Too bad... phone paging just isn't reliable for getting important
messages out. It can't properly handle open areas, you can't hear it
when you're on the phone, etc.
They'll spend the $ when something bad happens and everybody doesn't get
the message. Funny how disasters free up spending magica
Josh,
What happens with this DNS setup if the DNS server is down at one of the
sites? The phones would be configured with the DNS servers from the
other sites I would imagine but how does the 'match' effect what they
are returned for DNS values?
Thanks,
Mike
-Original Message-
F
Luckily, there is an unlimited supply of extension on the PBX. Cant beat
the price either.
From: Picher, Michael [mailto:mpic...@cmctechgroup.com]
Sent: Monday, December 21, 2009 11:44 PM
To: Todd Hodgen; Rene Pankratz
Cc: sipx-users@list.sipfoundry.org
Subject: RE: [sipx-users] Is a user ab
That might work... for polycom phones you could register that second line to
the phone as well and then the voicemail light will work.
Mike
From: Todd Hodgen [mailto:thod...@verizon.net]
Sent: Tuesday, December 22, 2009 2:39 AM
To: Picher, Michael; 'Rene Pankratz'
Cc: sipx-users@list.sip
How about a workaround. Assign a secondary extension with each user, and
give it a voicemail box. Then when they want to have the voicemail enabled,
they can simply go to the portal, and forward their extension to the
extension with the voicemail box.
Ive not tried this, but if you forward a
Rene,
So, the answer is "No, it is not an option now." If it's something you'd like
then you should probably create a feature request. Right now only the Admin
can enable / disable VM.
Mike
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org]
Well, I agree with your considerations about business systems and the
problems if any user would be able to control these features.
My problem is, if you buy a simple answering machine, then this machine will
have a button to enable/disable it. And if you spent some more money you
will also be abl
Nicely done Josh.
The location based gateways also make all of that work properly. Back
before location based gateways I had a trick for using multiple DNS
servers and creating 'phantom' gateways that allowed me to use the same
gateway names in the dial plan but resolve those gateway names to
dif
I too tested Skype under the beta program with sipXecs and it was easy breezy...
Mike
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of M. Ranganathan
Sent: Tuesday, December 22, 2009 1:46 AM
To: he...@tcd.ie
Cc:
Under system/dial plan/ site to site, you would have the 4 + 3 digits, and
then "" and suffix. That will give the resulting call over the trunk group
a digits dialed of the 3 digits.
But, under Devices/Gateways/unmanaged Gateway there is another section for
dialplan. On that screen, at the top,
If you are behind NAT at both locations use sipXbridge to do this.
See the 1 line table below the table of ITSP's in the WIKI in the
section titled 'Required Parameters for dialing another sipXecs system'.
http://sipx-wiki.calivia.com/index.php/SipXbridge_Overview_and_Configura
tion
Mike
2009/12/22 Sen Heng :
>
> Hi Rangn,
>
> Could you help with Skype-SIP trunk please. Dim is the first people I see in
> the list success connected Skype. I followed his instruction but no avail.
Not sure why you are having problems. We have tested successfully with
Skype ( rather extensively in fa
Did you put the prefix of 42 in you dial plan for the new gateway?
Here is how to use sipxviewer.
Got to /var/log/sipxpbx delete all the files in that area, at least that
is what I do so I don't have to wade through a bunch of old logs
Make your test calls
In /var/log/sipxpbx run merge-lo
When you select Gateway, select Unmanaged Gateway. You will be given the
opportunity to enter the IP address of the far end. There are notes below
it - "To interconnect two VoIP systems using SIP enter the IP address or
fully qualified name of the other system."
With this, each side is giving pe
Hi Rangn,
Could you help with Skype-SIP trunk please. Dim is the first people I see in
the list success connected Skype. I followed his instruction but no avail.
Do you have any suggestion about internal SBC file or the other things?
Thanks a lot,
Sen
-Original Message-
From: dimitris
Calling into a sip system would not require credentials. Calling out DOES,
which is why the user has to register first.
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431
Email: tgrazi...@myitdepartment.net
LAN/Telephony/Security and Control Systems Hel
Start with your DNS. Can you see the the Other side from each site?
When you connect these site to sites up, they are assumed to be trusted, so
no password is required. You couldn't get in if they didn't explicitely let
you in.
Your dial plan seem accurate.
-Original Message-
From: sip
You could add the public ip address as the gateway to the remote system and
have your dial plan use that gateway.
Site 1 ip 1.2.3.4
Site 2 ip 2.3.4.5
Site 2 uses a dialplan to dial site
On site 2, create an unmanaged gateway to dial, ipaddress 1.2.3.4. Etc.
Tony Graz
Hello!
I'd like to connect two SipXecs systems (system A and B) such that a user
on A could dial an extension on B and connect without a provider. Both
systems behind a NAT.
I have followed this:
http://sipx-wiki.calivia.com/index.php/HowTo_interconnect_two_sipX_PBXs
And still cannot get it go
I think I haven't seen a 64 bit iso in a while. Yes, using the 64bit yum
repo to install it is the right way.
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431
Email: tgrazi...@myitdepartment.net
LAN/Telephony/Security and Control Systems Helpdesk:
Tele
If anyone is interested or wishes to modify/edit/add to it, I posted my
document on setting up BIND views for location based DNS SRV in the new
confluence wiki. I realize it will probably get moved at some point, but
I wanted to get it online before I started traveling for the holidays.
http://
On Mon, 21 Dec 2009 22:11:18 -0500, Tony Graziano wrote:
> The answer is it will work on a 64bit os also, as you simply install using
> the 64 bit repo.
Tony, I can't recall 100% but I thought I was told that the ISO is 32bit only
right? I've always installed using the ISO only, which means I wou
The asterisk solution works much better with the money they are allowing
me to put toward paging: $0
Tony Graziano wrote:
> Well, putting in a valcom paging system and connecting it to overhead
> speakers works well.
>
> Tony Graziano, Manager
> Telephone: 434.984.843
The answer is it will work on a 64bit os also, as you simply install using
the 64 bit repo.
More importantly you would look at the upcoming versions from a performance
perspective to plan hardware accordingly.
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984
I know it would work on 64 bit hardware, but i meant 64 bit CentOS as
opposed to 32 bit that the ISO loads.
On 12/21/2009 9:01 PM, m...@grounded.net wrote:
> I asked a similar question long ago because hardware choices are important
> depending on what you're up to.
>
> I only have 64bit hardwar
I asked a similar question long ago because hardware choices are important
depending on what you're up to.
I only have 64bit hardware and have installed using the ISO several times now.
I also have a dev server on Centos 5.x with source and it all works fine.
Mike
> this? Is 64 bit fully sup
Well, putting in a valcom paging system and connecting it to overhead
speakers works well.
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 think its also contextual how involved you will be with the 32 or 64 bit.
For instance, in some builds, the 32 bit might be offered more frequently in
some distros. This makes testing some feature a little easier than waiting
for a different build. I'm not sure it matters though.
Absolutely
Once you add the sipX repo, install it (yum install sipXecs), and
reboot the server then it's essentially an ISO install. You simply have
to rerun the setup script (sipxecs-setup-system) and it will all be set
up. Use CentOS 5.4 (that's what I use)
mkitchin.pub...@gmail.com wrote:
I found a quirky way to get around this, and it's a bit of a kludge,
but it has worked in initial prototype testing. I hacked together some
asterisk dialplan to record a message, then call several different
paging groups, one after another, with the same message. It takes a
little while to make
I thought I throw this out there under a different subject since it sort
of split from my original question.
From here:
http://sipx-wiki.calivia.com/index.php/SipX_on_Different_Platforms
It seems like 64 bit is a fine option. I could have sworn the download
page used to say sipx was tested and f
Make sure there are no special characters in description (~...@#$%^&*,
etc...) or in the AA name... also the only other time I have had a
problem with AA's is if I had invalid options specified in them.
Mike
From: arda savran [mailto:ardasav...@yahoo.com]
Sent: Monday, December 21, 2009 7
This is the second time we are getting that recommendation but the thing is
that this is not the first time we are creating an autoattendant in this system.
We had no issue with the first two times. We are trying to figure out why it
didnt work this time...
--- On Mon, 12/21/09, Picher, Michae
Scott,
I'm looking forward to the new Wiki and I'm sure the rest of the
community will as well. This will be a good opportunity to reorganize
the content so things are much easier to locate.
On the organization of the Wiki, I guess I'll need to see how things
roll-up from version to version in t
Yeah, my dial plan is very simple.
Dialed Number
Prefix
and
Any number of digits 0 digits 1 digits 2 digits 3 digits 4 digits 5
digits 6 digits 7 digits 8 digits 9 digits 10 digits 11 digits 12
digits 13 digits 14 digits 15 digits 16 digits 17 digits 18 digits
Did you create a dialplan for 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:
http://w
Hi Dim,
Thanks a lot for prompt response. I did follow your instruction. But I still
got "ITSP timed out".
Would it possible you could send your internal SBC XML file to me please ?
Thanks in advance,
Sen
-Original Message-
From: dimitris [mailto:dim_ha...@yahoo.gr]
Sent: 2009年12月22日
Sen Heng wrote:
>
> Hi Dimitris,
>
> I bought a channel and one skype-sip online number. I put
> username/password into skype business control panel but my profile
> shows “SIP user not yet registered at sip.skype.com”
>
> I would appreciate if you could share some experience.
>
> Thanks in advan
Sorry, I mean I use Skype business username/password like
99051x/SBXA from business control panel...
Thanks,
Sen
-Original Message-
From: Tony Graziano [mailto:tgrazi...@myitdepartment.net]
Sent: 2009年12月22日 5:14
To: he...@tcd.ie; sipx-users@list.sipfoundry.org
Subject: Re: [si
You can't use a skype username/password. It needs to be a skype business
username/password.
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431
Email: tgrazi...@myitdepartment.net
LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: 434.984.842
Hi,
I try to setup skype-sip trunk.
On Sipx side, I use 3.sip.skype.com, 5060 as port , transport= auto. The
Sipx is behind firewall use internal ip add 192.101.1.4.
I use "sipxbridge-1" as ROUTE. Then I put in skype username/password.
On Skype side, I put in my public IP address. I al
Hi,
I try to setup skype-sip trunk.
On Sipx side, I use 3.sip.skype.com, 5060 as port , transport= auto. The
Sipx is behind firewall use internal ip add 192.101.1.4.
I use "sipxbridge-1" as ROUTE. Then I put in skype username/password.
On Skype side, I put in my public IP address. I
If possible can you (at the very least) give me access to start writing
in my personal space so I can start creating documents (and move them to
the appropriate places later)? I already have one written up to go in
the non-version-specific user docs area.
I think the proposed layout will work v
I'm definitely not using virtual. I'm moving everything else in my data
center (except by largest HP UX Oracle DBs) to VMware, but not sipx.
Maybe/hopefully one day, but not right now.
On 12/21/2009 2:16 PM, Josh Patten wrote:
> I only have 87 registrations on this system currently with an avera
Don't know, but I imagine if you aren't using any special services or
doing any BLF monitoring that you'd only be limited by your bandwidth.
Josh Patten
Assistant Network Administrator
Brazos County IT Dept.
(979) 361-4676
On 12/21/2009 2:07 PM, Abdul Hakeem wrote:
> Josh,
> Just out of curiosi
I only have 87 registrations on this system currently with an average of
about 8 concurrent calls, but I will soon be adding another 200 to the
mix with the addition of our Sheriff's office and Jail. I'm also running
it on bare metal hardware, I didn't want to risk virtualization on a
voice pla
Hi Dimitris,
I bought a channel and one skype-sip online number. I put username/password
into skype business control panel but my profile shows "SIP user not yet
registered at sip.skype.com"
I would appreciate if you could share some experience.
Thanks in advance,
Sen.
On Thu, Dec 3, 20
One of the weaknesses of many open source projects is documentation,
and I don't think that sipXecs can claim to be an exception (yet).
We've got a lot of material in the current wiki [1], but it has a
number of problems:
* Much of it is very dated.
There are warnings about problems that were
Thank you. I was under the impression that 32 bit was preferred/better
tested and I should stick with the version of CentOS that came with the
newest ISO install. Am I completely mistaken?
How many users do you have on that hardware? Given my understanding
regarding the 32 bit OS, I hadn't plann
I would install the x86_64 version of CentOS, run a full update, then
add the sipX repo in and install sipX, then you get to use all of your
RAM and if any of the sipX components were compiled with 64 bit
optimizations then you'll have a performance advantage. I use an HP
DL-360 G5 with 8 cores
I would assume there are no problems, but just wanted to double check to
see if anyone had any issues running the sipx ISO install on these
blades. I searched BL460 and Centos 5.2 and RHEL 5.2. I didn't see any
major issues. I know I'll need to go through some minor tweaks to get
all the HP man
Hi Mike:
SIPez builds custom end point applications using the sipXtapi user agent that
we maintain. We have a IVR server engine upon which we could write an auto
dialer as you describe. Please let me know if you would like to discuss a
possible solution.
Cheers,
Dan
--- On Fri, 12/18/09, m..
You could vote for it... I think the source routing feature if/when
implemented will do what you need.
http://track.sipfoundry.org/browse/XX-5077
Thanks,
Mike
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behal
Last year I had an Asterisk setup where the sales team wanted to have
calls to the sales extension routed to the sales team based on region of
the incoming NPA.
We did this with a custom script in Asterisk which used a MySQL table
NPA -> Internal Region -> Extension. If all else failed, the use
I believe that VM must reside on the same server as sipxconfig.
Mike
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of arda savran
Sent: Monday, December 21, 2009 9:12 AM
To: sipx-users@list.sipfoundry.org
Subject: [sipx-users] autoatt
No, I think that's the way it is on all of the Polycom phones... I
think it's a bit odd that it isn't in there at the group level.
Mike
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Burden,
Mike
Sent: Monday, December 21, 2009 10:
I see a couple of things that might be causing an issue. First, does
your ITSP take 10 or 11 digits? 10 digit would be 847-382-7754 and 11
would be 1-847-382-7754, you are sending 11 to them (not sure if this
matters). Second, you are sending your internal caller ID to the ITSP,
which is most l
I don't understand why if you create a new group and add a phone to it why
the moh uri would not be filled in. I've never seen this issue (unless the
DNS or something is actively wrong).
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431
Email: tgrazi...@
If you make changes to an individual phone, it will overwrite your Group
Configuration. If you configure it via a browser, it will overwrite the
sipXecs individual configuration.
It actually works very well at a group level, and no configurations are
required at an individual level unless you
Josh,
Here is a packet trace.
Duane
-Original Message-
From: Josh Patten [mailto:jpat...@co.brazos.tx.us]
Sent: Monday, December 21, 2009 10:00 AM
To: D. R. Lang; sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] ITSP outbound 602 Error
It sometimes does. I would need a more co
It sometimes does. I would need a more complete trace to see if
everything is right. Please don't forget to CC the list when responding
to a mailing list post.
Josh Patten
Assistant Network Administrator
Brazos County IT Dept.
(979) 361-4676
On 12/21/2009 9:54 AM, D. R. Lang wrote:
> Josh,
> I
You need to inform your ITSP that they need to point SIP traffic to you
on port 5080 and not 5060.
Josh Patten
Assistant Network Administrator
Brazos County IT Dept.
(979) 361-4676
On 12/21/2009 9:47 AM, D. R. Lang wrote:
> I have version 4.04 running internally. I can call the external trunk
I have version 4.04 running internally. I can call the external trunk lines
and they forward calls to the internal hunt groups. When dialing out I get
fast busy signals. When packet tracing the output of the gateway I see the
following results.
Registration to ITSP 200, viewing the results b
Good morning,
I deleted all of my phones and re-discovered them in order to undo any
individual configuration and make sure that all of my phones were
configured using the settings in the Phone Group.
All of the phones are Polycom IP550, and I have double checked that they
are all part of t
I have updated my logging level from INFO to DEBUG, and I have also
started a tshark trace with a circular buffer (tshark -b duration:1800
-b files:5 -w outfile.pcap), so I've always got two hours worth of
packet capture.
Next time it happens, I should have more info to pass along.
Mike
Good Day,
We have a cluster of 4 SIPX servers,
They are all connected to the same Layer2 switch and the system is operational.
The first two servers are running all the services except conferencing and
voicemail. The third server is conferencing and the fourth server is configured
as dedica
Sorry about that Ranga was trouble shooting all weekend with Tony and
got the diagnosis to a network issue at the customer site as I took the
system back to our office and threw it on a Linksys router with an
internet connection and it worked fine. The way the customer has their
network setup is by
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