is list working?
___
sipx-users mailing list
sipx-users@list.sipfoundry.org
List Archive: http://list.sipfoundry.org/archive/sipx-users/
ACK
___
sipx-users mailing list
sipx-users@list.sipfoundry.org
List Archive: http://list.sipfoundry.org/archive/sipx-users/
Content-Type: text/plain;
charset=utf-8
Content-Transfer-Encoding: 8bit
Organization: SipXecs Forum
X-FUDforum: 08063afcdd00a6e76393c5b9527381e8 61039
Message-ID: ee6f.4e03b...@forum.sipfoundry.org
Has anyone else noticed anything odd when setting a user's
call forwarding to ring a cell
I am running the following version:
sipXconfig (4.2.1-018932 2010-06-25T10:43:40 build33)
I have a problem with one account.
People call in and successfully leave messages for this user.
The account holder can connect to the web interface and see all the new
messages.
Attempts to Play any
Yea, you don't want to 'trust' dscp because sipXecs doesn't set it
This is actually a big problem as I see it. What could be more important
to VoIP than QOS? Yet SipX doesn't support dscp. I've had a jira (for
Polycoms at least) here:
http://track.sipfoundry.org/browse/XX-6018
Please vote for
resending...
On Wed, Jun 22, 2011 at 1:14 PM, Douglas Hubler dhub...@ezuce.com wrote:
Update #8 : Wed, 22 June 2011
==
- ** No security updates in this update **
- ISO has *not* been rebuilt as decided in release policy. Yum update
OK, I've reconfigured the port as follows to mark all ingress traffic
with a dscp value of 46 (ef):
interface GigabitEthernet1/0/7
stp edged-port enable
broadcast-suppression pps 3000
port access vlan 162
undo jumboframe enable
description sipXecs
traffic-priority inbound ip-group
Would you be able to post snort signatures you set up against these attacks?
Thanks in advance
___
sipx-users mailing list
sipx-users@list.sipfoundry.org
List Archive: http://list.sipfoundry.org/archive/sipx-users/
___
sipx-users mailing list
sipx-users@list.sipfoundry.org
List Archive: http://list.sipfoundry.org/archive/sipx-users/
Content-Type: text/plain;
charset=utf-8
Content-Transfer-Encoding: 8bit
Organization: SipXecs Forum
X-FUDforum: 08063afcdd00a6e76393c5b9527381e8 61037
Message-ID: ee6d.4e02e...@forum.sipfoundry.org
Hi,
When I tried to make an outgoing call from SipXezPhone
found that call failed with
Tony, you're heavy into development. Have you heard of any RTMP (like red5)
apps that would work with sipx in terms of being able to have a browser based
phone? I know of a few apps but none that look very usable to the average
person.
On Tue, 21 Jun 2011 18:55:06 -0400, Tony Graziano wrote:
On Thu, Jun 23, 2011 at 10:29 AM, m...@grounded.net m...@grounded.net wrote:
Tony, you're heavy into development. Have you heard of any RTMP (like red5)
apps that would work with sipx in terms of being able to have a browser based
phone? I know of a few apps but none that look very usable to
On Thu, Jun 23, 2011 at 3:20 AM, Nihar nihar.ra...@planet1world.com wrote:
When I tried to make an outgoing call from SipXezPhone
found that call failed with CAUSE_NO_RESPONSE event and call
is disconnected with remote phone.I can see that there is
nothing wrong with remote x-lite phone.It's
if it doesn't you need to update sipx so you get the fs patch that fixes
stutter on media server
On Jun 24, 2011 5:06 AM, Alex Brown alex...@bellsouth.net wrote:
OK, I've reconfigured the port as follows to mark all ingress traffic
with a dscp value of 46 (ef):
interface GigabitEthernet1/0/7
I've never tried. In the way paging is designed, its entirely possible that
latency is causing the audio to be too late and not used.
broadcast tone then go offhook and now deliver this audio to all
members.
try a paging group of one remote phone. my guess is it will work. the local
phones all
Testing to see if I can reach the list.
___
sipx-users mailing list
sipx-users@list.sipfoundry.org
List Archive: http://list.sipfoundry.org/archive/sipx-users/
Hi Tony,
Guessing you would know who maintains the list. My SMTP server is reporting
that the list server host drive is full so is rejecting emails. Not sure who to
contact about it that would get immediate attention but figured you would.
Mike
On Wed, 25 May 2011 12:14:40 -0400, Tony
Alex,
With regards to the Polycom phones the DSCP settings are set. I'll forward
you a doc off-list explaining how to read the Polycom settings.
With regards to sipXecs and setting RTP and SIP DSCP values, I agree.
Mike
On Fri, Jun 24, 2011 at 5:48 AM, Tony Graziano
yeah, its fixed now.
On Thu, Jun 23, 2011 at 4:37 PM, m...@grounded.net m...@grounded.net wrote:
Hi Tony,
Guessing you would know who maintains the list. My SMTP server is reporting
that the list server host drive is full so is rejecting emails. Not sure who
to contact about it that would
--
==
Tony Graziano, Manager
Telephone: 434.984.8430
sip: tgrazi...@voice.myitdepartment.net
Fax: 434.326.5325
Email: tgrazi...@myitdepartment.net
LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: 434.984.8426
sip: helpd...@voice.myitdepartment.net
Helpdesk
further I would expect this cannot be fixed on this current design. it would
have to be a join and silence until all members join, then allow a
broadcast design driven from the media server. How long do you wait and
what if members are on the phone? besides being complicated to design (I
have it
Well Paul, that is a good question! Fact is that Tele2 offers it in their
INVITE SDP anyway.
I learned today that according to the Tele2 engineers and Andreas from Unet,
the reason that the call is dropped is in the number '19' that is sent in the
SDP response:
m=image 0 UDPTL 19
19 is the
I'd raise a sipXecs JIRA (http://track.sipfoundry.org) as an external
problem so this is at least flagged. If you would like it addressed sooner
rather than later I'd also raise it over at FS.
On Wed, Jun 22, 2011 at 7:56 PM, Peter van der Salm
peter.vanders...@smart-future.nl wrote:
Well
On Thu, Jun 23, 2011 at 5:08 PM, Norman Branitsky
nor...@cherniaksoftware.com wrote:
I don't see much difference.
Suggestions, please.
Did you upgrade from 4.0.4? It's possible you hit this
http://track.sipfoundry.org/browse/XX-9461
there is a workaround posted
Am I the only that finds RFC3551 interesting in that it doesn't apply
to t38? Why would the ITSP reference RFC3551? It's for audio and video
only.
the port 0 with PT of 19 is sofia rejecting the sdp FS doesn't support
it. The only time it came up in FS is when an ITSP was not ignoring
port zero
Tony,
I'm not sure if this is the same, but...
There was a customer that had an issue where 50 percent of about 100
polycoms would randomly lockup. They were not on calls at the time,
just out of the blue lock-ups. I never tracked down the ultimate
problem, but did come up with a workaround.
OK, I've reconfigured the port as follows to mark all ingress traffic
with a dscp value of 46 (ef):
interface GigabitEthernet1/0/7
stp edged-port enable
broadcast-suppression pps 3000
port access vlan 162
undo jumboframe enable
description sipXecs
traffic-priority inbound ip-group
At a minimum you've cleaned up your QOS :-)
But as Tony suggests, I'd still do a 'yum update'.
On Wed, Jun 22, 2011 at 3:04 PM, Alex Brown alex...@bellsouth.net wrote:
OK, I've reconfigured the port as follows to mark all ingress traffic
with a dscp value of 46 (ef):
interface
I tried but could not get it to happen yesterday. The only change I've
made was putting the sip domain back into automatic mode (hence I did
not have the A record) as a result of taking in updates. I will have
to look back into this. It's on a vlan and separate subdomain. It IS
connected via an
As I've said, the FS update eliminated that same stutter we all had at
some point. You probably haven't moved to the most recent patch, be
aware it does make polycom template changes, so when the system comes
up it will reboot the phones.
On Fri, Jun 24, 2011 at 7:55 AM, Michael Picher
I had created a holding page for karoo docs here:
http://wiki.sipfoundry.org/display/sipXecs/karoo+SBC
On Wed, Jun 15, 2011 at 7:33 PM, Joegen Baclor jbac...@ezuce.com wrote:
On 06/16/2011 05:12 AM, Roman Gelfand wrote:
I get a successful ITSP registration. I am supplying the same
Well said Tony,
An offered stream MAY be rejected in the answer, for any reason. If
a stream is rejected, the offerer and answerer MUST NOT generate
media (or RTCP packets) for that stream. To reject an offered
stream, the port number in the corresponding stream in the
Update #8 : Wed, 22 June 2011
==
- ** No security updates in this update **
- ISO has *not* been rebuilt as decided in release policy. Yum update
after installation is recommended for getting these updates
- Thank you George for your
Let's make it more text crunchyone for the weekend (although I have
other plans :)
(Please tell me I am wrong, but I can't find proof that I am)
Officially m=image is not a defined SDP media type:
rfc 4566:
media is the media type. Currently defined media are audio,
video,
resending
On Wed, Jun 22, 2011 at 1:14 PM, Douglas Hubler dhub...@ezuce.com wrote:
Update #8 : Wed, 22 June 2011
==
- ** No security updates in this update **
- ISO has *not* been rebuilt as decided in release policy. Yum update
they claim to be running sipx on multiple interfaces. we all know this is
not a good idea.
On Jun 24, 2011 4:02 PM, Yuri Kurkarewicz sipx...@kapten.com.br wrote:
*Has anyone seen if this post is true?*
**
*http://qxip.net/mediawiki/index.php/SIPX_IPv6_HACK*
*Good morning!
*
*How do I enable SRTP and TLS on the extensions? I tried to look in
the wiki and
not found!
Thank you!
*
___
sipx-users mailing list
sipx-users@list.sipfoundry.org
List Archive: http://list.sipfoundry.org/archive/sipx-users/
Do not understand, what these multiple interfaces that you say?
thanks.
2011/6/24 Tony Graziano tgrazi...@myitdepartment.net
they claim to be running sipx on multiple interfaces. we all know this is
not a good idea.
On Jun 24, 2011 4:02 PM, Yuri Kurkarewicz sipx...@kapten.com.br wrote:
these are two different things
srtp was just added yesterday. you might consider waiting g for someone to
do test and create the wiki pages.
tls is in the wiki.
On Jun 24, 2011 4:09 PM, Yuri Kurkarewicz sipx...@kapten.com.br wrote:
___
sipx-users
On Fri, Jun 24, 2011 at 4:22 PM, Tony Graziano
tgrazi...@myitdepartment.net wrote:
srtp was just added yesterday. you might consider waiting g for someone to
do test and create the wiki pages.
it was confirmed that media relay doesn't properly proxy srtp. That
means SRTP won't work with NAT
*So what would be the recommendation to encrypt the data??
*
* Would be to use an IPSEC VPN connecting the remote extensions?*
* I'm testing Bria Android version 2.3.4!
*
* thanks!*
2011/6/24 Douglas Hubler dhub...@ezuce.com
On Fri, Jun 24, 2011 at 4:22 PM, Tony Graziano
What can we use on a SIP trunk to increase security? Of course not using TLS
and SRTP?
2011/6/24 Douglas Hubler dhub...@ezuce.com
On Fri, Jun 24, 2011 at 4:22 PM, Tony Graziano
tgrazi...@myitdepartment.net wrote:
srtp was just added yesterday. you might consider waiting g for someone
to
Ytray usingway Igpay Atinlay, itway eemssay otay elphay
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Yuri Kurkarewicz
Sent: Friday, June 24, 2011 4:49 PM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] TLS
On 11-06-24 6:41 AM, Douglas Hubler wrote:
Did you upgrade from 4.0.4?
Yes.
It's possible you hit this
http://track.sipfoundry.org/browse/XX-9461
there is a workaround posted
Only user howard is affected.
There are lots of brand new messages in his mailbox - XX-9461 talks
about messages
atm media does it it work at least with remote users for srtp. I suggest
waiting for a howto and examples of what does/does not work
On Jun 24, 2011 4:43 PM, Douglas Hubler dhub...@ezuce.com wrote:
On Fri, Jun 24, 2011 at 4:22 PM, Tony Graziano
tgrazi...@myitdepartment.net wrote:
srtp was just
See http://tools.ietf.org/rfc/rfc3362.txt
On 06/24/2011 10:42 PM, pscheep...@epo.org wrote:
Let's make it more text crunchyone for the weekend (although I
have other plans :)
(Please tell me I am wrong, but I can't find proof that I am)
Officially m=image is not a defined SDP media
46 matches
Mail list logo
