On the HD note, everyone seems to be promoting HD voice from polycom to
audiocodes etc.
What sort of bandwidth is that codec using between two HD enabled polycoms?
I'm still at the Polycom 330 g711u/a level - polycom phones and
audiocodes gateways.
The way you explain the tool sounds like we
You are right. I overcompensated just remembering that I saw 80k being used for
one call. Not all calls will achieve that, but after you add overhead, it's a
"safe number".
In my case I was watching this at my POP where we have a system installed that
can look at the SIP call setup, sees the c
May be off topic, but related to this particular thread discussing
bandwidth. When one is looking at Bandwidth requirements for lines, it's
important to discuss where those bandwidth requirements are. For example,
when you are talking about on an Ethernet wire, 800K for 10 trunks of G711
is reaso
You don't really need to testthe bandwidth will dictated by the
CODEC. g.711 plus the overhead of your underlying protocols. It is
what it is.
Here is one of many breakdowns of where the 80kbps comes from.
http://www.cisco.com/en/US/tech/tk652/tk698/technologies_tech_note09186a0080094ae2
Thanks for the reply Tony,
Forgive my ignorance but what is a DPI system? We have Polycom 330s all-over and it sounds like the test that we gotta do to see the real bandwidth consumption.
Is there any sw/links to this DPI system and documentation on how to go about testing it with
Polycom <->
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 3.11.x and 4 will have native SIP trunking, Do you
know when is the realistic release date for 4?
**No. I am checking the tracker to see what kin
Hi Tony,
Good to hear from you.
I didnt know only 3.11.x and 4 will have native SIP trunking, Do you
know when is the realistic release date for 4?
So for that kind of volume you need a minimum of 15 trunks and a 20-pack of DID numbers.
20-pack DID numbers because they come in 20? in you
>
> Interesting idea: are you thinking about sipXconfig UI for managing
> that
> ACD? You probably have 2 options: reuse existing UI or add a plug-in
> for a
> new one. Let me know on sipx-devel when you need some help with that.
> D.
No help needed at this stage, currently my ACD has no extr
>>> 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 understand that
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
Paul Scheepens wrote:
>
> Hi Guys,
>
> I wanted to build a new HA sipx server so I installed a 3.10.1 and yum
> updated it to 3.10.3-014144.
> Now the callresolver won't start. It fails immediately.
> I increased the log level to DEBUG, but the log file is not updated
> anymore.
> This is the con
Paweł Pierścionek wrote:
> On 2009-03-31, at 01:03, Pedro wrote:
>
>> Hi,
>>
>> I have the same problem, if you find a solution please send me by
>> mail.
>> I'll do the same
>>
>
> There are tons of problems with ACD when You try to max it out with 50
> agents or 30 connections.
> Nortel see
Hi Pawel,
Thanks for your answer.
It sounds good idea the plug&play that you’re developing.
I’m not involved in the development but I can help you testing.
You said that you developed patches;
do you think this could help with the issue
of segmentation fault? If so, where can I get these patch
>> Thanks Mike.
>>
> I am using 3.1.2 with Bootrom in one office. The difficulty we have there is
> there is on one person from my IT department there at most 3 days per week.
>
> The volume of calls is pretty good and we are finding our receptionists
> Polycom 650 reboots, so we slowly disconnec
>>> On 3/31/2009 at 11:28 AM, in message
, "Picher,
Michael" wrote:
>
>
>
> From: Picher, Michael
> Sent: Tuesday, March 31, 2009 11:28 AM
> To: 'Jim Canfield'
> Subject: RE: [sipx-users] Paging Issues
>
>
>
> I just haven't tested 3.1.2. I know Tony has and I think he was good
> with
I've had the same problem with 3.10.3. I think there's a memory leak
somewhere.
Restart the resource list service. If that doesn't work do a 'service
sipxpbx restart'
Mike
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of
dundow.
I've done this before with some Patton Gateways.
Are they both FXO gateways?
Mike
> -Original Message-
> From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-
> boun...@list.sipfoundry.org] On Behalf Of Massimo VIGNONE
> Sent: Tuesday, March 31, 2009 1:08 PM
> To: sipx-users@l
Hi everybody,
Take a look at this scenario.
Sipx with two gateways: gateway A is connected to another (traditional)
PBX and gateway B is connected to PSTN.
The traditinal PBX sends outgoing calls from extension 200 through the
gateway A, then Sipx routes calls according its dial plan.
In the di
Hello All,
We are currently having the following issue with Sipxecs Version 3.11.2:
"The agents cannot login into the ACD group. It says they are logged in but
when you call the ACD they don't ring".
Any help with be greatly appreciated :)
Thanks,__
From: Picher, Michael
Sent: Tuesday, March 31, 2009 11:28 AM
To: 'Jim Canfield'
Subject: RE: [sipx-users] Paging Issues
I just haven't tested 3.1.2. I know Tony has and I think he was good
with it.
I'd go with an ATA and the Valcom. Valcom is also great as a company at
helping you
Hi Guys,
I wanted to build a new HA sipx server so I installed a 3.10.1 and yum
updated it to 3.10.3-014144.
Now the callresolver won't start. It fails immediately.
I increased the log level to DEBUG, but the log file is not updated
anymore.
This is the contents although I tried to start/reboote
On 2009-03-31, at 01:03, Pedro wrote:
> Hi,
>
> I have the same problem, if you find a solution please send me by
> mail.
> I'll do the same
>
There are tons of problems with ACD when You try to max it out with 50
agents or 30 connections.
Nortel seems to ignore my patches which make things
Jim,
Go to 3.1.0 or 3.1.1 with 4.1.2 bootrom.
Page that many phones at once? Wow... Brave!
Mike
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Jim
Canfield
Sent: Monday, March 30, 2009 11:32 AM
To: sipx-users@list.sipfound
I simply meant we have them for a different need, so we don't use the lines on
the m3's. I also wanted to point out that we use snom phones but not in a way
that would be helpful to you in discovering provisioning or confiuration
issues.
-Original Message-
From: Keith Gearty
To:
S
Oh I understand now. Sorry for the confusion.
Tony Graziano wrote:
>I simply meant we have them for a different need, so we don't use the lines on
>the m3's. I also wanted to point out that we use snom phones but not in a way
>that would be helpful to you in discovering provisioning or conf
What did you mean when you said /"[We] only use the m3's for wireless in
the office, and those need to all be unique lines" /?
I'm using a Snom m3 registered to the same extension as a Snom 320 (so
that the user can dial/answer calls when at his desk and when roaming).
I previously asked the
The Linksys SPA phones I use all have it as well, but I've never really
had to use it so I've never played around with it. I have phones sharing
the same extension number and they work just fine without the setting. All
ring when a call comes in but the other users are still able to dial out
o
Snom and Polycom make SLA available, but the is feature onlyinteroperable with
a few switch types. If you don't have a supportedswitch (Broadsoft, Sylantro,
etc., it won't work). Snom buries itbecause the switches that support it can
implement it more easily viaautomatic provisioning methods and
Scott Lawrence wrote:
> I believe that using the phone menu key bypasses the proxy, and doesn't
> work in a sipxecs system even without the acls.
>
No, it worked before configuring Smartnode with the acl.
The problem is that Polycom phones configuration had an empty value in
the Outbound Prox
I just came across a setting called "User Shared Line", buried deep in
the Device>Line>SIP settings page. I'm not sure if it may be specific
to Snom phones.
Last week I asked a question on here about using multiple phones
registered to the same extension. Everyone said it was possible, but no
Hi,
I have a problem with Dialling Rules and Gateway combinations.
I am using an ITSP with multiple numbers, thus multiple gateways to the same
ITSP.
This is required as the different numbers are for different cities/regions and
lets me take advantage of LCR (Least Cost Routing).
So for examp
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