Dear All,
I am exploring the SipX integration with our intranet and would
appreciate your help;
User inside the intranet will click Check Voice Mail (or Phone Settings)
which will open a new window and log user into his/her User portal.
Before launching the window, user's login and pin is assu
Dear All,
We have been using the 48-port Linksys switch with Polycom 330 phones
and we have this constant phone freezing issue. Tried pretty much all
firmware and rom combination and the phone freezing issue is not solved.
In order to test if the issue is related with the PoE, we are looking to
al AC adapters on phones leaving no more than 12 POE
> devices on the switch and turning off POE on the AC powered phones and see if
> it still happens. I can say that though because all of my 650's came with
> power adapters and I have a box of them.
> I suspec
Thanks Scott. I will try to rent a HP Procurve and check out the Fluke
PoE analyzer.
However, can i run SIP 2.2.2 for Polycom 330s with Sipx 4.0.2 ?
I dont see the version number when editing phone profile anymore and not
sure what are the possible issues running 2.2.2
That was the only version
Hi Bernardo and Michael,
We use that model of switches too as well as the latest 48-port model.
I think whatever the problem is, it has almost the same pattern
somewhere. We are also using Polycom 330s.
Same firmware and ROM version (beside almost all version pair tests that
end up with same f
Hi Scott,
I have installed a CentOS ISO pub stable release today (4.0) and get it
up and running. Sipxbridge was functional.
I did a yum update for the CentOS operating itself early today (before
4.2 released) to get the latest OS updates.
When i followed your guide to update to 4.0.2, didn
Dear all,
I understand certain version (4.2.x?) of sipx supports an API(rest/
soap/other) to get() and set() User Portal data - where integration
with custom intranet and HR systems would be possible without asking
users to login sipxconfig
I can't seem to find any doc or example on this, i
rt
> settings seldom work and result in frequent reboots and freezes. I
> would make sure they have full class 3 power or use the external power
> supply for testing and see how it goes.
>
>
> On Tue, Jul 28, 2009 at 9:48 PM, Cuneyt M wrote:
>
>> Hi Huijun,
>>
Hi Huijun,
I have tried SIP 3.1.3 revision C (tried bot split and combined
separately) with ROM 4.1.3 and noticed few issues and intermittent freezing.
Then I've lowered the firmware 3.1.1(base version) and rom to 4.1.1(base
version), which solved the hanging on transfer issues but the
inter
Following the update to sipx 4.0.1 and SIP 3.1.1 with ROM 4.1.1 (base
versions, not revisions) device files for Polycom 330 phones, users are
reporting very frequent freeze - which occurs either during conversation
or simply when they come back to office.
1) which SIP & ROM would be more stable
I've recently updated 3.10.2 installation to latest stable 4.0.1 via yum
(as per the wiki tutorial )
I am running Polycom SIP 3.1.1 and ROM 4.1.1 (all base version, not the
revisions)on polycom 330 phones.
( spip_ssip_3_1_1_release_sig.zip and BootROM_4_1_1_release_sig.zip)
However, users repor
Dear All,
I have installed a new system with CentOS ISO from the *development*
downloads.
Sipx version 4.1.x
I am using Polycom 330 phones and would like to know which ROM and
firmware version of Polycom will best suit the Sipx 4.1.x (polycom 330
phones)
Thank you in advance,
All the best!
_
Dear All,
I am planning to upgrade a few installations running on 3.10.2 to latest
4.x (what is the latest-known-to-be-running-well version btw?) via yum.
I've tried installing a test system with the latest 4.0.1 centos iso few
weeks ago but that throw error 'sipxecs package cant be located'
*site3.*company.com.
_sip._udp.*site3.company.com*. IN SRV 1 0 5060
*site3.*company.com*.
*
Tony Graziano wrote:
Then I would check my routing between two of the servers (both
directions) to ensure it is taking a private path.
After that set a proxy to debug, and tail the proxy
ected site or
calls will fail.
-Original Message-
From: Cuneyt M
To:
To:
Sent: 7/9/2009 4:18:39 PM
Subject: [sipx-users] Connecting Multiple SipX PBX Sites on 3.10.3
Dear All,
I am still using 3.10.3, as my previous attempts to upgrade to 4.01
failed and had to leave that aside as I d
Dear All,
I am still using 3.10.3, as my previous attempts to upgrade to 4.01
failed and had to leave that aside as I didn't have more down time to
try updated Wiki page for 3.10.x to 4.0.1 yum update - yet.
The current issue on 3.10.3 briefly(!) when all sites are VPNed and i
-create a gate
Hi everyone,
Is there any example or sample script that does Click-to-Dial/Call by
using the sipX 4.x API - which can be used through a web based
(intranet in this case) contact database?
Would appreciate if someone can share a sample script that can be
integrated with third party (web) apps.
rom a statement by Scoot a few weeks ago? I would ask if anyone has
upgraded succesfully first. The 4.1 unstable is pretty good, but there are
still some problems to be fixed slated for the 4.01 release.
Cuneyt M 05/30/09 3:13 PM >>>
Hi there,
I have one sipX server running
Hi there,
I have one sipX server running on 3.10 which is upgraded from 3.8 via
YUM (3.8 was installed off the Centos ISO at the time).
Having briefly read the troubles reported in the list and the 4.01
announcement on sipx homepage, I am not sure how i should go about
upgrading to a stable 4.
o: cun...@entegra.com.sg
Cc: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] Bandwidth.com as SIP-TRUNK for SipX 3.10.3
Hi,
Hope things are going well for you. My comments are below.
Cuneyt M 03/31/09 5:35 PM >>>
Hi Tony,
Good to hear from you.
I didnt know only
<-> sipx->wan<->bandwidth.com
Would appreciate that very much!
all the best.
Tony Graziano wrote:
Hope things are going well for you. My comments are below.
Cuneyt M 03/31/09 5:35 PM >>>
Hi Tony,
Good to hear from you.
I didnt know only 3.11.x and 4 will
iano wrote:
On 3/31/2009 at 4:38 PM, in message <49d27f47.8030...@entegra.com.sg>, Cuneyt M
wrote:
Hello everyone,
We have existing setup of SipX 3.10.3 with AudioCodes gateways.
However, we will be testing/integrating Bandwidth.com US Sip-Trunk (with
DID) with SipX. I unders
Hello everyone,
We have existing setup of SipX 3.10.3 with AudioCodes gateways.
However, we will be testing/integrating Bandwidth.com US Sip-Trunk (with
DID) with SipX. I understand that we can add a Sip Trunk just like
another gateway in Sipx(?)
The main use-case would be Philippines satellit
Wed, 2009-03-25 at 09:13 +0800, Cuneyt M wrote:
Hi Scott,
I setup the Xlite as per the previous installations which worked
flawlessly.
Enabled Invite All to Proxy too.
Attached are the sshots of xlite config, the internal server error and
the log from MP118 (level 5 debug output).
DNS
...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] *On Behalf Of *Cuneyt M
*Sent:* Tuesday, March 24, 2009 6:51 PM
*To:* Scott Lawrence
*Cc:* sipx-users@list.sipfoundry.org
*Subject:* Re: [sipx-users] Sipx latest stable 3.10.3 issue: INTERNAL
SERVER
Hi Scott,
If you say
to look, any setting or what not to fix
this issue?
Kindly let me know.
All the best
Scott Lawrence wrote:
On Wed, 2009-03-25 at 09:13 +0800, Cuneyt M wrote:
Hi Scott,
I setup the Xlite as per the previous installations which worked
flawlessly.
Enabled Invite All to Proxy too
n Wed, 2009-03-25 at 09:13 +0800, Cuneyt M wrote:
Hi Scott,
I setup the Xlite as per the previous installations which worked
flawlessly.
Enabled Invite All to Proxy too.
Attached are the sshots of xlite config, the internal server error and
the log from MP118 (level 5 debug output).
DNS resol
eport this freezing problem and our Server Type findings to
> Polycom.
>
> Hope that helps!
>
>
> -Paul
> paul.moss...@nortel.com
>
>
>
>
>
>> -Original Message-
>> From: Cuneyt M [mailto:cun...@entegra.com.sg]
>> Sent: February 26, 2009
ul.moss...@nortel.com
>
>
>
>
>
>> -Original Message-
>> From: sipx-users-boun...@list.sipfoundry.org
>> [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Cuneyt M
>> Sent: February 25, 2009 11:38 PM
>> To: Gerald Drouillard; sipx-use
helps.
>
> -Paul
> paul.moss...@nortel.com
>
>
>
>
>
>> -Original Message-
>> From: sipx-users-boun...@list.sipfoundry.org
>> [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Cuneyt M
>> Sent: February 25, 2009 11:38 PM
>&
appreciate if someone can direct me to a reference XML file that i
can counter check
thanks in advance to all.
Gerald Drouillard wrote:
> On 2/25/2009 12:57 AM, Cuneyt M wrote:
>> Hi All,
>>
>> We have recently upgraded our Polycom 330s to latest ROM 4.1.2 and SIP
>
Hi All,
We have recently upgraded our Polycom 330s to latest ROM 4.1.2 and SIP
app. 3.1.1
these updates carried after we updated sipx from 3.8 to 3.10.3 via yum
update.
However, the issue we are having is the Polycom 330 phones freeze at
random times, there is no pattern that causes the freez
-maker'.
So it would really be helpfull to see a working configuration of other
Polycom 330 user(s).
All the best...*
*
Scott Lawrence wrote:
On Fri, 2009-02-20 at 04:15 +0800, Cuneyt M wrote:
However, i am facing with a bigger problem. When user A is on an active
call with outsider B, and
Hi again,
I wrote in an earlier email with great length the issue i am experiencing.
I will keep it short and simple with the hope that i can get some
pointers from the sipx mailiinglist members;
We have Polycom 330 phones connected to SipX 3.10.3 (recently upgraded
from 3.8 centos iso based
hello everyone,
Please excuse me for this rather long email as I wanted to explain the
steps taken in details which is a complete mystery to me and desperately
looking for solution.
We are experiencing some rather weird behaviors with the Polcyom 330 and
SipX 3.10.3 (upgraded via yum from Cent
open internet and have a firewall, it either
needs to be sip aware and configured properly, or you need to be able
to pass the traffic over a vpn and be able to resolve the DNS SRV
records accordingly.
Tony
>>> Cuneyt M 02/04/09 1:27 PM >>> Hi Michael,
I actually follow the
Hi Michael,
I actually follow the tutorial (and verified SRV records resolved at
each end) and setup the two end-points.
Used the full domain in the gateway address field (also tried with IP
addresses, using domain name proved to be working v.s. IP)
What happens is, when i dial from site A an
ul Mossman wrote:
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 c
Hi all,
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 (the UI dropdown stil
Hi Mike and Tony,
The door-relay idea sounds very interesting.
We have polycom phones and Mediant 1000 with ISDN and analog interfaces
(where most analog interfaces (fxo) are currently not used). Polycom
phones dial out and receive calls from ISDN lines.
How do you do a ring-down in such set
Dear All,
We have a new mediant 1000 with 2 E1 and 4 4-port FXO modules.
Under sipx 3.8, when i add the gateway Mediant 1000, under Ports I can
only see the E1 options but not the additional modules (i.e. FXO
modules). I have checked the settings and there seems to be no place to
key-in the c
Dear All,
I do recall reading about this issue sometime back, however the online
article i located didnt seem to work for me.
I have the (and i believe common for other polycom users) issue of
handset volume being not persistent (not remembered) and defaults to
'low' default. Therefore users a
Dear All,
We will be ordering new batch of phones. So i would like to get your
valuable feedback before i proceed with the order;
Polycom 330 still good for generic user/usage ?
Or do we have an alternate model, brand that works (or will work in very
near future) with sipX?
Would appreciate y
Dear All,
We had interim disconnections with Polycom 330s with bootrom 4.0 and
firmware 2.1.1, 2.1.2, 2.13 (respectively).
We checked the network, the gateways but didnt lead to any solution.
However, when we replace one user's Polycom 330 with Linksys 942 (same
extension, no dial plan - same
Hi everyone!
I am approached with the enquiry of being able to "record" calls on the
SipX level.
I would like to know if there is any current or near-future plans to
integrate such functionality to sipX.
If there is no current or near-term planned call-recording feature, i
would like to hear
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