create a custom rule to add the proper prefix to any 7 digit number
(i.e. prepend with your desired area code, etc.).
i typically create internal dial plans that dont require dialing 9
either, but thats me...
i also typically suspend the use of "1" and add that at the gateway,
so any 7 or 10 digi
I know this has to a common problem, but for the life of me I can't dig up
the correct documentation. I'm using sipx, voip.ms and a Linksys SPA-3102.
Long distance (9+1+10 digit number) is working fine, but I'd like to
configure local dialing so that I can dial a 7 digit number and have that
number
This will not work.
Sipx hunt groups do not support resending the call to the same phones
more than one time in a hunt group cycle.
the only reliable way to do this is:
1. ring phone lines 200, 201, 202
if no answer then
2. ring lines 203-210, and lines 300-302 (300-302 are secondary lines
on th
i do not find that using a speedial to park or retrieve is at all
reliable, in fact, the user aspect introduces a very unreliable
element with the way the park system and polycom interacts. i use a
polycom config which requires manual entry to put the call on "park".
the same reliably "unpark".
a
Hi,
I have an issue that I can't seem to figure out it in sipXecs:
I have an auto-attendant that forwards to a hunt group which rings 4 phones
from support staff.
If those phones are not answered within 10 seconds, I need all phones in the
office to ring including the original 4 phones.
I tried m
Not currently. See http://track.sipfoundry.org/browse/XX-8473 for more info.
From: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.org] on behalf of Chris Rawlings
[cm.rawli...@gmail.com]
Sent: Tuesday, April 19, 2011 7:15 PM
To: Discuss
i am looking into how to do Parking with BLF on either Yealink or Polycom
phones... basically what i need is the ability to press a BLF button... that
button parks the call in a predetermined location.. the light lights up red
on ALL phones that have the button setup on them... the call can be then
Which voicemail do you want it to go to - the cell phone, or the sipxecs.
If sipXecs, put the last line to forward to 8+ your extension number.
If you want it to go to your cell phones voicemail, have the ringing to your
cell phone longer that the period of time that your cell phone voicemail
k
Send it to your cellphone and turn off the cell phone voicemail by sending
bust/na settings at the cellular carrier. Then set the forwarding number to
a google voice number set to DND. One box to rule them all and in the cell
phone screen transcribe them...
On Tue, Apr 19, 2011 at 4:43 PM, Bob And
On Tue, Apr 19, 2011 at 11:55 AM, Michael Scheidell <
michael.scheid...@secnap.com> wrote:
> On 4/19/11 11:53 AM, Michael Scheidell wrote:
>
> On 4/19/11 11:39 AM, Tony Graziano wrote:
>
>
> what you you think of implementing an (optional) 'ping' (sip method
> 'OPTIONS') at specified intervals?
>
I am trying to set up the following:
The ability to give customers one number to reach me: I have sip trunk as
an alias to my extension(2101)-this is working. I have BRIA on my iphone
ext(2201) . I have call forwarding set on my main extension to send calls
to the BRIA ext.t the same time
On Tue, Apr 19, 2011 at 3:21 PM, Douglas Hubler wrote:
>
> sorry, this is not clear at all ! Let me restate this:
> - Change an agent's Security level to "SUPERVISOR" as found in the
> Call Center Agent's Edit page. Do *not* make the user superadmin
> privledges in the User Edit page.
Ha. I
On Tue, Apr 19, 2011 at 11:15 PM, Tom Frey wrote:
> Hi,
>
> I started testing sipXecs for my office but noticed there does not seem
> to be support for the Cisco SPA 5xx series phones.
> I found that someone apparently made a plugin for them a while ago:
> http://track.sipfoundry.org/browse/XX-744
On Tue, Apr 19, 2011 at 3:37 PM, Michael Scheidell
wrote:
> On 4/19/11 3:20 PM, Douglas Hubler wrote:
> If you make someone a supervisor, login to interface on port 5050 then add
> supervisor dashboard you should see all sorts of info.
sorry, this is not clear at all ! Let me restate this:
- Ch
Hi,
I started testing sipXecs for my office but noticed there does not seem
to be support for the Cisco SPA 5xx series phones.
I found that someone apparently made a plugin for them a while ago:
http://track.sipfoundry.org/browse/XX-7443
However, quite frankly I have no idea where to a) find the
On Tue, Apr 19, 2011 at 10:20 PM, Douglas Hubler wrote:
> If you make someone a supervisor, login to interface on port 5050 then add
> supervisor dashboard you should see all sorts of info.
>
> The format for the commands is quasi freeswitch but on your initial tests,
> ignore those commands and l
On 4/19/11 3:21 PM, Matt White wrote:
Look back in your proxy log for fake accounts like "Bob@sipdomain" etc. It has
gotten so bad we no longer allow 5060 from the internet. In that way, Sipxbridge 5080
port requirment/limitation has saved us as most attackers aren't looking for alternative
On 4/19/11 3:20 PM, Douglas Hubler wrote:
If you make someone a supervisor, login to interface on port 5050 then
add supervisor dashboard you should see all sorts of info.
The format for the commands is quasi freeswitch but on your initial
tests, ignore those commands and log into interface
>>> On 4/19/2011 at 08:00 AM, in message
, Chris Rawlings
wrote:
> are you using Unified Messaging on Exchange Server SP1 ?
> I did not have any issues with SipX other than random crashing. As for
I believe it has already been suggested but your random crashing could be due
to sip attacks. In
If you make someone a supervisor, login to interface on port 5050 then add
supervisor dashboard you should see all sorts of info.
The format for the commands is quasi freeswitch but on your initial tests,
ignore those commands and log into interface on port 5050 and go available
from there to take
On 4/19/11 2:59 PM, Kyle Haefner wrote:
Hi All,
Using the caller-Id field one can mask the caller-Id for external
calls. Is there a way to do this for internal calls as well? I have
users that want a forward facing number shown for all outbound calls.
is this two questions or one?
users can
Hi All,
Using the caller-Id field one can mask the caller-Id for external
calls. Is there a way to do this for internal calls as well? I have
users that want a forward facing number shown for all outbound calls.
Thanks!
Kyle
___
sipx-users mailing li
Hi folks,
Wanted to start getting familiar with openACD this morning and I have
a couple basic questions:
1) Where do you see queue status?
2) When writing commands, what is the proper format? For example the
wiki has the following for "Login":
agent_dialplan_listener testme@fedorabox age
I guess 'one liners' aren't that funny after all...
This message and any files transmitted with it are intended only for the
individual(s) or entity named. If you are not the intended individual(s) or
entity named you are hereby notified that any disclosure, copying, distribution
or reliance up
On 4/19/11 2:09 PM, Michael Scheidell wrote:
On 4/19/11 1:58 PM, Matt White wrote:
Yeah, that would work too but still requires the itsp to support responding to
the options. Ultimalty I think thats the more elegant solution is to support
that type of sip options and response.
The other th
On 4/19/11 1:58 PM, Matt White wrote:
Yeah, that would work too but still requires the itsp to support responding to
the options. Ultimalty I think thats the more elegant solution is to support
that type of sip options and response.
The other thing would be to simply build an alarm when a "
>>> On 4/19/2011 at 01:24 PM, in message <4dadc54e.9080...@secnap.com>, Michael
Scheidell wrote:
> On 4/19/11 12:27 PM, Matt White wrote:
>> However, it would be nice if the options was there in the itsp template to
> enable it for any itsp that supports it.
>>
>>
> download a centos5 rpm of sip
On 4/19/11 12:27 PM, Matt White wrote:
However, it would be nice if the options was there in the itsp template to
enable it for any itsp that supports it.
download a centos5 rpm of sipsak.
while /usr/bin/sipsak -s sip:ping@1.1.1.1 -vv; do sleep 30; done
you could script something to 'ping'
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Organization: SipXecs Forum
In-Reply-To:
X-FUDforum: 08063afcdd00a6e76393c5b9527381e8 <58761>
Message-ID:
well now i feel a little embarrassed. I tried again just now
and it worked perfectly!!
Thank you once again G
On Tue, Apr 19, 2011 at 7:30 PM, Manny wrote:
>
> Content-Type: text/plain;
> charset="utf-8"
> Content-Transfer-Encoding: 8bit
> Organization: SipXecs Forum
> In-Reply-To:
> X-FUDforum: 08063afcdd00a6e76393c5b9527381e8 <58759>
> Message-ID:
>
>
>
> Ok i was able to follow through with everythi
Content-Type: text/plain;
charset="utf-8"
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Organization: SipXecs Forum
In-Reply-To:
X-FUDforum: 08063afcdd00a6e76393c5b9527381e8 <58759>
Message-ID:
Ok i was able to follow through with everything you said.
except how to accept and save value 4 'enable'
meani
>>> On 4/19/2011 at 11:53 AM, in message <4dadb002.10...@secnap.com>, Michael
Scheidell wrote:
> On 4/19/11 11:39 AM, Tony Graziano wrote:
>>
>>> what you you think of implementing an (optional) 'ping' (sip method
>>> 'OPTIONS') at specified intervals?
>>>
>> There is an assumption that all its
On 4/19/11 11:53 AM, Michael Scheidell wrote:
On 4/19/11 11:39 AM, Tony Graziano wrote:
what you you think of implementing an (optional) 'ping' (sip method
'OPTIONS') at specified intervals?
There is an assumption that all itsp's support that... it would be
better to send it a keepalive, and
On 4/19/11 11:39 AM, Tony Graziano wrote:
what you you think of implementing an (optional) 'ping' (sip method
'OPTIONS') at specified intervals?
There is an assumption that all itsp's support that... it would be
better to send it a keepalive, and observe the response, parse it and
send an ala
On Tue, Apr 19, 2011 at 11:31 AM, Michael Scheidell
wrote:
> Looking for a way to be notified if an ip authentication based, or unmanaged
> sip trunk becomes unresponsive, network fails, etc.
>
The alarms function would be where this would fit in (nicely).
>
> for user/auth method, sipx will notif
Looking for a way to be notified if an ip authentication based, or
unmanaged sip trunk becomes unresponsive, network fails, etc.
for user/auth method, sipx will notify you if it can't keep registration
as per the registration interval. but this doesn't do anything for ip
based (if you have 'r
Chris,
You've probably already done this, but it is something that tripped me
up so may help others down the line. 4.2.1 did not seem to care if
Exchange was part of the intranet subnets, but 4.4 won't work with
Exchange unless Exchange falls within the intranet subnet.
Kyle
On Tue, Apr 19, 201
something like this might work. .
http://cpansearch.perl.org/src/SULLR/Net-SIP-0.55/bin/stateless_proxy.pl
--
Michael Scheidell, CTO
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ISN: 1259*1300
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are you using Unified Messaging on Exchange Server SP1 ?
I did not have any issues with SipX other than random crashing. As for
exchange i believe it is quite borked on 4.4. I setup a 4.2.1 machine the
same way i have always and all issues are now gone except for the AA on
exchange 2010 sp1. This i
On Tue, Apr 19, 2011 at 4:27 AM, Zhiping Wang
wrote:
> "i saw that 4.4 is still beta and 4.2.1 is the stable release"
>
> It's also my concern. It's hard to make the decision to move to 4.4 from
> 4.2.1 ...
>
>
> On Tue, Apr 19, 2011 at 8:52 AM, Chris Rawlings
> wrote:
>>
>> it looks like i fixed
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