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I read in the release notes that there was an integration
with dim dim. having become frustrated with the lack of
support an
Charles, what venerable scott is trying to get across is
0 means NO and 1 is yes. So change that.
Personally I think it could all be easier if software guys started coding
repos and things with Y or N.
Change that, yum clean all, have at it again.
Tony Graziano, Mana
On Fri, 2010-04-16 at 15:30 -0700, Charles wrote:
> enabled=0
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sipX
I forgot this but do I still have something wrong with my repo?
[CentOS5 base]
name=CentOS-5-Base
mirrorlist=http://mirrorlist.centos.org/?release=5&arch=$basearch&repo=os
baseurl=http://mirror.centos.org/centos/5/os/i386/
gpgcheck=0
enabled=0
gpgkey=http://mirror.centos.org/centos/RPM-GPG-KEY-Cen
Sorry for the silly question but how do you see if the repo url is functional
or how do you find a functional one
Charles
On Apr 16, 2010, at 3:21 PM, Tony Graziano wrote:
You need a functional Centos repo to get the deps.
Tony Graziano, Manager
Telephone: 434.
You need a functional Centos repo to get the deps.
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431
Email: tgrazi...@myitdepartment.net
LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: 434.984.8426
Fax: 434.984.8427
Helpdesk Contract Cu
Attempting to yum update to 4.2 via CLI from 4.0.4 but am having issues with
some dependencies...suggestions?
Error: Missing Dependency: libasound.so.2(ALSA_0.9.0rc4) is needed by package
sipx-freeswitch
Error: Missing Dependency: alsa-lib is needed by package sipx-freeswitch
Error: Missing De
Ok, sounds good. Thanks for all the hard work!
Ken
-Original Message-
From: Scott Lawrence [mailto:xmlsc...@gmail.com]
Sent: Friday, April 16, 2010 4:10 PM
To: Ken Fulmer
Cc: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] 4.2 Documentation
On Fri, 2010-04-16 at 15:29 -0500, K
Ok, thanks. Just checking.
Ken
-Original Message-
From: Scott Lawrence [mailto:xmlsc...@gmail.com]
Sent: Friday, April 16, 2010 4:11 PM
To: Ken Fulmer
Cc: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] 4.2 Cluster Scalability
On Fri, 2010-04-16 at 15:35 -0500, Ken Fulmer wrote
On Fri, 2010-04-16 at 16:10 -0400, Michael Scheidell wrote:
> We were waiting for 4.2 to 'go live', which it is, and along with
> regression testing, we will be testing inbound DID.
> previously, we just took the odd DID and put it in as the alias for
> the user, now that we have a 100 block of DID
On Fri, 2010-04-16 at 15:35 -0500, Ken Fulmer wrote:
> I’d asked a question regarding how many servers could exist in a
> cluster. The reply was no more than 5 had been tested.
>
>
>
> Has this number been scaled higher in 4.2?
There have been no significant changes in that area.
On Fri, 2010-04-16 at 15:29 -0500, Ken Fulmer wrote:
> Sorry if already asked:
>
>
>
> Is there a specific area in the wiki for 4.2 information?
There will be early next week I need a tool installed on the wiki to
clone the 4.0 documentation so that we can update it.
__
I'd asked a question regarding how many servers could exist in a cluster.
The reply was no more than 5 had been tested.
Has this number been scaled higher in 4.2?
Thanks,
Ken Fulmer
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Sorry if already asked:
Is there a specific area in the wiki for 4.2 information?
Thanks,
Ken Fulmer
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On 4/16/10 4:15 PM, Eric Varsanyi wrote:
The 'current' polycom boot rom is spip_ssip_BootROM_4_2_2_release_sig.zip , the
current SIP firmware (the application) is
spip_ssip_vvx_3_2_3_release_sig_combined.zip .
i upgraded 4.04 to 4.2 (needed to manually reboot polycom ONCE)
running boot 4
The 'current' polycom boot rom is spip_ssip_BootROM_4_2_2_release_sig.zip , the
current SIP firmware (the application) is
spip_ssip_vvx_3_2_3_release_sig_combined.zip .
Polycom provides a package of config files with each release that go 'behind'
(further down the viewpath) the files generated
We were waiting for 4.2 to 'go live', which it is, and along with
regression testing, we will be testing inbound DID.
previously, we just took the odd DID and put it in as the alias for the
user, now that we have a 100 block of DID's, I wanted to make sure we
did it right.
Internal calls seem to
> they are aware of it and working on a fix.
>
> As far as we know right now, this will not be a problem
> except with phones that are on networks that significantly
> delay responses from the phone (we found it with one of our
> remote users who's got a slow DSL link). The symptom you'll
> s
On Fri, 2010-04-16 at 15:18 -0400, Francis Tinio wrote:
> If we upgrade to 4.2, do we need to use bootrom 3.2.3 for polycom?
> That is the latest version for IP500.
We've done most of our testing with the 3.2.2 firmware (which is not the
same as the bootrom), I believe.
>sipXivr will look for {prefix}/etc/sipxpbx/sipxivr/EmailFormats.properties
>(including all EmailFormats_{locale}.properties variants) first, and if
>not found will use the default formats defined in
>org.sipfoundry.voicemail.EmailFormats.properties (in the sipXivr.jar file).
>Per user selectable
If we upgrade to 4.2, do we need to use bootrom 3.2.3 for polycom? That is the
latest version for IP500.
On Apr 16, 2010, at 3:10 PM, Scott Lawrence wrote:
> that there are other problems that
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sipx-users mailing list sipx-users@list.sipfoundry.
On Fri, 2010-04-16 at 14:49 -0400, Francis Tinio wrote:
>
> i'll try to update to 3.2.3 and update sipx to 4.2 again later.
I should have put this in the release notes and forgot...
There is an outstanding problem with Polycom version 3.2.3
http://track.sipfoundry.org/browse/XTRN-1046
they a
hmmm
we use 3.1.0 for our polycom ip500. we tried 3.2.3 before but it was crashing
on us, and 3.1 worked so we stuck to that.
i'll try to update to 3.2.3 and update sipx to 4.2 again later.
On Apr 16, 2010, at 2:41 PM, Eric Varsanyi wrote:
> When you get the reboot loop on the polycoms
When you get the reboot loop on the polycoms look in the tftproot directory for
the app.log file for the phone. When I got this recently (due to some changes I
had made to the polycom plugin) it was because of a parsing error in the
generated .cfg file. It may be that the new config changes for
in the user profile, its called "Groups". and you now have a Branch.
(Group didn't change)
under gateways, what used to be groups is now location, but it looks
like the only options now are what you entered into as a branch. (and
group doesn't exist anymore)
it doesn't look like you can put
On 4/16/10 1:43 PM, Tony Graziano wrote:
I would create the default branch. Assign the SBC to it, make sure that any
field for system devices that need branch get it assigned and restart
services as needed.
It looks like I have to go through the users, one at a time and assign
them to a br
You might switch from TUN to TAP interface bridged with the ETH0 of the SipX
machine. This would result in a VPN tunnel directly into the LAN and VPN
client will be able to connect to SipX with the "normal" IP address.
But also all other devices in LAN will become available to VPN client...
René
Yes, I noticed the branch issue last night, and did assign the branch
throughout. I recall that issue from when I first worked with 4.1.x
The phones are working fine internally. This seems to be an issue with the
sipxconfig, apache, or something else on the fringe. I just haven't had a
lot of ti
I would create the default branch. Assign the SBC to it, make sure that any
field for system devices that need branch get it assigned and restart
services as needed.
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431
Email: tgrazi...@myitdepartment.net
L
On 4/16/10 1:39 PM, Tony Graziano wrote:
Did you manually reboot the phone?
I've seen issues with the "browser" with 4.1x (or later) when you
don;t follow through everywhere and create a branch and assign it.
There are options that wouldn't come up until "branch" was addressed
in several place
On 4/16/10 1:35 PM, Francis Tinio wrote:
reverted back to 4.0.4. will try again tonight when office is closed
lol
anyway, did you have to do anything in particular to get the update?
or in the config files, like reinstall config files of the phones?
yes, just for fun, sent profiles to
Did you manually reboot the phone?
If the SBC is gone, I would disable the trunking role and restart
services, then enable the role and restart.
I've seen issues with the "browser" with 4.1x (or later) when you
don;t follow through everywhere and create a branch and assign it.
There are options t
can anyone send me their repo file for centos? I'm running centos 5.2
(installed initial server via sipxecs ISO 4.0.2 or 4.0.3). wehn I initially
ran yum update I was getting file not founds for the centos mirrors so had to
manually change.
On Apr 16, 2010, at 1:35 PM, Francis Tinio wrote
reverted back to 4.0.4. will try again tonight when office is closed lol
anyway, did you have to do anything in particular to get the update? or in the
config files, like reinstall config files of the phones?
On Apr 16, 2010, at 1:26 PM, Michael Scheidell wrote:
> On 4/16/10 1:17 PM, Fran
Interesting, I have a similar issue with a system that no longer runs.
I sent profiles on the system early this morning, and it was processing
calls internally, but I could never get the gateway working. When I went
into the system/internet there was no internal SBC available. When I looked
a
On 4/16/10 1:17 PM, Francis Tinio wrote:
ok updated via yum update.
nice gui interface cahnge. refreshing.
however, now my system is not running. I rebooted one phone, and now goes into
endless loop and says config error 0x0.
mine went fine, centos5, it sent profiles 'automatically' to
Make sure your server was sent profiles. Make sure to send profiles to
all phones. Check to see that all your roles are properly enabled that
you use and in a running state.
On Fri, Apr 16, 2010 at 1:17 PM, Francis Tinio wrote:
> ok updated via yum update.
>
> nice gui interface cahnge. refreshin
ok updated via yum update.
nice gui interface cahnge. refreshing.
however, now my system is not running. I rebooted one phone, and now goes into
endless loop and says config error 0x0.
On Apr 16, 2010, at 12:00 PM, Tony Graziano wrote:
> Can't use the gui. Read the release notes.
> =
With MoH now built on freeswitch, is the moh file format still restricted to
8kHz sampling, mono, and 16 bits per sample and save as Microsoft PCM .wav file?
-M
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On Fri, 2010-04-16 at 10:44 -0500, Josh Patten wrote:
> Any update on the 64 bit build?
I don't expect we'll have that posted any earlier than the middle of
next week
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Can't use the gui. Read the release notes.
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431
Email: tgrazi...@myitdepartment.net
LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: 434.984.8426
Fax: 434.984.8427
Helpdesk Contract Customers:
Release notes for 4.2 say to use the cli. There is another issue with some
package dependencies so a working OS (whataver distro you use, like centos)
repo might be needed.
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431
Email: tgrazi...@myitdepartment
32bit.
On Apr 16, 2010, at 11:59 AM, mkitchin.pub...@gmail.com wrote:
> Are you running 64 bit or 32 bit?
> Sent via BlackBerry from T-Mobile
>
> -Original Message-
> From: Francis Tinio
> Date: Fri, 16 Apr 2010 11:56:48
> To: Josh Patten
> Cc:
> Subject: Re: [sipx-users] sipXecs 4.2
Are you running 64 bit or 32 bit?
Sent via BlackBerry from T-Mobile
-Original Message-
From: Francis Tinio
Date: Fri, 16 Apr 2010 11:56:48
To: Josh Patten
Cc:
Subject: Re: [sipx-users] sipXecs 4.2.0 Released !
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sipx-users mailing list sipx
my bad it's 4.0.4.
It's not updating to 4.2 via sipx gui.
On Apr 16, 2010, at 11:56 AM, Francis Tinio wrote:
> we are in 4.0.2 I see update 4.2.0 available. However, when we try to
> update via sipx panel, it reboots but does not update.
>
>
> On Apr 16, 2010, at 11:44 AM, Josh Patten
we are in 4.0.2 I see update 4.2.0 available. However, when we try to
update via sipx panel, it reboots but does not update.
On Apr 16, 2010, at 11:44 AM, Josh Patten wrote:
> Any update on the 64 bit build?
> Josh Patten
> Assistant Network Administrator
> Brazos County IT Dept.
> (979)
Also, we are on 4.0.2 version. I'll try to update to 4.2.2 see if it helps.
Thanks
On Apr 16, 2010, at 11:45 AM, Francis Tinio wrote:
> basically, this is how we're setup with this client.
>
> The sipx server is in our Philly datacenter. It's not behind NAT and it has
> a public IP with fire
basically, this is how we're setup with this client.
The sipx server is in our Philly datacenter. It's not behind NAT and it has a
public IP with firewall turned on. All ports closed except the ones I said
earlier. Then we have 2 locations. Our client's office, and our office. We
have phon
Any update on the 64 bit build?
Josh Patten
Assistant Network Administrator
Brazos County IT Dept.
(979) 361-4676
On 4/15/2010 4:05 PM, Matthew Kitchin (public/usenet) wrote:
On 4/15/2010 3:42 PM, Scott Lawrence wrote:
At this time we don't have 64 bit rpms built (those build servers got
turn
On Fri, 2010-04-16 at 10:59 -0400, Francis Tinio wrote:
> Hi.
>
> Our issue with calls suddenly disconnecting after 18 secs has been
> fixed.
>
> However, it appears another problem that I initially thought was
> linked to this is not. We cannot seem to call another extension on a
> different lo
On 4/16/10 11:22 AM, an...@iguanait.com wrote:
Hi,
we have installed sipxecs 4.0.4 on centos 5. We installed also openvpn
server on this machine, because we want to create a tunnel for one of
our users. We want this user to use this tunnel to connect to sipxecs.
I can ping client's ip address a
Hi,
we have installed sipxecs 4.0.4 on centos 5. We installed also openvpn
server on this machine, because we want to create a tunnel for one of
our users. We want this user to use this tunnel to connect to sipxecs.
I can ping client's ip address and he can ping openvpn server.
But he is not abl
It was from the DID provider with incorrect settings.
On Apr 16, 2010, at 11:05 AM, JOLY, ROBERT (ROBERT) wrote:
>> Our issue with calls suddenly disconnecting after 18 secs has
>> been fixed.
>>
>> However, it appears another problem that I initially thought
>> was linked to this is not. W
Yeah! I can confirm that this forwarding issue between Sipx/Verizon is
resolved in 4.2. Thank you for all the hard work. It looks great from
what I have seen so far.
On 1/27/2010 2:59 PM, M. Ranganathan wrote:
> On Wed, Jan 27, 2010 at 3:53 PM, mkitchin.pub...@gmail.com
> wrote:
>
>> I hav
> Our issue with calls suddenly disconnecting after 18 secs has
> been fixed.
>
> However, it appears another problem that I initially thought
> was linked to this is not. We cannot seem to call another
> extension on a different location. Nor can we transfer calls
> between phones on differ
Hi.
Our issue with calls suddenly disconnecting after 18 secs has been fixed.
However, it appears another problem that I initially thought was linked to this
is not. We cannot seem to call another extension on a different location. Nor
can we transfer calls between phones on different locatio
> Hi, All,
>
> I am running 4.0.4-018580 in my lab and want to test
> Aggressive mode in Media Relay. My sipx stays in 192.168.2.4
> and two of my test end points reside respectively in
> 192.168.2.31 and 10.33.150.150.
>
> When I turn on the Aggressive mode in Medial Relay and then
> make a
On 4/16/2010 12:42 AM, Eric Varsanyi wrote:
> I can do a one off build of FC12 RPM's if anyone needs them (I 32 or 64 bit).
>
> -Eric
>
> On Apr 15, 2010, at 8:04 PM, Josh Patten wrote:
>
>
>> +1
>>
>> I have plenty of VM resources I can dedicate to getting the 64 bit RPM's
>> built.
>>
>> mkit
On Fri, 2010-04-16 at 13:03 +, Scott Richesson wrote:
> Quick question:
> During the upgrade from 4.0.4 to 4.2, will the legacy voicemail
> messages/greetings/names/etc. be moved to the new voicemail system, or
> is this information lost?
It should preserve all customizations (the storage of t
The voicemail directory structure and meta data (xml files) is identical
in 4.0.x and 4.2.x. Everything should be the same after you upgrade,
though I would make a backup of the system before you do anything.
Josh Patten
Assistant Network Administrator
Brazos County IT Dept.
(979) 361-4676
On
Quick question:
During the upgrade from 4.0.4 to 4.2, will the legacy voicemail
messages/greetings/names/etc. be moved to the new voicemail system, or is this
information lost?
Thank You,
Scott Richesson
IT Manager
Cincinnati Fan
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Wow, Thank you Tony.
You know what...when this system ran 4.0.4(upgraded from 4.0.3) it did work
without assigning the did as alias to aa in dial plan. That is why i assumed
it sort of caught all dids not specified elsewhere.
Now i know.
Thank you and have a nice weekend.
*Vänliga Hälsningar/Bes
> Yes, go to your etc/yum.repos.d/sipXecs.repo file to make edits.
> Change all lines that have 5.2 in it to 5
Didn't work for me when I tried that yesterday so I just enabled the one in the
repo file and was able to update. Any problem with that? Two of the servers
I've done are working fine i
Its an either you direct all inbound calls to an internal destination or you
route calls by DID, which means the DID should be an alias of the AA in the
fial plan.
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431
Email: tgrazi...@myitdepartment.net
LAN
Hi!I was under the impression that if i did not direct calls at SBC
level and did not assign dids as aliases to users,
that the autoattendant defined in the dialplan would catch all calls and
direct them
to which ever attendants that i had selected(based on schedule).
Is this not true?
Sending
On Thu, 2010-04-15 at 20:44 -0400, Matt White wrote:
>
> I tried to see if the listserv could be modified to have an exception
> for any email from the forums IP address, but I don't have access to
> the list server.
It does not have such an option.
_
Rhon, well, sipX to sipX isn't REALLY working then is it if you have to
go through the PBX.
Take a PC, configure it's IP info exactly in the same manner you would a
phone. If you are using static IP info, use the same static IP info, if
you are using DHCP, use DHCP.
Do the DNS tests from t
Ok, did
sipxconfig.sh --database drop create
could add route and now itsp account shows.
thank you
have a nice weekend.
*Vänliga Hälsningar/Best Regards*
/Ola Samuelson
/
2010-04-16 10:05, Ola Samuelson skrev:
Ran sipxecs-setup-system but that does not delete all settings.
How do i do that th
hi all,
my apologies for bothering you all with my previous post...already found
the problemforgot to add my gateway to the domain and specify its name
server.
these commands were the ones i neglected to include in the router
*ip domain name ourcompany.com*
*ip name-server 172.16.10.254*
Ran sipxecs-setup-system but that does not delete all settings.
How do i do that the best way without reinstalling everything?
*Vänliga Hälsningar/Best Regards*
/Ola Samuelson
/
2010-04-15 22:35, M. Ranganathan skrev:
Not sure what is happening but I would venture to guess it might be
cured
Thanks for the link and useful info. :)
On Fri, 2010-04-16 at 00:27 -0700, Todd Hodgen wrote:
> Yes, go to your etc/yum.repos.d/sipXecs.repo file to make edits.
>
> Change all lines that have 5.2 in it to 5
>
> On the last section, the sipXecs section, change it to the following:
> [sipxecs-stab
Yes, go to your etc/yum.repos.d/sipXecs.repo file to make edits.
Change all lines that have 5.2 in it to 5
On the last section, the sipXecs section, change it to the following:
[sipxecs-stable]
name=SIPfoundry sipXecs pbx - latest stable version
baseurl=http://sipxecs.sipfoundry.org/pub/sipXecs/L
an...@iguanait.com wrote:
> Hi,
>
> thanks for all your work done on sipxecs and big thanks for new release
> 4.2.0 :)
>
> I would like to ask if there are any special steps that we need to do in
> order to upgrade from 4.0.4 to 4.2.0 via yum?
>
> We have installed 4.0.4 on centos 5 system.
>
> Tha
Hi,
thanks for all your work done on sipxecs and big thanks for new release
4.2.0 :)
I would like to ask if there are any special steps that we need to do in
order to upgrade from 4.0.4 to 4.2.0 via yum?
We have installed 4.0.4 on centos 5 system.
Thanks in advanced!
__
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