here? reboot is not helping. everything (date
time, dial plans, services) looks fine normal. stumped (and
panicking slightly).
thanks
milosz
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to restart services as prompted. make sure
the time/date is correct on the system.
On Mon, Jul 23, 2012 at 2:53 PM, milosz mew...@gmail.com wrote:
hi all,
my autoattendants are suddenly all giving the message sorry, our
office is closed no matter what autoattendant alias/extension you
call
Use a better phone. The m3 is a consumer grade device.
i gave up on these things for customer use a long time ago... i had
four of them (which, no bitterness, i /did/ buy on your recommendation
a number of years ago =) and i've been trying to get them to work for
internal and work-from-home use.
hi sipx-users,
long time.
is anyone using snom m3's with the ta904 gateway? have you had an
issue with outbound calls being dropped after 30 seconds (ostensibly
because the phone rewrites the route header to blank)? have you found
a workaround? let me know.
thanks
milosz
ps. m3's are still
i have had an enormous amount of problems with the snom m3's. the
hardware is poor and the stack is abysmal. if you want an idea of
what you'll be dealing with, take a look at the firmware bugfix
inventory.
i moved to kirk handsets with kws-300 servers and haven't looked back.
it's more than
for dns!
milosz
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, all processes running
without complaint. any ideas for where to start troubleshooting?
upgraded using the 4.0.4 repos at sipxecssw.org
thanks,
milosz
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hi guys,
does 4.0+ allow the literal + delimiter in voicemail-to-email? 3.10.3 won't
let you input that, and it would be a very useful (for rule/filtering
purposes) and extremely simple to implement.
milosz
[image: untitled.JPG]
untitled.JPG___
sipx
I agree but sometimes, you just don't have that luxury :).
lol@ luxury.
so let's see, you want something that will work well.
you want something that is also easily configurable by lay people.
and you want it to be under $1000.
but you have not found anything.
however you continue to
however you continue to maintain that a grand is very realistic from
what you can tell (hmmm).
Um, ya, just look around on the net, tons of hardware, see above :).
um, ya, this is the point which you are missing... does it match your
criteria? no, it does not, because if it did, you
jonathan and carlos: how many users are you guys supporting with the pfsense
setups?
On Thu, Aug 13, 2009 at 1:42 PM, Jonathan Petersen
jonathan.peter...@ontraonline.com wrote:
I am using pfSense with sipXbridge and it works great! The only
“special” thing that was required was enabling
the
archives you'll find tons of threads about it.
the bottom line is: a whitebox pc with 2gb of ram and an old cpu running
sipx on bare metal will probably perform better than a virtual machine with
full access to your box's resources.
daniel: how much and what kind of testing have you done?
milosz
- Looking thru the other audiocodes4.csv - It just
didn't look right - so here is another one...
From: milosz [mailto:mew...@gmail.com]
Sent: Wed 5/13/2009 5:38 PM
To: Nathaniel Watkins
Cc: sipX users
Subject: Re: [sipx-users] Hunt Group 3 extensions
just read through this thread... nathaniel, try setting the syslog level on
the audiocodes to max, do a capture of a bad call, and send the file. also,
what's the firmware on the audiocodes?
On Wed, May 13, 2009 at 1:30 PM, Scott Lawrence
scott.lawre...@nortel.comwrote:
On Wed, 2009-05-13 at
there are only 3 extensions
in the huntgroup (i.e. everything works perfectly)...
--
*From:* milosz [mailto:mew...@gmail.com]
*Sent:* Wed 5/13/2009 1:41 PM
*To:* Nathaniel Watkins
*Cc:* sipX users
*Subject:* Re: [sipx-users] Hunt Group 3 extensions is not falling back
i recommend you email patton support about this, esp. since it is not a sipx
issue.
On Fri, May 8, 2009 at 9:26 AM, McCoy, Chris cmc...@cmctechgroup.comwrote:
I have a production sipx box with 2 patton smartnode gateways one is a
4560 the other is a 4118, the 4560 has a pri with dids on it,
have you tried doing an iperf udp test between the main site and the remote?
that'll give you an idea of whether it's a network vs. application issue.
On Wed, May 6, 2009 at 11:38 AM, Tim Byng t...@missioninc.com wrote:
There are two pieces that haven't been fully discussed here, the
i love how they can't spell cache. or maybe that is internal
audiocodes slang.
have you actually configured a dns server for the audiocodes? cause
it looks like you have not. set the dns server, then burn and reset
the gateway.
also a-records are way more reliable with the audiocodes gear
in protocol def proxy
registration. fyi my pbx vendor who recommends audiocodes actually
does not use dns with the audiocodes -at all- (this is impossible with
sipx, unfortunately). wonder why...
On Wed, Apr 29, 2009 at 4:00 PM, milosz mew...@gmail.com wrote:
i love how they can't spell cache
if you can't figure it out from there then you need to send a full
syslog capture. a few pasted lines is basically useless, i am just
basically throwing very mildly educated guesses at you.
On Wed, Apr 29, 2009 at 4:09 PM, milosz mew...@gmail.com wrote:
oh, wait a minute... it looks like
a
thread where you stated that you were concerned that unauthorized ua's
could make calls through the pattons unless you used acl's. what
mechanism (other than number obfuscation through a dial plan) would
you use to ensure that unauthorized users wouldn't exploit this?
milosz
wrote:
milosz wrote:
you can have the patton route the calls directly to the second gateway
(instead of going through sipx). not sure if that's what you want. i
believe there is an example of it in the smartware manual.
Yes, this is what I've already done, but in a large organization
:
On Thu, 2009-04-02 at 11:17 -0400, milosz wrote:
i wonder how easy it would be to
(1) pull the list of extensions off your pbx and throw them in an
excel sheet (or something)
(2) use this list to generate a bunch of insert statements for the
users table in sipxconfig
(3) use same list sip
you can have the patton route the calls directly to the second gateway
(instead of going through sipx). not sure if that's what you want. i
believe there is an example of it in the smartware manual.
the other thing you could do is create a user for the 200 extension in
sipx and have the 4960
you could also email supp...@patton.com with your question... but it
looks like your problem is you've blocked your phones.
On Sat, Mar 28, 2009 at 11:10 AM, Massimo Vignone
massimo.vign...@unimore.it wrote:
Hi everybody,
After configuring my Patton Smartonode gateways with the following ACL,
you can technically record calls by doing packet captures, but it's a
pretty clunky and high-overhead setup.
On Thu, Mar 26, 2009 at 8:24 AM, Keith Gearty ke...@glensound.co.uk wrote:
That's a shame. I figured out that you could record a call by putting
the user on hold while you dial your own
really? that is news to me, as i have 7960's working just fine on
netgear, dell, and adtran switches.
i'm guessing you are referring to cdp, which is a protocol often
supported by third-party vendors (netgear, for one) (a colleague once
characterized it as cisco destroy phone), or pre-standard
when i get a chance.
On Wed, Mar 25, 2009 at 7:16 PM, Damian Krzeminski dkrze...@nortel.com wrote:
milosz wrote:
er, running 3.10.3.
On Fri, Mar 20, 2009 at 7:31 PM, milosz mew...@gmail.com
mailto:mew...@gmail.com wrote:
hi all,
do i need to do something special to block caller-id
send an audiocodes syslog capture also.
On Tue, Mar 24, 2009 at 7:37 PM, Scott Lawrence
scott.lawre...@nortel.com wrote:
On Wed, 2009-03-25 at 06:30 +0800, Cuneyt M wrote:
Hi everyone,
I just make a new installation for a new site with sipx 3.10.2 Centos
ISO (updated via stable repo to
as i said in a previous email, which maybe wasn't clear enough,
blocking caller id on a per-group basis does not work for me. is it
working for other people?
On Wed, Mar 25, 2009 at 4:56 PM, Damian Krzeminski dkrze...@nortel.com wrote:
McCoy, Chris wrote:
I have a customer, they are looking to
i'm a little confused here. are you saying that gateways cannot send
inbound calls to users? or just analog gateways? how can that
possibly be true (unless you are talking about the specific software
limitations of specific gateways)?
On Mon, Mar 23, 2009 at 1:16 PM, Tony Graziano
:
As far as I could remember analog gateways could never call users, just hunt
groups or AA's (this was the case with both AudioCodes and Patton unless
someone can tell me something has changed within sipx to make it no longer an
issue).
milosz mew...@gmail.com 03/23/09 2:50 PM
i'm a little
.
Tony
milosz mew...@gmail.com 03/23/09 2:57 PM
seriously, no one but me (and tony) is using cordless phones with sipx?
On Wed, Mar 18, 2009 at 5:15 PM, milosz mew...@gmail.com wrote:
hi all,
what are people using for cordless handsets? i am not happy with my
snom m3's right now.
milosz
er, running 3.10.3.
On Fri, Mar 20, 2009 at 7:31 PM, milosz mew...@gmail.com wrote:
hi all,
do i need to do something special to block caller-id for an entire group
other than checking the box and hitting apply? it seems to work fine for
single users, but not user groups.
(why am i doing
software that causes my outgoing calls to
fail.)
milosz
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hi all,
what are people using for cordless handsets? i am not happy with my
snom m3's right now.
milosz
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this is a very common topic of discussion on the list and i recommend
you search the archives for more information.
pattons seem to be the gateways of choice at the moment.
On Tue, Mar 17, 2009 at 4:45 PM, natif na...@altern.org wrote:
Hello,
I'm looking for some type of mediagateway to use
] On Behalf Of milosz
Sent: Thursday, March 12, 2009 2:04 PM
To: sipx-users@list.sipfoundry.org
Subject: [sipx-users] patton smartnode 5.3 firmware
anyone having success using this with 3.10.3 right now? 4.2 is EOL
and i'm having an issue with early media (for some reason sprint and
vzw voicemail
, Jan 31, 2009 at 9:55 AM, Scott Lawrence
scott.lawre...@nortel.com wrote:
On Fri, 2009-01-30 at 20:21 -0500, milosz wrote:
hi all,
anyone else having this issue with snom m3 transfers and the patton
gateways? transfers from the snoms to other phones calls are failling
with 403's for incoming
try capturing the syslog output from the mediant for a bad forward
to see if it complains about anything.
try using manipulation tables in the mediant to have it pass 200 to
sipx rather than the full 555 number.
On Fri, Feb 27, 2009 at 7:24 PM, Chris St Denis
chris.stde...@vhisper.com wrote:
this list has suddenly crossed over from professional interest to
pretty entertaining for me.
but, on a non-antagonistic note, i would actually be interested to
know what will differentiate sipxecs 4.0 from its commercial
antecedent.
On Wed, Feb 25, 2009 at 3:05 PM, voice vo...@netgeneral.net
protocol management routing tables ip to hunt group routing
what do you have in there?
you should have trunk groups set up with your fxo channels in there.
if you have that just send me your config ini file and i'll look at it.
On Mon, Feb 23, 2009 at 10:05 AM, Mitchell, Kenny (Ineos)
what version of the mediant firmware are you running?
have you done a packet capture to determine what the traffic looks
like? debug syslog capture from the mediant? sipx call trace? all
these things can help you isolate the problem.
On Thu, Feb 19, 2009 at 3:15 PM, Cuneyt M
wouldn't that be kind of a big issue if your sites are basically on
opposite sides of the globe? are you able to keep your latency
jitter manageable?
On Wed, Feb 18, 2009 at 1:53 PM, Michael Picher
mpic...@cmctechgroup.com wrote:
The only reason you might not would be that all voicemail and AA
in order for us to help you you need to provide more information. for
example: what is it that is not booting? the install iso or centos
itself post-install? what is the exact error you are getting?
On Fri, Feb 6, 2009 at 2:57 PM, sabb01 @hotmail.com sab...@hotmail.com wrote:
Hi,
I have
what version of the mediant firmware?
what is your sip transport protocol set to on the mediant?
have you tried setting up a syslog capture (with the acsyslog tool,
for example)?
On Thu, Feb 5, 2009 at 7:24 AM, Mitchell, Kenny (Ineos)
kenny.mitch...@ineos.com wrote:
Hi team,
I saw a mail
owned two mediants for over a year and i haven't even
seen a link to a manual.
On Thu, Feb 5, 2009 at 9:27 AM, Mitchell, Kenny (Ineos)
kenny.mitch...@ineos.com wrote:
Hi Milosz
I'm currently running version 5.0 of the Mediant firmware, I'm awaiting
the latest version (5.2?) being sent to me
are you saying that the phone doesn't issue a dhcp request when it is
coming up unless its lease is expired or something? that is bizarre.
i actually recommend you do a wireshark capture to make sure that this
is what is really happening. are you sure you don't have something
weird like multiple
To: Meller, Milosz
Cc: sipx-users@list.sipfoundry.org
Subject: RE: [sipx-users] clearing false mwi?
On Fri, 2009-01-30 at 11:53 -0500, Meller, Milosz wrote:
i'm using sipxconfig, though i must have broken this while changing
the messaging settings to work with url dialing disabled. pretty
192.168.3.19;branch=z9hG4bK-sipXecs-6e5e1434db4e263edd1230d4418c825a2fd7
Via: SIP/2.0/UDP 172.20.3.108;branch=z9hG4bK2d77aee4DE9BADF3
From: \Milosz Meller\ sip:6...@oc.local;tag=BC0A382A-57F1F8F1
To: sip:1...@oc.local
Cseq: 2 SUBSCRIBE
Call-Id: 15cdcdbd-5d684c20-2f419...@172.20.3.108
Contact: sip:6
what you want to look for is _why_ the authentication is not accepted.
Look for WARN or ERR level entries in sipstatus.log
that is the odd thing... there don't seem to be any relevant ones.
actually there are no ERR or WARN messages in there at all since i
disabled vm subscription on the phones
putting multiple vm backends on a single raid 5 is a bad idea no matter
how many drives you have. have you done any kind of performance testing
on that array? i guarantee you the random i/o is garbage. i can't
imagine the performance on those other vm's is very good if they're at
all
,
milosz
-Original Message-
From: Scott Lawrence [mailto:scott.lawre...@nortel.com]
Sent: Thursday, January 01, 2009 2:33 PM
To: Meller, Milosz
Cc: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] clearing false mwi?
On Tue, 2008-12-30 at 12:33 -0500, Meller, Milosz wrote:
hi
given a gateway that supports it, theoretically it should be easy to
receive t.38 or g711 faxes, convert them to tiff, and email them without
any kind of extra boards or appliances. the box that tony's using is
$700, which isn't going to break the bank, but there's not really any
good reason i
. every time i make a change i have to email someone and tell
them to reboot their phone. or reboot their switch. awesome.
From: Chris St Denis [mailto:[EMAIL PROTECTED]
Sent: Tuesday, November 04, 2008 8:07 PM
To: Meller, Milosz
Cc: Tony Graziano; sipx-users
are you not having the polycom-polycom-polycom blind transfer issue with
this firmware? cause i am. supposedly it's fixed in 3.1. i'm
seriously thinking about just upgrading to get that out of my hair. no
one in that environment uses blf anyway.
also, anyone else having issues with their
that polycom is considered well-supported but
frickin' music on hold doesn't work.
milosz
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt White
Sent: Monday, July 14, 2008 2:16 PM
To: sipx-users@list.sipfoundry.org
Subject: [sipx-users] Alternative
yeah, that's what i was thinking, was hoping there was an easier way :)
From: Picher, Michael [mailto:[EMAIL PROTECTED]
Sent: Wednesday, June 18, 2008 7:30 PM
To: Meller, Milosz; sipx-users@list.sipfoundry.org
Subject: RE: [sipx-users] zero for operator prompt
have all our phones rebooting several
times a day, but i'll maybe do some work on it later. getting a major
kick out of this.
the phones are polycom 650's with 3.0.2c firmware/4.1.0 bootrom.
audiocodes mediant 1000 gateway. sipx 3.8.1.
milosz
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sipx
hey guys,
anyone know how to turn off the to reach the operator, dial zero at any
time prompt before voicemail greeting? my users HATE it.
haven't done any research on the phones rebooting, will go back to it
when i have some time. put in a workaround in the meantime.
thanks,
milosz
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