Hi George, here it is.
Seems OK but I don't know whether it needs/puts the (new or old) keys from
the primary in the tar as well.
In the /var/sipxdata/certdb are still the old primary keys that expired
(gssipx02.internal.epo.org.crt).
In the /etc/sipxpbx/ssl directory is the new ssl.crt that
Hi George,
I'm still struggling. The sipxecs-setup on the secondary still can't load
the initialisation file.
This is of course (when you look at it for the 6th time) just an https
connection.
I pasted
https://10.12.48.43:8443/sipxconfig/initial-config/th.internal.epo.org
.tar.gz
in a browser
Hi george, I deleted everything in /var/sipxdata/tmp
Then the initialconfig is not run anymore, so that is not good.
Put everything back except tar.gz's, then it still does not work.
These were the tar's:
sipx-configuration.tar.gz
sipx-snapshot-gmsipx02.internal.epo.org.tar.gz
I got back from holiday and discovered my primary server was not OK.
It showed the following:
/usr/lib/ruby/1.8/net/http.rb:586:in `connect': certificate verify failed
(OpenSSL::SSL::SSLError)
(the rest is gone because of user error :( )
I tried a sipxproc -R, this did not work.
Then I
I use the standard certs, I thought they would not expire tht soon.
I am running 4.4.0-287.gb0a66 btw.
Can you give me a quick hint how to check?
Paul
Douglas Hubler dhub...@ezuce.com wrote on 16-08-2012 18:00:37:
On Thu, Aug 16, 2012 at 11:55 AM, pscheep...@epo.org wrote:
I got back
Thanks,
Expired indeed:
Validity
Not Before: Jun 29 08:31:15 2009 GMT
Not After : Jun 28 08:31:15 2012 GMT
So I need to generate new keys, should I just follow this:
http://wiki.sipfoundry.org/display/sipXecs/SSL+Keys+and+Keystores
Paul
Douglas Hubler
Done that, that solves everything for the Primary server.
I think I can do the same on my 2 secondaries or should I use this:
libexec/sipXecs/initial-config secondaryServerHostName ,
mentioned in
http://wiki.sipfoundry.org/display/sipXecs/SSL+Keys+and+Keystores
Paul
Michael Picher
I am a bit stuck now.
I generated keys with /usr/bin/ssl-cert/gen-ssl-keys.sh for secondary on
primary.
Copied them over and installed them with /usr/bin/ssl-cert/install-cert.sh
Then tried to push profiles, doesn't work.
and tried restart services, doesn't work.
How should I install new keys on
Any help is greatly appreciated, I have now a working primary and 2
secondaries that don't talk to the primary.
How and where do I generate keys for the secondaries (on primary?)
How and where do I install these keys (on secondary?)
Paul
pscheep...@epo.org wrote on 16-08-2012 20:17:46:
I am a
Hi George,
Sorry for the late response but I first went home to have a look at my 16
year old son who has over 40deg celsius fever probably because of food
poisoning from a French road restaurant.
Thanks for the tip, but it doesnt work completely, I cant send profiles,
the jobs fail:
File
European coast is fine then as well :o)
Paul
BTW: I am very (as in VERY) close to Rotterdam airport, but our main
airport is Schiphol (Amsterdam), bummer :-(
Michael Picher mpic...@ezuce.com wrote on 11-07-2012 01:09:13:
Not really worried about that at the moment... Will be work from
Hi Chris,
Since you are a newbie I just want to remind you that (0.)4.5 is a
development release.
All odd releases are dev, all even ones are prod, so if you want to build
a stable system install 4.4 or wait for 4.6.
http://wiki.sipfoundry.org/display/sipXecs/Version+Numbers
Paul
Kumaran
Or delete and recreate all speeddials.. See this old issue:
http://forum.sipfoundry.org/index.php?t=msgth=16735start=0S=92cec18f83d528faecdbc0af9901c458
Paul
Michael Picher mpic...@ezuce.com wrote on 08-06-2012 11:55:45:
Certain phones can only have so many entries inside the phone...
Sven Evensen sven.even...@onrelay.com wrote on 07-06-2012 12:24:11:
I saw that and I believe we made a patch once to add SDP to the
reInvite. Will have to check that.
But the BYE from sipx says sipXbridge;cause=205;text=\Protocol
Error - 200 OK with no contact\. I am trying to understand
Yes, good point (same point was made by Geoff). It means using a special
voice subdomain, but it does make life easier.
I prefer to use the main company domain because john@company.com looks
better than john@voice.company.com
but normally you don't use the domain name anyhow because most
Welcome to the world of Counterpath.
Bria is a product that has (almost) everything you could wish for,
but the more advanced features don't work reliable enough
and aren't documented properly (a lot of trail and error).
Paul
Elwin Formsma e.form...@telecats.nl wrote on 29-05-2012 11:53:10:
Douglas Hubler dhub...@ezuce.com wrote on 25-05-2012 01:52:58:
On Thu, May 24, 2012 at 12:21 PM, Gerald Drouillard
gerryl...@drouillard.ca wrote:
List Archive: http://list.sipfoundry.org/archive/sipx-users/
I think you can get as crazy as you want with this. I believe the
ultimate in
Douglas Hubler dhub...@ezuce.com wrote on 25-05-2012 11:01:14:
On Fri, May 25, 2012 at 2:59 AM, pscheep...@epo.org wrote:
then we should make sure webmin works. Did you typically install
webmin on sipxecs systems before or just thinking ahead?
I think it is important that there is at
Douglas Hubler dhub...@ezuce.com wrote on 25-05-2012 14:44:01:
On Fri, May 25, 2012 at 7:23 AM, pscheep...@epo.org wrote:
2) Use UC DNS for the servers and the company DNS for the clients,
this
means putting the
necessary records in the company DNS. All records are available on the
UC
I think it is confusing to have a DNS Settings and an Unmanaged
Service panel.
The only difference between the 2 is the unmanaged tick.
I would place the tick box at the bottom of the first screen with a
clearer description, something like:
It is advised to let the UC system manage and run it's
Bria can work with RLS based on Workgroups.
In Bria, under the SIP account in the Presence tab enter the ~~ rls string
(sip:~~rl~C~username@domain) under Workgroup address.
In Bria's Main Menu select View, Workgroup.
Paul
Tony Graziano tgrazi...@myitdepartment.net wrote on 21-05-2012 13:35:47:
It might be nice to make this optional/configurable.
So by default SipXconfig will handle the iptables and updates it when
profiles are send etc.
The true freak can then in the GUI under System, Control (a new item)
tick the Let me control IP-tables box (or edit a file).
I think there would be
Hi Simon,
Please try to test with X-Lite.
This one is known to work OK with SRV records.
I never got Ekiga working properly.
Anyhow your DNS does not seem to be OK.
Is your SIP domain voiptest.netappsid.local or netappsid.local?
Please show the output of dig -t SRV
Thought about that as well, only A, B and C should transfer to 2 to make
it understandablestill a cludge maybe.
Paul
Tony Graziano tgrazi...@myitdepartment.net wrote on 23-03-2012 16:19:03:
Sounds like an enhancement request (to polycom). Alternately your
app might be configurable to
I agree as well.
ACME is a good product, probably doing more regarding SIP standards then
any other (SBC) vendor.
It is at least MUCH better then a product build in Norway, bought by John
C, for Video.
I had to implement their solution instead of an ACME because management
did not know Acme,
I think for a production environment it's much easier to run 4.4 from ISO
(thus Centos 5) and upgrade by
-backup
-ISO install of 4.6
-restore
BTW: NOT 4.5, all odd releases are development releases
Douglas Hubler dhub...@ezuce.com wrote on 08-02-2012 19:38:03:
On Wed, Feb 8, 2012 at 12:42
I thought it was more a question what would be the easiest (as in least
effort) way to move from CUCM to SipX 4.4 and then to 4.6.
I thought Jesse wanted to run 4.4 on CentOS6 just to have an easier
upgrade path.
But if he runs 4.4 on CentOS6 he's all alone, if he runs 4.4 on CentOS5
he's part
With Bria you can control a SIP phone, so in theory you can couple a
Polycom 335 with a Bria softphone.
It's called Deskphone mode in Bria, your mileage may vary.
Michael Picher mpic...@ezuce.com wrote on 25-01-2012 11:45:01:
True, but the problem I see with the CX300 is that it is about the
CX200 and CX300 are USB audio devices, so they can work with any
softphone.
CX200 has echo cancellation build in, but audio quality is relatively poor
because the echo cancellation is too strong.
I can't recommend the CX200 because of this.
CX300 has (almost) no echo cancellation, but audio
CX300 has all this.
Individual volume control for handset, speaker and headset.
Mute button, hook, hold.
The only weird thing is that all buttons only engage when you let go.
So you can't hang up by pacing the handset on hook because it is not
detected.
You can pick up the handset to answer
I thought with tls the sip messages would be encrypted and therefore an
alg would not be able to mess it up.
Paul
Tony Graziano tgrazi...@myitdepartment.net wrote on 12-01-2012 16:46:56:
I would imagine there would be no real difference between port 5060
and 5061 on most alg's...
On Thu,
That is, when the certificates are exchanged, so if that part works...
I thought with tls the sip messages would be encrypted and therefore
an alg would not be able to mess it up.
Paul
Tony Graziano tgrazi...@myitdepartment.net wrote on 12-01-2012
16:46:56:
I would imagine there
Use binary mode (bin) in FTP when you transfer the file.
Paul
I captured packets using this command on the sipX system: tcpdump -n
-s 0 -i eth1 -w sipx-tcpdump1.cap not port 22
Then I changed the command to: tcpdump -n -s 65535 -i eth1 -w sipx-
tcpdump2.cap not port 22
When I try to open
And a happy NEW YEAR...(I will be in 2012 hours earlier then most of
you :o)
And a BIG thanks to Tony and Todd and Michel and Dale and George and
Joegen and Martin and all the others that provide answers and fixes.
I am grateful that they support this community, learn from these people.
+1
I think everyone has made their point and it is time to move on.
Thanks to all who contributed to the project in 2011. Hopefully we can
expand that list in 2012 and make this an even better project!
Sincerely,
Dave Deutschman
-Original Message-
From:
Created http://track.sipfoundry.org/browse/XX-9995 to get this improved.
Paul
Worley, Dale R (Dale) dwor...@avaya.com wrote on 12-12-2011 18:29:02:
From: Michael Picher [mpic...@ezuce.com]
I think the default install has always installed with weight 0 SRV
records.
Well, it
The answer is in the 2 issues and Dale's answer:
I don't recall where the timeout is set, but it is a variable
parameter in the SIP stack and can be adjusted straightforwardly.
XX-6065:sipX seems to be configured with T1 timer set to 100msec
instead of the recommended default of 500msec.
It's all in RFC2543 (I lerned this from Acme Packet, their SBC's smooth
all INVITE's etc according to the RFC timers):
10.4.1 UDP
A SIP client using UDP SHOULD retransmit a BYE, CANCEL, OPTIONS, or
REGISTER request with an exponential backoff, starting at a T1 second
interval, doubling
If SipX would follow the RFC then it would send out 7 INVITE's totalling
to
0+0.5+1+2+4+8+16=31.5 seconds of INVITE's.
I don't think the client will wait that long, but the current SipX could
be improved.
If the client would terminate the call before the 31.5 seconds expire then
I assume the
This tool has a function for SNOM phones only.
I don't think it's anything special except cloudifi-ing phone
administration.
From the web-pade:
As a hosted, web-based service, snom Active offers VARs a highly
reliable platform
to manage and provision hundreds to thousands of
The question has been asked before by multiple people including Kumaran
about 3 weeks ago.
Answers were provided as well:
http://forum.sipfoundry.org/index.php?t=msggoto=63948S=cec29bee730a54d52045d0a59841961e
If the ip-address of the RTP stream was specified in the sdp part of the
INVITE then all is RFC-OK.
Don't know if this answers the question but no one else replied..
Paul
Tony Graziano tgrazi...@myitdepartment.net wrote on 25-10-2011
12:39:02:
I don't think it is, but I wanted opinions.
Maybe it's easier to build a video client yourself :)
Here are some more sip video clients to try:
http://www.androidzoom.com/android_applications/sip+video+client
Maybe tiviphone is good (looking at the ratings).
Joegen Baclor jbac...@ezuce.com wrote on 07-10-2011 04:44:40:
The closest to
One small remark, I don't think that the fact that for example spark does
not support SRV's should prevent SipX from supporting HA for IM.
There are other clients out there that do support SRV's, and you have to
start building the bridge from both ends.
SipX should support HA for as many
Agreed that we don't disagree.
My remark was more a general remark that you should not wait for the
client to support things before you touch the server and vice versa.
(and the SRV support is already there, just the HA part is missing, 50%
done :)
Michael Picher mpic...@ezuce.com wrote on
FWIW: I can populate all buttons, no problemo.
BTW: I need 19 speeddials to fill a 650+BEM, 5 for the 650 buttons and 14
for the BEM.
Paul
Douglas Hubler dhub...@ezuce.com wrote on 03-10-2011 16:03:35:
On Sun, Oct 2, 2011 at 1:38 PM, Becker, Jesse beck...@sunyulster.edu
wrote:
All,
I
Yep, seems like it, probably any value other then 0 would do, have you
tested this Cyril?
Could be nice info for the
http://wiki.sipfoundry.org/display/sipXecs/Counterpath+Bria+Softphone
page.
Paul
BTW: Brian is the Bria user!
Michael Picher mpic...@ezuce.com wrote on 21-09-2011 15:48:24:
Tony Graziano tgrazi...@myitdepartment.net wrote on 16-09-2011 14:28:08:
sorry traveling does that
On Sep 16, 2011 8:25 AM, Michael Picher mpic...@ezuce.com wrote:
Thanks, the Nehos from au andbe that threw me. I can't understand
why,
it's perfectly clear.
From:
My view on the world: Simplicity for an extra 4 extensions (as in 4
trunks) in a remote location can be achieved without an extra SipX.
Good points Todd.
One of my favorite sayings... simplicity breeds reliability.
On Wed, Sep 14, 2011 at 1:27 AM, Todd Hodgen thod...@frontier.com wrote:
Sorry, just trying to understand the situation, maybe (or probably) it's
me but I'm not sure what I am not understanding.
It's small.
It's not a lot of bandwidth.
It's 4 trunks.
How many users are we talking about?
BTW: I know that trunk != extension, that's why I stated it like that.
Michael
*78
See
http://forum.sipfoundry.org/index.php?t=msgth=12238goto=42374S=58bc77940ab3597c4c454ca47645680d#msg_42374
g...@runnet.ru wrote on 13-09-2011 11:04:19:
Salut!
I've got a question about call interception in sipX 4.2. I need to
build some functionality using my sipX 4.2
What I meant to say is that I would try to test whether Bria works
properly now.
I do have to change my test setup (or add a few VM's) for that (it's a 3
box setup as well where the third server is the xmpp server with FQHN =
XMPP domain).
From the tracker:
Hi Cyril,
You really should bang Counterpath on their heads, they are not doing what
they should be doing.
If I use XMPP on Bria I lose the XMPP session 5 to 10 times a day, so it's
not useable anyhow.
Some people have less issues, but that's probably because they only added
2 or 3 XMPP
It works for me, with a third SipX box as XMPP server, a Robert Joly
trick.
It's not nice, but the other option means XMPP domain != SIP domain.
Anyhow, I am getting fed up with Counterpath as well.
I had the feeling we were getting somewhere end of last year but the last
6 months things
This is what Robert had to say about it:
Me:
Other point: when Counterpath fixes the problem is the
provisioning of Bria then still OK?
If I am not mistaken Sipx now delivers
proxies:proxy1:domain=FQDN
and it should be
proxies:proxy1:domain=Sipx/XMPP domain
So I
I've got slightly better experience with Counterpath.
I just send mails to supp...@counterpath.com and then wait, remind them,
wait (reat n times) and eventually, maybe, the problem gets sorted.
But for example the XMPP part of Bria is still far from stable although
it's known for about half a
There is no relation between the provisioning server and the server where
a phone will register.
The provisioning server is just that; the place where a phone, in this
case Bria, gets its configuration files.
When it has the config it will register based on the information in there.
If the sip
See the wiki:
http://wiki.sipfoundry.org/display/sipXecs/Remote+User+NAT+Traversal
Paul
From:
Tony Graziano tgrazi...@myitdepartment.net
To:
Discussion list for users of sipXecs software
sipx-users@list.sipfoundry.org
Date:
26-08-2011 14:51
Subject:
Re: [sipx-users] SipX setup with remote
Hi Nathaniel,
Maybe you want to try this Notepad replacement (with TAB support) and Text
diff (for example by right-clicking on a Tab) in one (and many more
features):
http://www.pspad.com/en/
Paul
Nathaniel Watkins wrote on 24-08-2011 03:47:50:
Could also pull the current file and pull
Other thing: going through the list I observed the following, not sure
whether this is OK.
commit 2a72f5c0393ae1be9c4b336969c95e6944c9dd82
Author: George Niculae geo...@ezuce.com
Date: Mon Aug 1 13:57:05 2011 +0300
XX-9792: XML Error while reloading the Media services when distributed
So, now it's no longer Hubble telescope but Hubler microscope?
Thanks for all the good work anyhow.
Douglas Hubler dhub...@ezuce.com wrote on 08-08-2011 20:54:57:
On Mon, Aug 8, 2011 at 12:58 PM, Douglas Hubler dhub...@ezuce.com
wrote:
eZuce team has discovered and fixed a number of bugs in
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,
Why is T.38 offered in the first place, if it's just a phone call.
If it isn't be offered it won't be rejected.
And even if m = image is rejected, when m = audio is accepted the caller
could still accept the call.
Is the operator to blame here or the caller?
- who is creating the INVITE
- who
Douglas Hubler dhub...@ezuce.com wrote on 14-06-2011 22:43:12:
On Tue, Jun 14, 2011 at 3:26 PM, Adrien Guillon aj.guil...@gmail.com
wrote:
Anyways, I just wanted to voice that I'm here and ready for ipv6. I
have noticed that in many projects (I'm looking at you ClearOS) there
is this
Nope, nothing yet.
I just send a friendly reminder
From:
Opyzcinr opyzc...@gmail.com
To:
Discussion list for users of sipXecs software
sipx-users@list.sipfoundry.org
Date:
26-05-2011 10:55
Subject:
Re: [sipx-users] Bria 3.2 Contact Directory
Sent by:
sipx-users-boun...@list.sipfoundry.org
It works for me, but call pickup is a bit picky in my environment.
When Bria calls Bria then Tandberg C40 can pick up
When Patton calls Bria then Tandberg C40 can't pick up.
Maybe your problems are caused by similar effects.
For the rest most of my call pick ups work fine.(haven't tested
Why would the package contain a 64 bit file when a 32 bit file is
installed?
Is this a 32 or 64 bit machine?
Tony Graziano tgrazi...@myitdepartment.net wrote on 25-05-2011 17:22:19:
you have to yum remove the package in order to update properly i
think...
On Wed, May 25, 2011 at 11:11 AM,
Hi Matt,
Below you can find the support mail that I just send to Counterpath, it
gives some clarification what is happening:
Hi,
In 3.1 there were 2 places to configure webdav, one on Account
level and one on phone level.
With 3.2 this changed to Account-level only.
Hi Matt,
If you created speeddials with presence then they will show up in Bria in
the Workgroup, press View, Workgroup in Bria to see them.
Unfortunately in Bria 3.2 I don't see any BLA (or was that BLF)
information for those speeddials anymore,
this used to work in 3.1 (will send another
It could mean your hardware clock is buggered or you have a very weird set
of NTP servers.
The time jumps are really big and all over the place.
Use NTP, and use a reliable source.
Have a look at this thread:
http://list.sipfoundry.org/archive/sipx-users/msg26255.html
Some more information
(Maybe more a SIP question then a SipX question)
I am trying to get my Tandberg Cisco C40 codec under control.
Every second call it makes fails.
Wireshark showed me that with every second call the C40 makes it sends the
authentication nonces in the first INVITE (so without being challenged by
Hi Peter,
I think the problem is that sipxproxy thinks that the caller is local,
hence the request to authenticate.
Does it work when the call is not forwarded to VM, so when you have a
phone with that extension?
If that does not work then it would mean sipx thinks that 82.148.198.254
(from
Thanks,
I'll contact Cisco-berg on this.
BTW The previous software version (3.1.4) doesn't have this problem, it
just starts with normal INVITE's.
But that has other issues.it moves the screen output on the codec-unit
from the DVI to the HDMI output and the 42 screen has no HDMI input
Kyle,
What version of Spark?
My 2.5.8 has no problems.
Aaron,
What exactly are you using for Video/Audio/Desktopsharing if I may ask?
Regards, Paul
Kyle Haefner kyle.haef...@colostate.edu wrote on 13-05-2011 22:30:01:
Aaron,
Did you do anything special.I have not been able to get
I've got a Tandberg C40 codec and as it seems I have issues with GRUU.
I don't use NAT or anything.
I think http://track.sipfoundry.org/browse/XX-8834 will fully fix the
GRUU problem.
In the mean time it would be nice if we could get this beast to work
properly.
Does anybody (Staffan..) know
Hi Staffan,
My C40 is not behind NAT, still does not work.
The 200OK is simply not ACK-ed.
The 200OK makes it to the other side (Bria) and is ACK-ed so Bria is
sending audio and video.
The C40 shows it's accepting Audio and Video in its STATUS screen, but
it's not sending anything (because the
Sorry, the 200OK was tcp ACK-ed but not SIP-ACK-ed by Bria.
I will have to dig deeper (the C40 also does not work with a Tandberg
MXP3000).
Paul
- Forwarded by Paul Scheepens/EPO on 12-05-2011 14:21 -
sipx-users-boun...@list.sipfoundry.org wrote on 12-05-2011 14:11:38:
Hi Staffan,
OK, twice sorry, the wiresharks got a hold of me
Bria does ACK the 200OK, but this never reaches the Tandberg C40, so this
ACK is definitely eaten by SipX.
Also making a call from an MXP300 the same happens, the ACK is eaten.
Is this a new problem or is it related to X-8834.
As said, I
On 12 maj 2011, at 15.26, pscheep...@epo.org wrote:
Bria does ACK the 200OK, but this never reaches the Tandberg C40, so this
ACK is definitely eaten by SipX.
Also making a call from an MXP300 the same happens, the ACK is eaten.
The sympton sounds a lot like X-8834. You have to turn on
Sorry for the bad formatting, (I can't work with/hate Lotus Notes).
I've made a snapshot after putting sipXproxy and sipXregistrar to debug,
what files should I look at and what to look for.
Or do I need to put other processes in debug mode?
What is the real filename of your
If Mr Fat Fingerer can't get it right in 3 times then you could play
something like:
You have fat fingers (or better failed to make a valid selection), your
messages will not be sent, if you want to leave a message please call/try
again but refrain from using options.
Paul
On Thu, May 5,
I would for sure keep the delay active, so after 3 failed attempts there
is always a waiting period before you can try again (preferably doubling
with each attempt, so 1 minute = 2 minutes = 4 minutes etc).
I think most users don't use the GUI that often, it would be nice if the
user was
Hi Stefan,
Bria uses the hostname to connect to the XMPP domain, it only partially
uses SRV records.
It means that to get some sort of XMPP support in Bria you need an XMPP
server with hostname=XMPP-domain (=SIP-domain).
I installed a dedicated XMPP server for that purpose (third server in a
Here's my 2 sip server config part of my Patton.
Maybe you can use it.
Paul
sipx-users-boun...@list.sipfoundry.org wrote on 25-03-2011 09:46:41:
From:
Todd R. Hodgen thod...@frontier.com
To:
'Discussion list for users of sipXecs software' sipx-
us...@list.sipfoundry.org
Date:
Hi Tony,
See inline
Tony Graziano tgrazi...@myitdepartment.net wrote on 23-03-2011 22:56:25:
There are three different discussions/functions here:
1. What does sipx use a STUN server for, and what are we using as a
default and why.
It is perhaps simpler to enhance this current config
My 2 cents:
The better solution:
From the same web page where the STUN servers were listed I found the
following:
STUN may use DNS SRV records to find STUN servers attached to a domain.
The service name is _stun._udp or _stun._tcp
And SipX users LOVE SRV records.
So if the maintainer of the
Yep, true, but an out-of-the-box solution could be easier to
maintain/less likely to be forgotten.
Paul
Michael Picher mpic...@gmail.com wrote on 17-03-2011 13:34:39:
You could just rsync that folder (/var/sipx to another box (funny
how i say that without ever setting up rsync :-) )
On
It would be nice to have a secondary tftp server so that you could put a
slave provisioning box in the DMZ for remote workers.
Paul
Michael Picher mpic...@gmail.com wrote on 15-03-2011 20:41:23:
I guess I'm not sure why a duplicate tftp server would be needed
(unless you are doing custom
Created CDR list does not sort on the displayed information
http://track.sipfoundry.org/browse/XX-9499
Paul
George Niculae geo...@ezuce.com wrote on 03-03-2011 22:31:38:
From:
George Niculae geo...@ezuce.com
To:
Discussion list for users of sipXecs software
Most of my Requests are bigger then 1300 bytesand according to RFC
3261 that means switching to TCP (or another congestion controlled
protocol):
If a request is within 200 bytes of the path MTU, or if it is larger
than 1300 bytes and the path MTU is unknown, the request MUST be sent
Good find, George,
While you're at it, maybe also add STS (Site2Site) to the list, I can't
find these records when searching for To nor From or To,
or remove the callee_route = statements altogether??? or exclude the
once that are not needed.
I also don't see records with type= CUSTM for
In a few months time we will start a large teleworking project.
About 1500 people will work from home for 2 to 3 days/week.
The max number of concurrent users will be approx 1000.
In the pilot we used Bria over VPN, simple and good enough for the pilot.
For the real thing we want to offer the
Thanks Tony, see my remarks/questions inline:
Tony Graziano tgrazi...@myitdepartment.net wrote on 11-02-2011 14:51:53:
I would enjoy seeing a vpn option on the polycom. I like the concept of
the
vpn client in the phone.
Yep, VPN on Polycom would be really nice, maybe, one day
I had
Currently the pipe is limited to 100 Mb, but it is delivered on a 1Gig
ethernet, so we can quickly scale up.
That's also part of the plan (including new firewalls).
regards and have a good weekend, Paul
From:
Nathaniel Watkins nwatk...@garrettcounty.org
To:
Discussion list for users of
I think that if Bria is doing everything according to standards when
Firewall traversal is on Automatic then SipX should be able to cope with
it.
BTW: The patch was mine... I will put it in place myself now.
Regards, Paul
From:
Tony Graziano tgrazi...@myitdepartment.net
To:
Discussion list
Sorry, some extra info:
10.1.248.31 is the GW (patton).
The Patton doesn't like to use the same port (5060) to 2 different SipX
clusters.
So I configured it to use port 5061 as source port for my second SipX
HA-pair.
See part of the Patton config below:
context sip-gateway gssipx02
Made myself floating again.
rport was enabled on Bria by accident.
Disabled it and now it seems to be OK again..
Still weird that this happened.
rport was invented for UDP, not for TCP, and I cant really find in the RFC
http://www.faqs.org/rfcs/rfc3581.html what the behaviour
For the second time in a looong time I have the problem that only 50% of
the calls from a GW to a SIP-phone come through, the other 50% go to
voicemail directly.
I have an HA setup and the GW is distributing the calls round-robin to the
2 SipX servers that form an HA-cluster.
Also calls from
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