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sip
it somewhere friendly?
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I think that makes it all fixed!
On Jul 27, 2012 7:00 AM, Douglas Hubler dhub...@ezuce.com wrote:
On Thu, Jul 26, 2012 at 8:35 PM, Bryan Anderson shadow...@gmail.com
wrote:
Ok, Thanks for the update on the Polycom, now I have something regarding
updating DNS records/configuration.
I had
Does anyone have a good link for using Homer 3.25 web interface? I am now
able to login, but can't seem to generate any capture. I am up on the
latest update and trying to understand the interface and wondering if I am
doing something wrong.
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I had sipx shut down when I did the update then started sipx back up.
On Fri, Jul 27, 2012 at 8:05 AM, Douglas Hubler dhub...@ezuce.com wrote:
On Fri, Jul 27, 2012 at 7:45 AM, Tony Graziano
tgrazi...@myitdepartment.net wrote:
Does anyone have a good link for using Homer 3.25 web interface
thanks. not much in there yet!
Does anyone have an example of a homer working capture and what they had to
do in the interface to make the capture successful as an example?
On Fri, Jul 27, 2012 at 8:18 AM, Tony Graziano tgrazi...@myitdepartment.net
wrote:
I had sipx shut down when I did
at 8:27 AM, Tony Graziano tgrazi...@myitdepartment.net
wrote:
thanks. not much in there yet!
Does anyone have an example of a homer working capture and what they had
to do in the interface to make the capture successful as an example?
On Fri, Jul 27, 2012 at 8:18 AM, Tony Graziano
tgrazi
, Tony Graziano
tgrazi...@myitdepartment.net wrote:
my homer db is empty and is not capturing any data. I do not see that the
versions are the same between 32/64 bit in the repo. It appears the 64 bit
is newer by 17 hours. Is the current 32 bit supposed to contain the fix as
well (my test is 32
and immediately had more.
-Bryan Anderson
On Fri, Jul 27, 2012 at 7:57 AM, Tony Graziano
tgrazi...@myitdepartment.net wrote:
what are you seeing that tells you the system is capturing?
On Fri, Jul 27, 2012 at 10:54 AM, Bryan Anderson shadow...@gmail.comwrote:
My 64bit is capturing. I cant search
Does anyone know when 4.6 (32-bit) will have the latest homer fix?
On Fri, Jul 27, 2012 at 11:17 AM, Tony Graziano
tgrazi...@myitdepartment.net wrote:
right. I have none. I browsed the data already. I suspect the version in
the 32 bit repo still does not have the fix.
On Fri, Jul 27, 2012
.
Geoff Van Brunt
From: sipx-users-boun...@list.sipfoundry.org [mailto:
sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tony Graziano
Sent: Wednesday, July 25, 2012 5:19 PM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] FW: MS Lync and Exchange 2010
Certificates
clients. Lock in is not the way to
go.
Geoff Van Brunt
From: sipx-users-boun...@list.sipfoundry.org [mailto:
sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tony Graziano
Sent: Thursday, July 26, 2012 10:28 AM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users
ldap, etc.
George
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It appears that you might be refiaterunf via the IP address instead of the
sipdomain. If so that's a deal breaker.
On Jul 25, 2012 12:22 AM, Mark Dutton repl...@datamerge.com.au wrote:
I did some checking in the logs. It seems that the mwi
service is not responding to subscribe requests. The
I think it needs to the in the sip stack.
I think using ip tables might have an unintended consequence of blocking
other legitimate traffic that is sip based like mwi, registrations and
perhaps resource list stuff.
Putting this in the sip stack also means it needs to be determined whether
or not
Damn auto correct.
Are you REGISTERING via IP or manually configuring the phone? Realize your
posts are being manipulated by your mail client too.
On Jul 25, 2012 6:37 AM, Mark Dutton repl...@datamerge.com.au wrote:
[quote title=Tony Graziano wrote on Wed, 25 July 2012
17:02]It appears
Admission control only works if fully thought out. If you have a remote
user and that user makes a call via the pstn it has to be properly
accounted for. Simply putting a limit on the trunk handles only part of a
wide area network telephony system. While the trunk call is accounted for
what about
You unsubscribe yourself via this link:
http://list.sipfoundry.org/mailman/listinfo/sipx-users
On Jul 25, 2012 8:19 AM, Mark Roseboom mark.roseb...@38media.net wrote:
Please remove me from this list
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On Jul 25, 2012, at 12:26 AM, Mark Dutton repl...@datamerge.com.au
The n9 has a limited sip stack. As does android 4.1 (jelly bean). I don't
suggest using either. I find 3rd party applications have a far better
compatibility and would reserve to re-think that when the is embedded
sipstacks mature in these devices.
The n9 is a limited release model (not in US and
config file from there to see of the problem
persists. Otherwise you might be chasing nothing.
On Jul 25, 2012 8:46 AM, Mark Dutton repl...@datamerge.com.au wrote:
[quote title=Tony Graziano wrote on Wed, 25 July 2012
18:56]Damn auto correct.
Are you REGISTERING via IP or manually configuring
Graziano wrote on Wed, 25 July 2012[/color
18:56]Damn auto correct.
Are you REGISTERING via IP or manually configuring
the
phone? Realize your
posts are being manipulated by your mail client
too.
On Jul 25, 2012 6:37 AM, Mark Dutton
mailto:repl...@datamerge.com.au wrote
Albershardt k...@nv.net wrote:
Anyone have insights from this trace?
Thanks~
On Jul 24, 2012, at 14:32 , Kurt Albershardt wrote:
Attached - thanks.
merged.xml
On Jul 23, 2012, at 19:29 , Tony Graziano wrote:
A graphic won't help anyone troubleshoot it. Post the trace file
should be publicly visible.
John Lightfoot
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on the dev list when I have the resources in place.
Geoff Van Brunt
From: sipx-users-boun...@list.sipfoundry.org [mailto:
sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tony Graziano
Sent: Wednesday, July 25, 2012 4:11 PM
To: Discussion list for users of sipXecs software
Subject: Re
No real services are enabled by default. That's normal.
When you restored from backup did you restart the server? If the backup was
from a prior version also it may not work due to the way passwords were
stored and encrypted with the domain.
You might check the jira for the poly com issue and
occurred. Click here to
continue. All red text nothing else on the screen.
I will check the jira for a polycom issue.
-Bryan Anderson
On Wed, Jul 25, 2012 at 2:29 PM, Tony Graziano
tgrazi...@myitdepartment.net wrote:
No real services are enabled by default. That's normal.
When you restored
I also wonder if the poly com and NTP issues were a byproduct of the
restore. I wouldn't put much effort into that if it does not happen on
fresh non-restored system.
On Jul 25, 2012 5:51 PM, Tony Graziano tgrazi...@myitdepartment.net
wrote:
The restore was not compatible. I am not sure how
for testing. Now inbound calls are
working.
Back to 4.6 for another round of testing...
On Jul 25, 2012, at 12:51 , Tony Graziano wrote:
Explain what user or service has a did or alias of 5755195606. That is
what is. Being sent in the invite. Normally you would dial this 10 digit
number
...@ezuce.com wrote:
On Tue, Jul 24, 2012 at 6:53 AM, Tony Graziano
tgrazi...@myitdepartment.net wrote:
mongo is running
sipxconfig is running
What else can I provide?
Are you overriding any configs like DNS? Anything unique about this
system? Change the domain name? host name? You using
: 0.0.0.0, config:
/etc/mongod.conf, dbpath: /var/lib/mongo, fork: true, logappend:
true, logpath: /var/log/mongo/mongod.log, port: 27017, replSet:
sipxecs }
On Tue, Jul 24, 2012 at 8:35 AM, Tony Graziano tgrazi...@myitdepartment.net
wrote:
confused
is there a cli mechanism to project server
Graziano
tgrazi...@myitdepartment.net wrote:
confused
is there a cli mechanism to project server profile?
project server profile, no, but just running
sipxagent
should rebuild files. Try deleting the zone file, then run sipxagent.
Even after update/restart the manual zone serial number
:
On Tue, Jul 24, 2012 at 9:23 AM, Tony Graziano
tgrazi...@myitdepartment.net wrote:
sipdomain and domain are uc.myitdepartment.net, host name is
pbx.uc.myitdepartment.net
deleted
zone file and restarted with sipxagent. zone not re-created.
mongod still dead
still? Earlier you
You are better off just getting a pots line installed at that location
On Jul 24, 2012 11:07 AM, m...@grounded.net wrote:
Anyone know of an ITSP which also supports modems?
When switching old PBX hardware to SIP via a SIP to Analog gateway, there
is sometimes a problem in that the PBX
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Me thinks more bandwidth would be a better idea. In this type of instance
nothing will fix the problem without hitting it hard enough. Normally we
use a separate pass thru appliance to groom the internet bandwidth to keep
voice happy and prioritize by application (layer 7 stuff). We've never
found
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http
There are known issues with homer and it is being worked on.
You should still be able to put the logging level at debug for the
component(s) you need. Logs for sipx are (always have been) in
/var/log/sipxpbx
On Jul 23, 2012 12:00 PM, Mark Dutton repl...@datamerge.com.au wrote:
Hi all
I am
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A graphic won't help anyone troubleshoot it. Post the trace file.
On Jul 23, 2012 8:28 PM, Kurt Albershardt k...@nv.net wrote:
Ack - sorry about the large attachment.
On Jul 23, 2012, at 18:26 , Kurt Albershardt wrote:
Getting closer - have AON configured (thanks, Tony) on pfSense and am
, Tony Graziano wrote:
Read this
http://blog.myitdepartment.net/?p=37
On Jul 20, 2012 9:06 PM, Tony Graziano tgrazi...@myitdepartment.net
wrote:
Your outbound Nat type needs to be set for static port before your Nat
rules are created.
On Jul 20, 2012 8:03 PM, Kurt Albershardt k...@nv.net
. In this
instance 10443 is still safe but do what you want.
On Jul 21, 2012 1:32 PM, Kurt Albershardt k...@nv.net wrote:
On Jul 21, 2012, at 11:08 , Tony Graziano wrote:
The NAT rules have to be created AFTER the outbound Nat rule otherwise
they stay randomized.
I do understand that, and was planning to do
Contact: sip:70.57.247.39:5080
Content-Length: 0
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sip
at this
site anyway.
I'm curious about the suggestion to use a sipxbridge instance (rather than
an unmanaged gateway as was suggested yesterday) and choosing
bandwidth.com instead of Vitelity as the template?
thanks...
On Jul 20, 2012, at 11:38 , Tony Graziano wrote:
Then you would have set this up
I refer to the bandwidth.com template because it has all the correct
settings in there for non-registration (ip based only) and works fine in
such cases.
On Fri, Jul 20, 2012 at 2:05 PM, Kurt Albershardt k...@nv.net wrote:
On Jul 20, 2012, at 12:01 , Tony Graziano wrote:
I think if you
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Correct
On Jul 20, 2012 4:30 PM, Kurt Albershardt k...@nv.net wrote:
Got it.
I assume inbound RTSP gets negotiated during the SIP call setup and that
sipXbridge sends a packet to dynamically open the needed port on a per-call
basis?
On Jul 20, 2012, at 14:21 , Tony Graziano wrote
You will need to make sure the DID call number format in sipx is the
correct format +1npanxx 1npanxx npanxx etc.
I would look through the sipXbridge log (tail -f) when the call comes in to
see what is in the invite.
On Jul 20, 2012 5:11 PM, Kurt Albershardt k...@nv.net wrote:
:
Gotcha - thanks.
I did notice a whole lot of extraneous consumer-like config junk in the
Vitelity template.
On Jul 20, 2012, at 12:06 , Tony Graziano wrote:
I refer to the bandwidth.com template because it has all the correct
settings in there for non-registration (ip based only
sending might be messing
up the firewall state table. Is there some way to turn off the keepalives
since we have a static NAT mapping to allow inbound? Or is there a good
reason to leave them running?
On Jul 20, 2012, at 15:46 , Tony Graziano wrote:
You will need to make sure the DID call
:36.889707 IP sipx.domain.com.5080 66.241.X.X.5060: SIP, length: 4
17:48:56.896991 IP sipx.domain.com.5080 66.241.X.X.5060: SIP, length: 4
On Jul 20, 2012, at 17:36 , Tony Graziano wrote:
What is the firewall?
On Jul 20, 2012 7:22 PM, Kurt Albershardt k...@nv.net wrote:
Packets appear
Read this
http://blog.myitdepartment.net/?p=37
On Jul 20, 2012 9:06 PM, Tony Graziano tgrazi...@myitdepartment.net
wrote:
Your outbound Nat type needs to be set for static port before your Nat
rules are created.
On Jul 20, 2012 8:03 PM, Kurt Albershardt k...@nv.net wrote:
Forgot to mention
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this evening to do any testing.
Philippe
On Thu, Jul 19, 2012 at 4:06 PM, Tony Graziano
tgrazi...@myitdepartment.net wrote:
Can you shoot me a sip debug of the transaction?
Since it is a sipx ivr (version 4.4/latest, right?), both transactions
are attended transfers from the IVR (I am
to
reach out and hard reset, especially with the relatively new VVX500.
Don't see any 3.2 downgrades for the VVX. Looks like the 4.0x series is
all there is to choose. Next time I'm by that location, I could pop in an
IP650 and see how it behaves.
On Thu, Jul 19, 2012 at 4:28 PM, Tony Graziano
Setup an unmanaged gateway so sipXbridge knows to allow those calls.
On Jul 19, 2012 7:01 PM, Kurt Albershardt k...@nv.net wrote:
Thanks - in this case it's a carrier account that uses IP-based
authorization so there is no registration.
How would I associate the source IP with the gateway
The onboard reports are written in jasper. You might look at whether they
have a CLI option there and the reports can be customizable as well.
On Jul 18, 2012 1:31 AM, m...@mattkeys.net m...@mattkeys.net wrote:
That may need to be greater than or equal to ( = ) instead of just
greater than,
I think the system can email the user their relevant passwords upon account
creation...
On Jul 18, 2012 3:14 PM, Kurt Albershardt k...@nv.net wrote:
On Jul 18, 2012, at 1:15 , Todd Hodgen wrote:
Just for a point of discussion, setting these two password fields to
random numbers doesn’t seem
Graziano wrote:
I would install sipx as a subdomain (I.e. uc.sipdomain.TLD). I would point
the firewall DNS forwarder for both the host and subdomain to sipx and let
sipx run its own DNS.
Working towards that now, and I have been using sipx.domain.com as the
subdomain from the beginning
I think this deserves a wiki how to...
On Jul 16, 2012 6:15 AM, Michael Picher mpic...@ezuce.com wrote:
There's no 'automatic' way to do this.
You will need to run on 2 servers.
Put sipXbridge on the second server, turn it off on the master.
1-to-1 NAT each server to a different outside IP
are checked from a workstation before I
try to register phones. The phones resolve the name in order to pull the
config file.
On Jul 14, 2012, at 18:01 , Tony Graziano wrote:
Please verify the DNS zone is running. It sounds like DNS is not working.
There is a known issue with DNS in regard
as a subdomain (I.e. uc.sipdomain.TLD). I would point
the firewall DNS forwarder for both the host and subdomain to sipx and let
sipx run its own DNS.
I would not hard code the phones AT ALL.
On Jul 15, 2012 5:55 PM, Kurt Albershardt k...@nv.net wrote:
On Jul 15, 2012, at 3:08 , Tony Graziano wrote:
I'd
I think the thing you are leaving out is the phone config generated by sipx
can load a different DNS setting than what you intended, which is why I
suggested this deployment option to avoid a conflicting configuration on
the UA itself.
On Jul 15, 2012 7:06 PM, Tony Graziano tgrazi
Please verify the DNS zone is running. It sounds like DNS is not working.
There is a known issue with DNS in regard to the build. If it is not
running please change the zone serial number to MMDDXX save the zone
and ensure named is running then try again.
On Jul 14, 2012 7:51 PM, Kurt
with our Leased Books collection using the copy
buckets to make floating changes. But I can't imagine using the copy
bucket method for our entire collection. I want a magic on/off
switch!
Thank you,
Sue Graziano
Collection Development Librarian
Santa Cruz Public Libraries
117 Union St.
Santa Cruz
, 7 Jul 2012 14:40:01 -0400
(EDT)
From: Voicemail Notification Service
postmas...@localhost.uc.myitdepartment.net
To: tgrazi...@myitdepartment.net
Message-ID: 4784962.2.1341686401465.javamail.s...@pbx.uc.myitdepartment.net
Subject: Voice Message: Tony Graziano
MIME-Version: 1.0
Content-Type
...;)
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but you cant get in via sipxconfig via the browser because of the cert
error to begin with... so I am suggesting there should be a way to
rebuild this from cli...
On Fri, Jul 13, 2012 at 10:25 AM, Douglas Hubler dhub...@ezuce.com wrote:
On Fri, Jul 13, 2012 at 10:02 AM, Tony Graziano
tgrazi
...@drouillard.ca wrote:
On 7/13/2012 10:37 AM, Douglas Hubler wrote:
On Fri, Jul 13, 2012 at 9:24 AM, Tony Graziano
tgrazi...@myitdepartment.net wrote:
I think in this above case it was a manual install from a centos 6.2
netinstall (minimal) and sipx did not add or require sendmail (hmmm
, 2012 at 10:40 AM, Tony Graziano
tgrazi...@myitdepartment.net wrote:
but you cant get in via sipxconfig via the browser because of the cert
error to begin with... so I am suggesting there should be a way to
rebuild this from cli...
On Fri, Jul 13, 2012 at 10:25 AM, Douglas Hubler dhub...@ezuce.com
in this at this stage of
the release.
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Using or developing for sipXecs from SIPFoundry
built-in SIP Trunk SBC
Is this the method of managing sipXbridge in 4.4?
--thanks
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skills on 4.4 first, then start
tinkering with 4.6 once I get the build process down.
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choosing 'unmanaged service'
did something?
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On Sat, Jul 7, 2012 at 11:11 PM, Douglas Hubler dhub...@ezuce.com wrote:
On Sat, Jul 7, 2012 at 8:31 AM, Tony Graziano
tgrazi...@myitdepartment.net wrote:
FWIW - The certificate is not being regenerated, as it still have the
original create date/time on it.
I'll look into this. What if you
, 2012 at 5:40 AM, Tony Graziano
tgrazi...@myitdepartment.net wrote:
On Sat, Jul 7, 2012 at 11:11 PM, Douglas Hubler dhub...@ezuce.com wrote:
On Sat, Jul 7, 2012 at 8:31 AM, Tony Graziano
tgrazi...@myitdepartment.net wrote:
FWIW - The certificate is not being regenerated, as it still have
could also be totally clueless.
I am still seeing errors logging into webhomer and still unable to access it.
On Wed, Jul 11, 2012 at 5:43 AM, Tony Graziano
tgrazi...@myitdepartment.net wrote:
also..
sendmail sends as “localhost.{hostname}”, instead of “hostname”.
proxy does not answer calls
Would it be better to discuss the blf/rls standards available to see if
there is a more efficient standard for a wan environment and whether they
are functionally possible.
On Jul 11, 2012 8:49 AM, Nathaniel Watkins nwatk...@garrettcounty.org
wrote:
Are there any realistic options that would
They don't think Ethernet overhead. After you wrap the packet its 32k. Why
would you care what their sales monkey says?
On Jul 11, 2012 9:28 AM, m...@grounded.net m...@grounded.net wrote:
Appia doesn't hard code the codecs at all (if they limited it to G.729,
AA/VM wouldnt work...its ulaw
and they just announced g722 support...
On Wed, Jul 11, 2012 at 12:26 PM, m...@grounded.net m...@grounded.net wrote:
On Wed, 11 Jul 2012 10:09:22 -0400, Tony Graziano wrote:
They don't think Ethernet overhead. After you wrap the packet its 32k. Why
would you care what their sales monkey says
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is created? This would force a delete and re-create.
On Jul 11, 2012, at 13:19 , Tony Graziano wrote:
the system will project the config file with the mac address that was
configured. Don't manipulate the file name. Delete the phone and
create it with the proper mac address and project the file
.
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.
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Maybe a brief wiki page on the dos and dont's with proxmox and state your
user and media server load so people have the proper expectations if they
try this also.
On Jul 10, 2012 4:56 PM, Kurt Albershardt k...@nv.net wrote:
On Jul 10, 2012, at 8:29 , Kurt Albershardt wrote:
There seems be to
through B2BUA for remote worker support -- not necessarily
features that sipx would care about as it supports its own scheme for
remote worker).
On 7/4/12, Tony Graziano tgrazi...@myitdepartment.net wrote:
I know where you are coming from ranga. Still nice to hear from you
2184:51 java
On Jul 8, 2012, at 19:44 , Tony Graziano wrote:
How much ram did you allot?
On Jul 8, 2012 6:16 PM, Kurt Albershardt k...@nv.net wrote:
On Jul 8, 2012, at 16:12 , Kurt Albershardt wrote:
after a restart Java is munching ~95-99% of all available CPUs again.
FYI -- I
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Using or developing for sipXecs from SIPFoundry? Ask me
Without looking I recall opening 200 tickets with nexvortex using an ingate
in less than a year.
So if it were me, and its not, I'd be really sure to not take their word
for it and do my own verification.
On Jul 9, 2012 2:10 PM, Gerald Drouillard gerryl...@drouillard.ca wrote:
On 7/9/2012 12:53
and that is
the Comcast modem/router. If it were the ALG/Firewall wouldn't this impact
all calls?
-Bryan Anderson
On Mon, Jul 9, 2012 at 11:23 AM, Tony Graziano
tgrazi...@myitdepartment.net wrote:
Without looking I recall opening 200 tickets with nexvortex using an
ingate in less than a year.
So
I did the same install over the weekend without these kind of issues.
Centos 6.2 minimal install.
Set network info
Yum install epel-release
Yum update
Reboot
Yum groupinstall sipxecs
I did use a vmware environment though. SINCE this is a Linux host, I think
maybe you need a leap second fix for
How much ram did you allot?
On Jul 8, 2012 6:16 PM, Kurt Albershardt k...@nv.net wrote:
On Jul 8, 2012, at 16:12 , Kurt Albershardt wrote:
after a restart Java is munching ~95-99% of all available CPUs again.
FYI -- I allocated another proc to the VM, and CPU jumped from 197% (2
CPUs) to
, Douglas Hubler dhub...@ezuce.com wrote:
On Fri, Jul 6, 2012 at 6:13 PM, Tony Graziano
tgrazi...@myitdepartment.net wrote:
it did not ask for any input, just said it was done. certificate still
unusable and not updated. So I should think there would be a way to
generate ssl keys and a script
/index.php(33):
include('/var/www/html/w...')\n#3 {main}\n thrown in
/usr/share/www/webhomer-auth-sipx/auth.php on line 13, referer:
http://10.255.252.59/webhomer/index.php?component=search
On Sat, Jul 7, 2012 at 7:40 AM, Tony Graziano
tgrazi...@myitdepartment.net wrote:
ran
service sipxconfig db
a newer ISO since this one requires a yum
update before setup to get around the certificate issue.
On Sat, Jul 7, 2012 at 8:31 AM, Tony Graziano
tgrazi...@myitdepartment.net wrote:
FWIW - The certificate is not being regenerated, as it still have the
original create date/time on it.
It also seems
For some reason the DNS zone fails due to the serial number. Manually
changing the bind config to Date Based instead of running number
and putting a starting serial number of 2012070700 in allows bind to
start.
On Sat, Jul 7, 2012 at 2:25 PM, Tony Graziano
tgrazi...@myitdepartment.net wrote:
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