I took the plunge tonight and upgraded from 4.0.4 to 4.2. So far, everything
that was working before seems to be working fine. Great!
I thought I'd try out the IMAP synchronization stuff. However, if I go to a
user group and try to add an IMAP server host into the Unified Messaging page,
I
Hi,
We're running on the same problem. And we've been in that dilemma in 2 weeks
now.
Any chance you make any progress please share them.
Best regards,
Rhon
On Tue, Apr 20, 2010 at 8:44 AM, Nathan Nieblas wrote:
> Ran into some firmware compatibility issues after upgrading from 4.0 to
> 4.2
On Thu, Apr 15, 2010 at 3:42 PM, Scott Lawrence wrote:
> Build 4.2.0-018575 is posted and stable!
>
> This release resolves over 900 issues - some of the highlights are listed
> below.
>
Just finished what seems to be a flawless upgrade from 4.0.4.
Fantastic work folks!
__
Good enough for me. It's just odd that they have been working (with the
exception of outbound caller-id) until today. Which coincides with the 4.2
upgrade this morning.
I have a callwithus account that I've routed outbound calls thru for the time
being.
I'll tweak the keepalive settings and,
Ditch the Cisco phones and buy Polycoms.
(You did not specify :-P )
Nathan Nieblas wrote:
Just FYI… If anyone has anything to
share Cisco
related I’m all ears J
___
sipx-users mailing list sipx-users@list.sipfoundry.org
List A
I would simply ask them why they are sending the disconnect. It could be
something simple like they want to see another method for keepalive.
If they are giving you problems, there are several good templates for ITSP's
in 4.x.
Tony Graziano, Manager
Telephone: 434.984.
Ran into some firmware compatibility issues after upgrading from 4.0 to
4.2
Firmware 8.3.5 on 79xx would only ring internal extensions but not dial
over the trunk, it would just hang until there was a reorder.
Firmware 8.5.4 on 79xx works with no known issues so far, however we run
into the Po
Session Timer Interval is set to the default (1800 seconds)...
From: Tony Graziano [tgrazi...@myitdepartment.net]
Sent: Monday, April 19, 2010 7:30 PM
To: Nathaniel Watkins
Cc: M. Ranganathan; sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] 1:28 - disconnec
telnet to the IMAP server in question on its IMAP port
ex:
telnet mail.mydomain.com 143
It will return some info, example:
* OK [CAPABILITY IMAP4rev1 UIDPLUS CHILDREN NAMESPACE THREAD=ORDEREDSUBJECT
THRE
AD=REFERENCES SORT QUOTA IDLE ACL ACL2=UNION STARTTLS] Courier-IMAP ready.
Copyr
ight 1998-
FWIW the freeswitch project (which provides the media services for sipxecs) now
sells an officially licensed g.729 codec. You only need the license if you need
to transcode or terminate channel formats (ie: you want voicemail/AA in g.729
or the conference bridge).
-Eric
On Apr 19, 2010, at 6:1
BYE
10 /tmp/trace.DZS25547/_.sipxbridge.trace.xml:1506
216.82.225.24:5060
false
-
-
attached is the phone call log - session timeout...any idea why?
Also - I've had issues with outgoing caller-id from this ITSP - I've called and
they've stated the caller-id is just garbage...that may be another discussion
however.
Thanks
From: Tony Gra
I am noticing that in 4.2, users cannot see any calls in the CDR with
the exception of "internal calls". If they are a member of the admin
group, they can see them.
Is it just me?
--
==
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431
Email: tgrazi...@myitdep
I think media server will now do G722, but that's not what he's looking for.
On Mon, Apr 19, 2010 at 7:11 PM, Picher, Michael
wrote:
> Not to my knowledge.
>
>
>
> You can set your phones and gateways to prefer g.729 over other Codecs and
> then fall-back to g.711u as a secondary choice. But to
Yes. Perhaps thy needeth to get a Winmachine and IE to see whats up?
On Mon, Apr 19, 2010 at 6:17 PM, Michael Scheidell wrote:
> Ill check logs tomorrow. Yes have a branch. Sipxbridge assigned to branch.
> No other template causes internal error page to show up. Does it work for
> you. ?
>
>
Not to my knowledge.
You can set your phones and gateways to prefer g.729 over other Codecs
and then fall-back to g.711u as a secondary choice. But to my knowledge
all conversations with the media services on the PBX are still limited
to g.711.
Mike
From: sipx-users-boun...@list.sipfou
Ill check logs tomorrow. Yes have a branch. Sipxbridge assigned to branch. No
other template causes internal error page to show up. Does it work for you. ?
-Original Message-
From: Tony Graziano
Sent: Monday, April 19, 2010 5:59 PM
To: scheid...@secnap.net ; thod...@verizon.net
Cc: s
Viewing the sipxconfig log would be handy. Did you try creating a branch and
making sure sipxbridge was assigned to the branch?
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431
Email: tgrazi...@myitdepartment.net
LAN/Telephony/Security and Control Sys
I think a wiki page would be more fruitful if it gives you the commands to
determine if your server/provider supports IDLE and SEARCHES. The server
might be compatible but the function may not be enabled.
Teach me to fish, don't feed me.
Tony Graziano, Manager
Telephon
Has to support SEARCHES with headers and IDLE.
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 Custom
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Thanks again Josh, and all others who have responsed. Will
update you on how
On Mon, Apr 19, 2010 at 4:55 PM, Ken Fulmer
wrote:
> Ok, so I set the priority on the primary server to 1 and the secondary
> server to 2. The phones all register to the 1st server as expected but the
> inbound audio is getting sent to the 2nd server.
>
> 1st server = 10.10.3.10.
>
> 2nd server =
Same as before...
Josh Patten
Assistant Network Administrator
Brazos County IT Dept.
(979) 361-4676
On 4/19/2010 4:36 PM, Francis Tinio wrote:
> Hi.
>
> Just wondering, has anyone tried or is using the ACD feature in 4.2? It was
> buggy at 4.0.4. Just wondering if it's useable now with the ne
Hi.
Just wondering, has anyone tried or is using the ACD feature in 4.2? It was
buggy at 4.0.4. Just wondering if it's useable now with the new version out.
Thanks
___
sipx-users mailing list sipx-users@list.sipfoundry.org
List Archive: http://list.s
Doesn't appear that has made it to the wiki yet...Do this:
For whatever groups you want to set IMAP sync on:
* Go to that group
* Go to Unified Messaging
* Set the IMAP server host, port, and TLS setting (the last one is
optional)
Now, set up each user:
* Go to the user you wi
On Mon, 2010-04-19 at 15:21 -0400, Abdul Mayat wrote:
> I noticed the new branch setting in 4.2, and wanted to
> clarify its usage with the forum...Currently we are able to
> apply the branch setting to users, phones, gateways. If
> this was extended to dialplans too, would it be a way of
> suppo
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Sorry guys, i may be a little slow on the uptake here, but
can you give me so
So can Cyrus version 2.3.x
Josh Patten
Assistant Network Administrator
Brazos County IT Dept.
(979) 361-4676
On 4/19/2010 4:07 PM, Pizza Napoletana wrote:
> We have been using it with dovecot. No issues.
> I don't know if there is a wiki page that lists IMAP servers that are known
> to work wel
We have been using it with dovecot. No issues.
I don't know if there is a wiki page that lists IMAP servers that are known to
work well / not well. If there is, dovecot can be added to the good list.
On Apr 19, 2010, at 1:11 PM, Josh Patten wrote:
> What IMAP server are you using? Your IMAP serv
As far as I know, up through at least Lotus Notes 8.5.1, the IMAP IDLE
command is not available. You can open a telnet session to your IMAP
server, and test using some of the syntax described:
http://tools.ietf.org/html/rfc2177
This would be a way of confirming one way or another.
Kurt Siegfrie
Ok, so I set the priority on the primary server to 1 and the secondary
server to 2. The phones all register to the 1st server as expected but the
inbound audio is getting sent to the 2nd server.
1st server = 10.10.3.10.
2nd server = 10.10.3.11
Phones are registered to 10.10.3.10. Inbound RTP is
On 4/19/10 4:21 PM, Todd Hodgen wrote:
There was no template for voip.ms when I started using it, so I didn't
not use the drop-down list.
I would love to compare the template with what I used.
and constantly, different configuration tests, voip.ms static
registration (change voip.ms sub acc
No. Everything is stored on the sipX server. The IMAP sync is purely for
syncronization, not as a storage engine. Also, don't know about Lotus.
It really depends on how well they support the IMAP protocol.
Josh Patten
Assistant Network Administrator
Brazos County IT Dept.
(979) 361-4676
On 4/1
it could be my upgrade. My copy of firefox. but its curious that its
just voip.ms.
if someone could confirm this, I will open a JIRA
just to to gateway, new sip trunk, type some name in, select
'sibxpridge', type some ip in, then under template, drop down to voip.ms.
if you get an internal
IF the template is errorring out, I'd recommend documenting it and creating
a JIRA as it doesn't seem to be the desired results.
From: Michael Scheidell [mailto:scheid...@secnap.net]
Sent: Monday, April 19, 2010 12:54 PM
To: Todd Hodgen
Cc: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-use
There was no template for voip.ms when I started using it, so I didn't not
use the drop-down list.
From: Michael Scheidell [mailto:scheid...@secnap.net]
Sent: Monday, April 19, 2010 12:54 PM
To: Todd Hodgen
Cc: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] and I am stupid: Re: more
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Thanks Josh, I was hoping to test this with Lotus Notes. Im
sure I saw it lis
On 4/19/10 3:43 PM, Todd Hodgen wrote:
And you are making the change both in sipXecs and on their configuration
server?
best I can tell, i appears that STATIC registration does NOT work (You
can't transfer a call), you must use (at least with voip.ms), user/password
and I still can't cre
What IMAP server are you using? Your IMAP server must support IDLE, and
searches within headers. Two IMAP servers that are known NOT to work are
GroupWise and legacy Cyrus versions (2.2.x and older). I know because I
tested both.
Josh Patten
Assistant Network Administrator
Brazos County IT Dept
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Has anybody managed to get the IMAP integration working for
email. Appreciate if anybody can point me in the direction
of a
On 4/19/10 3:43 PM, Todd Hodgen wrote:
And you are making the change both in sipXecs and on their configuration
server?
yes
can you create a new sip trunk with voip.ms from the drop down list? and
NOT get an internal error?
--
Michael Scheidell, CTO
Phone: 561-999-5000, x 1259
> *| *SE
And you are making the change both in sipXecs and on their configuration
server?
-Original Message-
From: Michael Scheidell [mailto:scheid...@secnap.net]
Sent: Monday, April 19, 2010 12:40 PM
To: Todd Hodgen; 'Michael Scheidell'; sipx-users@list.sipfoundry.org
Subject: RE: [sipx-users] an
Then its their servers. Ny works, Dallas doesn't. Still gives me an internal
error if I select voip.ms when creating a new gateway.
-Original Message-
From: Todd Hodgen
Sent: Monday, April 19, 2010 3:22 PM
To: 'Michael Scheidell' ; sipx-users@list.sipfoundry.org
Subject: RE: [sipx-u
You do not need separate sub-accounts for each DID with VOIP.ms. I have 4
on one account, pointing to their Dallas server, and I have one DID for a
foreign exchange in a sub-account that points to their Los Angeles Server.
Works like a champ.
One terminates on the AA, one is pointed to an exte
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Hi All,
I noticed the new branch setting in 4.2, and wanted to
clarify its usage with the forum...Currently we are able to
a
Yes. Trace would show you if the disconnect (BYE) was sent from which
side. More importantly it might show the call not getting a proper
acknoweldgement and may sipstation does not know the call was actually
"connected".
Alternately, if this is a recent upgrade, you might consider deleting
and rec
On Mon, Apr 19, 2010 at 2:44 PM, Ken Fulmer
wrote:
> 1. I was under the impression that RTP will flow through the server where
> the phones are registered - in both the inbound and outbound directions. We
> are not seeing this behavior for inbound calls.
SipXbridge will load balance between the s
On Mon, Apr 19, 2010 at 2:35 PM, Nathaniel Watkins
wrote:
> I just made a call thru sipstation – should be very similar to bandwidth.com
> – and it disconnects at 1:28…
>
>
>
> I’m assuming there is a setting that was changed on the upgrade…
>
>
>
> Survey says:__
Survey says
On 4/19/10 2:38 PM, Michael Scheidell wrote:
VERY interesting.
if I set the sbcbridge to destination 100 (AA/IVR), and it answers (aa
picks up), and I dial the extention, and it picks up, and I transfer,
then it works.
but if DID it to an extention, it can't answer. I am thinking it
might h
1. I was under the impression that RTP will flow through the server where
the phones are registered - in both the inbound and outbound directions. We
are not seeing this behavior for inbound calls.
2. We have three servers - one primary, one secondary, and a third server
that only has the voice ma
VERY interesting.
if I set the sbcbridge to destination 100 (AA/IVR), and it answers (aa
picks up), and I dial the extention, and it picks up, and I transfer,
then it works.
but if DID it to an extention, it can't answer. I am thinking it might
have to do with MOH, and codex's.
still looki
I just made a call thru sipstation - should be very similar to bandwidth.com -
and it disconnects at 1:28...
I'm assuming there is a setting that was changed on the upgrade...
Survey says:__
This message and any files transmitted with it are
On Mon, Apr 19, 2010 at 12:22 PM, Ken Fulmer
wrote:
> We have 4.2 installed with three servers in a cluster. The first two have
> sipXbridge running and the third only has voice mail.
>
>
>
> 1. Outbound calls flow through the server where the phone is
> registered. This isn’t the case for i
We have 4.2 installed with three servers in a cluster. The first two have
sipXbridge running and the third only has voice mail.
1. Outbound calls flow through the server where the phone is
registered. This isn't the case for inbound calls. Inbound calls load
balance the RTP stream betwee
Ok, than it is working as expected I guess. Too bad support for the MyBuddy
function was dropped, since we were really charmed by that feature. Any plans
to re-introduce such a feature in a different form?
Cheers, Eelco
-
Eelco Brölman | Sr. Product Specialist | Telecats bv
-Original M
I'm using user/password and sipxbridge. I've just tested here is the flow:
1) Dialed 866 number (from cell) - aa picks up
2) Press 3 to transfer to my internal extension (plays hold music during
the transfer)
3) Pickup my internal extension
4) transfer to another phone/ex
attempting to recreate my itsp account for voip.ms.
I get an internal error every time I select 'VOIP.MS' as the ITSP account.
evertyhing else seems to work.
(as not cause internal error)
steps: new gateway: SIP Trunk
name it.
select sipxbridge
(wiich enabled a new box: Use Provider Template)
s
I have a voip.ms account for inbound 800 calls -- I updated to 4.2
last night -- and after reading your post, noticed that I was no
longer registered to voip.ms -- The Register on initialization
checkbox was unchecked...re-checking this box solved this issue on the
ITSP account settings...
I have a voip.ms account for inbound 800 calls - I updated to 4.2 last night -
and after reading your post, noticed that I was no longer registered to voip.ms
- The Register on initialization checkbox was unchecked...re-checking this box
solved this issue on the ITSP account settings...
I use t
On 4/19/10 11:19 AM, Tony Graziano wrote:
I've found you cannot do any entries on the gateway screen out of
order (have to go top to bottom) and you have to use firefox or IE
with 4.2.0. Grumble.
Im screded now!
deleted old gateway, tried to put a new one in (in order!)
got 'internal error'
h
On 4/19/10 11:19 AM, Tony Graziano wrote:
I've found you cannot do any entries on the gateway screen out of
order (have to go top to bottom) and you have to use firefox or IE
with 4.2.0. Grumble.
'out of order'?
and I am using firefox.
so, I should NOT need to wipe clean, start over?
--
I've found you cannot do any entries on the gateway screen out of
order (have to go top to bottom) and you have to use firefox or IE
with 4.2.0. Grumble.
On Mon, Apr 19, 2010 at 11:17 AM, Michael Scheidell
wrote:
> I tried to create a sipx account (just in case voip.ms was now supported
> with a
On 4/19/10 11:05 AM, Michael Scheidell wrote:
I will try changing from static to user/auth and see.
didn't help. even though voip.ms see this on port 5080:
Account
Server
State
IP/Port
Next Registration
ljlkjljlkjlkjljlk
sip.us2.voip.ms (Dallas)
I tried to create a sipx account (just in case voip.ms was now supported
with a config)
Funny thing. new gateway does not ask what itsp, and when I look at
cnfig after I add it, there is no option at left on new itsp for itsp
account.
I suppose a backup/ wipe clean, install from ISO and res
On 4/19/10 11:06 AM, M. Ranganathan wrote:
On Mon, Apr 19, 2010 at 11:02 AM, Michael Scheidell
wrote:
On 4/19/10 10:59 AM, Scott Lawrence wrote:
The key thing is that voip.ms needs to send the INVITE to port 5080 -
you have to get them to reconfigure.
anyone using voip.ms know how to
On Mon, Apr 19, 2010 at 11:02 AM, Michael Scheidell
wrote:
>
> On 4/19/10 10:59 AM, Scott Lawrence wrote:
>
> The key thing is that voip.ms needs to send the INVITE to port 5080 -
>
> you have to get them to reconfigure.
>
>
>
> anyone using voip.ms know how to get them to send invites to port 508
I will try changing from static to user/auth and see.
On 4/19/10 11:02 AM, Tony Graziano wrote:
Or, in some cases make sure you send to port "0" or auto for port "5060" for
them. When you register, attempt to register on port "5080", which I find
works with sime ITSP's that require registratio
guess I mean 4.2.0?
not 4.1.2?
latested stable?
On 4/19/10 10:58 AM, Scott Lawrence wrote:
On Mon, 2010-04-19 at 10:18 -0400, Michael Scheidell wrote:
Trying to regress test 4.12. ok, this didn't work on 4.04 either, I
just decided to wait.
Michael... please try to b
Or, in some cases make sure you send to port "0" or auto for port "5060" for
them. When you register, attempt to register on port "5080", which I find
works with sime ITSP's that require registration (the send calls to you on
the port you register on).
Tony Graziano, Ma
On 4/19/10 10:58 AM, Scott Lawrence wrote:
So did you mean that you are testing 4.1.2 ? That's very very old...
yes, newest, had CENTOS5 4.0.4 then edited reposits as is documented,
did a yum update.
Got pretty new GUI screen and everything.
4.0.4 would not recover from the failed x
On 4/19/10 10:59 AM, Scott Lawrence wrote:
The key thing is that voip.ms needs to send the INVITE to port 5080 -
you have to get them to reconfigure.
anyone using voip.ms know how to get them to send invites to port 5080?
guess this is just another case of an ITSP NOT following the specs.
Ensure the service is enabled and the users have IM enabled. If the users
can chat with each other, then it is working. The "MyBuddy" function was
suddenly "proprietized" by Avaya and abrupty removed from the codebase,
sadly.
Tony Graziano, Manager
Telephone: 434.984.84
Has support for g.729 improved with the 4.2 version?
Thanks,
Ken Fulmer
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On Mon, 2010-04-19 at 10:42 -0400, Michael Scheidell wrote:
> On 4/19/10 10:21 AM, M. Ranganathan wrote:
> >
> >
> > If you are seeing a REFER sent out to the ITSP then it likely is a
> > misconfiguration on your sipxecs.
>
> sipxbridge shows
> > Are you sure that the signaling is
> > being SE
On Mon, 2010-04-19 at 10:18 -0400, Michael Scheidell wrote:
> >
> > > Trying to regress test 4.12. ok, this didn't work on 4.04 either, I
> > > just decided to wait.
> > >
> > Michael... please try to be more careful in your descriptions. There is
> > no such thing as sipXecs 4.12 or 4.04 (
Hi,
I have a problem with the MyBuddy IM account.
I just installed a fresh instance of sipXecs 4.2 (ISO install). I created a
group, enabled all the options in the "Instant messaging" section for that
group. Next I created 2 users (and phones for that matter), and added them to
the newly creat
On 4/19/10 10:21 AM, M. Ranganathan wrote:
If you are seeing a REFER sent out to the ITSP then it likely is a
misconfiguration on your sipxecs.
sipxbridge shows
Are you sure that the signaling is
being SENT by the ITSP to port 5080 and what does the Contact header
of the INVITE specify?
Use Pandion or Spark for your IM client. Pidgin is like a cheap swiss
army knife: It may have all the tools you'll ever need, but none of them
are any good.
Josh Patten
Assistant Network Administrator
Brazos County IT Dept.
(979) 361-4676
On 4/19/2010 9:31 AM, Matt White wrote:
Ok, just pla
Ok, just playing around with IM.
I can signin with pidgin, but I can't add a buddy. The buddy shows as "Not
Authorized".
If I try and get status of u...@domain.com it shows me the full name and name
of the IM user, so it does appear to be resolving the buddy name properly with
openfire.
Problem solved!
Looks like updating to 3.2.3 bootrom and equivalent sip software 2.1.3, and
upgrading sipx to 4.2 did the trick.
Now so far I was able to contact the other location without issues.
I haven't tested all functionality but apparently this was the biggest issue
with multiple locati
On Mon, Apr 19, 2010 at 10:18 AM, Michael Scheidell
wrote:
> On 4/19/10 10:07 AM, Scott Lawrence wrote:
>
> On Mon, 2010-04-19 at 09:46 -0400, Michael Scheidell wrote:
>
>
>
> Trying to regress test 4.12. ok, this didn't work on 4.04 either, I
> just decided to wait.
>
>
> Michael... please try t
On 4/19/10 10:07 AM, Scott Lawrence wrote:
On Mon, 2010-04-19 at 09:46 -0400, Michael Scheidell wrote:
Trying to regress test 4.12. ok, this didn't work on 4.04 either, I
just decided to wait.
Michael... please try to be more careful in your descriptions. There is
no such thing as
On Mon, 2010-04-19 at 09:46 -0400, Michael Scheidell wrote:
> Trying to regress test 4.12. ok, this didn't work on 4.04 either, I
> just decided to wait.
Michael... please try to be more careful in your descriptions. There is
no such thing as sipXecs 4.12 or 4.04 (get the dots right).
> Call f
On 4/19/10 9:46 AM, Michael Scheidell wrote:
Trying to regress test 4.12. ok, this didn't work on 4.04 either, I
just decided to wait.
Call from outside to a voip.ms hosted DID. Answer inside (behind
firewall, natted, all else seems to work).
receiver is polycom 650, 3.13c Firmware, Sipx
Trying to regress test 4.12. ok, this didn't work on 4.04 either, I
just decided to wait.
Call from outside to a voip.ms hosted DID. Answer inside (behind
firewall, natted, all else seems to work).
receiver is polycom 650, 3.13c Firmware, Sipx 4.12.
Blind Xfer to internal phone extension, (
On Mon, 2010-04-19 at 08:45 -0400, Tony Graziano wrote:
> Well, there are two things that should make their way to the codebase IMO.
>
> One would be registration attempts, the other invites. I think people
> probably see more invites as a way of probing than they do registration
> attempts.
>
>
Well, there are two things that should make their way to the codebase IMO.
One would be registration attempts, the other invites. I think people
probably see more invites as a way of probing than they do registration
attempts.
Either way, getting this information out for the ip addresses becomes
On Mon, Apr 19, 2010 at 8:34 AM, Tony Graziano
wrote:
> I've been thinking and am curious if there is a way to log these attempts
> only to a separate file for use with a firewall. Or send those to a syslog
> which could periodically parse them for idebtifying attacks.
>
> Since these are on port
I've been thinking and am curious if there is a way to log these attempts
only to a separate file for use with a firewall. Or send those to a syslog
which could periodically parse them for idebtifying attacks.
Since these are on port 5060, that would make me assume they will be in the
proxy log,
On Mon, Apr 19, 2010 at 6:38 AM, Michael Scheidell wrote:
> you would be pleased to see discussions on some of the security lists
> about just totally isolating amazon's cloud due to constant attacks of
> ALL types.
>
> google for 'emerging threats + amazon' and you should see some. I just
> forw
On Mon, 2010-04-19 at 10:35 +0800, Rhon wrote:
> Hi Scott,
>
> Attached is the siptrace for your reference. Tried to interpret it,
> but just don't know what it means. :(
You did not reset your logging to INFO level - none of the messages from
the phones are in the trace (you must restart the co
seems log rotate, or something else wasn't upgraded when going from 4.04
to 4.2.
see this in daily and weekly: (wonder why a .gz file isnt in gzip format?
/etc/cron.weekly/makewhatis.cron:
zcat: /usr/share/man/man8/stunnel.8.gz: not in gzip format
'file' command shows its just a troff file.
you would be pleased to see discussions on some of the security lists
about just totally isolating amazon's cloud due to constant attacks of
ALL types.
google for 'emerging threats + amazon' and you should see some. I just
forwarded your post to the list as well.
yes, if you are using a firew
Seems to me they have a quick and effective way to deal with this on
Asterisk.
http://www.teamforrest.com/blog/171/asterisk-no-matching-peer-found-bloc
k/
Feature request?
Mike
> -Original Message-
> From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-
> boun...@list.sipf
http://www.voiptechchat.com/voip/457/amazon-ec2-sip-brute-force-attacks-on-rise/
--
==
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.
I figured this out. :)
Thanks
On Mon, Apr 19, 2010 at 1:05 PM, Rhon wrote:
> Hi Tony,
>
> Thank you for your reply.
>
> This is what I see in the SipXeconfig:Conference > New Conference
>
> Name:
> Extension:
> Description:
> Conference Owner
> Participant PIN:
> Maximum legs:
> Music On Hold s
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