Does SIPX support ZRTP?
--
havesoftware, Inc.
http://www.havesoftware.com
Jakson Kalsson
Senior Programmer
jakkals...@havesoftware.com
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Probably late in this discussion, but just to confirm the obvious, did you
reset the flash on this phone to factory defaults to ensure there are no
corrupted configurations in the phone(s) themselves.
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] O
I guess it is because sipXecs will not insert a record-route if it is
merely acting as a relay. You need to make the B2BUA send the correct
domain. However, it must be prepared to authenticate against sipXecs if
it does that.
On 05/18/2011 03:16 AM, Laurent Schweizer wrote:
Hello,
I have
The only thing at that point which can get into the way of answer
getting stuck is proper receipt of ACK or PRACK. Check the logs if
these two requests are reaching the Polycom properly. I have also seen
polycoms act weird in call answer when using TCP so checking whether it
is sending 200 Ok
On Tue, May 17, 2011 at 5:35 PM, Carl Farrington wrote:
> Thanks very much. It worked, although there is some inconsistency on the
> operator AA and voicemail - once says "square" and the other says "hash".
> I'll probably find the square.wav or whatever, and symlink to hash.wav.
In 4.6 localiz
Request for clarification or correction on Wiki entry regarding Karel IP11x
Phones being an option from the Add new phone drop down menu. The only
options for Karel Phones are Karel NT321, Karel NT341, and KPhone. Karel
IP11x is NOT an option.
From:
http://wiki.sipfoundry.org/display/sipXecs/Karel
I think you are confusing the principles here.
Your outgoing call will be placed on a trunk based on the dial plan that you
set up.The phone is registered with the Proxy, and based on permissions
it has, it can dial out on multiple dial plans in the system. It will grab
the first route
I understand that sipX work like a proxy so he must forward the call to the
destination present in the domain part.
As phone I have snom phone configured with the default auto-provisioning and
they seems to store the domain present in the From header and not only the
number
so the question is, if
Thanks very much. It worked, although there is some inconsistency on the
operator AA and voicemail - once says "square" and the other says "hash". I'll
probably find the square.wav or whatever, and symlink to hash.wav.
From: sipx-users-boun...@list.sipfoun
From: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.org] On Behalf Of Laurent Schweizer
[laurent.schwei...@peoplefone.com]
if I start the call with a correct domain (the Sipx domain) all is ok.
how can I solve this issue ?
__
On 5/17/2011 2:42 PM, Gerald Drouillard wrote:
>> They ring, but she cannot answer them. She physically answers the phone,
>> but the 'answer' option is still there on the handset.
>>> "You cannot pick up the call at all"
>>> So the buttons on the polycom buttons appear to not be responding?
>>> Ph
> They ring, but she cannot answer them. She physically answers the phone,
> but the 'answer' option is still there on the handset.
>> "You cannot pick up the call at all"
>> So the buttons on the polycom buttons appear to not be responding?
>> Phone continues to ring but user w/extension 10 smash
Logging level was set too low :)
Just got a chance to restart the services, and will wait for the
next failed call.
On 5/17/2011 1:17 PM, Matthew Kitchin (public/usenet) wrote:
When we were on 4.2.1, we had it limited to g711. With 4.4, we are
a
Hello,
I have an SIPx running sipXconfig 4.2.1-018971 with snom phone ( snom 820
V8) and I have a problem with some calls.
When the snom start a call with a not correct domain like
0212556677@45.121.20.20 (the ip 45.121.20.20 as nothing to do with the SIPx
IP ) the call is forwarded
When we were on 4.2.1, we had it limited to g711. With 4.4, we are
able to change it back to allow g729. Something changed in sipx to
resolve this issue for us. This has been consistent in all our
setups. The problem is the same when they were 4.2.1 and since going
to 4.
Yes. Sorry. I'm merging my issues for the day in my head :)
On 5/17/2011 12:56 PM, Melcon Moraes wrote:
Did you mean Audiocodes BOOTP tool?
-
MM
On Tue, May 17, 2011 at 12:23, Matthew Kitchin (public/usenet)
mailto:mkitchin.pub...@gmail.com>> wrote:
FYI, after de-bricking it with with t
I would be surprised if you have just one issue here. I think looking at
logs will give you the best clues as to what might be going on.
Have you considered limiting Codecs? Tried sending the calls to an auto
attendant to see what results you get with that?
From: sipx-users-boun...@list.
Did you mean Audiocodes BOOTP tool?
-
MM
On Tue, May 17, 2011 at 12:23, Matthew Kitchin (public/usenet) <
mkitchin.pub...@gmail.com> wrote:
> FYI, after de-bricking it with with the Polycom BOOTP tool, it took the
> latest 5.8 fine. No more 6.x for me!
>
>
> On 5/16/2011 7:24 PM, Todd Hodgen wr
On 5/17/2011 12:13 PM, Tony Graziano wrote:
I would do a siptrace and see if the call is being PRACK'd or not. Is
this a HA system?
I will try next time she gets me some details on a failed call. It is
not HA. Plain vanilla setup.
On Tue, May 17, 2011 at 1:03 PM, Matthew Kitchin (public/usenet
On 5/17/2011 12:15 PM, Douglas Hubler wrote:
> On Tue, May 17, 2011 at 1:03 PM, Matthew Kitchin (public/usenet)
> wrote:
>> Ok. This problem. "About 7-8 times a day when the main line rings, I go
>> to answer it and no one is there, but on the LED screen the first button
>> says “answer”. You can
On Tue, May 17, 2011 at 1:03 PM, Matthew Kitchin (public/usenet)
wrote:
> Ok. This problem. "About 7-8 times a day when the main line rings, I go
> to answer it and no one is there, but on the LED screen the first button
> says “answer”. You cannot pick up the call at all and the caller hangs
> up
I would do a siptrace and see if the call is being PRACK'd or not. Is this a
HA system?
On Tue, May 17, 2011 at 1:03 PM, Matthew Kitchin (public/usenet) <
mkitchin.pub...@gmail.com> wrote:
> On 5/17/2011 11:56 AM, Douglas Hubler wrote:
> > On Tue, May 17, 2011 at 12:51 PM, Matthew Kitchin (usenet
On 5/17/2011 11:56 AM, Douglas Hubler wrote:
> On Tue, May 17, 2011 at 12:51 PM, Matthew Kitchin (usenet/public)
> wrote:
>> Sorry. We switched from auto attendant to user at ext 10 handling most
>> calls. That is when the problems began. 7 to 8 calls per day now are unable
>> to be answered, a
On Tue, May 17, 2011 at 12:51 PM, Matthew Kitchin (usenet/public)
wrote:
> Sorry. We switched from auto attendant to user at ext 10 handling most calls.
> That is when the problems began. 7 to 8 calls per day now are unable to be
> answered, and 50% of pages are unsuccessful.
focusing on proble
Sorry. We switched from auto attendant to user at ext 10 handling most calls.
That is when the problems began. 7 to 8 calls per day now are unable to be
answered, and 50% of pages are unsuccessful.
-Original Message-
From: Douglas Hubler
Sender: sipx-users-boun...@list.sipfoundry.org
Da
I'm having trouble understanding when the problem began. Can you
reformat this focusing on the very last change you made when the
problem started to occur.
On Tue, May 17, 2011 at 11:39 AM, Matthew Kitchin (public/usenet)
wrote:
> This is a fun one. I need some pointers on where to start.
> I hav
Dang it, I meant 82 (not 90) is a hunt group and 10 was member of
that. Sorry.
On 5/17/2011 11:02 AM, Matthew Kitchin (public/usenet) wrote:
One more additional note. I forgot about this one setting. The
initial call here is routed to a phantom exten
One more additional note. I forgot about this one setting. The
initial call here is routed to a phantom extension that then uses
scheduling and call forwarding to route to different handsets or
hunt groups depending on the time of day. These problems did occur
before the
This is a fun one. I need some pointers on where to start.
I have about 15 sites that are all roughly identical in their setup, so
I know the general design is ok.
This particular site was Sipx 4.2.1 and Polycom 450/550 firmware 3.2.4
and bootrom 4.3 up until last week.
They are now sipx 4.4 (lat
FYI, after de-bricking it with with the Polycom BOOTP tool, it took the
latest 5.8 fine. No more 6.x for me!
On 5/16/2011 7:24 PM, Todd Hodgen wrote:
Can you run them on 5.8 or 5.6 and not mess with updating them to 6.0?
Seems like an issue needs to be opened with Audiocodes.
*From:*sipx-use
On Tue, May 17, 2011 at 5:49 PM, George Niculae wrote:
> Hi All,
>
> I am trying to access sipXconfig from polycom microbrowser but I'm
> getting the following error:
>
> MicroBrowser - Polycom
>
> Error:
>
> SSL certificate problem, verify that the CA cert is ok.
> Details: error:140900086:SSL r
Hi All,
I am trying to access sipXconfig from polycom microbrowser but I'm
getting the following error:
MicroBrowser - Polycom
Error:
SSL certificate problem, verify that the CA cert is ok.
Details: error:140900086:SSL routines:
SSL3_GET_SERVER_CERTIFICATE:certificate verify failed.
I installe
I think I was one of the "bad power supply" issue Tony describes. It
isn't the same, but I'll recap for anyone as I have some new info that
may be of use to some.
Periodically we would hear a loud crackling noise (nothing made my blood
pressure shoot up faster). Upon inspection the power supply
It also helps not to use the combined firmware which takes far longer to boot.
Am 17.05.2011 um 08:43 schrieb Jan Fricke:
> I remember that our polycom phones took several minutes to boot when there
> was no default gateway configured on the SipX.
>
>
> While receiving configuration files fro
I do have a snapshot, anywhere I can send this to?
No alarms
Henry
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of George
Niculae
Sent: dinsdag 17 mei 2011 10:33
To: Discussion list for users of sipXecs softw
On Tue, May 17, 2011 at 11:28 AM, Henry Dogger wrote:
> Georgen,
>
> I did as proposed but again this morning my primary server went down...
> Any ideas?
>
Can you make a snapshot available? Is there any alarm raised before going down?
George
___
sipx-
Georgen,
I did as proposed but again this morning my primary server went down...
Any ideas?
Henry
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of George
Niculae
Sent: maandag 16 mei 2011 12:44
To: Discussion lis
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