On Wed 13.Jan.10 16:24, Dale Worley wrote:
>On Tue, 2010-01-12 at 20:17 -0700, Dan McDaniel wrote:
>> On Mon 11.Jan.10 15:18, Dale Worley wrote:
>> >On Sun, 2010-01-10 at 23:00 -0700, dan wrote:
>> >> I've set up an old SPA2002 to do some testing. It registers with my sipx
>> >> server and I can ca
I'm working on a project that uses Thomson ADSL2+ residential gateways - QoS
on these gateways is fine, basically the QoS is very similar to the LLQ
implementation on Cisco.
How were you saturating your upload and download?
On Thu, Jan 14, 2010 at 10:05 AM, Tony Graziano <
tgrazi...@myitdepartmen
A week or so ago I posted a msg about Polycom phones getting
disconnected while listening to VM. Thinking something might be wrong
with the PBX I rebuilt it and restored the database.
Now I'm having exactly the same problem. The only thing slightly
different about this system than all of the
Thank you. That looks like it will work for me.
On 1/13/2010 2:46 PM, Robert Joly wrote:
You can add an 'if no response' forwarding rule for extension 8
which will ring 's voicemail box directly.
*From:* m
Is the sipXpark server getting replaced in version 4.2 or simply being
improved upon? I haven't heard a definite answer yet.
--
Josh Patten
Assistant Network Administrator
Brazos County IT Dept.
(979) 361-4676
___
sipx-users mailing list sipx-users@li
On Tue, 2010-01-12 at 20:17 -0700, Dan McDaniel wrote:
> On Mon 11.Jan.10 15:18, Dale Worley wrote:
> >On Sun, 2010-01-10 at 23:00 -0700, dan wrote:
> >> I've set up an old SPA2002 to do some testing. It registers with my sipx
> >> server and I can call the operator or voicemail, but it cannot dial
> I looked at that trace, and removed it from the issue because it's
> unrelated (it does cause the ACK too loop, but not for the same reason).
>
> What you hit was a different bug:
>
> http://track.sipfoundry.org/browse/XX-4782
>
> which is fixed in main.
Sorry, but anyway, this is great news! =)
What I think you are experiencing is this: Consumer grade equipment and
consumer grade dsl.
I've never used consumer grade routers except for one way... turn off dhcp
and add a static local route and enable the wireless on it as an access
point (like at home, pfsense with a linksys router configur
On Wed, 2010-01-13 at 21:32 +0100, Staffan Kerker wrote:
> Hi
>
> I once again stumbled over the misrouted ACK problem when it comes to GRUU
> contacts and updated the issue tracker with a new trace.
> I'm not totally sure though if the error is in SipX or in the endpoints.
> However, the trace sh
On Wed, 2010-01-13 at 14:39 -0600, mkitchin.pub...@gmail.com wrote:
> Thank you. That makes a lot of sense. So, that brings me to a question
> on the web/gui based forwarding. Sometimes the users specifically
> wants the the call to go to the destination mailbox as well. If
> forwards his call
Thanks for the feedback! I was leaning towards pfSense, so this makes me
feel a bit better about that. Tony, I saw your blog posts and they seem very
helpful (I'm subscribed to your blog).
I've been running a number of tests with QoS and remote workers with the
D-Link DIR-655 and the PAP2T-NA adap
You can add an 'if no response' forwarding rule for extension 8
which will ring 's voicemail box directly.
From: mkitchin.pub...@gmail.com
[mailto:mkitchin.pub...@gmail.com]
Sent: Wednesday, January 13, 2010 3:39 PM
To: Joly,
On Wed, 2010-01-13 at 15:31 -0500, Robert Joly wrote:
>
> The issue with the set-based call forwarding is as follows:
> Suppose that A calls B and B and does set-based call forwarding to C.
> In that scenario, what sipXecs ultimately see is A calling C and that
> call will be successful only i
Thank you. That makes a lot of sense. So, that brings me to a question
on the web/gui based forwarding. Sometimes the users specifically wants
the the call to go to the destination mailbox as well. If forwards
his calls to , and doesn't answer, it still goes to s
voicemail, ri
Hi
I once again stumbled over the misrouted ACK problem when it comes to GRUU
contacts and updated the issue tracker with a new trace.
I'm not totally sure though if the error is in SipX or in the endpoints.
However, the trace shows the ACK loop within the SipXproxy.
http://track.sipfoundry.org/b
I suppose that if none of your dialplan rules required permissions, you
would not run into these issues but no sane person would do that in a
real environment.
The issue with the set-based call forwarding is as follows:
Suppose that A calls B and B and does set-based call forwarding to C.
In th
Not that it matters...
We have a site with a need for a flexible attendant to be able to forward
their calls off site. There is no set schedule for this, because they have
someone in at various times at night and weekends to handle calls without
forwarding them.
We use the polycom forward button,
Thanks. So, is that a definitive "it can't be done"? I would like to be
able to present all the options. There are only a few users I would like
to be able to enable it for. I would like to be able to give them all
the options, and then explain why they should do what.
On 1/13/2010 2:13 PM, Ro
This is another case of stale information on the wiki. This statement
no longer applies.
Also, the use of set-based call forwarding features is strongly
discouraged as its usage brings about the permission problems that you
are seeing. Instead, sipXconfig or the user portal should be used to
co
I'm pretty sure this is the restriction I have come across.
http://sipx-wiki.calivia.com/index.php/How_to_configure_User_Call_Forwarding
"In order to be authorized to forward calls to external PSTN numbers,
the individual user or the group to which the user belongs to has to
have the permission
Did you have any luck resolving this? I think I have run across the same
issue.
On 1/7/2010 11:58 AM, Simon Moore wrote:
> Hi,
>
> I am having trouble with getting the Auto Attendant forwarding to
> external number.
>
> 2 Methods tried
>
> 1) Press 2 in the auto attendant and it has a UK mobile n
I'm not sure if this is maybe a security setting, a misconfiguration on
my part, ignorance on my part, or something else.
We have Polycom 450s and 550s. They are behaving the same.
We are trying to use the call forwarding feature at the handset level. I
enabled the feature to activate soft button
Hi all,
Regarding the "Polycom Firmware 3.2.1" thread we had going back in
October of last year...
1. I've finally raised a JIRA to cover the enhancement detailed in the
http://list.sipfoundry.org/archive/sipx-users/msg17622.html post.
http://track.sipfoundry.org/browse/XX-7401 : Activate mu
On Wed, 2010-01-13 at 15:35 +, Dylan Ebner wrote:
> We are building up a SIPX system for deployment and we have run into a
> very nasty bug. Apparently the Cisco phones do not work properly with
> ACD. This was entered as xx-7227, but it looks like SIPX will not be
> fixing the issue.
FYI ...
We are building up a SIPX system for deployment and we have run into a very
nasty bug. Apparently the Cisco phones do not work properly with ACD. This was
entered as xx-7227, but it looks like SIPX will not be fixing the issue.
So, has anyone gotten Cisco phones working with ACD? If so, how? Do
Probably for the same reason I did: an incomplete SIP stack on the
gateway. I had to do this when I had my Adtran Total Access 908e in
place because it would choke on two things: directed call pickup and
attended transfer.
Josh Patten
Assistant Network Administrator
Brazos County IT Dept.
(979)
Hello!
I am having an issue with my new sipx setup and am hoping that someone may
have a solution. I am currently running sipx server 4.0.4 with audio code
mp118(FXO) firmware 5.6a. Hooked up into the server is a polycom 550
running firmware 3.1.3. The issue that I am having is that when I attem
On Tue, 2010-01-12 at 20:14 -0600, Gabe Casey wrote:
> In the case of calls originating from the pstn to my media gateways an
> operator trying to perform a consultative transfer will always fail
> because of this , what workaround options would you suggest given that
> the issue is slated to fix v
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