working in this configuration except for the DID
the call out is made on. This still continued to be incorrect.
Are the users in the correct user group. In the user group have you
assigned the caller id.
--
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--
Gerald Drouillard
Technology Architect
this problem doesn't exist!
Could I conclude this is not seesion timers on my or ITSP side?
Can you call in and leave a voicemail longer than 60sec?
--
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which I believe is a capacity issue on their side.
Other ITSP players you should check are http://www.voipinnovations.com/
and Appia if you have a lot of patience.
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--
Gerald Drouillard
Technology Architect
Drouillard Associates, Inc.
http
had some recent
complaints of busy signals as well...
That should go away if you can use IP auth.
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--
Gerald Drouillard
Technology Architect
Drouillard Associates, Inc.
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anything by
SSH'ing with the PlcmSpIp user, whether it's a public IP or not.
I would recommend all your ssh servers have sshd_config with at least:
AllowUsers user1name,user2name
PermitRootLogin no
I am also a big fan of fail2ban
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--
Gerald Drouillard
of the server.
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to
the sipxecs box.
~Noah
Do you understand that if your sipx smtp server is only running on
localhost that you will not be able to connect to it via
telnet/ssh/whatever?
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--
Gerald Drouillard
Technology Architect
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http
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voip.ms gateways in sipx. You just add the order
of pref. in the dial plan(s).
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On 11/5/2012 2:16 PM, Tony Graziano
wrote:
I am looking at a strange issue with a system which
had a drive failure. We replaced the drive and reloaded (did not
restore) the system, then updated it to the latest update. We see
the proxy staying steady at
weeks of quite with #22 now. All local clients are not
using ssl.
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, but it
appears that there were a few resent versions of 4.4 that had issues.
We have been using patch 22 for about a week now and it seems to be
working fine.
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Gerald Drouillard
Technology Architect
Drouillard Associates, Inc.
http
no
AllowUsers yoursecretusername
MaxAuthTries 3
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/tcp also?
Is your ITSP sending the calls to your based on your registration is it
IP based. If IP the call has to come in on 5080 to be able to transfer.
Did you do a yum update?
Send profiles to the server: System|Servers
Reboot
--
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--
Gerald
.
- Thank you all for your continued testing and fixes.
Thank you. Seems to be working great on 4 different systems now.
--
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--
Gerald Drouillard
Technology Architect
Drouillard Associates, Inc.
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an extension I hear this:
Please hold while i transfer your call, but nothing happen, the call is not
transfer to extension.
Any idea?
Do you have patch #20 installed? If not, you should. Did this work
before and now it is not?
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--
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.
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.
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. Jitsi works
for us.
On Wed, Sep 26, 2012 at 11:36 AM, Gerald Drouillard
gerryl...@drouillard.ca wrote:
On 9/26/2012 11:14 AM, Tony Graziano wrote:
I sent server profiles. I deleted and recreated the user. In the
registrar it shows the PassTokenDB='plain text password'
authTypeDB='DIGEST
of
stability before working it back in.
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will not work on a server with 2 NIC
cards even if only one is being used.
Am I correct in my assumption?
It will work. Just configure only one of them.
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--
Gerald Drouillard
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We have better luck with difficult locations using tcp instead of udp.
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--
Gerald Drouillard
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http
the changes, ntp server may take a few minutes before it
is synchronized.
*
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sipx
service providers.
-Bryan Anderson
Sounds like you are have an ALG/firewall issue on one of your routers.
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as a gateway? Send server profile and reboot sipx?
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that was
working and now is not after a phone reboot. Setting to TCP or UDP did
not help. Even tried the TCP Fast Failover that seems to been added
recently.
also, see if sending server profile helps.
I'll try that this weekend.
--
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--
Gerald Drouillard
profile and restarting fixed it.
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in dhcp like:
sip-servers-name
boot-server
Dynamic DNS reverse domain and Dynamic DNS domain name could also
come into play for both DNS and dhcp.
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--
Gerald Drouillard
Technology Architect
Drouillard Associates, Inc.
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packages.
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.
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been problems with those numbers before the switch.
On Fri, Apr 27, 2012 at 6:58 AM, Gerald Drouillard
gerryl...@drouillard.ca mailto:gerryl...@drouillard.ca wrote:
On 4/26/2012 6:45 PM, m...@grounded.net mailto:m...@grounded.net
wrote:
Is anyone else using appia on port 5080
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)*
**
*Usually first password doesn't work. Tell them to Reload trunk group.*
*When ordering specify 10 digit calling - no +1{phonenumber}*
--
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--
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Technology Architect
Drouillard Associates, Inc.
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,fax_enable_t38_request=true,proxy_media=false,bypass_media=false,fax_disable-v17=true}sofia/gateway/$DOMAIN/$DESTtxfax($TIF)
$DOMAIN
Now I just have to figure out if I want to do all the work to migrate the
t38modem/hylafax server.
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days now and have been impressed. We
had a site that could not connect with 3cx (probably because of alg in
some upstream router) and Jitsi worked instantly. Call quality is
better than 3cx, due to the many encodings available.
--
Regards
--
Gerald
where you have over $200/month in services you may
want to consider:
http://www.voipinnovations.com/
--
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--
Gerald Drouillard
Technology Architect
Drouillard Associates, Inc.
http://www.Drouillard.biz
hair pin transfers. Who is your ITSP?
--
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--
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with forwarding rules/schedule.
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On 4/19/2012 2:58 PM, Simon Brûlé wrote:
I added 192.168.175.0/24 http://192.168.175.0/24 to the intranet
subnet and I still have the same problem.
2012/4/19 Gerald Drouillard gerryl...@drouillard.ca
mailto:gerryl...@drouillard.ca
On 4/19/2012 2:37 PM, Simon Brûlé wrote:
Hi, I know
?
That would be the only thing inbetween your softphone and the sipx
server... right?
http://screenshots.portforward.com/Cisco/Linksys_E2500/Management.htm
2012/4/19 Gerald Drouillard gerryl...@drouillard.ca
mailto:gerryl...@drouillard.ca
On 4/19/2012 2:58 PM, Simon Brûlé wrote:
I added
trying it out now and so far I am impressed.
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a email to fax gateway.
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that was a heavy hylafax user with usb
modems. We are now using sipx for receiving faxes and hylafax for sending.
http://www.drouillard.biz/blog/sipxecs-hylafax-and-t38modem/
--
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--
Gerald Drouillard
Technology Architect
Drouillard Associates, Inc.
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or cancel a transfer. I have verified in my
voip.ms account that it is registered with the public IP and port 5080.
So it looks like we are back to a firewall problem, correct?
Yes. What kind of firewall do you have?
--
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--
Gerald Drouillard
Technology
to the same number within a minute of each other.
Is this a new install? A capture at the firewall usually tells all.
Probably some kind of ALG / sip helper.
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--
Gerald Drouillard
Technology Architect
Drouillard Associates, Inc.
http
Intranet Domains in System|Internet Calling
* Send profiles to the server?
* Send profiles to the phones?
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--
Gerald Drouillard
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On 3/26/2012 11:00 AM, Stiles Watson wrote:
Anyone else have an idea why I lose audio when retrieving a call from
hold or canceling a transfer?
Sounds like the call did not come in on port 5080?
--
Regards
--
Gerald Drouillard
Technology Architect
On 3/26/2012 1:41 PM, Todd Hodgen wrote:
Internal Blind transfer seems to work fine for me.
It works ok for us if one of the endpoints is a polycom phone or via a
DID. 3cx to 3cx transfer does not work.
--
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--
Gerald Drouillard
Technology
if it works for you
now. I'm running version 5.x, the version now being downloaded is
version 6.x
I am running 6.0.2.0943.0 which is the newest I believe.
--
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--
Gerald Drouillard
Technology Architect
Drouillard Associates, Inc.
http
with voip.ms
http://www.drouillard.biz/blog/ip-authentication-with-sipxecs-and-voip-ms/
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--
Gerald Drouillard
Technology Architect
Drouillard Associates, Inc.
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On 3/23/2012 2:11 PM, Josh Patten wrote:
VPN or possibly a session border controller with TLS capabilities.
I have not had any success with TLS getting through firewalls or access
points that have ALG or Verizon MiFi.
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Gerald Drouillard
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. Try sending profiles to
the server. I eventually got it to start working without have to dig
into the log files ;-).
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--
Gerald Drouillard
Technology Architect
Drouillard Associates, Inc.
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ACD checked under server roles?
Maybe uncheck and check again
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.
When ordering specify 10 digit calling - no +1{phonenumber}* (if you
want voip.ms style)
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ahead and do that.
Sit back and enjoy your registration free connection.
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the frequency of the problem. It appears to affect
just how quickly the registration recovers. IMO the registration system
on voip.ms can get overloaded. I had captures showing voip.ms rejecting
the registration.
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--
Gerald Drouillard
Technology
switching to
NAT=yes made the accounts come back up. IMO something had changed at
voip.ms. Anyway it works. And I am sure it works both ways, I just
lean to setting it to Yes now.
--
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--
Gerald Drouillard
Technology Architect
Drouillard Associates
On 2/9/2012 8:02 AM, Philippe Laurent wrote:
Thanks Gerald! Working on the set up today after end-of-business
day. I'm assuming that for IP Auth that I will have to port forward
5080 to 5060?
No. 5080 to 5080
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Gerald Drouillard
Technology
$359.99
19-115-094 CPU INTEL|CORE I3 2120T 2.6G 3M R
1
$134.99
$134.9
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--
Gerald Drouillard
Technology Architect
Drouillard Associates, Inc.
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On Mon, Feb 6, 2012 at 9:31 PM, Gerald Drouillard
gerryl...@drouillard.ca wrote:
On 2/6/2012 3:10 PM, Nathaniel Watkins wrote:
Has anyone else seen the below behavor? I seem to get this about once a
day - one minute later I get a message that the ITSP account has recovered.
I was quick
, 2012 at 8:23 AM, Gerald Drouillard
gerryl...@drouillard.ca mailto:gerryl...@drouillard.ca wrote:
On 2/5/2012 12:20 AM, Tony Graziano wrote:
Fail2ban requires the firewall use iptables I think.
You can and should run it on the sipx server.
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: http://list.sipfoundry.org/archive/sipx-users/
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On 2/4/2012 11:41 PM, Gerardo Barajas wrote:
Hi members of the list.
¿Is Fail2ban useful in this situation??
Yes
Search the list and you will see how.
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--
Gerald Drouillard
Technology Architect
Drouillard Associates, Inc.
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On 2/5/2012 12:20 AM, Tony Graziano wrote:
Fail2ban requires the firewall use iptables I think.
You can and should run it on the sipx server.
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--
Gerald Drouillard
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confirm this behavior?
Are you sure your MOH is encoded properly?
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the 5080 port then you will not be able to transfer the
call.
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, the registration method is a waste of everyone's resources if you
have a static IP.
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about outbound from the sipx server? I would
imaging sipx would bounce to the next gateway if voip.ms was unreachable.
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Gerald Drouillard
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with Verizon MiFi 4G.
I am thinking TLS may do it?
Seems like VPN's would be a lot of work.
I have tried different ports with not luck. Using x-lite and 3cx as the
clients.
I noticed 3cx has a tunnel feature.
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Gerald Drouillard
Technology
On 1/12/2012 4:43 PM, Michael Picher wrote:
Tls is currently broken afaik
That is what I thought. Thanks. It would be nice to get it working
if it allows the client to sneak around their ALG routers.
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Gerald Drouillard
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is the boot/firmware on the polycom
phones? Were the phones provisioned via sipx or did you do it manually?
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Gerald Drouillard
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to cd to '${deleted_dir}' 12
fi
done
else
echo Mailstore '${MAILSTORE_DIR}' not found 12
exit 1
fi
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. That is always a good starting point when
troubleshooting.
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this is something that people have had
experience with.
Have you found evidence that your server went to sleep? Is it a laptop?
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Gerald Drouillard
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On 12/20/2011 2:41 AM, Todd Hodgen wrote:
My own Voip.ms account seem to be functioning fine.
You may not notice if you don't have the alarms being emailed to you, or
you don't have constant voice traffic.
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Gerald Drouillard
Technology
. It is getting bad enough now that we are
probably going to switch away from them. It is effecting 4.2.1 and 4.4
systems.
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Gerald Drouillard
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, but with the 8000 there are no FXO ports.
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suggest using software raid instead of the hardware raid. That
way you can take the drives and plug them into a new machine without too
much concern about adapter compatibility.
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--
Gerald Drouillard
Technology Architect
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http
kernel has sip helpers
loaded and kicked in only for the t.38. The following clears up the
problem:
http://www.shorewall.net/FAQ.htm#faq77
or basically:
rmmod nf_nat_sip
rmmod nf_conntrack_sip
Maybe a nice note for the wiki?
--
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--
Gerald Drouillard
hitting the IVR will have this parameter to false.
In an ideal world I would vote for this to be a parameter on the gateway.
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--
Gerald Drouillard
Technology Architect
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I added a JIRA. Please vote for it here:
http://track.sipfoundry.org/browse/XX-9958
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Gerald Drouillard
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Drouillard Associates, Inc.
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to the dev list.
On Fri, Nov 11, 2011 at 6:14 PM, Gerald Drouillard
gerryl...@drouillard.ca mailto:gerryl...@drouillard.ca wrote:
On 11/11/2011 5:36 PM, Tony Graziano wrote:
I also decided to test it with other media services (conferencing),
and it only seems to be with voicemail
Does anyone know of a sipx rtp keepalive setting during voicemail?
I just wanted to share this experience for anyone in the future having
this problem:
02:37:02 PM *[Gerald Drouillard]* Yes, and the voicemail message is
capped at exactly 1 min everytime with the 2 dids on this account
, because they seem to treat all pop's differently or
with favoritism. I'm sure those POP's got what they deserve though.
On Fri, Nov 11, 2011 at 4:13 PM, Gerald Drouillard
gerryl...@drouillard.ca mailto:gerryl...@drouillard.ca wrote:
Does anyone know of a sipx rtp keepalive setting during
/tmp/test.cap
--
Regards
--
Gerald Drouillard
Technology Architect
Drouillard Associates, Inc.
http://www.Drouillard.biz
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there.
I'll be testing a new trunk/account with appia today and keep this in mind.
--
Regards
--
Gerald Drouillard
Technology Architect
Drouillard Associates, Inc.
http://www.Drouillard.biz
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