Strange, I never saw this reply in the list.
By roberts358
>Thank you for the help in regarding this needed change.
>I was able to get things (Kinda) figured out enough to get
>this to work out. Needed to restart the service after the
>changes to the file were made.
>Thanks again for the postings
Hi !
I am using Polycom 331 phones, 3.2.3 SIP load, 4.2.2 bootrom. Using
the 4.2.0 ISO install.
When I have a shared user assigned to a few 331 phones, and I put a
call on hold, the screen shows two calls. If I take another call and
put it on hold (2 calls total), the screen of that phone will sh
> The call doesn't make it to the mediant so never makes it out of sipx. Do I
> need to enable something else?
Operator error, everything works fine.
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I think the approach is to use "branch" as the location.
On Mon, May 10, 2010 at 5:52 PM, Austin Curry wrote:
> I would like to know if anyone has a list or directions on the proper
> configuration for phones, users, gateways and dial plans for creating a
> location based dial plans.
>
> Thanks
I would like to know if anyone has a list or directions on the proper
configuration for phones, users, gateways and dial plans for creating a
location based dial plans.
Thanks,
Austin
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I guess one more little issue.
I have the new polycom phone and another phone on the same UserID.
The old phone is able to make and take calls while the poly is only able to
take calls.
Outgoing calls have dead air after dialing and nothing else happens.
The call doesn't make it to the mediant
On Mon, 10 May 2010 15:23:09 -0500, Josh Patten wrote:
> That is the awesome new Polycom auto provision feature. Now you can plug
Got it. She's all good to go now. Fancy, and awesome as well.
Sveet!
> phones in without first provisioning them, then assign them a user and a
> phone group onc
Related to: http://track.sipfoundry.org/browse/XX-8350
got that from the sipx developers list
Josh Patten
Assistant Network Administrator
Brazos County IT Dept.
(979) 361-4676
On 5/10/2010 3:42 PM, Marc Richardson wrote:
It's clear that the format for SOAP calls to manage users has changed
fr
It's clear that the format for SOAP calls to manage users has changed from
4.0 to 4.2, however the documentation for 4.2:
http://wiki.sipfoundry.org/display/xecsuserV4r2/Configuration+SOAP+APIs
points to the same WSDL, source and sample code as 4.0.
The sample_soap.pl and python scripts which wo
That is the awesome new Polycom auto provision feature. Now you can plug
phones in without first provisioning them, then assign them a user and a
phone group once they are online. To view the autoprovisioned phones
select -unassigned- from the "Filter By" drop down menu. Once you're
finished wi
Make sure you push the profile. The tftp server does not activate until the
first profile is pushed, I've found out.
On Mon, May 10, 2010 at 4:15 PM, m...@grounded.net wrote:
> Am I missing something again? I set up a new phone, uploaded device files,
> added a line.
> When it boots however, it j
Am I missing something again? I set up a new phone, uploaded device files,
added a line.
When it boots however, it just gets a Line 1: ID: 7M6
Haven't seen that before, must have missed something in the 4.2 update?
On Mon, 10 May 2010 14:59:14 -0500, Josh Patten wrote:
> Did you use the backup
AFAIK device files are not included in backups.
Josh Patten
Assistant Network Administrator
Brazos County IT Dept.
(979) 361-4676
On 5/10/2010 3:13 PM, m...@grounded.net wrote:
> On Mon, 10 May 2010 14:59:14 -0500, Josh Patten wrote:
>
>> Did you use the backup/restore method of upgrading?
Hi there, has anyone thought of or developed (unless I missed on
somewhere) a sipxecs feature listing or RFP tool question and answer
tool? If one does not exist, I was going to propose that we develop
one with a wiki format.
Does one exist today?
B
__
On Mon, 10 May 2010 14:59:14 -0500, Josh Patten wrote:
> Did you use the backup/restore method of upgrading? If so, your uploaded
> files aren't copied to the backup and you have to reload them all manually.
I didn't restore it since it would have brought in the ssl/cert problems I had
to start w
Did you use the backup/restore method of upgrading? If so, your uploaded
files aren't copied to the backup and you have to reload them all manually.
Josh Patten
Assistant Network Administrator
Brazos County IT Dept.
(979) 361-4676
On 5/10/2010 2:58 PM, m...@grounded.net wrote:
Ok, what gives?
Ok, what gives? Changing the device files out did the trick.
Phone is updating, downloading, etc as I type this.
Thanks.
On Mon, 10 May 2010 14:54:13 -0500, Josh Patten wrote:
> *facepalms*
>
> That's what happens when you have 2948529067247823 SSH sessions open
> (like I do)
>
> Josh Patten
>
*facepalms*
That's what happens when you have 2948529067247823 SSH sessions open
(like I do)
Josh Patten
Assistant Network Administrator
Brazos County IT Dept.
(979) 361-4676
On 5/10/2010 2:52 PM, m...@grounded.net wrote:
> Please disregard the xientd comment, I was looking at the wrong server
Please disregard the xientd comment, I was looking at the wrong server.
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On Mon, 10 May 2010 11:09:55 -0400, Michael Scheidell
wrote:
> If you don't have one, or you arn't using it as a remote, then just
> laugh at me and let me suffer for being stupid :-)
>
> But if you are using cisco 7960's natted in a remote office, using sipx
> natted at home office, and are NO
> I would go ahead and delete your polycom firmware/bootrom from sipXconfig
> then load firmware 3.2.3/bootrom 4.2.2 back in, then send profiles to the
> phones.
Currently, my devices files are spip_ssip_vvx_3_1_3RevC_release_sig_split.zip
and spip_ssip_vvx_BootROM_4_2_0_release_sig.zip. Those we
I would go ahead and delete your polycom firmware/bootrom from
sipXconfig then load firmware 3.2.3/bootrom 4.2.2 back in, then send
profiles to the phones.
Try that and post back results.
Josh Patten
Assistant Network Administrator
Brazos County IT Dept.
(979) 361-4676
On 5/10/2010 2:38 PM,
You could do that in a Patton gateway, but the PBX itself can't do it.
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Burleigh,
Matt
Sent: Monday, May 10, 2010 3:19 PM
To: sipx-users@list.sipfoundry.org
Subject: [sipx-users] Forward to
Both tftp and ftp from another host are able to get the syncinfo.xml file.
Next, wiki says: If this fails, check /etc/xinit.conf and /etc/xinit.d/* files
for proper access.
Neither exists.
I then cleared the config again using keys 1,3,5 and 7.
Same thing, 'application is not present'. Is this a
What problem are you seeing with the Polycom phones?
Josh Patten
Assistant Network Administrator
Brazos County IT Dept.
(979) 361-4676
On 5/10/2010 2:29 PM, m...@grounded.net wrote:
On Mon, 10 May 2010 14:23:08 -0500, Josh Patten wrote:
try the following:
wget ftp://PlcmSpIp:plcms.
On Mon, 10 May 2010 14:23:08 -0500, Josh Patten wrote:
> try the following:
>
> wget ftp://PlcmSpIp:plcms...@uc.domain.com/syncinfo.xml
Ok, that helped on that problem :).
So the polycom's have a different problem then?
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To be honest, I'm not that familiar with iptables. I'm waiting for it to do
the same thing so I can check where the packets get stuck.
On May 10, 2010, at 2:17 PM, WORLEY, Dale R (Dale) wrote:
>
> From: sipx-users-boun...@list.sipfoundry.org
> [sipx-u
try the following:
wget ftp://PlcmSpIp:plcms...@uc.domain.com/syncinfo.xml
Josh Patten
Assistant Network Administrator
Brazos County IT Dept.
(979) 361-4676
On 5/10/2010 2:18 PM, m...@grounded.net wrote:
> Can't seem to log into ftp to get polycom phones working on a 4.2 setup.
>
> The files ar
Is there a way to append or add the extension of an external number to a
forward? In the old modem days, we used a comma for pauses before
dialing the additional digits.
For example, I want to call 555-123-1212 and extension 710:
555-123-1212,,710
Thanks!
_
Can't seem to log into ftp to get polycom phones working on a 4.2 setup.
The files are in /var/sipxdata/configserver/phone/profile/tftproot.
The config tests all pass so am not sure what is wrong.
# wget ftp://plcmspip:plcms...@uc.domain.com/syncinfo.xml
--2010-05-10 14:11:50-- ftp://plcmspip:*p
From: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.org] On Behalf Of Francis Tinio
[fti...@toqen.com]
I just noticed something weird with our setup. After every couple days or so,
our sipx server stopped receiving any incom
"/app" should not be needed these days.
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:
ht
IF you hit it with
http://
it will redirect you to https://:8443/sipxconfig
you should not be typing https://:8443 and expect it to work
without the extra click.
If you do not forward BOTH prot 80 and 8443 from your firewall, the
experience will be the same/
On Mon, May 10, 2010 at 1:42 PM, Ja
I usually just setup 8443 through the firewall and know that I have to
hit: https://iporname:8443/sipxconfig/app
> -Original Message-
> From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-
> boun...@list.sipfoundry.org] On Behalf Of Jake Ballamis
> Sent: Monday, May 10, 2010
I'm actually not hitting it remotely most of the time. I get that
message whether I hit remotely or locally.
Jake Ballamis
Technical Support Manager
p. 801-566-TECH (8324)
f. 801-208-9317
jballa...@alliancetechsolutions.com
This e-mail is intended solely for the person or entity to which it is
Here is the sipxconfig log entry when I try to set a call forward.
I get the "Internal Error has occurred" message. This is an install from the
4.2 ISO.
Everything with sipxconfig works to this point, so it is not an httpd or
postgres issue.
Does anyone else have a similar issue with 4.2 install
So, never did get a solution on this, is there a way to make the page lists
longer? :)
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you lost me there.
How can iptables cause the states not to reset? Any workaround aside from the
cron? :)
On May 10, 2010, at 10:12 AM, Tony Graziano wrote:
> Well, I wouldn't do it the way you are. It sounds like the states are not
> resetting. I don't use iptables for this type of environm
If you don't have one, or you arn't using it as a remote, then just
laugh at me and let me suffer for being stupid :-)
But if you are using cisco 7960's natted in a remote office, using sipx
natted at home office, and are NOT using a VPN, then tell me what you
configured for nat_enabled, nat_r
Well, I wouldn't do it the way you are. It sounds like the states are not
resetting. I don't use iptables for this type of environment, because it is
limiting. You obviously can create a cron job to run in an off hour to do
this for you, but it will break any connections you have at that time.
Hi.
I just noticed something weird with our setup. After every couple days or so,
our sipx server stopped receiving any incoming calls. In order to fix we need
to flush iptables. Then it works again without a hitch.
We have enabled built-in internal firewall since our sipx is sitting on a
p
On Mon, 2010-05-10 at 07:58 +0200, Rene Pankratz wrote:
> Scott, thanks for your answer.
>
> I set up a site-to-site dialplan rule for the exact match on 298 going
> through my gateway but sipX still wants an authentication if 298 is
> dialed.
> When removing the unmanaged GW from another dialing
Hi
I solved this issue by performing two encoding steps:
1. Encode the 16-bit/48KHz source to 8-bit/8KHz G.711a format (my voicemail
system is using G.711a)
2. Encode the 8-bit/8KHz G.711a format to required 16-bit/8KHz signed WAV
format for SIPXecs
Strange but glad I managed to work around th
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