Its seeing the request and making an offer but not getting an ack back,
quite possibly some other dhcp server is running and also making an offer so
the ack won't be sent by the phone because its busy receiving another
offer...
Look at your network, this usually means a vlan config issue or an
ove
If you turn off your dhcp server does the "network initialize" on the phone.
Either you have vlans crossed or more than one dhcp server in the mix
somehow.
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431
Email: tgrazi...@myitdepartment.net
LAN/Teleph
Tony,
Well I restored the system from a backup from when it was initially
installed and before we added the Cisco 7960s to the mix. (BTW, the
7960's still make and receive calls because their old profiles are still
loaded,until the next power glitch anyway.)
The good thing is the Web UI no longe
It doesn't matter how cool it is if the backup has the flaw in it too.
If the corruption is the database (I really think it is) and the
backup is possibly corrupted too, depending on how far back you go...
I've pulled a bad db down into sql editors, razors, ran reindexes and
more. Never did find
Actually, the backup and restore process is pretty cool. The backup
creates a .dat file for each table that can be restored one by one with
the copy command. I don't really think it is a 'corruption' issue, it
really feels like a disconnect between DB values and some XML stored
values. Can be co
You can backup voicemail as a zip file and restore it after you build
the new system and import the users/phones you have exported.
Dialplans. Gateways. AA prompts need to be re-created and/or
downloaded/uploaded manually.
Tony
On Fri, Jun 11, 2010 at 6:01 PM, Thomas Packert wrote:
> OK - I wil
Tony,
Yes, when the Firefox browser crashes (v 3.6.3) (IE does same thing) then
this error appears in the SipXconfig.log I am doing tail -f on
/var/log/sipxpbx/sipxconfig.log
No, I can not create a fake phone - Any time I try to select the Polycom
550 is crashes the browser.
Tom
-Original M
OK - I will start planning a rebuild.
BTW - the delete commands were entered by me to 'delete' a polycom in
order to attempt to re-create it. But anytime I touch a polycom phone it
crashes the browser.
Thanks for all the suggestions will keep you posted.
Tom
-Original Message-
From: To
Sincerely, I think you have a DB issue.
This:
delete from phone_group where phone_id = 12;
delete from line where line_id = 15;
delete from phone where phone_id = 12;
And this:
org.sipfoundry.sipxconfig.phone.polycom.PolycomPhone incompatible with
org.sipfoundry.sipxconfig.phone.cisco.CiscoPhon
OK, I see it is from the sipxconfig.log. While there are a lot of
similarities, don't necessarily interchange the term SCS500 and
sipxecs. They use different version number and there are subtle
differences, which are changing daily.
On Fri, Jun 11, 2010 at 5:19 PM, Tony Graziano
wrote:
> Your sip
Your sipxconfig.log is where you saw these errors?
You are using IE or Firefox for your sipxconfig session? Chrome and
others may not work properly.
Do you have a recent backup?
Can you create a "fake" phone of the same make/model? No line needs to be added.
On Fri, Jun 11, 2010 at 4:57 PM, Tho
I am thinking I've tested against this and found the only way presence
was supported in bria was if you used BOTH the bria phone and the Bria
XMPP client.
Example:
Login as user 200 on Bria, login as pidgin on bria (also user 200).
Login as user 201 (Spark).
user 200 calls AA, 201 does not see
We are on version 4.0.4 (specifically 4.0.4-017289 2009-11-19)
It has not been upgraded recently, it is a recent install on this version.
-Original Message-
From: Tony Graziano [mailto:tgrazi...@myitdepartment.net]
Sent: Friday, June 11, 2010 3:32 PM
To: Thomas Packert
Cc: Josh Patten; s
> How does this affect presence with the phone. As I understood
> it the phone needed to support dialog events AND the IM
> client from Bria had to be used to observe those events. Is
> this not correct?
Not sure I fully understand the question but let me venture on an answer.
A user's telep
How does this affect presence with the phone. As I understood it the
phone needed to support dialog events AND the IM client from Bria had
to be used to observe those events. Is this not correct?
On Fri, Jun 11, 2010 at 4:38 PM, JOLY, ROBERT (ROBERT) wrote:
>
>
>> -Original Message-
>> Fr
> -Original Message-
> From: sipx-users-boun...@list.sipfoundry.org
> [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of
> Tony Graziano
> Sent: Friday, June 11, 2010 11:40 AM
> To: Paul Scheepens
> Cc: sipx-users@list.sipfoundry.org
> Subject: Re: [sipx-users] IM group not d
I have seen this before on systems upgraded from 4.0.4 to 4.2, it
related to users, but the error was similar. Out of order data,
meaning a corrupt database.
Your messages:
delete from phone_group where phone_id = 12;
delete from line where line_id = 15;
delete from phone where phone_id = 12;
Ma
What version SCS is this? Was it recently upgraded?
On Fri, Jun 11, 2010 at 3:11 PM, Thomas Packert wrote:
> All, Thanks for all the ideas and suggestions.
>
> 1 - The FTP server is running the Cisco 7960 phones can reboot just fine,
> and I see the DHCPDISCOVER, DHCPOFFER and TFT request perfe
All, Thanks for all the ideas and suggestions.
1 - The FTP server is running the Cisco 7960 phones can reboot just fine,
and I see the DHCPDISCOVER, DHCPOFFER and TFT request perfectly.
Jun 11 15:00:43 phonesystem in.tftpd[2083]: RRQ from 10.123.1.215 filename
SIP001B0C183702.cnf
Jun 11 15:01:1
The ftp server won't run until you push the first profile via sipxconfig,
neither does tftp. Or, it will run but nothing will be reachable "until" the
file is there, so...
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431
Email: tgrazi...@myitdepartment.
Or the FTP server isn't running
Josh Patten
Assistant Network Administrator
Brazos County IT Dept.
(979) 361-4676
On 6/11/2010 11:55 AM, Tony Graziano wrote:
> If the phone isn't booting it can't be discovered. It likely can't boot
> because its being offered incompatible bootrom/firmware.
> ===
If the phone isn't booting it can't be discovered. It likely can't boot
because its being offered incompatible bootrom/firmware.
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431
Email: tgrazi...@myitdepartment.net
LAN/Telephony/Security and Control Sys
On Fri, Jun 11, 2010 at 11:40 AM, Thomas Packert wrote:
>
> We have a new Nortel SCS500 system and have an error with the Polycom
> Phones. If they reboot they will not come back up. We have 9 Polycom 550
> phones. The Cisco 7960s boot up fine and are working perfectly. It is just
> the 9 Poly
First: Ensure you have the proper device files uploaded via the
sipxconfig>devices>device files.
You should have bootrom version 4.x (any 4.x version is fine) and
firmware 3.1.3RevC. The phones should load that without a config file
when they boot up.
Make sure you create the phone by the phone m
Hello all,
This is my first request for help on the SipFoundry list serve.
We have a new Nortel SCS500 system and have an error with the Polycom
Phones. If they reboot they will not come back up. We have 9 Polycom 550
phones. The Cisco 7960s boot up fine and are working perfectly. It is
From: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.org] On Behalf Of Osmose
[panagiotisbekia...@gmail.com]
But inside the list I cannot find an option for unmanaged phones.
___
Why
On Fri, Jun 11, 2010 at 11:28 AM, Tony Graziano
wrote:
> From Counterpath Support:
>
> "That's right we do not support XMPP via DNS_SRV lookup. and
> unfortunately there is no plan adding that on the current version."
>
>
> On Fri, Jun 11, 2010 at 9:27 AM, Paul Scheepens wrote:
>> I am trying to
>From Counterpath Support:
"That's right we do not support XMPP via DNS_SRV lookup. and
unfortunately there is no plan adding that on the current version."
On Fri, Jun 11, 2010 at 9:27 AM, Paul Scheepens wrote:
> I am trying to work out what would be the best method to set up presence.
> I've l
I thought I had tried that combination and still had early media.
I'll look again and report back.
On 06/11/2010 09:39 AM, Saint, David (David) wrote:
>
>
>
>> -Original Message-
>> From: sipx-users-boun...@list.sipfoundry.org
>> [mailto:sipx-users-boun...@list.sipfoundry.org] On Beha
> -Original Message-
> From: sipx-users-boun...@list.sipfoundry.org
> [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of
> Josh Patten
> Sent: Thursday, June 10, 2010 7:00 PM
> To: sipx-users@list.sipfoundry.org
> Subject: Re: [sipx-users] Audiocodes gateways disable voice de
Tried it, didn't work, caused the call to stay in early media no matter
what which caused it to drop at 60 seconds every time.
On 06/11/2010 02:39 AM, Nikolay Kondratyev wrote:
> Josh,
> Will the following
> ===
> ?? Answer Supervision: The Answer Supervision feature enables the FXO d
> I am trying to work out what would be the best method to set
> up presence.
> I've learned quite a lot already and will try to make some
> wiki doc in the future, but..
>
> One of the things I played with was Instant Messaging for a
> User Group.
> I created a UserGroup "GroupChat", in Use
I am trying to work out what would be the best method to set up presence.
I've learned quite a lot already and will try to make some wiki doc in the
future, but..
One of the things I played with was Instant Messaging for a User Group.
I created a UserGroup "GroupChat", in User Group settings, und
> Hi
>
> Click-to-call sends an initial INVITE with an SDP offer that
> only includes G.711u as audio codec. Once this call is
> answered, a REFER is then sent which effectively blind
> transfers the call to the remote party. During the transfer
> the two parties can negotiate new media howe
Hi
Click-to-call sends an initial INVITE with an SDP offer that only includes
G.711u as audio codec. Once this call is answered, a REFER is then sent
which effectively blind transfers the call to the remote party. During the
transfer the two parties can negotiate new media however if the initial
>>> Osmose 06/11/10 5:59 AM >>>
>>>There is a new fw available released the day before
>>>yesterday for all the yealink phones.
>>>I am contacting yealink to see what can be done.
>>>
>>>I will keep you up to date.
We had tested up to the beta firmware v50.25 and looks like these are the final
v
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Organization: SipXecs Forum
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There is a new fw available released the day before
yester
As I understand this feature, click to call uses "REFER", and is not
designed like a 3PCC B2BUA. Once the invite goes out the
phones/endpoints involve negotiate the media based on an offering.
Since click-to-call sends the refer, the "INVITE" to your phone does
not have media involved until it is n
Josh,
Will the following
===
?? Answer Supervision: The Answer Supervision feature enables the FXO device
to
determine when a call is connected, by using one of the following methods:
. Polarity Reversal: device sends a 200 OK in response to an INVITE only
when it
detects a polarity re
Hi
I have a small issue with Click-to-Call from Phonebook in SIPXecs. Problem
is that SIPXecs always sends a proposal for only G.711u in SIP INVITE to the
"calling" party. My deployment uses G.711a and ideally I'd like to be able
to control this if possible.
Any way to change this?
Thanks
Just
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