Hi all,
I'd like to better understand how Phone Groups are actually used. It
seems to me that Phone Groups can do two things:
1. "Filter by..." when listing Phones. i.e. If you frequently "Send
Profiles" or "Restart" certain sub-groups of phones, then you can group
them and save a little tim
Hi all,
Regarding the "Polycom Firmware 3.2.1" thread we had going back in
October of last year...
1. I've finally raised a JIRA to cover the enhancement detailed in the
http://list.sipfoundry.org/archive/sipx-users/msg17622.html post.
http://track.sipfoundry.org/browse/XX-7401 : Activate mu
Scott wrote:
...
> I propose that as one of our New Years Resolutions for 2010
> we resolve to improve our project documentation dramatically.
+1
> I think that using Confluence offers us some real advantages
> toward that goal:
>
> * Generally I think Confluence is a nicer wiki tool than
Josh wrote:
> Are there plans to add configuration support for the
> SoundPoint IP 335 to 4.2? The only configuration differences
> between the 335 and 330/331 are that the 335 supports HD
> Voice/G.722 and uses an RJ-9 headset port.
> The nicest thing about this new model, though, is the
> ad
WITCH said "good luck with that" in
fs_cli whenever sipX tried it). I posted my frustration with it on this
list some time within the last month.
Paul Mossman wrote:
http://sipx-wiki.calivia.com/index.php/Using_Sangom
http://sipx-wiki.calivia.com/index.php/Using_Sangoma_Telephony_Cards_wit
h_sipXecs
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Josh Patten
Sent: November 25, 2009 4:07 PM
To
Scott wrote:
> On Mon, 2009-11-09 at 16:08 -0800, Ashwin Kumar wrote:
> > Hi,
> > I was able to fix the issue by re initializing the database after
> > setting the encoding. Its working fine now.
>
> Can you please summarize what you did in detail? Someone
> else is sure to hit this
Also c
> Does the EDE provide me the GUI, that is the main reason I
> want to install sipX...and I want the option to keep modify
> the code too.
The EDE has some Eclipse "readiness" features:
http://sipx-wiki.calivia.com/index.php/Express_Development_Environment_S
etup#Eclipse_readiness
You will nee
> The express_dev_setup looks to be for Fedora 10...I have a
> CentOS 5 machine... Will they work..
Yes, CentOS 5.2 and 5.3 should both work.
> Like the wget is specific to Fedora.
>
> wget
> http://sipxecs.sipfoundry.org/rep/sipXecs/main/extras/ede/ede_
> fedora_staticip_root.sh
> chmod +x
Hi Ujjval,
Not that it's going to salvage your current development environment, but
there's a much easier and predictible way to start coding with sipXecs:
http://sipx-wiki.calivia.com/index.php/Express_Development_Environment_S
etup
-Paul
paul.moss...@nortel.com
_
This will be resolved with Polycom 3.2.2.
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Joly,
Robert (CAR:9D30)
Sent: November 4, 2009 12:45 PM
To: jermaine pinder; Lawrence, Scot
The Polycom KWS 300 and KWS 6000 are both known to work well with sipXecs.
("Registered Product" status for Nortel SCS -
http://www.nortel.com/prd/dpp/product/scs.html#P)
-Paul
paul.moss...@nortel.com
> -Original Message-
> From: sipx-users-boun...@list.sipfoundry.org
> [mailto:
Mike wrote:
> I figured that's how we'd have to do it... I just wasn't
> sure if the dev team was hatching a scheme for multiple
> versions of Polycom firmware being active on the PBX.
Scott and I have talked about how to do it, but there are currently no
plans.
I can open a JIRA, but let's fi
Mike wrote:
> If that's true, how will we deal with a mixed environment of phones?
Legacy phones will simply not download 3.2.X firmware. This is Polycom
BootROM behaviour.
On the sipXecs side, the config files generated for these models are
compliant with Polycom 3.1.3RevC. (The latest 3.1.X.)
Josh wrote:
> It appears Polycom Firmware 3.2.1 is now available to the
> general public. Are there currently plans to get sipXconfig
> outfitted with the newest config options in sipX 4.2 or will
> that have to wait until a later release?
Done. (I'm hoping to ultimately support Polycom 3.2
tos/RPM-GPG-KEY-CentOS-5
[sipxecs-stable]
name=SIPfoundry sipXecs pbx - latest stable version
baseurl=http://sipxecs.sipfoundry.org/pub/sipXecs/LatestStable/CentOS/5/
$basearch/RPM
gpgcheck=0
On Oct 15, 2009, at 8:20 AM, Paul Mossman wrote:
&g
9 PM, William Otten
wrote:
Paul
The only repo I have is the sipxecs repo under
/etc/yum.repos.d/sipxecs.repo and nothing elsethere are no other
extras, addons, etc....
Paul Mossman wrote:
The yum localinstall
To: Paul Mossman
Cc: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] How to change
sendmail smtp server
On
Hi William,
Error 404 when using yum? Sounds like you've hit XX-6529:
sipxecs-release RPM installs sipxecs.repo file that points at the wrong
repository.
The JIRA now has work-around instructions which could probably fix the
problem. (They assume you've installed from the 4.0.2 ISO, which run
Scott wrote:
...
> The sipXpbx RPM requires 'bind'
Not in 4.0.2.
That dependency is only in 4.2. (See XX-6101.)
-Paul
paul.moss...@nortel.com
___
sipx-users mailing list sipx-users@list.sipfoundry.org
List Archive: http://list.sipfoundry.org/archi
Hi all,
The high-end Polycom models (VVX 1500 and SoundStation IP 6000/7000) have an
overwhelmingly large number of codec options. Many are simply bit rate
variations of the same underlying codec. sipXecs however exposes all options,
thus cluttering the codec selection table.
I recently noti
Hi all,
FYI, there is a discussion regarding automated phone provisioning on
sipx-dev that may interest you:
http://list.sipfoundry.org/archive/sipx-dev/msg19481.html
Please feel free to join in with your questions and feedback. Thanks.
-Paul
paul.moss...@nortel.com
Hi Matt,
The VVX profile in complete in 4.2, but not in 4.0.1.
I've used these phones, they work well. They use the same firmware as
the SoundPoint IP phones. The interface is identical, except that the
touch screen replaces many buttons.
If you're familiar with the 670, then you should f
Hi Arda,
Have you seen this article?
http://sipx-wiki.calivia.com/index.php/HowTo_configure_AudioCodes_Stand-
Alone_Survivability_Feature
-Paul
paul.moss...@nortel.com
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@l
Ranga wrote:
> On Fri, Jul 31, 2009 at 5:02 PM, Yakout
> Esmat wrote:
> > As Paul mentioned. You would want to consume the least amount of
> > bandwidth per call over your choked up DSL link, while on
> the LAN you
> > want to use the better quality G711 CODEC.
> >
> > The way Cisco does it i
Ranga wrote:
...
> > Can we set these up in a way that forces the phones to use alaw
> > (g711) between each other over the LAN and use G729 over
> the SIP trunk
> > (SIP trunk over DSL) to provider?
>
>
> No you cannot do that. May I ask you why you want to do that?
> We used to have code
Hi all,
Here's the root article on Sangoma cards, which has links to both the
FreeSWITCH and NetBorder instructions.
http://sipx-wiki.calivia.com/index.php/Using_Sangoma_Telephony_Cards_wit
h_sipXecs
And yes, still a work in progress at this point.
-Paul
paul.moss...@nortel.com
___
Keith wrote:
> Am I right in assuming that a standard ISO installation of
> 4.0.0 would be set up to use the stable repos?
Unfortunately no.
That won't be available until 4.0.2, with XX-5638.
But the repo file from XX-5638 is attached, and should do the trick.
-Paul
paul.moss...@nortel.com
Damian wrote:
...
> The PlcmSpIp password needs to match the default password for
> Polycom phones. Otherwise you'd have to manually reconfigure
> every phone before it could retrieve its configuration.
>
> It's far from perfect: Paul is working on HTTPs provisioning
> and Scott is advocating a
Hi Charles,
The change to blank by default was introduced in 3.10.3 - see XX-3140
and http://list.sipfoundry.org/archive/sipx-dev/msg12014.html
If the default values worked for you in a pre-3.10.3 system, then
re-programming them on your 4.0 system should give you the same
behaviour you had be
Tony wrote:
> I am able to use the system without issue except I cannot
> upload Polycom soundpoint files. I created a 'deactivated'
> placeholder for my files, then tried to upload the bootrom
> (smallest file) through sipxconfig. i was unable to do so
> (not getting any progress bar in fire
.
> I am going to rebuild the users that I had on this box and
> see what happens Thanks Chris
>
> -Original Message-
> From: Paul Mossman [mailto:paul.moss...@nortel.com]
> Sent: Monday, May 04, 2009 1:03 PM
> To: McCoy, Chris; sipx-users@list.sipfoundry.org
ror has occurred Click here to continue, and
> when you click on it, I get directed back to the log in screen
>
>
> -Original Message-
> From: Paul Mossman [mailto:paul.moss...@nortel.com]
> Sent: Monday, May 04, 2009 11:24 AM
> To: McCoy, Chris; sipx-users@list.sipfo
m that attempt. It will
> take me a while to get set up for the polycom trace as I am
> running an unmanaged switch and do not have a hub I can put
> in line with the phone to get a capture.
> Thanks
> Chris
>
>
>
> -Original Message-
> From: Paul Mossma
x-users] polycom 501 not registering
>
> Nope only have one nic in the pc.
> In my eyebeam setting I do have the primary dns server set to ip
> 192.168.99.2 which is my sipx box, I have to do this as my
> laptop dns is pointed to 4.2.2.1 and 4.2.2.2.
> Thanks
> Chris
>
>
> Thanks
> Chris
>
>
>
> -Original Message-
> From: Paul Mossman [mailto:paul.moss...@nortel.com]
> Sent: Monday, May 04, 2009 10:34 AM
> To: McCoy, Chris; sipx-users@list.sipfoundry.org
> Subject: RE: [sipx-users] polycom 501 not registering
>
>
> Ch
Chris wrote:
> I have one polycom 501 on my system, it is my inhouse system so I
> only have one phone, but since the update to 4.0.0 the phone will
> not register with the server, I have updated the sip.ld files and
> boot rom to
> Sip to 3.1.2.0392
> Boot rom to 4.1.2.0037
> I have formatte
Nathaniel wrote:
> I am running bootrom 4.1.1 and firmware 3.0.2
>
> As soon as this install is finished, I will attempt an upgrade.
Hi Nathaniel.
I see you're using TFTP Option 66, but can you confirm that the Polycoms
are indeed using Server Type FTP? (Menu - 3 - 2 - 456 (password) -
Enter
Damian wrote:
> Keith Gearty wrote:
> > Thanks for your support Paul.
> >
> > Those posts are 2 years old. Has nothing happened on this
> issue since
> > then? Is there an issue in the tracker that I can vote for?
> >
> >
>
> Thanks for finding this Paul.
> I could not find any issue that
Keith wrote:
...
> It seems to me that this is a fundamental problem with the way that
> groups are implemented in SipX. Here are two possible alternatives
> that I would suggest:
>
> 1.Provide a boolean for each setting that defines if the value in
> the setting should take affect or not
Tony wrote:
> Correct. If your harware phone or software phone supports
> h.264 and can successfully register and use sipx then the
> video stream is "part of" the phone call if both sides are
> compatible.
>
> Counterpath is a good example of a softphone that supports
> video (try x-lite,
Hi again,
Just some follow-up on this:
1. Scott pointed out that this should be in the Wiki, so I did:
http://sipx-wiki.calivia.com/index.php/HowTo_configure_AudioCodes_Stand-
Alone_Survivability_Feature (Linked from
http://sipx-wiki.calivia.com/index.php/HowTo_configure_AudioCodes_SIP_Ga
tew
Hi all,
Here are the configuration steps for a typical way to use the Audiocodes
Stand-Alone Survivability (SAS) Feature with sipXecs 4.0.
This feature can be useful for a branch office that has a local
Audiocodes Gateway, but uses a remote sipXecs server. In "Normal" mode
the Audiocodes Gateway
t; Anything You can advise?
>
> Looking forward.
>
> All the best!
>
> Paul Mossman wrote:
> > Hi Cuneyt,
> >
> > Excellent, hopefully that resolves the freeze problem.
> >
> > The Server Type option is only configurable on the phoneside. All
&g
see if the freeze will go away.
>
> Is there a section under SipX which i can set this as
> "default" for all Polycom 330 phones (we have %99 330s and two 430s) ?
>
> I couldnt locate while browsing the settings. If there is
> that would be a life saver!!!
>
&
Hi Cuneyt,
Did you happen to change the Server Type (under Server Menu) to
TrivialFTP? If so then change it back to FTP.
I just saw the freeze problem on a 670, and switching back to FTP fixed
it.
I also saw this problem on a 330, and again switching back to FTP fixed
it:
http://knowle
Dale wrote:
...
> After thinking about this, I think a better strategy would be
> to: (1) change the new minimum to 1800 seconds (30 minutes),
> and (2) make this change permanent.
>
> My thinking is that we do not want phones, under any
> circumstances, subscribing to any sipX server with sort
Dale wrote:
> On Thu, 2009-02-05 at 16:03 -0500, Tony Graziano wrote:
> > Is this supported in the current stable version? I am using only
> > Polycom phones, and would like to know if this is actually
> "supported'
> > or supposed to work. If it is, are there any known issues with it?
>
> Dia
>>> On 1/15/2009 at 3:38 PM, in message
<87ac5f88f03e6249aea68d40bd3e00be194da...@zcarhxm2.corp.nortel.com>,
"Paul Mossman" wrote:
Hi Matt,
Yes, the latest SIP versions of AdvaTel PhoneEasy products use with BLF
resource lists.
Mike
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Cuneyt M
Sent: Monday, February 02, 2009 10:30 PM
To: Paul Mossman
Cc: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-u
Cuneyt M wrote:
> I have recently upgraded to latest stable (thanks shruthi for the
> pointers) 3.102 via yum update.
>
> We have Polycom 330 and we were using 2.1.2 with bootrom 4
> while we were at sipx 3.8
>
> As the natural step (and understanding that 3.10 supports
> firmware 3 config inis
Akshata wrote:
>
> I agree.
> When we add an internal line with the phone, DHCP parameter
> is disabled (default). Whereas while adding external line to the
> phone, Registration Server is defined anyways. So no where DHCP
> option has to be enabled/used.
>
> Its good to
blf subscriptions, sorry
> I thought they were using the buddy lists.
> Thanks
> Chris
> --Original Message--
> From: Paul Mossman
> To: McCoy, Chris
> Subject: RE: [sipx-users] Polycom Buddy list problem
> Sent: Jan 26, 2009 12:35
>
>
> Chris wrote:
> &g
Damian wrote:
> Scott Lawrence wrote:
> > On Mon, 2009-01-26 at 10:11 -0500, Paul Mossman wrote:
> >> Hi all,
> >>
> >> Can anyone think of a good reason to keep the Polycom "SIP
> Settings
> >> in DHCP" screen around?
> >>
>
Hi all,
Can anyone think of a good reason to keep the Polycom "SIP Settings in
DHCP" screen around?
It is cumbersome to use with sipXecs, and I don't think there's a good
use case for doing so anyway.
Basically, it can be used to configure the Polycom to get the SIP server
address from DHCP. It
Hi Chris,
> I have a Pingtel user that uses the buddy list to show the
> status of the people that are in the local office, under 3.8
> the list worked, upgraded the customer to 3.10.3 and the
> list seems to be broken, is there some setting that I need
> to turn on in the pbx or a setting I
Ranga wrote:
> If you are testing sipxbridge, make sure you delete
> etc/sipxpbx/sipxbridge.xml and re-activate the dial plan
> after you update to r14521, ( This will update your copy of
> etc/sipxpbx/sipxbridge.xml )
I glossed over the "If you are testing sipxbridge" part of Ranga
email...
Hi all,
Polycom recently clarified for me some details on what happens when
invoke the various options under "Reset to Default". (Menu - 3 - 2 -
- 1 - 4.)
These are described in Polycom Quick Tip 18298, but I think some
important details are missing.
[http://knowledgebase.polycom.com/kb/search.
unications with them it was slow, normally 2 days for a response. Do
they US support yet?
>>> On 1/15/2009 at 3:38 PM, in message
<87ac5f88f03e6249aea68d40bd3e00be194da...@zcarhxm2.corp.nortel.com>,
"Paul Mossman" wrote:
Hi Matt,
Yes, the lates
Hi Matt,
Yes, the latest SIP versions of AdvaTel PhoneEasy products use with BLF
resource lists.
-Paul
paul.moss...@nortel.com
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Matt White
ller is getting music from the park
server, which is not necessarily the same setting at MOH and "worked"
without MOH before MOH worked on Polycom phones. Right?
>>> On 1/13/2009 at 9:05 AM, in message
<87ac5f88f03e6249aea68d40bd3e00be1942b...@zcarhxm2.co
razi...@myitdepartment.net]
Sent: Tue 1/13/2009 8:34 AM
To: Picher, Michael; sipx-users@list.sipfoundry.org; Paul
Mossman
Subject: RE: [sipx-users] Usefulness of Attended Transfers with
PolycomSoundPoint IP phones?
I've found the following
n function,
pulling it away makes for grumpy coworkers/users.
>>> "Paul Mossman" 01/12/09 4:40 PM >>>
Hi all,
Due to XTRN-229 (Polycom Attended Transfers can't reach trans
target's VM) we have disabled Attended Transfers on Polycoms in 4.0.
Hi all,
Due to XTRN-229 (Polycom Attended Transfers can't reach trans target's
VM) we have disabled Attended Transfers on Polycoms in 4.0.
Given that we can perform Blind and Consultative transfers without
problem, how desirable is the Attended Transfer functionality?
For background, here's t
Milosz wrote:
> i vote disabled by default also. i don't have a single user
> (including me) who prefers to see a phone number display that
> includes the sip: and @domain portions.
>
> speaking of which, anyone know what setting to change in
> 3.10.2 to fix the "messages" button after you d
Damian wrote:
> Paul Mossman wrote:
> > Hi all,
> >
> > We are uncertain about this issue, and are looking for
> input from the
> > user community. (XCF-2837)
> >
> > Should Polycom SoundPoint IP profiles have URL dialing enabled or
> > dis
Hi all,
We are uncertain about this issue, and are looking for input from the
user community. (XCF-2837)
Should Polycom SoundPoint IP profiles have URL dialing enabled or
disabled by default?
The benefit of having URL dialing disabled is that the Call ID Number on
the display is shown without
Thank you Tony and Michael for your valuable feedback.
Tony wrote:
> One touch voicemail is problematic when you have more than
> one line registered on the phone.
Yes, I can see how that could be confusing. But, I've observed that
Polycoms are smart enough to disregard a One Touch Voicemail
Max wrote:
> I'm testing a Polycom IP 330, but I can't figure out hot to
> do directed call pickup.
>
> In my production system (based on SipX 3.8.1-011577) I can do
> directed call pickup between Linksys phones, but I can't do
> directed call pickup between Linksys and Polycom phones.
>
> I'
Hi all,
I've found that a few of the Polycom default User Preferences values are
not... well... to my preference. I think the defaults should be
changed, and I wanted to see what you think.
1. Headset and Handset Volume Persistence to On. This means the phone
will remember the user's volume sel
Hi all,
When you add a new Polycom SoundPoint IP phone in sipXconfig, you
currently have a choice of firmware version "1.6" of "2.0". In
practice, the "2.0" options actually means "any version greater than
1.6".
I'd like to have support for "1.6" removed.
Polycom firmware is up to 3.0.3 now,
Hi all,
I can see that the Call Pick-up feature will indeed attempt to perform a
pick-up of a call that has triggered a Dial Plan rule and is heading out
through a Gateway. This works great when the remote Gateway is another
sipX machine.
However, I tried this with a 4 port FXO Gateway, and it d
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