host=dynamic
>
> I just did this and did a 'r
On 2009-06-14 at 00:19, Jeff LaCoursiere (j...@jeff.net) wrote:
> > [hft0]
> > type=friend
> > username=hft0
> > secret=mysecret
> > context=outtrunk-office
> > host=192.168.200.99
>
> Change the above to host=dynamic
I just did this and did a 'reload'.
> >reg.1.server.1.address="jtsd05"
>
On Sat, 13 Jun 2009, Jim Gottlieb wrote:
> I'm evaluating using Polycom phones for our call center and I've set
> up my first phone (a SoundPoint 560) to give it a try.
>
> The phone is working and can successfully place and receive calls.
> But every minute, there's an error in the log file:
>
>
I'm evaluating using Polycom phones for our call center and I've set
up my first phone (a SoundPoint 560) to give it a try.
The phone is working and can successfully place and receive calls.
But every minute, there's an error in the log file:
chan_sip.c: Registration from '' failed for
'19
I'm working on replacing a SoundPoint 600 with a 650. I need to merge
these two sets of digitmaps in the polycom sip.cfg file, because the 650
locks up when I try to use the digitmap from the 600. I've included the
default digitmap from a 3.1.3 RevB polycom release.
I'd like to merge these two
A client of mine asked about a Polycom IP321..anyone else heard about it?
-Matt
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mai
> Yes with EFK in the latest firmwares you are able to change the on
> screen button layout. I used it to bring a Do Not Disturb button to
> the main screen of the SoundPoint IP330's. I may just be dense but
> paired with the Administrator and Developer guides from Polycom it was
> still rather fru
> I wish Polycom would hire someone with ergonomics skills. The whole
> menu system is the most painful ever designed outside entry-level
> phones. Polycom is an acknowledged leader in sound quality and robust
> hardware but their idea of a menu sucks rocks and always has. Most of
> their menus req
On 05/21/2009 09:11 AM, Barry L. Kline wrote:
> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
>
> Robin Rodriguez wrote:
>
>> still rather frustrating getting the EFK working. If needed I could
>> post that portion of sip.cfg to get you started.
>
> Please do! Just having the example could be he
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Robin Rodriguez wrote:
> still rather frustrating getting the EFK working. If needed I could
> post that portion of sip.cfg to get you started.
Please do! Just having the example could be helpful for those of us
preparing to tackle this kind of pr
st firmware revision 3.1.x and
> newer.
> -Karl
>
>
>
>
> - Original Message -
> From: "Matt Darnell"
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>
> Sent: Thursday, May 21, 2009 3:04 AM
> Subject: [asterisk-users] Po
From: "Matt Darnell"
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Thursday, May 21, 2009 3:04 AM
Subject: [asterisk-users] Polycom Productivity Suite
> Has anyone been able to do the following:
>
> 1. Set the phone to automatically record all
On Thu, May 21, 2009 at 10:04 AM, Matt Darnell wrote:
> 1. Set the phone to automatically record all calls to the USB stick,
> now you have to press three keys.
Not possible AFAIK.
> 2. Put Record on the main screen when a call is active. This would
> eliminate having to press the 'more' softkey.
Has anyone been able to do the following:
1. Set the phone to automatically record all calls to the USB stick,
now you have to press three keys.
2. Put Record on the main screen when a call is active. This would
eliminate having to press the 'more' softkey.
Thanks,
Matt
I have a polycom soundpoint ip 501 phone I was having problems getting
some of the functionality working like the **number to pick up calls to
other phones
then it just started working (when it couldn't update itself for a while)
Unfortunately then I got that issue sorted and it updated to the la
Hello,
I have so far Polycom 501 and 403 which displays the label of a key and
the name of a buddy near its key. Today I received new 330's and they do not
display the name, only "1" & "2". Besides that they work correctly. Anyone
has an idea, or is it a known "feature"?
Justin Phelps wrote:
> dialplan.digit
> map="[2-9]11T|[*]xxT|891xxxT|[1-7]xxxT|8[0-46-8]xxT|8500T|851xxxT|9,1[2-9]xT|9,xxxT"
>
> dialplan.digitmap.timeOut="3|3|3|3|3|3|3|3|3"/>
>
> Do the above changes look in line with common practice JohnM?
Short Answer:
They do.
Longer answer,
Y
Do the above changes look in line with common practice JohnM?
--
Justin Phelps
www.onitato.com
850.866.6864
Date: Thu, 07 May 2009 08:57:13 -0400
From: John Millican
Subject: Re: [asterisk-users] Polycom Dialplan Digitmaps
To: Asterisk Users Mailing List - Non-Commercial Discussion
Justin Phelps wrote:
> I'm replacing a SoundPoint IP 600 with a SoundPoint IP 650.
>
> I attempted to simply reuse the existing config files for the old phone
> on the new phone, but the new phone would lock up on the 4th digit when
> attempted to dial out certain numbers. So, I downloaded the n
I'm replacing a SoundPoint IP 600 with a SoundPoint IP 650.
I attempted to simply reuse the existing config files for the old phone
on the new phone, but the new phone would lock up on the 4th digit when
attempted to dial out certain numbers. So, I downloaded the newest
firmware and config temp
.
Michael Graves
mgraves mstvp.com
o(713) 861-4005
c(713) 201-1262
sip:mjgra...@mstvp.onsip.com
skype mjgraves
FWD 54245
> Original Message
> Subject: Re: [asterisk-users] Polycom wideband codecs?
> From: randulo
> Date: Tue, April 21, 2009 1:40 pm
> To: Asterisk Use
On Tue, Apr 21, 2009 at 4:40 PM, Steve Underwood wrote:
> Which Polycom supports G.722.2? I think they are only supporting G.722,
> G.722.1 and G.722.1C right now.
Could someone enlighten me, what is the difference (the result part
that matters, not the spec)?
r
mgra...@mstvp.com wrote:
> Doing a little research before Friday's Voip Users Conference call with
> Dan Behringer.
>
> Are any of the newer Polycom wideband codecs implemented in v1.6?
> Specifically, G.722.1 or G.722.2?
>
Which Polycom supports G.722.2? I think they are only supporting G.722,
mgra...@mstvp.com wrote:
> Doing a little research before Friday's Voip Users Conference call with
> Dan Behringer.
>
> Are any of the newer Polycom wideband codecs implemented in v1.6?
> Specifically, G.722.1 or G.722.2?
Asterisk 1.6 has passthrough/record/playback support for G.722.1
(Siren7) a
Doing a little research before Friday's Voip Users Conference call with
Dan Behringer.
Are any of the newer Polycom wideband codecs implemented in v1.6?
Specifically, G.722.1 or G.722.2?
Thanks,
Michael Graves
mgraves mstvp.com
o(713) 861-4005
c(713) 201-1262
sip:mjgra...@mstvp.onsip.com
skype
Just watned to kick this back out here one more time.
Anyone done this or ever looked into a work-around?
Thanks!
Steve
> I have meetme working with BLF on polycom phones however when
> meetme is not actually being used by anyone the 'status' of meetme
> becomes "idle".
>
> Which the Polycom
On Mar 19, 2009, at 9:05 AM, Ken D'Ambrosio wrote:
> Hey, all. I'm all over MWI, but I gotta say that I think the
> Polycoms go
> a bit over the top. The blinking LED is enough for me; how do I
> disable
> the stuttered dialtone and the audible warble? I've looked through
> the
> config
Hey, all. I'm all over MWI, but I gotta say that I think the Polycoms go
a bit over the top. The blinking LED is enough for me; how do I disable
the stuttered dialtone and the audible warble? I've looked through the
config files, but there are a HELL of a lot of options, and I haven't been
able
I have meetme working with BLF on polycom phones however when
meetme is not actually being used by anyone the 'status' of meetme
becomes "idle".
Which the Polycom phone sees and produces a clock symbol and FLASHING red
LED.
Are there any 'tricks' or work-arounds to change this status to something
On Wed, 2009-02-25 at 11:37 -0500, M Hulber wrote:
> So I'm thinking, would this work if I had a sip_.conf as well as a
> sip_.conf? What the relationship between the LINEs in the
> sip_.cfg and the Reg on the phone? What's the relationship
> between the AUTH and the LINEn_AUTH? Th
Bob,
Ok, that's the route I ended up taking where all lines are the same
user. I put the AUTH an LINEn_AUTH in the phone instead. I wanted to
be able to set up so that each line is a different peer like below:
sip_.cfg:
AUTH = ; secret
LINE1 =
LINE1_PROXY = 1
LINE1_CAL
I agree with the comments on the intended target market for this phone.
In defense of Polycom, if your TFTP server is external you could connect
to a remote access point by setting up WEP/WPA fairly easily from
Starbucks or wherever you are. If it requires web authentication to get
through th
On Wed, 2009-02-25 at 15:13 +, Jeff LaCoursiere wrote:
> Aha! Mind posting that config?
My sip_allusers.cfg looks like this:
CODECS = g711u, g711a
PROXY1_TYPE = Asterisk
PROXY1_ADDR = 192.168.8.1:5060
#PROXY1_KEYPRESS_2833 = enable
PROXY1_KEYPRESS_INFO = disable
PROXY1_HOLD_IP0 = disable
#PR
On Wed, 25 Feb 2009, Bob Pierce wrote:
> Mark,
>
> Are you still having trouble with your 8002? I had a lot of trouble with
> mine initially, but after playing with it for about 8 hours I figured it
> out. Now it works great all around our office. Our NOC technician loves
> it!
>
> There is a pro
On Tue, 24 Feb 2009, Michael Graves wrote:
> It seems to me that based upon your comments you miss the point of the
> product. It's design targets large commercial concerns, school
> campuses, corporate parks, etc...not making free calls from Starbucks.
Completely right. I assumed it was a gene
Mark,
Are you still having trouble with your 8002? I had a lot of trouble with
mine initially, but after playing with it for about 8 hours I figured it
out. Now it works great all around our office. Our NOC technician loves
it!
There is a problem with the sample configs that Polycom publishes. I
It seems to me that based upon your comments you miss the point of the
product. It's design targets large commercial concerns, school
campuses, corporate parks, etc...not making free calls from Starbucks.
I had one under test for several months and it behaved really well on
my WLAN using a Netgear
I have one of these seemingly useless devices too. Please let me know if
you get anywhere with it. I bought it thinking it would be a good phone
to take around to various hotspots and keep my extension. Turns out it
really wants to be only in its home "hotspot" and has some stringent
restri
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
>
>
&
risk-users-boun...@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
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 an
een current 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]
interesting - I haven't noticed this with any of my installs. What
> version of firmware and SIP?
> >
> >
> >
> >
> >
> > From: Barry D. Hassler
> > To: Asterisk Users Mailing List - Non-Commercial Discussion <
> a
Since I have a partial answer for completeness:
When I moved (duplicated) the PROXYn_ADDRESS definition to the user
specific config it then registers with the proxy. It seems like a
defect to me.
I'd still like to see how people have set up different users/lines using
LINEn and LINEn_AUTH. I
I have a new Polycom Spectralink 8002 and am having trouble with the
configuration or the unit but I can't see what's wrong. The unit does
not seem to even attempt to register with the Asterisk proxy but I can
make calls to it. I have viewed the syslog from the device which it
will actually w
half Of Dave Fullerton
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
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
ry 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 after beingup
> a LONG time
>
>
>
> That's interesting - I haven't noticed this with any of m
f Of Danny
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
ne (extension).
_
From: asterisk-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-u
n of firmware and SIP?
>
>
>
>
>
> From: Barry D. Hassler
> To: Asterisk Users Mailing List - Non-Commercial Discussion
>
> Sent: Friday, February 20, 2009 8:41:33 AM
> Subject: [asterisk-users] Polycom Phones start to break up after being up a
> LO
That's interesting - I haven't noticed this with any of my installs. What
version of firmware and SIP?
From: Barry D. Hassler
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Friday, February 20, 2009 8:41:33 AM
Subject: [aste
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 --
Hello!
I am selling hardware I have been using in various Asterisk test
installations.
If you are interested in buying all or any of the following items please
reply off list.
I got a few
* X100P clones
* Sirrix PCI4S0 ISDN BRI cards (+ 1 Sirrix S0 voltage/power supply)
* 1 Polycom Soundpoint
Hold down 2,4,6,8 and * at the same time. This is the 501 reset key
sequence.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Thursday, December 04, 2008 6:24 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> [EMAIL PROTECTED]
> Sent: Thursday, December 04, 2008 7:24 PM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] polycom no menu
>
> Was messing with a pol
Was messing with a polycom 501 and changed the IP from dhcp to static. Working
with a user remotely. Now, the user says the phone does not show anything on
the LCD and does not respond to any buttons.
When rebooting, there is text shown as it proceeds. ??
Is there a way to reset this to a de
At 15:26 11/17/2008, hin lee wrote:
>Anybody know why the volume on calls are so low? How can I increase
>the volume?
>
>
>--- On Sat, 11/15/08, hin lee <[EMAIL PROTECTED]> wrote:
>
>> From: hin lee <[EMAIL PROTECTED]>
>> Subject: Re: [asterisk-u
Anybody know why the volume on calls are so low? How can I increase the volume?
--- On Sat, 11/15/08, hin lee <[EMAIL PROTECTED]> wrote:
> From: hin lee <[EMAIL PROTECTED]>
> Subject: Re: [asterisk-users] Polycom low volume
> To: "Asterisk Users"
> Date: Sat
ile
>http://docs.google.com/Doc?id=dggrkn86_4csdzthf9&hl=en
>
>Server cfg file
>http://docs.google.com/Doc?id=dggrkn86_6gp7hr9fg&hl=en
>
>SIP cfg file
>http://docs.google.com/Doc?id=dggrkn86_3djzb86d7&hl=en
>
>
>
>--- On Sat, 11/15/08, hin lee <[EMAIL PR
<[EMAIL PROTECTED]> wrote:
> From: hin lee <[EMAIL PROTECTED]>
> Subject: Re: [asterisk-users] Polycom low volume
> To: "Doug" <[EMAIL PROTECTED]>, "Asterisk Users"
>
> Date: Saturday, November 15, 2008, 10:40 PM
> Attached, my configuration f
ones. Hope I
>provided enough information.
Why don't you post a link to your sip.cfg?
Typical PhoneXX.cfg?
>
>Thanks!
>Hin
>
>
>--- On Sat, 11/15/08, Darrick Hartman <[EMAIL PROTECTED]> wrote:
>
>> From: Darrick Hartman <[EMAIL PRO
, Darrick Hartman <[EMAIL PROTECTED]> wrote:
> From: Darrick Hartman <[EMAIL PROTECTED]>
> Subject: Re: [asterisk-users] Polycom low volume
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>
> Date: Saturday, November 15, 2008, 1:44 PM
> Ac
Actually, it could be within Asterisk, but only if you have Zaptel
hardware. If you are only using SIP devices, then the problem is with
the phone configuration. You really don't provide enough information to
determine what is causing your problem. How are you provisioning the
phones? What
Probably has nothing to do with Asterisk. You can set the volume and
persistence in the phones config files.
Michael
On Fri, 14 Nov 2008 22:43:45 -0800 (PST), hin lee wrote:
>Using a Polycom 550 and 650 phones on my Asterisk server for testing. I can't
>figure out why the volume is so low. Ho
Using a Polycom 550 and 650 phones on my Asterisk server for testing. I can't
figure out why the volume is so low. How can I adjust the volume control on
Asterisk? It's at max on the handset phones.
Thanks!
Hin
___
-- Bandwidth and Colocat
I found out what the problem was.
It appears to be a bug in the Polycom 430 firmware.
I have 2 lines on the phone and both of them use the same auth id but with
different servers.
It seems that if you make an outgoing call from the phone on line 2 and then
called party hangs up. Asterisk says B
Hi,
I have a really strange problem with a Polycom 430 phone and Asterisk
1.4.20.
Currently If I dial the Polycom from my mobile phone answer the call on the
Polycom and then hangup the mobile the call ends fine on the Polycom.
But if I call from the Polycom to my mobile and then I hang up the mo
check digimap into polycom web interface and check the digmap rules
for your voip system
On Fri, Oct 10, 2008 at 9:07 PM, Ed DeHart <[EMAIL PROTECTED]> wrote:
> I have four Polycom 330 phones connected to an asterisk system. There are
> other VoIP phones connected too. All of the extensions are
rule with prefix, gives second dialtone after
dialing 9
* [1-8]xx: A regular 3 digit extension is dialed immediately ("9" excluded
as a prefix)
--- On Thu, 10/9/08, Ed DeHart <[EMAIL PROTECTED]> wrote:
> From: Ed DeHart <[EMAIL PROTECTED]>
> Subject: [asterisk-
Ed DeHart schrieb:
> I have four Polycom 330 phones connected to an asterisk system. There
> are other VoIP phones connected too. All of the extensions are four
> digits beginning with 11. From any of the phones, except the Polycom,
> picking up the handset to call extension 1103 for examp
You need to investigate the digit map in the phones configuration. This
determines what dialing patterns the phone will accept. See deails
here:
http://sipx-wiki.calivia.com/index.php/Digit_Maps_used_to_Define_the_Dia
l_Plan
Michael
--Original Message Text---
From: Ed DeHart
Date: Thu, 9 Oct 200
I have four Polycom 330 phones connected to an asterisk system. There
are other VoIP phones connected too. All of the extensions are four
digits beginning with 11. From any of the phones, except the Polycom,
picking up the handset to call extension 1103 for example works fine.
With the
Could someone please tell me where to download Polycom 3.1.0RevB?
Polycom.com is not possible. Thanks.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.
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=displayKC&docType=kc&externalId=12856&sliceId=SAL_PUBLIC_1_2&dialogID=7620671&stateId=1
We have set tcpIpApp.port.rtp.mediaPort
We've used the devstate backport with the snom phones for this. The
buttons toggle log in and out with one and pause/unpause with another.
We use the astdb to store current status and add/remove/pause/unpause
queue member functions. Works great
On 9/6/08, James Sneeringer <[EMAIL PROTECTED]> wrote
I have not applied the 1.4 backport to my system, so I haven't used
DEVSTATE, but this page appears to show how to do what you want:
http://www.voip-info.org/wiki/view/Asterisk+func+Devstate
That page also has a link to the backport.
-James
On Thu, Sep 4, 2008 at 11:34 PM, Lee, John (Sydney)
<
Seems that this got it working as suggested in the thread - thank you
all for replies.
I took out the attendant.uri option as you dont need it. It seems to
be that you can set up a buddy watch for one endpoint using this
option - dont know exactly what this option is supposed to be. I was
rea
> Sorry, needed to add one more note. To clarify, my agent phones have a
> speed dial assigned for their login, and another to pause/unpause. I
> could then use DEVSTATE to enable or disable the light next to their
> speed dial button based on their status. I can't use it to update
> anything on th
Sorry, needed to add one more note. To clarify, my agent phones have a
speed dial assigned for their login, and another to pause/unpause. I
could then use DEVSTATE to enable or disable the light next to their
speed dial button based on their status. I can't use it to update
anything on the LCD scre
I believe DEVSTATE() in 1.6 (backported to 1.4 in various places) will
let you arbitrarily control BLF, so you could control it in the
dialplan when an agent logs in or out (or pauses, or whatever).
Separately, you might be able to use sipsak (http://sipsak.org/) to
construct a SIP message that es
> It's not perfect, because it
> doesn't display DND or queue login/pause status, but it's better than
> nothing.
James, on a different note, is it true that at this stage, we can never
get any queue login status/light on Polycom phone?
I posted a query a few days ago but I have got 0 reply.
Any
I'm using a similar feature on 550 and 650 phones, also running 2.2.2.
I've never used the option to do it, though, so I'm not
sure how it differs from what I'm doing. Instead, on the phones that
are allowed to do this, I have the following in their XML config. You
could just as easily enable it i
> I believe that this is what I need to enable more than one buddy icon?
> Can you please point me in the right direction. Only the polycom
> screen, I can only see 1 buddy icon despite having 2 speed dial
> entries.
>
I have been able to successfully turned on "presence" (which is the term
use
Hi,
Can anyone please comment on what the issue may be with this. I am
trying to set up an Polycom IP601 with multiple buddy icons displaying
endpoint status.
I am using a polycom IP601, sip 2.2.2.0084
In the phone1.cfg file I set:
Using this, I get a valid SUBSCRIBE and NOTIFY from the s
Adam,
We have the exact same issue occurring on one of our networks. I haven't had
time to dive into it much, but here is what I've found out:
Calling from a softphone via IAX2 or SIP, transfers work fine.
Calling from a Polycom 501 outside the network, transfers work fine.
However, the Polycom I
Thanks for responding Kate.
I do have a transfer button on the phone, and I follow the transfer
process as described in the user's guide. When I press "transfer" the
first caller is placed on hold and then I call the party I want to
transfer to. At this point I'm supposed to press "transfer"
I think it should work standard (i.e. no special setup) Do you have a
transfer button on the phone?
Kate
Adam Moffett wrote:
> I can't transfer calls with my polycom 501's. Do I need to set up
> something in particular in the asterisk dialplan to make the feature work?
>
>
>
>
I can't transfer calls with my polycom 501's. Do I need to set up
something in particular in the asterisk dialplan to make the feature work?
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25
1:20 PM
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> Subject: [asterisk-users] polycom with http/https basic authentication
>>
>> Hi,
>>
>> I apologize that this is not directly associated with Asterisk, I have
>> been trying to solve this
301 - 400 of 2690 matches
Mail list logo
