I am seeing success as well. Let's hope the remote worker issue with
3.2.3 that's blocking the 4.2 release will be resolved quickly.
Eric Varsanyi wrote:
> I tested the shared line feature on Polycom firmware 3.2.3 with sipxecs SVN
> 18544 and I can't find any problems (!).
>
> A user is on 10 p
I tested the shared line feature on Polycom firmware 3.2.3 with sipxecs SVN
18544 and I can't find any problems (!).
A user is on 10 phones and (some 650's and some 335's) and I have no problem
putting calls on hold on one phone and picking up on another. There's a weird
short delay (500ms?) an
Yes. I got that. I have a 5 year old and a 3 year old, so we are watching those
again.
Sent via BlackBerry from T-Mobile
-Original Message-
From: Tony Graziano
Date: Tue, 13 Apr 2010 21:04:52
To: Matthew Kitchin (public/usenet)
Cc: Josh Patten;
Subject: Re: [sipx-users] REFER problem,
On Tue, 2010-04-13 at 17:52 -0700, Shawn Westerhoff wrote:
> Thanks but we are setup as suggested:
>
>
> > 1. Configure the siptrunking role for the server (enable it).
> > 2. No configuration is needed on the SBC.
> > 3. Configure the gateway as a siptrunk, making sure you use the SBC
> > route.
I am still the person with the most humor on the list. Come on, ever
watch road runner as a kid? Acme. Wile E. Coyote (super genius). Give
me some credit.
On Tue, Apr 13, 2010 at 8:53 PM, Matthew Kitchin (public/usenet)
wrote:
> http://www.acmepacket.com/default.asp
> I know several large provide
BUT if the ITSP is not sending on port 5080, it bypasses sipXbridge.
You need to check with the provider and ask if they can send on port
5080. If not, conside an ingate SBC. We install them and they work
well for these instances.
On Tue, Apr 13, 2010 at 8:33 PM, Shawn Westerhoff wrote:
> I see i
One of the very first things would be to simply change the color scheme so that
light text is not on light background.
Who's idea was that? :)
On Tue, 13 Apr 2010 20:01:27 -0400 (EDT), Administrator wrote:
>
>
> Content-Type: text/plain;
> charset="utf-8"
> Content-Transfer-Encoding: 8bit
> Org
http://www.acmepacket.com/default.asp
I know several large providers use them. I believe Verizon does.
On 4/13/2010 6:45 PM, Tony Graziano wrote:
> What the heck is an
> ACME. Was it installed by a long nosed tech name Wile E. Coyote?
>
>
> On Tue, Apr 13, 2010 at 7:25 PM, Josh Patten wrote:
>
I see in the doc:
"Supports call transfers locally: Call transfers are supported without sending
the REFER to the ITSP. Therefore, it can handle both blind and consultative
transfers and it is possible to transfer in or outbound calls via an ITSP back
out to the ITSP (hair-pinned transfers)."
Content-Type: text/plain;
charset="utf-8"
Content-Transfer-Encoding: 8bit
Organization: SipXecs Forum
In-Reply-To:
X-FUDforum: 08063afcdd00a6e76393c5b9527381e8 <44587>
Message-ID:
If anybody would like to create a new theme for use in the
forums we would welcome the involvement.
We are mor
Correct...
1. Configure the siptrunking role for the server (enable it).
2. No configuration is needed on the SBC.
3. Configure the gateway as a siptrunk, making sure you use the SBC
route. Point this trunk or gateway to the ACME thingy or whatever you
had been pointing to before.
4. Change your d
sipXbridge, the SBC built into sipX, should be handling this for you.
Your ITSP must be able to send the SIP traffic to port 5080 on your
server.
Shawn Westerhoff wrote:
Thanks Tony,
The SIP provider is O1 Communications in Sacramento, CA and they say they can not support REFER method with
Thanks Tony,
The SIP provider is O1 Communications in Sacramento, CA and they say they can
not support REFER method with the ACME (1000 I think) they are using. We
connect to O1 via private MPLS so no NAT is involved, no Internet.
We are on 4.04. Are you saying we can change the method used to
What version are you using? Who is the ITSP and what SBC are you using
to separate sipx from the Internet?
sipXbridge is capable of doing that in 4.0.4 (current stable version),
with a properly configured firewall. 4.1.7 offers only minor
improvements to sipXbridge.
On Tue, Apr 13, 2010 at 6:52
We use a SIP provider that can not handle REFER during a call
transfer. We are seeking a way around this short of moving to 4.1.7
development release where we understand the issue has been fixed by
using a RE-INVITE (or INVITE again, not sure if RE-INVITE is even a
term we should be using).
Our p
In upcoming 4.2 there are three levels of detail the admin can choose for
the email notification.
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431
Email: tgrazi...@myitdepartment.net
LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: 434.9
Version: 4.1.16
Feature: Voicemail notification via email
Request #1
Currently, email notifications are formatted as follows:
From: Voicemail Notification Service
Subject: Voice message from ()
I think it will be more useful if formatting is as follows:
From:
Subject: Voicemail
:-P
It's all good, I just hope that when I'm banging my head against the
wall over some obscure thing that everyone will be patient with me.
Josh Patten
Assistant Network Administrator
Brazos County IT Dept.
(979) 361-4676
On 4/13/2010 3:55 PM, Matthew Kitchin (public/usenet) wrote:
Now I se
Now I see. I'm going to really frustrate you now. I don't see it because
I'm trying on my phone. I don't monitor any extensions on my phone. I
didn't realize you had to be monitoring something for that option to
show up. I just added a test line on my phone, and now I see it. I have
sent the in
ok did the trace and vieiwing via sipviewer.
I'm seeing 487 Request Terminated. That is where the call gets dropped and
does not get forwarded to VM or next hunt.
On Apr 13, 2010, at 4:13 PM, Scott Lawrence wrote:
> On Tue, 2010-04-13 at 15:43 -0400, Francis Tinio wrote:
>> my bad
>>
>>
o_O
How can you be monitoring the presence of the park queue and not have
that option?? It just does not make sense.
Have you cleared out all locally stored preferences with the "Polycom 4
finger salute ( 4 6 8 * all at the same time for 3 seconds)"?
There has to be something wrong at the p
Hmm. Step 5 doesn't exist for me. After 4, I have the list of ring
tones, but I don't see where I can select "Attendant Calls"
This site seems to say the same thing you said
http://knowledgebase.polycom.com/knowledgebase/End%20User/Tech%20Alerts/Audio/SoundPoint_IP_Enhanced_BLF_QT37381.pdf
but it
On Tue, 2010-04-13 at 15:43 -0400, Francis Tinio wrote:
> my bad
>
> I'm not doing it in the hunt group.
>
> I'm doing this in the ghost extension of 5007.
>
> it's not even pointing to huntgroup for now since I'm triyng to figure
> out why call gets disconnected after 20 seconds.
Trace i
my bad
I'm not doing it in the hunt group.
I'm doing this in the ghost extension of 5007.
it's not even pointing to huntgroup for now since I'm triyng to figure out why
call gets disconnected after 20 seconds.
On Apr 13, 2010, at 3:41 PM, Tony Graziano wrote:
> You can't do that.
>
>
You can't do that.
First destination
5001
5002
5003
Do not do forwarding or voicemail. FALLBACK DESTINATION of 85001 (5002's
VM). If you do it the way you described you WILL break the nesting exception
and it will not work.
LOOK at the huntgroup config at the bottom. I'm repeating myself too ma
you lost me there.
1 should dia1 5001 first right? then if no answer go to number 2, which is
85001?
On Apr 13, 2010, at 3:30 PM, Tony Graziano wrote:
> #2 is backwards. Should be 5001. No answer should be fallback destinatuion
> of 85001.
>
> Tony Graziano, Manage
#2 is backwards. Should be 5001. No answer should be fallback destinatuion
of 85001.
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:
On Tue, 2010-04-13 at 14:57 -0400, Francis Tinio wrote:
> This is how I configured the incoming calls.
>
>
> 5007 - ghost - no phone, no line, alias of 1234567890
> call forwarding rules of 5007
> 1. workhours - dial 5001 for 30 secs
> 2. workhours - dial 85001 for 1 30 sec if no response
> 3. a
This is how I configured the incoming calls.
5007 - ghost - no phone, no line, alias of 1234567890
call forwarding rules of 5007
1. workhours - dial 5001 for 30 secs
2. workhours - dial 85001 for 1 30 sec if no response
3. afterhours - dial 5011
5001 - my extension
5011 - afterhours recording
ok I think I got it.
How many seconds is the default before a call goes to VM?
On Apr 13, 2010, at 2:25 PM, Tony Graziano wrote:
> You really need a call trace to see where the call is going.
>
> On Tue, Apr 13, 2010 at 2:24 PM, Francis Tinio wrote:
> ok i disabled the softkey.
>
> in th
yeah working on that, currently don't have physical access and ssh is disabled
On Apr 13, 2010, at 2:25 PM, Tony Graziano wrote:
> You really need a call trace to see where the call is going.
>
> On Tue, Apr 13, 2010 at 2:24 PM, Francis Tinio wrote:
> ok i disabled the softkey.
>
> in the hun
You really need a call trace to see where the call is going.
On Tue, Apr 13, 2010 at 2:24 PM, Francis Tinio wrote:
> ok i disabled the softkey.
>
> in the hunt group there's o duplicate.
>
> still the same result. i can reject the call and it goes to fallback...but
> if i let it ring, it turns
ok i disabled the softkey.
in the hunt group there's o duplicate.
still the same result. i can reject the call and it goes to fallback...but if
i let it ring, it turns to busy tone
also, the phone settings are pretty much default settings and I saw a
server wide forwarding feature, defaul
Don't do that. Disable that softkey.
In your hunt group make sure you have no line number showing up more than
once.
Make your fallback destination a dump into voicemail by prefixing it with an
"8".
On Tue, Apr 13, 2010 at 2:05 PM, Francis Tinio wrote:
> I enabled the softkey forward function,
I enabled the softkey forward function, but it's not forwarding.
On Apr 13, 2010, at 1:46 PM, Scott Lawrence wrote:
> On Tue, 2010-04-13 at 13:31 -0400, Francis Tinio wrote:
>> Thanks, I'll get the trace info once I get to login to the box
>> directly since I disabled remote access.
>>
>> Anywa
Since installing a build based on SVN 18544 I've been getting messages from
cron about a report job failing:
requesting call stats from -01-01T00:00:00+00:00 (class = DateTime)
/usr/lib/ruby/1.8/net/http.rb:560:in `initialize': Connection refused -
connect(2) (Errno::ECONNREFUSED)
fr
On Tue, 2010-04-13 at 13:31 -0400, Francis Tinio wrote:
> Thanks, I'll get the trace info once I get to login to the box
> directly since I disabled remote access.
>
> Anyway, FWIW, if I reject the call from the polycom, the call gets
> diverted to VM successfully. But if I don't pick up, it gets
Thanks, I'll get the trace info once I get to login to the box directly since I
disabled remote access.
Anyway, FWIW, if I reject the call from the polycom, the call gets diverted to
VM successfully. But if I don't pick up, it gets a busy tone.
On Apr 13, 2010, at 1:01 PM, Scott Lawrence wr
I wanted to see if anyone is using a call accounting package with SIPX.
The software would be used in a law firm environment to bill outgoing
calls to clients by user.
I recall the topic being discussed before, but would like to know what
other sipx users are using.
Thank You,
Austin Cu
On Tue, 2010-04-13 at 12:01 -0400, Francis Tinio wrote:
>
> Our system has been working flawlessly for a couple weeks now.
> However, it has stopped forwarding calls to the VM system when not
> answered.
>
> I have not made any changes to the config for a while now, so I'm a
> bit unsure why VM s
yup.
I can have the 1st forward to 85001 and it will go straight to VM.
However, anything beyond that 1st forward does not work.
I tried 5001 as the 1st, the fallback of 85001 if not answered, I get a busy
tone.
Weird since this was working correctly before.
On Apr 13, 2010, at 12:59 PM, T
You are sure the line has voicemail permissions and that it does not a
forwartd on 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
I have disabled hunt group for troubleshooting purposes.
Right now when the number is dialed, it rings a specific extension and that
extension rings fine. And I assume by default that will forward to that
extensions VM if not picked up. However, it does not and goes to busy tone.
I have tried
Well, how you have your hunt group DOES matter.
You can never call the same extension twice.
Use a fallback destination for the hunt group. Make the fallback destination
8+mailboxnumber (example box 200, fallback destination is 8200).
On Tue, Apr 13, 2010 at 12:46 PM, Francis Tinio wrote:
> I
I can dial 101 and it goes to VM system
if I do top -U sipxchange cpu usage is close to none at 0.2% usage
I even tried disabling hunt for now and just forwarding to my extension
directly, after a couple rings it just goes busy and does not get forwarded to
VM still.
On Apr 13, 2010, at 12:05
1.) Can you call the VM system locally?
If you run...
top -U sipxchange
2.) Is your CPU usage running at a normal level? Are you using any SWAP?
3.) Is this for any voicemail transaction or just hunt group destinations?
On Tue, Apr 13, 2010 at 12:01 PM, Francis Tinio wrote:
> Hi.
>
> Our sys
Hi.
Our system has been working flawlessly for a couple weeks now. However, it has
stopped forwarding calls to the VM system when not answered.
I have not made any changes to the config for a while now, so I'm a bit unsure
why VM stopped working.
Any suggestions on how I can troubleshoot the
On Tue, Apr 13, 2010 at 10:08 AM, Francis Tinio wrote:
> Hi.
>
> We are not using ACD as per suggestions that this is still buggy. So we
> are using hunt groups and the calls go directly to hunt. Since we
> configured sipx this way, when someone calls, he is not greeted and just
> gets a ringin
Hi.
We are not using ACD as per suggestions that this is still buggy. So we are
using hunt groups and the calls go directly to hunt. Since we configured sipx
this way, when someone calls, he is not greeted and just gets a ringing sound
until someone picks up.
Question is, can I change this
I second everything Tony said (especially the advise that you upgrade)
> Server 2 is my ISDN gateway. I.E I configured a sip trunk on sipXecs
> which connects to the SIPControl Software
When you say 'a sip trunk' do you mean that you configured the ISDN
gateway as a gateway, or that you configu
Nice that its not arbitrary in timing. I would suggest two things:
A sip trace of the failed call.
A debug log at your ISDN gateway.
It needs to be established why the disconnect is happening and where it is
coming from. I would urge you to update to 4.0.4 first.
Tony
Hello,
For every user I configured the following call forwarding rule:
"Extension 861 will ring first -> Always -> Enabled -> At the same time ->
forward to *Cell Phone Number* -> ring for 30 seconds.
User A (Outside Number) calls User B (sipXecs user)
User B doesn't pick up. Therefore the cal
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