If you use SIP trunks, they will automatically know when the trunks can't
connect to the PBX. You can have your provider set it up to forward on
failure, it would be automagic for you.
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Ton
On Wed, Jul 14, 2010 at 11:02 PM, Gary Luca wrote:
> Thank you for the link regarding FTP. Do you know of any place where I can
> find detailed instructions on how precisely to configure sipX for whatever
> it needs in order for FTP to work for phone provisioning? On the local
> network I use TF
Ok. So I just got a whole TON of replies with questions to answer. Here
goes:
Nathaniel...
DNS/DHCP at the remote site are handled by the Westell ADSL2 modem. It
deals out just IP, Subnet mask, gateway (itself) and DNS (also itself). DNS
relays to what I imagine are Verizon DNS servers. The r
On 7/14/2010 9:46 PM, Douglas Hubler wrote:
> On Wed, Jul 14, 2010 at 5:29 PM, Matthew Kitchin (public/usenet)
> wrote:
>
>> Where is the template file I would need to look at? I'm familiar with
>> polycom_sip.cfg, and I don't see anything there.
>> I have only changed 1 or 2 polycom settings
On Wed, Jul 14, 2010 at 5:29 PM, Matthew Kitchin (public/usenet)
wrote:
> Where is the template file I would need to look at? I'm familiar with
> polycom_sip.cfg, and I don't see anything there.
> I have only changed 1 or 2 polycom settings total that had to be done
> outside of the sipx web inter
From: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.org] On Behalf Of Gary Luca Jr
[garyluc...@gmail.com]
Sent: Wednesday, July 14, 2010 8:24 PM
The phone fails to register with sipX.
___
D
This is called a PIC code . Your carrier should be doing this for you (your
lec anyway).
No. Just add that at long distance dialing rule as a prefix to send to the
gateway.
The LEC should know who your L D carrier is and do this for you. It's
nonsensical. The reason a user would dial it is to dia
I just found out we are supposed to add a '1010xxx' prefix to our long distance
calls. Is the correct way to do this by entering a separate gateway to our
Patton - and use the Prefix field in the Dial Plan?
[cid:image001.png@01CB2394.03BB9C70]
Or is there a better way to accomplish this?
On 2010-07-14 20:18, Miguel Jesús Díaz Macedo wrote:
> Hello all
>
> Can any one describe an example for the sipx-trace command line
> utility options
means "any string that would be found in the message of interest"
so if you wanted a trace showing all calls from FooBar brand phones, you
mig
That's funny. After rereading my message I realized I was vague. The symptom
was mentioned in the subject only and I forgot to reiterate it.
The phone fails to register with sipX. I'm on my phone right now but I'll check
that link when I get in front of a computer.
On Jul 14, 2010, at 8:13 PM
User lines are numbers:
I did re-try it internally and it appears to be working. Externally is another
story - I tried it by dialing 7485 via my cell phone (dialed the whole number -
goes to PRI - rings his extension - dialing *787485 simply kills the connection
on my cell phone and at both
Ummm...
How are you resolving with xlite?
Why don't you run a dyndns agent somewhere on the LAN and use real records?
The phone MUST be able to lookup SRV and it must point back to your sipx
server.
Your issue is dns related I think. You need to explain your xlite
configuration and WHY it works.
Hello all
Can any one describe an example for the sipx-trace command line
utility options
Thanks in advance
Miguel Díaz
___
sipx-users mailing list sipx-users@list.sipfoundry.org
List Archive: http://list.sipfoundry.org/archive/sipx-users
Unsubscribe:
Um I dunno what you mean. The user "line" must be a number. If sipxconfig
shows a name all bets are off.
What firmware?
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431
Email: tgrazi...@myitdepartment.net
LAN/Telephony/Security and Control Systems He
On 2010-07-14 17:16, Gary Luca wrote:
> I already posted this on the forum, but haven't gotten any replies
> yet. I know i'm not supposed to send to the list as well, but i'm not
> sure if it got sent out. Now that i'm aware of the list i'll use this
> from now on.
>
> Good evening all,
>
>
On Wed, Jul 14, 2010 at 4:43 PM, McIlvin, Don
wrote:
> The error occurred on both the "device file" page and "upload" page
> after clicking .. "An internal error has occurred. Click
> (here) to continue". The (here) brought me to the SipXecs config 4.2
> login page. All the other device file sets
On Wed, Jul 14, 2010 at 5:23 PM, Tony Graziano
wrote:
> 2. You need to provision the phone remotely via ftp (best choice) or tftp.
Be sure to consider this
http://track.sipfoundry.org/browse/XX-8401
community build has the fix, but if you're not running the community build
http://docs.google.c
What is handling DHCP/DNS at the remote site? What are your SRV records
pointing towards (what is your sip domain)? Can you resolve your sip domain
from the remote site?
http://sipx-wiki.calivia.com/index.php/DNS_Configuration
This message and any files transmitted with it are intended only
I'll check on the latency tomorrow as well. I have to imagine it's not bad as
both the main and remote offices are getting bandwidth through Verizon and they
are within miles of each other.
The outgoing proxy host name is a dynamic dns host (through no-ip.com). There
are no publicly resolvable
I would also like to pre-emptively slap myself on the wrist - Sorry Tony - beat
you to it :)
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Nathaniel Watkins
Sent: Wednesday, July 14, 2010 7:42 PM
To: Josh Pat
I tried it between 3 polycom phones as well (all on the same subnet).
Phone a calls phone b
Phone c dials *78 - call gets dropped.
To be fair - the phone to phone test - happened with alpha characters in the
userid. I have since changed those phone to the extension as the userid and
have n
The downgrade would on ly help for normal operation, but not keep it from
registering.
If you have more than 30 or 40ms of latency between the remote user and the
sipx system, it will cause the phone to ring once and go to voicemail.
It is also IMPORTANT your public DNS records are resolvable by
That sounds like your gateway is killing the call...Remind me what
gateway your using along with what firmware.
Josh Patten
Assistant Network Administrator
Brazos County IT Dept.
(979) 361-4676
On 7/14/2010 5:41 PM, Nathaniel Watkins wrote:
> I've actually not ever tried this - apparently it's
I've actually not ever tried this - apparently it's a big deal to be able to
answer someone else's phone from your desk :)
I was thinking that *78 should answer that call. When I do this - it just
kills the call (as in, disconnects the caller and no one gets the call).
I thought I read that yo
Tony. Thank you. I'm going to try re-provisioning tomorrow and downgrading back
to 3.1.3. I was going to try the downgrade anyway as my next step. So hopefully
that does the trick.
I'll let you know how I make out.
On Jul 14, 2010, at 5:23 PM, Tony Graziano wrote:
> 3 things.
>
> 1. Far en
If there is no power then YES it would. This way it can be forwarded off
site, but do as you want.
On Wed, Jul 14, 2010 at 5:40 PM, Thomas wrote:
>
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> In-Reply-To:
> X-FUDforum: 08063af
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Message-ID:
I am glad you mention this more detail. What I plan on doing
is that if the system goes down either in a power
Where is the template file I would need to look at? I'm familiar with
polycom_sip.cfg, and I don't see anything there.
I have only changed 1 or 2 polycom settings total that had to be done
outside of the sipx web interface.
On 7/14/2010 4:08 PM, Tony Graziano wrote:
> I don't know, but if you ca
3 things.
1. Far end does not need anything forwarded.
2. You need to provision the phone remotely via ftp (best choice) or tftp.
Set the public ip of the sipx system in its server menu and make sure the
port is forwarded at the sipx side.
3. Don't uise above 3.1.3RevC on a remote deployment! Ypu
I already posted this on the forum, but haven't gotten any replies yet. I
know i'm not supposed to send to the list as well, but i'm not sure if it
got sent out. Now that i'm aware of the list i'll use this from now on.
Good evening all,
I am working with sipX 4.2. I set this up about 2 months
I always suggest using a call forward variable on the main number in the
event of a disaster. This way a standard analog handset can be plugged in
and the main number forwarded off site.
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431
Email: tgrazi...@
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ok, thanks for the info
___
sipx-users mailing list sipx-users@list
I don't know, but if you cannot find it I suggest...
I would suggest looking in the template (vm) file and matching that up
against the polycom parameter and see if it is in there.
If there is no polycom parameter, its only a user setting. If there is a
parameter, then perhaps you could add the de
I'm guessing this is out there somewhere, but I can't find it.
Can sipx set the option on a Polycom Soundpoint 450? I would like to be
able to set the feature that is accessible on the handset by pressing
Menu, 3, 1, 1, 3, 3.
I'm running Sipx 4.0.4, Polycom bootrom 4.2.1 and firmware 3.1.3C spli
ALL you want is caller ID and possibly a call forward variable for the main
number. You don't want their vociemail interfering, and the handsets (sip)
won't be user friendly to hook flash to use these features, instead you'll
hang up on your callers. Don;t do it.
On Wed, Jul 14, 2010 at 4:51 PM, T
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I am designing a new system that will use POTS lines from
Verizon.
I have been on the phone with Verizon to get a Quote on
You should only be uploading the polycom ZIP file. If not, perhaps you hit
this.
http://track.sipfoundry.org/browse/XX-6856
On Wed, Jul 14, 2010 at 4:43 PM, McIlvin, Don
wrote:
> The error occurred on both the "device file" page and "upload" page
> after clicking .. "An internal error has occu
The error occurred on both the "device file" page and "upload" page
after clicking .. "An internal error has occurred. Click
(here) to continue". The (here) brought me to the SipXecs config 4.2
login page. All the other device file sets where deactivated at the
time. Only by deleting list entries s
I tried this on an EDE as well (not the community build) running rev
18948 and the issue still persisted. I attached the EDE debug at
http://track.sipfoundry.org/browse/XX-8438
Josh Patten
Assistant Network Administrator
Brazos County IT Dept.
(979) 361-4676
On 7/14/2010 11:27 AM, Douglas Hubl
On Wed, Jul 14, 2010 at 12:12 PM, Josh Patten wrote:
> *bump*
I just attached a file to the issue
backported-huntgroup-pickup-by-alias-fix.patch
that I believe josh is running. It was my attempt to backport the fix
from 4.2.1 dev to 4.2.0 stable as it was known regression, I just
don't know
*bump*
Josh Patten
Assistant Network Administrator
Brazos County IT Dept.
(979) 361-4676
On 7/13/2010 1:07 PM, Josh Patten wrote:
> http://track.sipfoundry.org/browse/XX-8438 is the new location
>
> Josh Patten
> Assistant Network Administrator
> Brazos County IT Dept.
> (979) 361-4676
>
>
> On
These are telephony terms. They are not sip OR sipxecs specific.
Huntgroup is when a call comes inand is sent to one, or multiple, extensions
to be answered. That's it in a nutshell for huntgroups.
ACD is auotmated Call Disctribution. It's more sophisticated and can
encompass NOT ringing an exten
Hello
I would like to know which is the diference between ACD and Huntgroup
and when to use one or another. Also be very usefull to know about
some internet references on this topic.
Thanks in advance
Miguel Diaz
___
sipx-users mailing list sipx-users@l
The user in question is just being setup on sipXecs - I installed a 650 at his
desk over the weekend. He is a heavy phone user - but I'm not sure if he needs
to place calls while away from his desk - or if a headset would suffice. I'm
assuming he'd prefer a wireless handset - but will verify.
It's more essential to outline the feature set desired...
Will this be a primary phone or for use when away from the desk? Will a
wireless headset with EHS work or do they need to dial while away from the
desk.
You'll find with any wireless phone the button sequences and such to to
transfers, bli
I have a user that really wants a wireless phone. Are there any recommended
wireless phone options that work well with sipxecs? I'm not partial to dect or
wifi (or if I have to go to an analog phone via an fxs gateway).
Thanks - Nathaniel
This message and any files transmitted with it are int
>
> On Mon, Jul 12, 2010 at 1:18 AM, Graeme Allen
> wrote:
> > Has the Grandstream GXV3140 been added to the
> > configuration/provisioning interface of SipX, if not, are
> there plans to add it?
>
> As an initial test, have you tried configuring it as a GXV3000?
> Can you (or anyone reading
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