Displaying 20 results from an estimated 2000 matches similar to: "Strange Polycom Issue"
2015 Mar 10
1
Strange Polycom Issue
On Mon, Mar 9, 2015 at 9:40 AM, David Wessell <david at ringfree.biz> wrote:
> Welcome to our hell.
>
> We ran into this on VVX 300 and 400 phones running UCS 5.2.x. We finally
> got Polycom to issue a hotfix firmware version. I'll be happy to share it
> with you offlist, just email me.
>
> Officially Polycom will fix the issue in 5.3 in a few months..
>
>
2014 Aug 11
1
401 Unathorized
I have an asterisk 1.8.x box that intermittently returns a 401. Calls come
through the same peer all the time, from the same carrier. However
intermittently the asterisk box returns a 401.
Below is the output of a failed call (1st) and a successful call (2nd). I
can't see any difference until we get to these lines.
Bad call:
--- (17 headers 14 lines) ---
Sending to carrierIP:5060 (no NAT)
2014 Sep 17
1
Polycom DND + Intercom/Paging Override?
Greetings-
As many of your are Polycom "experienced", I was hoping some kind soul could provide direction on a specific issue.
On a system running Asterisk 11.11.0 (and latest FreePBX), I'm finding an instance where, using intercom/paging functionality of FreePBX, I need to override an end user's 'Do Not Disturb' selection on the handset. By default, DND simply
2015 Mar 09
0
Strange Polycom Issue
Welcome to our hell.
We ran into this on VVX 300 and 400 phones running UCS 5.2.x. We finally
got Polycom to issue a hotfix firmware version. I'll be happy to share it
with you offlist, just email me.
Officially Polycom will fix the issue in 5.3 in a few months..
Thanks
David
On Mon, Mar 9, 2015 at 9:34 AM, Andrew Colin <andrew at convergedgroup.net>
wrote:
> Hi Guys,
>
>
2013 May 11
1
Which channels are required for FAX, GTALK and Jaber
Hello;
To be able to send and receive faxes through asterisk and to be able to have trunk with google voice and to be able to have integration with those that support Jaber .. What are the channels and libs that I have to be sure that they are existed?
Regards
Bilal
2013 Jun 18
2
Is Asternic.net out of business (Flash Operator, Call Center Stats)?
We have licensed both products and sent a support request on 6/11, with
zero reply or any activity on it at all so far. No replies to subsequent
ticket updates or e-mails.
--
Carlos Alvarez
TelEvolve
602-889-3003
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2014 Jan 10
1
CTI
Hi people
I'm just mailing to see what people are using for CTI solutions with
asterisk. Aslos, has anyone managed to integrate asterisk with Salesforce?
Thanks in Advance
Ish
--
Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
f: +44 (0)161 660 9825
e: ish at pack-net.co.uk
w: http://www.pack-net.co.uk
Registered Address: PACKNET LIMITED, Duplex 2,
2013 May 01
1
multiple provider for incoming
Matt,
At some point you need to consider how much is too much...
I run a call center with more then 125 commissioned phone sales reps and more than 60 customer service reps. We run dual servers, fiber from one provider and 6 bonded T1's from another provider. We purchase our so trunks from a wholesale company who is a major provider to resellers. Being so, their network is extremely
2005 Feb 09
2
Problem using TDM400P FXS card
Hello, everyone
After having spent several time to look for any
solution for my problem, I decided to write here.
Here is the problem got a Digium X101 FXO card and
Digium TDM400P alias freshmaker (1 fxs module on it on
first port ) in my asterisk box. The X101 works
perfectly.
The problem comes from the TDM400, when i plug a
tested working phone on it , i get no dialtone; the
goal is to
2011 Mar 22
3
Act! Integration
Is there any integration for ACT! and asterisk? I've googled for hours and haven't been able to find anything.
Thanks
David
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2013 Apr 05
1
Accessing examplars in apcluster (apcluster package)
Hi,
I was wondering how it was possible to access the actual cluster exemplars
from the APResult class. Currently it only spits it out onto the terminal
if you type the object but there is no other way to see which one is the
examplar.
Would appreciate any help.
Thanks,
Sachin
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2020 May 28
2
Notification when on the phone
Everybody,
I've had a request from my manager that I figure out how to get our Asterisk 13.x system using chan_sip to be able to display on the Polycom VVX series phone display (firmware 5.9.5), when an extension is called and the person on the other end is on the phone.
He said, "Our old Analog phone system could do it, how hard can it be?"
I've gone down the path of trying
2014 Mar 14
1
Working Config for Polycom VVX and Auto Answer
Hi -
Just wondering if anyone has gotten a Polycom VVX phone to
successfully do an Auto Answer with asterisk. I have an older
generation of Polycom phones that do this just fine, but I can't seem
to make the VVX phones work.
I tried the guide here:
http://community.polycom.com/t5/VoIP/FAQ-How-can-I-change-my-Ringtone-or-Ring-in-a-special-manner-for/td-p/5167
And I have this in my diaplan:
2015 Oct 19
2
Modify Contact in PJsip
Do you know if this can be achieved with the standard sip stack in asterisk?
Kind Regards
Andrew Colin
Converged Telecoms (Pty) Ltd.
Licensed Telecoms Operator : (0258/IECS/JAN/09) (0258/IECNS/JAN/09)
Switchboard: +27 (0)10?591 4600
Email:? andrew at convergedgroup.net
Web: ?http://www.convergedgroup.net
75 Witkoppen Road, Northriding, Johannesburg, 2169
P O Box 7246, Weltevredenpark, 1715
This
2012 Jun 08
2
Running Asterisk on VMware ESX
A point of clarification. The "get started video" on asterisk.org says that running asterisk on a virtual platform is a popular option. "Go for it!" the presenter says. However, the 2011 third edition of Asterisk, The Definitive Guide says they don't recommend it for production use, though many do so successfully. I also don't find may recent discussions of this
2015 Jul 08
2
Call Return
Hi Guys
I am trying to write a macro for a call return so for example
Anyone in the company transfers a call to another extension and it is not
answered etc it must return to the person who did the transfer
I have got it working but if the call originates externally for example
someone calls in to the switchboard and they transfer it then it tries to
return to the outside caller.
As
2015 Oct 19
2
Modify Contact in PJsip
Hi Joshua
If i put the default_user option per endpoint would it work??
So what exactly does the contact_user option do?
I know that in freeswitch there is the option extension-in-contact.We ?basically need to achieve the same functionality?
Thanks<div>
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2014 Nov 11
1
/etc/locale.conf is ignored
It seems that /etc/locale.conf is ignored in Centos 7.
As a traditionalist who prefers things sorted lexicographically rather
than indiscriminately with case ignored and dates to be displayed in
the form "Sep 11 2008", I have always added lines to this file:
$ cat /etc/locale.conf
LANG="en_US.UTF-8"
# Fix collating sequence for sort and ls
export LC_COLLATE=C
#
2019 Jan 15
2
MWI Delayed on Polycom VVX phones
Hi all,
When moving from a self compiled Asterisk 13.23.1 to Asterisk 13.24.0, has resulted in a MWI clearing delay of around 5 minutes.
After listening to a voicemail and deleting it, the Polycom VVX 601's MWI light is left on for around five minutes, before clearing.
Installing Asterisk 13.24.1 did not fix this.
Moving back to 13.23.1 allows the MWI to clear immediately. I see a note in
2004 Apr 23
4
call initiation
Users withing the office can dial a 3 digit extension and that will ring
a phone. The problem I'm running into is you have to press xxx then
press 'send or 'dial'. The pbx doesn't recognize a 3 digit number as an
internal extension and automatically dial it the user has to initiate
that call. Asterisk automatically initiates calls w/ 9+7 digits and LD
calls,