Agreed - once the technology matures, this will be a great feature - until
then, it's not for a production environment - the results are good for a laugh
though.
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Michael Picher
Sent:
I have found that automated system that call me get perfect translation
with services like google voice to text. When there is a lousy cell
connection, lots of background noise or heavy accents it doesn't work so
well. I use gvoice for my stuff and it works well enough I can sometimes
make out
Well, so then it translates things fairly well!
:)
On Tue, Sep 11, 2012 at 8:25 AM, Tony Graziano tgrazi...@myitdepartment.net
wrote:
I have found that automated system that call me get perfect translation
with services like google voice to text. When there is a lousy cell
connection, lots
Whatever Comcast is using is usable. It attached the WAV file was
well. Usually, the text is close enough, and I listen to the WAV file if
I need more detail.
On 9/11/2012 7:20 AM, Nathaniel Watkins wrote:
Agreed -- once the technology matures, this will be a great feature --
until then,
Hello,
I'm trying to test Yealink telephon in Sipxecs 4.6, but it isn't in the
list for adding. I have read that github.com is working in this project. Is
it possible to test this telephon with other telephone configuration of the
existing list? Do you know when will be ready the github.com
Will https://github.com/siplabs/sipxyealink be added to the sipx repository
(http://github.com/dhubler/sipxecs) and will it be updated to be compatible
to sipx 4.6?
2012/9/11 Daniel Peinado Lopez daniel.pein...@iant.de
Hello,
I'm trying to test Yealink telephon in Sipxecs 4.6, but it isn't
I have contacted my ITSP and we will be doing some
troubleshooting today to see if they are causing anything or
if they are seeing any strange. I'll post back with any
significant results and especially if we find a resolution.
I'm still open to any suggestions anyone here on the forums
may
Where is the original post?
On Tue, Sep 11, 2012 at 11:17 AM, Geoff Musgrave
geoff.musgr...@cacionline.net wrote:
I have contacted my ITSP and we will be doing some
troubleshooting today to see if they are causing anything or
if they are seeing any strange. I'll post back with any
Think special needs. There are some required uses for it. Seems
technology hasn't caught up to that need yet.
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Nathaniel
Watkins
Sent: Tuesday, September 11, 2012 5:21 AM
To: Discussion
Hello all,
I have just tried to set up LDAP authentication on my 4.6 install.
Everything worked fine when importing the users. But now I am not able
to login to the webUI anymore - even with superadmin...
I always get the following 500 error:
http error: 500
Root DNs must be the same when using
Good question. I'm not sure. I was on the forums looking for an answer to
another question when I added this.
I added the text below:
First post so bare with me. I've been searching off and on all day today with
nothing to show for it. So now I'm posting for some help.
For starters my build
Explain who the ITSP is and how you have trunking setup. Typically calls
from providers come to your system on port 5080. I would assume this is
working because the call DOES transfer to the AA (and audio is heard,
correct)? If not it is likely the call is not coming to the system on port
5080,
Tony, thank you for your response and your questions. I hope I answer them with
enough detail.
My ITSP is TouchTone and they are signaling on 5080.
Yes, the call will connect to the AA and will transfer to the queue and the
caller will hear the MOH for a couple seconds and then the call is
It is more likely the ACK is not being OK'd by the ITSP. I would try to
simplify the call inbound to see if this applies to all calls or not.
In other words, send the call to the AA (dont ring) and then see if the
call will stay up for longer than 20 seconds.
Hint: If the call answers and audio
You can say you attended the first meeting. That's why I'm going ;)
http://www.meetup.com/sipXecs-User-Group/
Wed Sep 19
6:00 PM
Meadhall
4 Cambridge Center, Cambridge, MA
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Thanks Tony. I was thinking that yesterday when I requested the troubleshooting
with the ITSP but I had other godaddy related fires to put out yesterday. We
will be doing some captures and go from there this afternoon and I'll report
back. I'll also try going direct to the AA and we'll see how
Ugh. Go daddy. i only use them for a registrar but cant even stomach their
DNS and moved everything. Good to know I missed that headache yesterday too!
On Tue, Sep 11, 2012 at 1:44 PM, Geoff Musgrave
geoff.musgr...@cacionline.net wrote:
Thanks Tony. I was thinking that yesterday when I
On Tue, Sep 11, 2012 at 10:16 AM, Daniel Peinado Lopez
daniel.pein...@iant.de wrote:
Will https://github.com/siplabs/sipxyealink be added to the sipx repository
no need, it can remain where it is. Might need to create couple tags
to be compatible w/build scripts.
I did a clean OS/app install of the 4.6 build from last
night. Setup went fine but I get a 404 when I launch the
web interface (/sipxconfig/app Not Found). There are a slew
of Mongo-related errors in the sipxconfig.log (attached).
Do we know if there's a problem with this build?
sipxconfig.log
On Tue, Sep 11, 2012 at 9:48 PM, John Lightfoot j...@vizhn.com wrote:
I did a clean OS/app install of the 4.6 build from last
night. Setup went fine but I get a 404 when I launch the
web interface (/sipxconfig/app Not Found). There are a slew
of Mongo-related errors in the sipxconfig.log
I eliminated the dummy ring piece and still the same result. I was able to
verify that I'm getting 2 way audio before the call drops though. My ITSP is
being difficult, but I think it's more because I wasn't available to respond to
their response soon enough to further explain the situation and
whatever answers the call, if/when configured properly is going to ACK
it. Assuming the ITSP is competent they will see the call as answered.
I simply don't know enough about your environment, but I would suggest
simply having the calls route to the AA (it shows answered this way), then
let it
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