Probably a dumb question, but if its only a small portion of people,
why not go for a hosted fax solution, one that does fax to email
inbound, and email or print driver for fax outbound ?
The last time I installed Rightfax (While working for a Rightfax
partner), It was lovely, and had a pricetag t
On Sat, Dec 19, 2009 at 12:31 PM, m...@grounded.net wrote:
> On Sat, 19 Dec 2009 12:07:32 +0000, shouldbe q931 wrote:
>> I've got basic interoperation working, which I've detailed here
>> http://sipx-
>> wiki.calivia.com/index.php/HowTo_configure_Avaya_CM_5.
I've got basic interoperation working, which I've detailed here
http://sipx-wiki.calivia.com/index.php/HowTo_configure_Avaya_CM_5.1_with_sipX
Cheers
On Fri, Dec 18, 2009 at 3:16 PM, Jordan Turner
wrote:
> Should this project be safe with Avaya? Or should be go ahead and reconsider
> going to
I was hoping they were a decent quality TTS...
Oh well, I'll probably shanghai somebody in the office :-)
Cheers
Arne
On Mon, Nov 23, 2009 at 5:30 PM, Alfred Campbell wrote:
>> Hi All,
>>
>> Having got enough functionality to be able to run sipXecs as a basic
>> conference bridge against our A
Hi All,
Having got enough functionality to be able to run sipXecs as a basic
conference bridge against our Avaya PBX, before I bring it into
service I'd like to add a new Auto Attendant prompt.
To keep the overall experience as smooth as possible, I'd like to have
the new Auto Attendant prompt in
Hi All,
With a change to the Avaya config which I've updated on the wiki, I
can now do a blind transfer to a conference!
Cheers
Arne
On Wed, Oct 28, 2009 at 6:03 PM, shouldbe q931
wrote:
> Apologies if I'm reading like an idiot (well, at best an uneducated
> person), but I
ed around gateway/group/user settings and how
> this gets processed.
> If the gateway/group/user is blank on caller ID, it send "6635". If I mark
> any anonymous it sends the full # for the main TN (which is probably the
> carrier). So if it doesn't like what the proxy
Hi Tony,
Granted I know more about UK ISDN than yours, but usually, if you
don't set the ID, or try to set an invalid one, the carrier would set
the main number for the trunk to be the ID. To restrict this you would
need to set "restrict" on the outbound call. Not a clue on how you'd
set this on s
Amazingly Avaya look as if they are going to accept the ticket :-)
I'm still interested in the difference between the two refers on sipXecs
Cheers
Arne
On Wed, Oct 28, 2009 at 6:03 PM, shouldbe q931
wrote:
> Apologies if I'm reading like an idiot (well, at best an uneducated
&
09 at 5:28 PM, Scott Lawrence
wrote:
> On Wed, 2009-10-28 at 14:59 +, shouldbe q931 wrote:
>> Point taken.
>>
>> Does that mean that when the "transfer" to 6810 happens as per the
>> attached pcap, where it just uses the domain rather than the F.Q.D.N
>>
anager
> 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://www.myitdepartment.net/gethel
Any further suggested next steps for seeing where the problem lies ?
Cheers
Arne
On Tue, Oct 27, 2009 at 9:31 AM, shouldbe q931
wrote:
> Just to clarify, that works for both cases, calling 6810, or calling
> 6815 (the AA), and then doing a blind transfer to 6810
>
> Cheers
>
&g
Just to clarify, that works for both cases, calling 6810, or calling
6815 (the AA), and then doing a blind transfer to 6810
Cheers
Arne
On Tue, Oct 27, 2009 at 9:21 AM, shouldbe q931
wrote:
> Hi Tony,
>
> Having a forward on 6810 to 6812 works, I get the "please enter your
>
0 from the avaya?
>
> On Mon, Oct 26, 2009 at 4:57 PM, shouldbe q931
> wrote:
>>
>> okay, I'm probably coming at this from the wrong angle.
>>
>> I'm seeing the "blind transfer" work when it goes to 6810 (a user
>> "extension" without a
r to use tcpdump/wireshark to check that what is going
over the wire is the same, if it is, then I can look at opening a case
with Avaya,...
Cheers
Arne
On Mon, Oct 26, 2009 at 8:25 PM, Scott Lawrence
wrote:
> On Mon, 2009-10-26 at 20:09 +, shouldbe q931 wrote:
>> I can certainl
onference bridge. It's just
> another extension on the system.
>
> -Original Message-
> From: sipx-users-boun...@list.sipfoundry.org
> [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Scott Lawrence
> Sent: Monday, October 26, 2009 12:06 PM
> To: should
noted, and done, also listing what _does_ work so far :-)
On Mon, Oct 26, 2009 at 7:06 PM, Scott Lawrence
wrote:
> On Mon, 2009-10-26 at 18:40 +0000, shouldbe q931 wrote:
>> Hi Tony,
>>
>> I don't have any phones connected to sipXecs. All of this has been
>>
Tony Graziano
wrote:
> Stupid question for me to ask this late, but do both systems have the
> ability to dial each other natively via dialplan and are setup as gateways
> in each others system? Would that matter?
>
> On Mon, Oct 26, 2009 at 2:31 PM, shouldbe q931
> wrote:
&g
aya system, but c'est la vie :-)
Cheers
Arne
On Mon, Oct 26, 2009 at 5:25 PM, Scott Lawrence
wrote:
> On Mon, 2009-10-26 at 16:59 +, shouldbe q931 wrote:
>> Oh, bother.
>>
>> I didn't quite understand the "proxy" limitation before now, I was
>> ho
ses were not
changed.
Cheers
Arne
On Mon, Oct 26, 2009 at 4:04 PM, Scott Lawrence
wrote:
> On Mon, 2009-10-26 at 15:14 +, shouldbe q931 wrote:
>> Hi Scott,
>>
>> SIP logging on the Avaya is a little "complex", but I'll see what I can find.
>>
>> I
I might be missing something here, but why would the call be handed
back to the Avaya system when its already been passed across to
sipXecs ?
Cheers
Arne
On Mon, Oct 26, 2009 at 3:14 PM, shouldbe q931
wrote:
> Hi Scott,
>
> SIP logging on the Avaya is a little "complex",
Mon, 2009-10-26 at 12:36 +0000, shouldbe q931 wrote:
>> Hi Scott,
>>
>> I'm a little unsure of how to use a Token in the filter, so I just
>> used "6" I'm also too much of a newbie to find the call-id for the
>> failing calls :-(
>
> The ea
Hi All,
I've created a new AutoAttendant on 6815, a conference on 6812 and a
user mailbox on 6810
If I dial 6812 directly, it asks for the PIN, if I dial 6810 I am get
the "The owner of extension 6810 is not available"
If I dial the AA on 6815, I can transfer to 6810 and get the "please
hol
I hadn't seen that capability in the documentation for the Auto
Attendant, but it sounds nearly perfect, the only issue that I can
see, is how to "capture" incorrect conference extensions...
I'll certainly try it on Monday
On Fri, Oct 23, 2009 at 7:54 PM, Scott Lawrence
wrote:
> On Fri, 2009-10-
Hi All,
After some fun and games, I have sipXecs 4.0.3 working against our
Avaya CM 5.1 server using SIP trunks.
My interest is in the conference "bridge" capability rather than
connecting phones (as that's what we have the Avaya for).
I've got basic conference working, but what I'm really after
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