On Wednesday, December 12, 2012, Tommy Laino wrote:
>
>
> Tony I am not sure what top is. I do not believe that we are
> using swap at all. Todd I would think that maybe you culd be
> right but this is all the phones and it was ll of a sudden.
> I did notice that somehow my FQDN ended up in the Do
Tony I am not sure what top is. I do not believe that we are
using swap at all. Todd I would think that maybe you culd be
right but this is all the phones and it was ll of a sudden.
I did notice that somehow my FQDN ended up in the Domain
Alias section not sure why it was in there or what I was
d
This is a line that I got from the sipstatus log file
"2012-12-11T23:59:34.892934Z"
:34:SIP:WARNING:www.mydomain.com:SubscribeServerThread
handleSubscribeRequest() - voicemailCGI GET failed with
-1."
"2012-12-12T01:02:34.224658Z"
:44:SIP:ERR:www.mydomain.com:HttpServer-2:40604940:SipStatus
:
"
Maybe a stretch - Checked the POE switch? Had a phone resetting just the
other day when they hit the messages button on the phone - as bizarre as
that sounds, it turned out to be a cable issue between switch and phone. Go
figure?!? Still scratching my head.
-Original Message-
From: sipx
I suggested top not the Mrtg statistics. Are toy using any swap?
On Dec 11, 2012 5:40 PM, "Tommy Laino" wrote:
>
>
> I do not see either of those logs in my snapshot. I checked
> the statistics for the server and I am not having any issues
> with resources
> --
> Tommy Laino
> Dome Technologies
>
On Wed, Dec 12, 2012 at 12:37 AM, Tommy Laino wrote:
>
>
> I do not see either of those logs in my snapshot. I checked
> the statistics for the server and I am not having any issues
> with resources
>
>
Sorry /var/log/sipxpbx/sipstatus.log and sipxivr.log
George
_
I do not see either of those logs in my snapshot. I checked
the statistics for the server and I am not having any issues
with resources
--
Tommy Laino
Dome Technologies
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Also co spider looking g at top and seeing if you have. Resource issues in
general.
On Dec 11, 2012 5:17 PM, "George Niculae" wrote:
> On Wed, Dec 12, 2012 at 12:15 AM, Tommy Laino wrote:
>
>>
>>
>> I got SipXecs 4.4 with all Polycom 335/550 phones with 3.2.6
>> firmware. Everything was working
On Wed, Dec 12, 2012 at 12:15 AM, Tommy Laino wrote:
>
>
> I got SipXecs 4.4 with all Polycom 335/550 phones with 3.2.6
> firmware. Everything was working just fine until today.
> Suddenly people would take their voicemails and the light
> would not extinguish. On the other hand others are receiv
I got SipXecs 4.4 with all Polycom 335/550 phones with 3.2.6
firmware. Everything was working just fine until today.
Suddenly people would take their voicemails and the light
would not extinguish. On the other hand others are receiving
messages and not getting any MWI. What gives? I checked the
S
in a pcap, dtmf will appear to press multiple times and reading that is
actually quite confusing. are you using the same ITSP or gateway that she
is using? If not, you are not actually recreating her call.
I have seen this in the past with itsp's who use a different payload number
for dtmf. If it
Actually, I think it might be that this is a new ITSP and I fat fingered
something in the config.
The ITSP (appia) platform (cardinal using opensips) requires sending calls
to them on the SRV record OR I have to define the gateway(s) by A records
(hostname).
It seems I had the proxy entered in as
I have 1 user with a softphone that calls into several automated systems
throughout the day and for some reason she cannot navigate through them. From
the captures I've done it appears that when she presses the numbers either on
the keyboard or the dial pad built into the softphone the signal is
Highly important that you confirm that this works for you pre dec-5 update.
On 12/12/2012 12:01 AM, Tony Graziano wrote:
Is it just me?
I have a dialplan rule when I use "xx" plus 10 digits to strip the
"xx" and send the 10 digits to a specified gateway. When I do this
with the latest stage (
in my case permissions match in both dial rules.
On Tue, Dec 11, 2012 at 11:21 AM, Elwin Formsma wrote:
> Hi Tony,
>
> Mabye slightly related to this:
>
> Rule 1: Dialplan rule with XX plus 10 => does not require permissions
> Rule 2: There is also a dialplan with 10 digits => does require
> p
Hi Tony,
Mabye slightly related to this:
Rule 1: Dialplan rule with XX plus 10 => does not require permissions
Rule 2: There is also a dialplan with 10 digits => does require permissions YY
User doesnt has permissions YY
When you dial XX+10 digits (follows Rule 1) you can dial it without problem
Is it just me?
I have a dialplan rule when I use "xx" plus 10 digits to strip the "xx" and
send the 10 digits to a specified gateway. When I do this with the latest
stage (dated dec. 5) I get a 500 internal server error from sipx.
Exception Info Unexpected error creating INVITE at SipUtilities.j
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