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Hi,
A friend has a few hundred deployed LeadTek BVA8055's and needs to bulk
re-provision them. There isn't much documentation on the web.
Anyone have documentation explaining the LeadTek provisioning process and the
provisioning file format?
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On Nov 24, 2009, at 6:17 AM, Tilghman Lesher wrote:
Sounds like your local DNS resolver isn't answering queries promptly.
Thanks, I'll look into it. Our CURL function only calls one hostname over and
over.
Would setting CURLOPT dnstimeout be of use in this situation?
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additional calls to CURL also hang.
After a few minutes, tcpdump will show the CURL traffic going to the web server.
And CURL begins functioning normally for a while.
Has anyone else seen this? Or have any suggestions on how to debug this?
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Something that didn't require flash (works on the iPhone) would be nice.
blip.tv may be an option.
On Oct 22, 2009, at 3:34 PM, John Todd wrote:
I'm doing some quick research on how to get our videos from AstriCon
available in a reasonable format that allows easy viewing, reduces
our
was negotiated and tell the user)
echo-test-g...@example.com (forces g711)
echo-test-g...@example.com (forces g729)
echo-test-...@example.com (forces gsm)
...
echo-test-i...@example.com (forces ilbc)
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On Oct 14, 2009, at 1:04 AM, Dan Journo wrote:
Thanks Eric,
I'd love to be able to make it to an Astricon one day. At the
moment, its a bit out of my price range.
Do you happen to know whether RackspaceCloud.com offers a Kernel
with a timing device enabled?
Many thanks and good
/2009/10/15/freepbx-in-a-cloud-with-a-click-1436
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.
Here's a writeup of our experience with Asterisk on Amazon EC2 http://voxilla.com/2009/02/18/asterisk-on-the-cloud-with-a-click-1405
.
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and with minimal impact on the server.
If Asterisk can't do this, is anyone using anything else to handle the
registrations for Asterisk?
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might also want to consider shipping a welcome packet to the
customer, that may cover you under PayPal's physical goods terms.
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clients.
Anyone have any recommendations?
Contact me off-list and I can get you a beta version of RF Dial.
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The pem file should contain both the private key and the certificate.
On Jul 24, 2009, at 4:08 PM, John A. Sullivan III wrote:
Hello, all. After many pages of googling and testing in the lab, I'm
still a bit perplexed about how to implement tls protection for the
asterisk manager.
Hello,
I'm running into an issue with TLS transport and I am probably missing
something obvious.
We are trying to configure an extension to use TLS for the transport.
The extension can make outbound calls using TLS, but inbound calls fail.
The extension configuration in sip.conf is set to
achieving this. It works in my
local-lan testing, but not on the public servers.
Any ideas?
Do your SIP clients support SIP redirects?
If so, you might want to consider configuring server A to issue 301
redirects pointing to server B.
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of the firewall shows the INVITE
going out, but the packets never make it to the Asterisk server.
Any ideas on how to configure pfSense to work with a SIP client using
STUN and ICE, without having to install siproxyd?
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Eric Chamberlain, Founder
RF.com - http://RF.com
.
Thanks, I just tried using the static-port option.
The source ports aren't randomized any more, but the INVITEs still
disappear after the initial 401-INVITE response.
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the other SPA series
phones.
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route specific. Amazon has good bandwidth, but I
would avoid proxying media if possible.
We haven't had any reliability problems with EC2 hosting our Asterisk
real-time application servers.
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-cloud-asterisk-1-6-0-5-optimized-amazon-ec2-33857.html
.
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On Apr 13, 2009, at 8:52 AM, Tilghman Lesher wrote:
On Sunday 12 April 2009 02:34:10 am Julian Lyndon-Smith wrote:
Eric Chamberlain wrote:
On Apr 11, 2009, at 12:53 AM, Julian Lyndon-Smith wrote:
Eric Chamberlain wrote:
[snip]
Thank you, that bug does have useful information.
We
On Apr 11, 2009, at 12:53 AM, Julian Lyndon-Smith wrote:
Eric Chamberlain wrote:
Is there any documentation that explains res_config_curl?
We use the 1.4 backported version - it works so well I just can't sing
it's praises enough. We use it for realtime voicemail and realtime
queues
Is there any documentation that explains res_config_curl?
Specifically, the format of realtime calls made to the web server and
what the return string for each call should look like?
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is an 8-port unit with a low per port cost.
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web interface.
Full story available at Voxilla.com
http://voxilla.com/2009/02/23/freepbx-in-a-cloud-with-a-click-1436
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do a call
history RSS feed for each phone.
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yourself, or you can use Voxilla’s pre-built image to eliminate a lot
of the heavy lifting.
Learn more at
http://voxilla.com/2009/02/18/asterisk-on-the-cloud-with-a-click-1405
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storage; and off-site backup, EC2 starts getting more cost
effective.
Storing Asterisk realtime data in Amazon's SimpleDB and voicemail in
S3, would make for a very interesting and scalable solution.
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://voxilla.com/2009/02/12/amazon-ec2-voip-1096
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for under $50 on eBay. An old 1600
or 1700 series router with an IOS that supports SIP wouldn't cost much
either.
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Yehavi,
You might want to check out some of the EDUCAUSE http://www.educause.edu
mailing-lists to find out what other universities are doing.
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a Cisco VPN from the iPhone.
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| regseconds
Also is there a way to enable debugging to show the SQL calls being
made to the database?
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To answer my own question after reviewing chan_sip.c.
sipusers has been de-implemented in 1.6.0.1 and doesn't do anything
anymore other than appear in sip show settings.
On Nov 12, 2008, at 9:04 AM, Eric Chamberlain wrote:
I'm trying to get sipusers working with a realtime odbc database
with the server that has open vpn
server?
You might have better luck looking for mobile devices that natively
support Cisco's VPN client, iPhone, etc.
Also, your VPN may not work the way you think it will if Fring is in
the middle of your call traffic flow.
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or digital phones?
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for
every provider.
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We need to include the domain information in the Authentication digest
username SIP header field.
Using SIP/username[:password[:md5secret[:[EMAIL PROTECTED]:port] in
the dialplan breaks if authname needs [EMAIL PROTECTED], is there a
way to specify this value from the dialplan?
--
Eric
On Oct 11, 2008, at 8:28 PM, Rob Hillis wrote:
Eric Chamberlain wrote:
Is there a particular reason you /can't/ register? It would seem
that
registration would provide the functionality you require, even if
you're
only making outbound calls.
In the case of a server like Asterisk
or call center, time on
call, etc. to Analytics.
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gateways for their carriers. Then sending an SMS is no different than
sending an e-mail.
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Is there a SIP packet that a SIP client can send to Asterisk to
confirm that the credentials entered by the user are correct, without
placing a call?
We'd like to test the credentials when the user enters them, rather
than wait until they try to make their first call.
--
Eric Chamberlain
-
From: Eric Chamberlain [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Saturday, October 11, 2008 5:20 PM
Subject: [asterisk-users] Is there a way to test SIP credentials
withoutmaking a call?
Is there a SIP packet
On Oct 11, 2008, at 1:41 PM, Rob Hillis wrote:
Eric Chamberlain wrote:
I should have clarified, we're only making outbound calls, not
inbound, so there is no registration.
Is there a particular reason you /can't/ register? It would seem that
registration would provide the functionality
,+410001,pop-inbound,1510555
510555,SIP/10.10.10.170-b7d94f78,SIP/
voipprovider.com-089ae8a0,Dial,SIP/1510555:password::[EMAIL PROTECTED]
,,M(post-dial),2008-10-09 20:59:00,2008-10-09
20:59:03,2008-10-09 20:59:08,
8,5,ANSWERED,DOCUMENTATION,1223585940.35
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.
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On Sep 10, 2008, at 7:11 PM, Tilghman Lesher wrote:
On Wednesday 10 September 2008 19:55:15 Eric Chamberlain wrote:
On Sep 10, 2008, at 2:01 PM, Tilghman Lesher wrote:
On Wednesday 10 September 2008 13:22:51 Ricardo Melendez wrote:
Hi to all, I actually have an asterisk server configured
Here's the use case: call comes in, extension match is made on caller
ID and dialed number, dial plan dials a number and connects the two
call legs.
Is there a way to get the Call-ID from the SIP header of the outbound
call leg and store it in the CDR?
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Eric Chamberlain
cdr_adaptive_odbc.com doesn't explain how to set things like usegmtime
or loguniqueid.
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Is there a way to include a linefeed in the message sent by JabberSend?
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Is there anything already out there that can efficiently convert a
JSON string into Asterisk dialplan variables?
Our current backend speaks JSON and we need to parse the response to
construct the dialstring.
--
Eric Chamberlain
to
decrypt the password as part of the call out process.
Has anyone developed something like this?
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Eric Chamberlain
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On Aug 20, 2008, at 10:19 AM, Tzafrir Cohen wrote:
On Wed, Aug 20, 2008 at 10:00:55AM -0700, Eric Chamberlain wrote:
We are exploring using Asterisk for a project and we are looking
for a
way to encrypt/decrypt the peer passwords stored in the realtime
database (postrges).
Ideally, we
is available to asterisk it will be
available
to anyone else in the system with sufficient privileges.
Assume I'm using a FIPS 140-2 Level 4 HSM, now, how can I protect my
passwords when they are in the database?
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Eric Chamberlain
Founder
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http://RF.com
of outages over the past year.
Have you considered other hosting solutions? There are a number of
high quality hosting offerings, that offer 24/7 phone support, without
requiring any long term contracts.
--
Eric Chamberlain
Founder
RF.com
http://RF.com
. He only has a
SPA3102 and a windows machine to code for, so offering space on an
asterisk box could go a long way.
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Eric Chamberlain
Founder
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,
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Philippe Sultan
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asterisk-users
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asterisk-users mailing list
You can use DNS SRV records to specify more than one proxy and their order of
usage.
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Eric Chamberlain, CISSP
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_
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert McNaught
Sent: Wednesday, November 21, 2007
on signal strength and availability.
--
Eric Chamberlain, CISSP
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-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of bilal ghayyad
Sent: Wednesday, November 21, 2007 1:29 AM
We use VoicePulse Connect. They now have a POP in San Francisco.
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Eric Chamberlain, CISSP
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_
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Adrian Marsh
Sent: Saturday, November 17, 2007 5:33 AM
To: asterisk
Philippe,
Thanks for the info.
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-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Philippe Sultan
Sent: Friday, November 09, 2007 2:39 AM
To: Asterisk
, log into the administrative
interface on the SPA942 and go to each Ext tab. On each Ext tab, there will be
a NAT Settings section, change the NAT Keep Alive Enable to No.
--
Eric Chamberlain, CISSP
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-Original Message-
From: [EMAIL
Hello,
I'm looking for a SIP to XMPP Jingle voice gateway.
I see that Asterisk has Jabber and Jingle support, but it looks like Asterisk
acts as a Jabber client.
Are there any Jabber server solutions, where Jabber users can call SIP users by
using the SIP URI and vice versa?
--
Eric
What is the use case?
Linksys, Polycom, Snom, and Aastra all have their strengths and weaknesses.
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Eric Chamberlain, CISSP
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-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf
Convert the voicemail to a mp3 file.
As of firmware version 1.1.1, the iPhone mail application will recognize, but
not play wav attachments. But the mail application does, recognize and play
mp3 file attachments.
--
Eric Chamberlain, CISSP
Chief Technical Officer
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I tested this again, and wav files do play as attachments with firmware 1.1.1.
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Eric Chamberlain, CISSP
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-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Eric Chamberlain
Sent
.
Typically you want to associate your devices with other records (logs, etc.),
making a relational database a much easier to manage solution with fewer moving
parts.
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Eric Chamberlain, CISSP
Chief Technical Officer
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_
From: [EMAIL PROTECTED] [mailto:[EMAIL
You should probably post that question on the Asterisk business forum.
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Eric Chamberlain, CISSP
Chief Technical Officer
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-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Andrew Joakimsen
Sent: Saturday
speeddials that are
automatically dialed when the line goes off-hook.
--
Eric Chamberlain
On Sep 27, 2007, at 11:14 AM, Mojo with Horan Company, LLC [EMAIL
PROTECTED]
wrote:
err... you'd set them to 'yes', right? Sorry if I'm missing the
obvious.
Eric Chamberlain wrote:
You can do
You can do this with any of the Linksys SPA series ATA's or phones, just set
Make Call Without Reg and Ans Call Without Reg to no.
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Eric Chamberlain, CISSP
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-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users
to convince people to pay close to a
thousand dollars per megabyte for SMS messages.
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Eric Chamberlain, CISSP
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not be voice
traffic in the GSM traffic in the IP tunnel. Calling it VoIP would be a
stretch.
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.
The cell providers don't even offer you a cheaper rate when you use the
infrastructure you paid for. Notice how the device supports up to three
simultaneous calls? You're even paying them to provide a better signal for
your neighbors.
--
Eric Chamberlain, CISSP
Chief Technical Officer
Voxilla - http
guys and know Asterisk extremely well.
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Eric Chamberlain, CISSP
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-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Jay R. Ashworth
Sent: Tuesday, September 11, 2007 5:57 AM
).
http://www.google.com/products?q=med3531dgrls=com.microsoft:en-us:IE-SearchBoxie=UTF-8oe=UTF-8sourceid=ie7rlz=1I7HPICum=1sa=Ntab=wf
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Eric Chamberlain, CISSP
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From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf
phones.
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-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of William Stillwell (Ki4swy)
Sent: Sunday, September 02, 2007 5:45 PM
To: asterisk-users
On the SIP side of things, we have a how-to guide for the Nokia E series and
Asterisk.
http://voxilla.com/voxilla-stories/voxilla-how-to-guides/using-the-nokia-e-series-phones-with-asterisk-865.html
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Eric Chamberlain, CISSP
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Using the phone itself as a GSM-SIP gateway is not possible with the native
VoIP application, but it looks like it should be possible with a custom
application for the phone.
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Eric Chamberlain, CISSP
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-Original Message-
From
Carlos,
Do you have the NAT keepalive options enabled on the PAP2T? It sounds
like the router is timing out the connection and dropping the port
mapping.
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Eric Chamberlain, CISSP
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-Original Message-
From: [EMAIL PROTECTED
Shouldn't you ask your attorney these questions?
Any answers you receive here will not legally protect you.
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Eric Chamberlain, CISSP
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_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Zeeshan
Zakaria
Sent
with using the user web interface is that the manufacturers
quite often change the interface with new firmware releases, so you are
constantly updating the scripts.
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Eric Chamberlain, CISSP
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_
From: [EMAIL PROTECTED]
[mailto
.
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Eric Chamberlain, CISSP
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_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Olivier
Sent: Monday, August 06, 2007 10:54 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Login
A phone system for under $100 is asking a lot.
It can be done, but what is your time worth.
You might want to consider some other phone system if all you need is
IVR and analog support or look at hosted solutions.
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Eric Chamberlain, CISSP
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Andrew,
Could you elaborate on how you configure the MWI of the mobile device to
use asterisk voicemail?
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Yes it's possible.
It's also possible to have Asterisk try and find the person in the field
and either connect the call or deliver the message.
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Eric Chamberlain, CISSP
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[mailto:[EMAIL PROTECTED
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