Nathan,
The best thing to do right now is to get a complete trace of these failed
calls. Clear your logs, regenerate the failed calls and grab a merged Xml
file. With that, I'm sure someone on this list will be able to point you in
the right direction fairly quickly. My guess is that you
My guess is that you do not have a single problem here, but several
small ones.
Not the least of which could be running sipXecs on VMWare ESXi 4.x...
Mike
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Todd Hodgen
Sent:
Thanks Todd, did the trace and checked it out in Sipviewer. The Cisco is
returning a 406 error which brought me to this thread:
http://www.mail-archive.com/sipx-...@list.sipfoundry.org/msg05465.html
I am attempting to take the phone back down to 8.2.x now.
From: Todd Hodgen
I was waiting for that J I read something about it being frowned upon
because of the lack of a realtime clock at high call load - but what
exactly would be considered high call load? We're doing maybe a maximum
of 5-6 simultaneous calls.
From: Picher, Michael [mailto:mpic...@cmctechgroup.com]
Confirmed 8.2.2SR4 resolves the 406 issue. 3491672850*# is my friend.
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Nathan
Nieblas
Sent: Tuesday, January 05, 2010 2:01 AM
To: Todd Hodgen; sipx-users@list.sipfoundry.org
Subject: Re:
On Thu, 2009-12-10 at 13:11 +1300, Justin Menga wrote:
I've come across interop issues with various SIP UAs and now a
commercial SBC (Cisco ASR 1000 Series SBC SP Edition) and SIPXecs,
which stems from the SIPXecs use of the Tag parameter in the From
Header for generating nonce values for SIP
On Mon, 2010-01-04 at 13:29 -0600, Josh Patten wrote:
I guess now that the holidays are over with it's time to start getting
back in the swing of things and one thing I think all of us want to see
is better documentation.
I've been waiting until people were back from the holidays to take up
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of
Lawrence, Scott AVAYA (BL60:9D30)
Sent: Tuesday, January 05, 2010 8:11 AM
To: Josh Patten
Cc: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] Wiki
On Tue, Jan 5, 2010 at 7:57 AM, David Saint sai...@avaya.com wrote:
It would be nice if there was some way of organizing the documentation
so that it is obvious which sipXecs release it applies to. Some
components may behave differently in different releases so it would be
good if, for
As a user/developer who might contribute to the wiki it would be good to have a
well known single obvious place to put info about a specific piece of hardware
in a hierarchy with the vendor at the top and the product family below it.
On the detail page for each device or family it might be good
On Tue, 2010-01-05 at 08:30 -0600, Eric Varsanyi wrote:
As a user/developer who might contribute to the wiki it would be good
to have a well known single obvious place to put info about a specific
piece of hardware in a hierarchy with the vendor at the top and the
product family below it.
My
As a follow-on question, when I look at the trace with sipviewer, which
component should be talking to the ISTP? I'm seeing sip-proxy contacting
the ISTP. Shouldn't sipxbridge be doing that?
On Mon 04.Jan.10 20:59, Dan McDaniel wrote:
I'm trying to get my Vitelity trunk working on my freshly
On Tue, Jan 5, 2010 at 11:12 AM, Dan McDaniel d...@dm3.us wrote:
As a follow-on question, when I look at the trace with sipviewer, which
component should be talking to the ISTP? I'm seeing sip-proxy contacting
the ISTP. Shouldn't sipxbridge be doing that?
You have identified the problem.
On Tue, 2010-01-05 at 02:45 -0800, Nathan Nieblas wrote:
I was waiting for that J I read something about it being frowned upon
because of the lack of a realtime clock at high call load – but what
exactly would be considered high call load? We’re doing maybe a
maximum of 5-6 simultaneous calls.
So I made some changes to our phone system last night and found that I
now have this issue ONLY if I make a call to our DID from an internal
phone.
Originally I had it set up as:
DID -- Phantom 120 -- Forward to x200 for 30 sec -- If no answer ring
hunt group 501 -- Dump to VM for x200.
That
On Tue 05.Jan.10 11:21, M. Ranganathan wrote:
On Tue, Jan 5, 2010 at 11:12 AM, Dan McDaniel d...@dm3.us wrote:
As a follow-on question, when I look at the trace with sipviewer, which
component should be talking to the ISTP? I'm seeing sip-proxy contacting
the ISTP. Shouldn't sipxbridge be doing
Make sure you push your profiles and restart any services that prompt to be
restarted.
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431
Email: tgrazi...@myitdepartment.net
LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: 434.984.8426
I'll raise my hand and disagree here a bit. If I want info on a
vendor's products I don't want to look in 2 different places depending
on whether it is managed by sipXecs or not.
I'm not 100% sure what it ought to be but in my head it looks something
like:
sipXecs -+- Version 3.x
|
On Tue, 2010-01-05 at 12:23 -0500, Picher, Michael wrote:
I'll raise my hand and disagree here a bit. If I want info on a
vendor's products I don't want to look in 2 different places depending
on whether it is managed by sipXecs or not.
To some extent that may be hard to avoid, and in the
A table.
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431
Email: tgrazi...@myitdepartment.net
LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: 434.984.8426
Fax: 434.984.8427
Helpdesk Contract Customers:
http://sipx-wiki.calivia.com/index.php/HowTo_configure_AudioCodes_SIP_Gateway_with_sipX
I have been working on a mediant 2k (5.6) configuration in order to allow voip
calls to go to sipx and fax calls to go to a fax server. I have not found much
documentation on how to do this so have yet to
Is there anything special needed in SipX to add lines to the expansion
module or does the phone add the lines to the BEM once the 6 lines on
the phone are assigned?
Austin Curry
___
sipx-users mailing list sipx-users@list.sipfoundry.org
List
It's automatic. As you add lines and speed dials to the phone they will
spill over to the expansion console. No extra configuration is required.
Josh Patten
Assistant Network Administrator
Brazos County IT Dept.
(979) 361-4676
On 1/5/2010 12:14 PM, Austin Curry wrote:
Is there anything
On Mon, 2010-01-04 at 20:56 -0800, steven warner wrote:
Can I get the definitive answer to the uri for music on hold?
I have seen:
~~...@host.domain
and
m...@host.domain
Which is correct for sipxecs? Do I need a special port?
Currently, the default MOH URI is sip:~~...@[domain].
Under each sipx (not the dev side... not sure what you guys want
there)...
I'd have Quick Start (quick how to on getting a system up and running
from ISO with links to more in-depth information), Administration
(installation, adding users, adding phones, adding gateways, system
features, dial
I've already got a basic web portal guide written up in my personal
space. Check it out: http://wiki.sipfoundry.org/display/~dead_again/Home
Josh Patten
Assistant Network Administrator
Brazos County IT Dept.
(979) 361-4676
On 1/5/2010 1:05 PM, Picher, Michael wrote:
Under each sipx (not the
(for steve)
Realize the sip:~~mh~[us...@[domain] does not work in the current 4.0.x
release, and is being worked in the 4.1x development version.
On Tue, Jan 5, 2010 at 2:01 PM, Dale Worley dwor...@avaya.com wrote:
On Mon, 2010-01-04 at 20:56 -0800, steven warner wrote:
Can I get the
On Mon, 2010-01-04 at 22:14 -0600, Josh Patten wrote:
Just wanted to know if there were any technical limitations to the
current RLS server (for BLF) such as limit on number of
monitored/monitoring extensions and/or stability issues when scaled to a
large number of extensions. Just trying
I am deploying Soundpoint 450s and 550s. I'm using Sipx 4.0.4. The 450s
worked out of the box. No major changes. Per the posts on here and the
documentation I found, I had to downgrade the 550s to make them work at
all. I added spip_ssip_vvx_3_1_3RevC_release_sig_split.zip to my Sipx
server
All of the phones the 3.1.1revc firmware support were unzipped and copied to
the tftp root directory when you unloaded it.
The phone knows the filename it is looking for, as each model is a uniquely
different name.
No magic, just logic provided by Polycom on that one.
When the phones boot, or
Thanks. So my 450s should be pulling it? They definitely don't seem to
be. I have deployed 6 handsets today and 12 total. My 450 bootrom is
4.1.2.0037, so it seems like I should have been ok with this requirement
'Split download file is recommended, but requires that all phones are
running
I agree with your responses although it would be nice to have say a
configuration option like Replay Attack Protection or similar which turned
on/off the validation rules currently used. I guess it's just an issue I
came across with two SIP UAs and a commercial SBC. My use of SIPXecs is
atypical
I believe they should be pulling it.
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431
Email: tgrazi...@myitdepartment.net
LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: 434.984.8426
Fax: 434.984.8427
Helpdesk Contract Customers:
Anyone got any tips on how to troubleshoot?
On 1/5/2010 2:31 PM, Tony Graziano wrote:
I believe they should be pulling it.
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431
Email: tgrazi...@myitdepartment.net
LAN/Telephony/Security and Control
Are phones set for TFTP or are they set for FTP or something else?
Try manually setting one for TFTP. Also, are they getting DHCP from the
same place?
Mike
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of
I'm going to backup and restore the system to a physical machine tonight and
see what happens. I don't think anything is setup incorrectly with my network
infrastructure or the system configuration itself at this point. We are still
experiencing strange issues; dropped calls, 1 way audio,
They set to default, but I will check what that as soon as I get home (kids
basketball practice right now). DHCP, DNS, TFTP, everything, etc is running on
the sipx server configured by the sipx wizard.
Sent via BlackBerry from T-Mobile
-Original Message-
From: Picher, Michael
Also - that is the firmware polycoms page shows when I follow their menus. I
assumed that meant it was supported. I will go through some release notes and
see what I can find.
Sent via BlackBerry from T-Mobile
-Original Message-
From: Tony Graziano tgrazi...@myitdepartment.net
Date:
Matrix:
http://downloads.polycom.com/voice/voip/sip_sw_releases_matrix.html
http://downloads.polycom.com/voice/voip/sip_sw_releases_matrix.htmlRelease
Notes:
http://downloads.polycom.com/voice/voip/relnotes/spip_ssip_3_1_3RevC_relnotes.pdf
I am a newbie to sipXecs and everything VOIP. I'd appreciate any help I can get.
I just installed the 4.0.4 ISO. But, sipXecs can't discover the spanking new
Snom 870s that I have on the network. Reading some posts from back in May 2008,
I gather these phones may not be supported yet. Is there
You will have to manually configure it. You might want to read up on
disabling GRUU with it also. It has its own web interface to manually
configure it. In general, SNOM has known interoperability issues, and the
mfr doesn't put a lot of emphasis on fixing them, hence the lack of
management
I guess I'll have to wait until tomorrow. I'm at home and I was hoping I
could tell the firmware version from the polycom web interface to see if
it had upgraded, but I can't find that anywhere. Oh well
On 1/5/2010 5:57 PM, Tony Graziano wrote:
Matrix:
No way I know of except query the phone menu itself.
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431
Email: tgrazi...@myitdepartment.net
LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: 434.984.8426
Fax: 434.984.8427
Helpdesk Contract
I'm not sure forward softkey is the same as an EFK blind transfer.
Go to the user line, on the phone, click diversion' and enable forward
and resend the profile to the phone. Alternately this can be done from the
sipxconfig ui by the user.
Is that what you were looking for or did I misunderstand
Whoops I guess I went a bit overkill on that post.
Tony Graziano wrote:
I'm not sure forward softkey is the same as an EFK blind
transfer.
Go to the user line, on the phone, click "diversion' and enable
"forward" and resend the profile to the phone. Alternately this can be
done from
As I was testing our setup with Verizon VOIP, they pointed out that we
were using G.711. I had left our Polycom 450s and 550s with the default
codec settings. Verizon stated all of their customers use G.729 for
general voice use. Their servers do immediately renegotiate to G.711
when they
Why not have the G722 as first option, G729 as second option, and then G711
as third option. This should put internal G722 to G722 compliant phones
working on that CODEC. Then, when they leave the system to systems that
don't support G722, they will go to G729 next. The devices will do their
I don't see G722 as an option for the most part. I only see that for the
650. I'm using 450s and 550s. Am I missing something?
On 1/5/2010 9:44 PM, Todd Hodgen wrote:
Why not have the G722 as first option, G729 as second option, and then G711
as third option. This should put internal G722 to
I am trying to get pricings on T1 and SIP Trunk in one deal. We have a current
T1/PRI connecting to our Panasonic and we need to move off of it in 2 months.
We are looking to replace it with another T1 from another provider and a SIP
Trunk to go along with it. Any recommendation? Also,
Perhaps I completely misunderstood the Polycom codec configuration page.
I just read this:
http://track.sipfoundry.org/browse/XX-6785
So the 450 uses the IP_650 codec group. I didn't realize that. I just
assumed the numbers on this page corresponded to the model of the phone.
This could be a no
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