Hi @ all,
in the reason on having some agents logged in in more than one queues I need to
get the state if an Agent goes in
Postprocessing State to do this for this Agent in all other Queues he is logged
on.
For this I tried to catch this Event above the AMI. But there is never thrown
the
Hi,
Is it possible? I tried AddQueueMember(SIP/$TRUNK/number) but it tries
to call SIP/$TRUNK instead.
Cheers,
Darkbasic
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Hello
In case a NAT firewall prevents using STUN to open SIP/RTP ports, a
solution is to first connect the phone to the Asterisk server through
a tunnel, and then have data go through the tunnel.
Are there hardphones that support OpenVPN?
If none, what about SSH? Is this a good
Gilles wrote:
Are there hardphones that support OpenVPN?
I've seen people mention snom with OpenVPN:
http://wiki.snom.com/Networking/Virtual_Private_Network_%28VPN%29
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary Safety,
deserve
Hi Gilles,
You can't tunnel UDP through SSH.
Some of the newer Grandstream handsets support OpenVPN and are a bit cheaper
than the Snom alternatives.
-
Regards,
AJ Stanfield
t: 0161-850-4001
e: a...@dmcip.com
w: http://www.dmcip.com
- Original Message -
From: Gilles
hello,
yeallink T26 and T28 support OpenVPN too
Regards
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On Tue, 31 Jan 2012 12:54:41 + (GMT), Arthur Stanfield
a...@dmcip.com wrote:
You can't tunnel UDP through SSH.
Some of the newer Grandstream handsets support OpenVPN and are a bit cheaper
than the Snom alternatives.
Thanks for the infos. So the only way to use SIP through locked-down
NAT
On Tue, 31 Jan 2012 07:57:22 -0500, bakko asannu...@gmail.com
wrote:
yeallink T26 and T28 support OpenVPN too
Thanks for the infos.
If someone tried the Snom, Grandstream, or Yeallink, how good is their
OpenVPN connection?
--
Hi Daniel,
Thank you very much for your responses! At least I only wasted 5 hours
on the chained certificate issue.
I have some responses / questions below.
The certificate is a GeoTrust Rapid SSL certificate. I have received
the my server specific crt file and also an intermediate certificate.
On 01/31/2012 12:17 AM, Ira wrote:
Tonight I tried 4 versions of Asterisk; 10.0.0, 10.0.1, 10.1.0 and trunk.
On 10.1.0 and trunk, I can't successfully enter the password for any
mailbox in voicemailmain on my Aastra 480i phones. All four version work
with a Snom cordless SIP phone. In 10.0.0
Use local channel
2012/1/31 Niccolò Belli darkbas...@gmail.com:
Hi,
Is it possible? I tried AddQueueMember(SIP/$TRUNK/number) but it tries to
call SIP/$TRUNK instead.
Cheers,
Darkbasic
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C F wrote:
Use local channel
You'll also want to keep track of the number of active calls, since, I
believe, the queue app will not be able to see signaling on that line.
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary Safety,
Alternately, you could use a SIP channel with followme or the newer releases
have some Bluetooth capabilities.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle
Sent: Tuesday, January 31, 2012 8:46 AM
Hello
To cut down on the number of hackers trying to break into an Asterisk
server, I'd like to simply move the SIP port from the standard UDP
5060 to something non-standard.
Since this server must be able to receive INVITEs from any SIP UA
(server or client), it appears that I must add an SRV
On Tue, 2012-01-31 at 14:13 +0100, Gilles wrote:
On Tue, 31 Jan 2012 07:57:22 -0500, bakko asannu...@gmail.com
wrote:
yeallink T26 and T28 support OpenVPN too
Thanks for the infos.
If someone tried the Snom, Grandstream, or Yeallink, how good is their
OpenVPN connection?
Using
On Tue, 31 Jan 2012 10:03:46 -0600, Jeff LaCoursiere
j...@sunfone.com wrote:
Using Yealink T-28 with OpenVPN works fine - about three weeks now with
no issues. Bummed that it seems to only support one tunnel, though. I
asked their support team if they could make whatever changes necessary
to
Jeff LaCoursiere wrote:
Bummed that it seems to only support one tunnel, though
As in you can't register the phone to more then 1 remote Asterisk server
via 2 different VPN tunnels or you can't have more then 1 call per VPN link?
Doug
--
Ben Franklin quote:
Those who would give up
On Tue, 2012-01-31 at 11:29 -0500, Doug Lytle wrote:
Jeff LaCoursiere wrote:
Bummed that it seems to only support one tunnel, though
As in you can't register the phone to more then 1 remote Asterisk server
via 2 different VPN tunnels or you can't have more then 1 call per VPN link?
On Tue, 2012-01-31 at 17:23 +0100, Gilles wrote:
On Tue, 31 Jan 2012 10:03:46 -0600, Jeff LaCoursiere
j...@sunfone.com wrote:
Using Yealink T-28 with OpenVPN works fine - about three weeks now with
no issues. Bummed that it seems to only support one tunnel, though. I
asked their support
On Tue, 31 Jan 2012 10:44:12 -0600, Jeff LaCoursiere
j...@sunfone.com wrote:
No - the phone allows you to register with multiple servers, and I would
like to reach each server over its own tunnel. It won't do that today.
Thanks for the info.
--
On 31/01/12 16:16, Gilles wrote:
Hello
To cut down on the number of hackers trying to break into an Asterisk
server, I'd like to simply move the SIP port from the standard UDP
5060 to something non-standard.
Something more appropriate for your goal might be a move to TLS, it is
definitely
I've created a page on wiki.asterisk.org outlining some changes we're
proposing to make to the Asterisk release and support cycles. As always,
before implementing any changes of this type, we'd like to collect some
community feedback on the proposal.
The page is here:
I am having problems with a deadlock in Asterisk 1.8.9.0.
The system is a x86_64 machine that is being used as a callcenter. The agents log in via the AgentLogin application, and each Agent/ channel is assigned to one or more queues. A custom separate process generates calls into the queues
From my perspective this makes a lot more sense than the current cycle. My
big issue is with patches that have new features. Not having them in a
trunk released version adds a lot of issues trying to support them in
house. I like the idea of LTR release more often that would have the
feature
I'm attempting to configure an H.323 trunk (using chan_h323) between an
Asterisk box and an Avaya IP office. It mostly works. Calls from Polycom SIP
devices registered to Asterisk can place calls over the trunk to IP Office
extensions and everything works great. However, calling from an IP
Il 31/01/2012 15:42, C F ha scritto:
Use local channel
Thanks, I completely forget about local channel.
Darkbasic
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Il 31/01/2012 15:46, Doug Lytle ha scritto:
You'll also want to keep track of the number of active calls, since, I
believe, the queue app will not be able to see signaling on that line.
I'm sorry but, how to?
Darkbasic
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I like the idea of LTR release
more often that would have the feature patches baked in. Case
in point the new conference app requires a jump to version 10
while the 1.8 conference app is quite useless but 1.8 is my LTR
version so I am stuck without the
Hello *,
is here someone with an experience of the Eicon Diva PRO 3.0?
I need 2 or 3 cards to connect an Anlagenanschluß (I do not know the
name in englisch, but is 2 or more lines with the same numbers) to my
intranet servers (IBM x335/x345 - requires PCI-X or at least PCI 2.1).
Thanks,
On 31-01-12 22:47, Michelle Konzack wrote:
Hello *,
is here someone with an experience of the Eicon Diva PRO 3.0?
I need 2 or 3 cards to connect an Anlagenanschluß (I do not know the
name in englisch, but is 2 or more lines with the same numbers) to my
intranet servers (IBM x335/x345 -
Why the short life on Asterisk 10?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P.
Fleming
Sent: Tuesday, January 31, 2012 11:39 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
On 01/31/2012 06:57 AM, Gilles wrote:
Thanks for the infos. So the only way to use SIP through locked-down
NAT routers is to use OpenVPN, either with the few hardphones that
support it or with a softphone on a computer.
You can also setup OpenVPN to connect a remote subnet (remote office)
On 01/31/2012 04:07 PM, Danny Nicholas wrote:
Why the short life on Asterisk 10?
Can you rephrase your question? This proposal does not change the
planned lifetime of Asterisk 10 at all.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP:
1.8 and 11 forward all seem to have a timeline of around 5 years. 10 only
runs for two. Since the code is available that isn't a biggie to me, but
the appearance of a short life could cause some customer discomfort.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
On 01/31/2012 05:14 PM, Danny Nicholas wrote:
1.8 and 11 forward all seem to have a timeline of around 5 years. 10 only
runs for two. Since the code is available that isn't a biggie to me, but
the appearance of a short life could cause some customer discomfort.
You must be looking at a
To properly phrase the question, why is 10.X not an LTS release?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P.
Fleming
Sent: Tuesday, January 31, 2012 5:20 PM
To: asterisk-users@lists.digium.com
The LTR releases have a shorter support/life cycle and 10 is not an LTR.
That accounts for the shorter life on 10. This is why I like this proposal.
We would get faster LTR releases and that would allow us to have newer
features sooner but still offer existing deployments security fixes on LTR
On 01/31/2012 05:20 PM, Danny Nicholas wrote:
To properly phrase the question, why is 10.X not an LTS release?
Umm... because it wasn't planned to be, and the decision was made nearly
two years ago? Asterisk 1.8 is an LTS release, and now (with this
proposal) Asterisk 11 would also be an LTS
On Tue, 31 Jan 2012 16:25:37 -0600, Dale Noll dn...@wi.rr.com wrote:
You can also setup OpenVPN to connect a remote subnet (remote office)
and it will route all traffic between subnets. Configure the hard/soft
phones on the remote subnet to route through the OpenVPN. This works
pretty well for
On Tue, 31 Jan 2012 18:22:41 +0100, Daniel Pocock
dan...@readytechnology.co.uk wrote:
Something more appropriate for your goal might be a move to TLS, it is
definitely needed for any external connectivity
[...]
As a further safety measure, you could use something like repro or
Kamailio as a SIP
I like it!
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asterisk-users mailing list
I have an Asterisk server it runs great with SIP phones, soft SIP phones
(twinkle) and a soft SIP phone app on my Android phone but I am having
problems getting two ATA boxes working. I have a Linksys PAP2T, it is
unlocked and I have used them before with no problems. I was able to
receive calls
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