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'm still bothered by
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
> person), but I'm
Apologies if I'm reading like an idiot (well, at best an uneducated
person), but I'm still bothered by this.
I can see that for each of the two cases sipXecs is sending a
different REFER, my point was why does the extension just have the
domain name, while the conference has the full hostname ?
A
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
> is incorrect ?
No, it's correct because in that trace the Refer-To header is:
On Wed, 2009-10-28 at 14:11 +, shouldbe q931 wrote:
> This is a trace from the sipXecs server with the simple filter of
> "host 10.201.1.5" the IP of the Avaya
>
> In it I call from 6999 on the Avaya to 6815 which is an AA on sipXecs,
> and then after listening to the prompt, dial 6812 which i
On Wed, 2009-10-28 at 11:16 +, shouldbe q931 wrote:
>
> It would be much easier for me to to run a tcpdump on the sipXecs box
> than it would be on the Avaya one, and I would be quite happy to.
That's unlikely to help - the message that we need to see never gets to
the sipXecs box, so it prob
p/
>
> - Original Message -
> From: shouldbe q931
> To: Tony Graziano ; sipXecs users
>
> Sent: Wed Oct 28 06:14:05 2009
> Subject: Re: [sipx-users] conferencing capabilities
>
> Any further suggested next steps for seeing where the problem lies ?
>
> Cheers
Helpdesk Contract Customers:
http://www.myitdepartment.net/gethelp/
- Original Message -
From: shouldbe q931
To: Tony Graziano ; sipXecs users
Sent: Wed Oct 28 06:14:05 2009
Subject: Re: [sipx-users] conferencing capabilities
Any further suggested next steps for seeing where the problem
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
>
> Arne
>
> On Tue,
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
> conference number"
Hi Tony,
Having a forward on 6810 to 6812 works, I get the "please enter your
conference number" prompt.
Cheers
Arne
On Mon, Oct 26, 2009 at 9:33 PM, Tony Graziano
wrote:
> THEN if you take user 6810 and do a permanent forward to 6812 in sipx, does
> it work dialing 6810 from the avaya?
>
> On
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 phone attached, so it goes to voicemail), but
not working when the destination is 6812 (a conference extension), and
having trouble seeing why it would wo
On Mon, 2009-10-26 at 20:09 +, shouldbe q931 wrote:
> I can certainly try adding a softphone to the mix tomorrow.
>
> Granted its from an external call (I'm not in the office this late),
> but the following tac trace shows a working blind transfer to the
> "goes to voicemail" extension (6810),
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 +, shouldbe q931 wrote:
>> Hi Tony,
>>
>> I don't have any phones connected to sipXecs. All of this has been
>> from the Avaya system to sipXecs
>>
>>
:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Scott Lawrence
Sent: Monday, October 26, 2009 12:06 PM
To: shouldbe q931
Cc: sipXecs users
Subject: Re: [sipx-users] conferencing capabilities
On Mon, 2009-10-26 at 18:40 +, shouldbe q931 wrote:
> Hi Tony,
>
> I don't have any
On Mon, 2009-10-26 at 18:40 +, shouldbe q931 wrote:
> Hi Tony,
>
> I don't have any phones connected to sipXecs. All of this has been
> from the Avaya system to sipXecs
>
> I have started to document the configuration on the wiki
> http://sipx-wiki.calivia.com/index.php/HowTo_configure_Avaya_
Hi Tony,
I don't have any phones connected to sipXecs. All of this has been
from the Avaya system to sipXecs
I have started to document the configuration on the wiki
http://sipx-wiki.calivia.com/index.php/HowTo_configure_Avaya_CM_5.1_with_sipX
Cheers
Arne
On Mon, Oct 26, 2009 at 6:34 PM, Tony
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:
> well, a "list trace tac" showed more than I thought it
well, a "list trace tac" showed more than I thought it would. Trunk 60
is the SIP trunk linking the Avaya to sipXecs
list trace previousPage 1
LIST TRACE
timedata
18:21:58 Calling party station
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
> hoping to be able to run it as a reasonably contained conference
> bridge, but if it requires each conference to have an "extension
> number" that is direc
Oh, bother.
I didn't quite understand the "proxy" limitation before now, I was
hoping to be able to run it as a reasonably contained conference
bridge, but if it requires each conference to have an "extension
number" that is directly "dialable" from the PBX, it's quite a
limitation, at least until
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 have however just seen this
> http://track.sipfoundry.org/browse/XX-5643 and wondered if it might be
> applicable.
No... it's not - that's a
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", but I'll see what I can
Hi Scott,
SIP logging on the Avaya is a little "complex", but I'll see what I can find.
I have however just seen this
http://track.sipfoundry.org/browse/XX-5643 and wondered if it might be
applicable.
Cheers
Arne
On Mon, Oct 26, 2009 at 2:50 PM, Scott Lawrence
wrote:
> On Mon, 2009-10-26 at 1
On Mon, 2009-10-26 at 12:36 +, 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 easiest filter would probably be '6999' - the caller number you
u
On Mon, 2009-10-26 at 10:47 +, shouldbe q931 wrote:
> 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"
>
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-
On Fri, 2009-10-23 at 14:31 -0400, Geoff Van Brunt wrote:
> Auto attendant allows you to dial an extension at any time including
> conference extensions. It will then prompt for the pin or say not found
> if it doesn't exit.
So all you have to do to set up your 'lobby' is create an autoattendant
w
lf Of shouldbe
q931
Sent: October-23-09 1:28 PM
To: sipx-users@list.sipfoundry.org
Subject: [sipx-users] conferencing capabilities
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" ca
>
> I'm also after the capability to have have conference calls
> recorded, and then emailed to the owner of the conference.
>
> Any constructive suggestions, or pointers to documentation
> that I've missed would be appreciated.
>
> Cheers
>
> Arne
The conference recording feature is current
...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of shouldbe q931
Sent: Friday, October 23, 2009 10:28 AM
To: sipx-users@list.sipfoundry.org
Subject: [sipx-users] conferencing capabilities
Hi All,
After some fun and games, I have sipXecs 4.0.3 working against our
Avaya CM
ginal Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of shouldbe
q931
Sent: Friday, October 23, 2009 1:28 PM
To: sipx-users@list.sipfoundry.org
Subject: [sipx-users] conferencing capabilities
Hi All,
After some fun and games, I
Helpdesk:
Telephone: 434.984.8426
Fax: 434.984.8427
Helpdesk Contract Customers:
http://www.myitdepartment.net/gethelp/
- Original Message -
From: sipx-users-boun...@list.sipfoundry.org
To: sipx-users@list.sipfoundry.org
Sent: Fri Oct 23 13:28:13 2009
Subject: [sipx-users] conferencing
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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