Yea, this isn't pretty…
Backup… export…
Build fresh with proper domain.
Add in your gateways, dial plan entries, user groups, phone groups, settings
for all of your features & services, then import your users (resetting all of
their PINs to some known value… or search the wiki for how to de
Looking for anyone that is using an Android Device on Sprint with a SIP
client in 3G or 4G mode. If you are currently using it, can you provide
the model number, and did you have to Root it in order to get it working?
Access via 3G/4G has been disabled in some models and I'm looking for one
t
Simplest way, if you can afford it:
1 - Remove/rotate all your logs on /var/log/sipxpbx
2 - Do the call test scenario you want to investigate
3 - Run sipx-trace -a -o merged.xml
One not-so-hard way to get the call-id is to look at the
CDR(Diagnostics->Call Detail Records) for the call then cli
>>> "m...@grounded.net" 05/10/12 2:23 PM >>>
>>When I ran the trace, I started it just before we made the call and ended it
>>as soon as the call failed. Should I wait a little longer as well?
No, but the callid you entered when you ran the command was not valid.
-M
__
Tony,
yes, phones are registered straight to the server and media path is our LAN...
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tony Graziano
Sent: Thursday, May 10, 2012 2:14 PM
To: Discussion list for u
I don't have an answer to this entire question, but until you get more info
from others, at a minimum, your pin number for each account is tied to the
domain name and will need to be reset. Try resetting the Admin password
from the command line and logging in. I believe it is documented in the
W
I need to change the domain on my sipxecs 4.4.0- 2012-04-13EDT09:33:36 with
800 phones and two HAs - all attempts on the test bench have lead to being
locked out of the gui config with the message "User ID and PIN combination
is not valid. Please try again."
I have tried changeing the domain in dom
When I ran the trace, I started it just before we made the call and ended it as
soon as the call failed. Should I wait a little longer as well?
On Thu, 10 May 2012 14:15:00 -0400, Matt White wrote:
"m...@grounded.net" 05/10/12 1:59 PM >>>
>
>
>> Yes, I see that in the trace; Max-Forwards:
On Thu, 10 May 2012 14:08:19 -0400, Tony Graziano wrote:
> You never explained your call flow. If the call originates from the
> ITSP, they set this value. They should not be sending a value that
> low.
I guess based on my earlier posting, I figured the call flow was obvious but
it's as follows.
On Thu, May 10, 2012 at 2:02 PM, Douglas Hubler wrote:
> On Thu, May 10, 2012 at 12:18 PM, Culbert, Chris wrote:
>> I have been noticing sporadically that when calling into voicemail that when
>> prompted to enter my pin it does not recognize any of the touch pad keys?
>> It doesn’t see any chang
>>> "m...@grounded.net" 05/10/12 1:59 PM >>>
>
>
>Yes, I see that in the trace; Max-Forwards: 13
>
>I don't see this as a setting in the gateway, is this something I can change?
>And even if so, why would it need to be >changed when sipx works fine with
>the other ITSP's? Does that p
The CANCEL is there because you have 2 phones registered under the same
user and one of them answered the call first. Not a real issue.
I didn't see the 488 Not Acceptable Here in the trace, but I noticed there
was only g.729 codec in the first INVITE w/ SDP and after sipXbridge
PCMU/PCMA. Maybe s
You never explained your call flow. If the call originates from the
ITSP, they set this value. They should not be sending a value that
low.
Sipx uses a sip friendly value of around 69 i think, which is the RFC
recommendation (as I recall).
On Thu, May 10, 2012 at 1:55 PM, m...@grounded.net wrote
On Thu, May 10, 2012 at 12:18 PM, Culbert, Chris wrote:
> I have been noticing sporadically that when calling into voicemail that when
> prompted to enter my pin it does not recognize any of the touch pad keys?
> It doesn’t see any change of state.
>
> Any quick fixes? Again very sporadic
Got a S
On Thu, 10 May 2012 13:30:10 -0400, Tony Graziano wrote:
> Why does the call come in with only 13 forwards? It's running out. See
> frame 11. It should have 60'ish (69) as a default. Either way 13 is
> way to low.
Yes, I see that in the trace; Max-Forwards: 13
I don't see this as a setting in t
On Thu, May 10, 2012 at 12:08 PM, George Niculae wrote:
> On Thu, May 10, 2012 at 7:05 PM, Douglas Hubler wrote:
> On this note, there are only 10 seats on such google hangout and eZuce
> team counts about 5 that will join, so this means ~5 seats available.
But please join hangout, if we ever ge
Why does the call come in with only 13 forwards? It's running out. See
frame 11. It should have 60'ish (69) as a default. Either way 13 is
way to low.
On Thu, May 10, 2012 at 1:24 PM, m...@grounded.net wrote:
> Here is the trace anyhow.
>
> Mike
>
>
> _
I've got a trace and am looking at it in sip viewer. The two things that seem
to be important are as follows.
The call answered elsewhere is strange. I don't see any mention of a codec
problem so far.
Mike
Time: 2012-05-10T17:11:46.402637Z
Frame: 40 sipXproxy.xml:3411 sipXproxy.xml:3412
Sourc
Hello all,
I have been noticing sporadically that when calling into voicemail that when
prompted to enter my pin it does not recognize any of the touch pad keys? It
doesn't see any change of state.
Any quick fixes? Again very sporadic
Thanks,
Chris Culbert
Messiah College
___
> As an FYI, a sipx-trace is the starting point with all failed calls/trunk
> issues. So spend time learning to read them.
Thanks. I'll do just that right now.
Mike
___
sipx-users mailing list
sipx-users@list.sipfoundry.org
List Archive: http://list.s
sipx-trace -a -o tracefile.xml CALLID
if you don't know the call id the to/from phone number works too.
rotate the logs before you make the test call and then when you use the phone
number as the call-id you wont pick up any other calls in the trace.
As an FYI, a sipx-trace is the starting poin
On Thu, May 10, 2012 at 7:05 PM, Douglas Hubler wrote:
> On Wed, May 9, 2012 at 10:37 AM, Douglas Hubler wrote:
>> Would anyone be interested in joining a google hangout this friday to
>> talk about sipxecs? In the spirit of a hangout out, no one is really
>> going drive the hangout, but it migh
On Wed, 9 May 2012 17:50:29 -0500, m...@grounded.net wrote:
> Have not had the chance to do a trace but I did get their pcap.
Ok, so I don't often have to run traces so I've forgotten how to.
Searching, I've come up with lots of examples but am not sure which one is best
for this purpose.
Can so
On Wed, May 9, 2012 at 10:37 AM, Douglas Hubler wrote:
> Would anyone be interested in joining a google hangout this friday to
> talk about sipxecs? In the spirit of a hangout out, no one is really
> going drive the hangout, but it might be useful to prepare some topics
> and some of us will be p
About 18 months ago I was experimenting with a 4610sw running SIP off
SipX. As I recall, codec negotiation on the Avaya phone was not working
consistently. In the 46XXsettings file, I needed to restrict the phone
to using a single codec G.711u in my case. I did get it to work.
don
From: sip
yess let me give you an example:
curl --insecure -X PUT -u 2010:2010
https://2010:2010@192.168.69.1:8443/sipxconfig/rest/my/conference/StanzaG/change
--data-binary @prova.xml
and in data binary file "prova.xml" do you need to report the same xml
format you get from
curl --insecure -X GET -u
Ohhh, thank you. I will look into it.
Any particular requirements (patch level) before applying patch? Do you have a
description of the commands?
~~
Carl B. Constantine, IT Analyst
Server, Network and Telecomm Infrastructure
IT Servic
here i've post a patch:
http://track.sipfoundry.org/browse/XX-10052
that allow "change" command on the jet create conference rooms:
you can change every paramenter from the pin to the owner
On Tue, May 8, 2012 at 8:55 PM, Mircea Carasel wrote:
>
>
> On Tue, May 8, 2012 at 9:43 PM, Carl Constan
Hi,
I have tried with a snom 300 and and all incoming call made from Bria or
Jitsi are working so it's looks to be related to Avaya.
So that's mean that Avaya phone 4620 with latest firmware doesn't support
direct RTP communication, how can I prove it ?
Best Regards.
2012/4/30 Cyril Constanti
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