Hi, team, does sipx support incoming call screening now ? Or how can a sipx
user divert a specific incoming call to his personal auto attendant ?
Cheers,
Jun
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Thanks Scott, that way is pretty efficient.
-Original Message-
From: Scott Lawrence [mailto:scott.lawre...@nortel.com]
Sent: Friday, August 21, 2009 10:58 AM
To: jun,wen
Cc: sipx-users@list.sipfoundry.org
Subject: RE: [sipx-users] Outgoing PSTN call blacklist by permission and
dial plans
On Fri, 2009-08-21 at 09:59 +0800, jun,wen wrote:
> Yes, Scott, I did forget to put a FXO gateway in my blacklist dial plan.
> When I added my FXO gateway, my sipx responded "486 Busy here" when I dialed
> the called party number inside my blacklist and my Grandstream sip phone got
> busy tone. Is
Using 4.0.1-015823:
I uploaded a 122 second queue audio file to an ACD queue. When setting
the audio interval to 122 corresponding to the length of the audio file,
ACD will only play max 30 seconds of the queue audio before restarting
from the beginning. I tested various queue audio interval
Yes, Scott, I did forget to put a FXO gateway in my blacklist dial plan.
When I added my FXO gateway, my sipx responded "486 Busy here" when I dialed
the called party number inside my blacklist and my Grandstream sip phone got
busy tone. Is it the designed or expected behavior on this scenario ? If
Normally you would troubleshoot call using sipviewer. The link is below. It
combines the logs for the call inot a single xml file. Sipviewer (as a client)
can be run on a windows PC and open the file to trace it.
http://sipx-wiki.calivia.com/index.php/Display_SIP_message_flow_using_Sipviewer
At t
Hi All,
I come from the Asterisk world where there is a console that you can see
almost everything happen and errors when there are errors. Also and handy
sip debug where you can see all the sip headers.
What is the best way to troubleshoot SIP issues with SipX? I feel so blind
troubleshooting v
Interesting, it's just quiet and I'm not missing anything. Just seemed strange,
at least since I signed up. Thanks for the URL :).
Mike
On Thu, 20 Aug 2009 19:30:57 -0400, Tony Graziano wrote:
> See the wiki:
>
> http://sipx-wiki.calivia.com/
> http://list.sipfoundry.org/archive/sipx-users/
>
See the wiki:
http://sipx-wiki.calivia.com/
http://list.sipfoundry.org/archive/sipx-users/
>>> "li...@grounded.net"
NaN. 08/20/09 7:25 PM >>>
Is there some way of checking the list to see posts that are coming in? I can't
believe there have been no posts what so ever today so am wondering if my
Is there some way of checking the list to see posts that are coming in? I can't
believe there have been no posts what so ever today so am wondering if my
list/email is messed up.
Thanks.
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To be honest, I'm surprised that not more people are using the TNT, it's a
carrier class box, serious users often look at it as a solution. Overkill for
most of course.
On Mon, 17 Aug 2009 15:48:25 -0400, Craig Fields wrote:
> We tested the TNT and APEX extensively. While it will work with sta
On Thu, 2009-08-20 at 16:49 +0800, jun,wen wrote:
> Hi, Team,
>
> I am trying to implement a blacklist to users to bar some specific
> call destination. when created a permission of "Blacklist
> Access", I made a new dial plan of "Blacklist" with specific barred
> PSTN number by a prefix and che
Damian Krzeminski wrote:
>In many places sipXconfig is using a validator that accepts either "dial
>digits" or "sip URIs". I am pretty sure that it can/should be relaxed.
>Please open an issue.
>D.
>
Done. http://track.sipfoundry.org/browse/XX-6358 I wasn't sure which
component to put it under,
Keith Gearty wrote:
> Keith Gearty wrote:
>
>> I just found an annoying little bug under User > Call Forwarding. The
>> "forward to" field is supposed to accept any potentially valid SIP
>> extension, but it seems to only accept numeric extensions or fully
>> qualified SIP URIs. If you put in
Check the 'Redirection' menu settings in the phone config. Hit the
dropdown and select On Busy and put in the extension...
Mike
-Original Message-
From: Keith Gearty [mailto:ke...@glensound.co.uk]
Sent: Thursday, August 20, 2009 6:50 AM
To: Picher, Michael
Cc: Scott Lawrence; sipx-users
Picher, Michael wrote:
> I don’t think you can… see my previous response.
>
> Set an if-no-answer forward option to the wireless phone. In the phone
> configuration look for the call forward on busy option and make that
> dial 8+ext to forward immediately to VM.
>
> I’m fairly certain that shoul
I don't think you can... see my previous response.
Set an if-no-answer forward option to the wireless phone. In the phone
configuration look for the call forward on busy option and make that
dial 8+ext to forward immediately to VM.
I'm fairly certain that should work the way you want exce
Keith Gearty wrote:
>I just found an annoying little bug under User > Call Forwarding. The
>"forward to" field is supposed to accept any potentially valid SIP
>extension, but it seems to only accept numeric extensions or fully
>qualified SIP URIs. If you put in a non-numeric extension without
jun,wen wrote:
Hi, Team,
I am trying to implement a blacklist to users to bar some specific
call destination. when created a permission of "Blacklist
Access", I made a new dial plan of "Blacklist" with specific barred
PSTN number by a prefix and checked the "Blacklist Access" as the
requi
I just found an annoying little bug under User > Call Forwarding. The
"forward to" field is supposed to accept any potentially valid SIP
extension, but it seems to only accept numeric extensions or fully
qualified SIP URIs. If you put in a non-numeric extension without the
SIP domain part, it
Hi, Team,
I am trying to implement a blacklist to users to bar some specific call
destination. when created a permission of "Blacklist Access", I made a new
dial plan of "Blacklist" with specific barred PSTN number by a prefix and
checked the "Blacklist Access" as the required permissions .
A
Scott Lawrence wrote:
On Wed, 2009-08-19 at 16:24 +0100, Keith Gearty wrote:
I have a need to limit the number of incoming calls on a particular line
to 1. So if that user is already handling an incoming call, any further
incoming calls go straight to voicemail or next user on hunt group
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