Are there any plans to implement AMQP directly in Asterisk or is it best
to use a third party bridge like Mule?
https://jira.amqp.org/confluence/display/AMQP/Advanced+Message+Queuing+Protocol
--
Henry J. Cobb
http://www.io.com/~hcobb
Has anybody created a crontab script to check the health of an Asterisk
server?
The part I'm struggling with is some sort of IAX ping to test the
connection to each provider without making a call.
-HJC
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Asterisk 1.4.5 full log:
[Jul 2 09:31:16] VERBOSE[2682] logger.c: == Refreshing DNS lookups.
[Jul 2 09:31:16] NOTICE[2682] dnsmgr.c: host 'outbound1.vitelity.net'
changed from 64.2.142.17 to 64.2.142.29
[Jul 2 09:31:23] DEBUG[2711] jitterbuf.c: Attempting to exceed
Jitterbuf max 600 timeslots
You will likely have latency issues - causing choppiness. Start with a
traceroute to validate latency.
Anybody tried IAX trunking on G.729 with jitter buffer internationaly?
-HJC
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If the system is running away then I'd suggest looking deeper into it -
is it opening a file and never closing it again, etc. Hard to track down
unless you have a good knowlege of what's running, etc.
If you think it might be asterisk itself, then check which files it has open.
lsof -p `ps h
If the system is running away then I'd suggest looking deeper into it
- is it opening a file and never closing it again, etc. Hard to track
down unless you have a good knowlege of what's running, etc.
lsof -p `ps h -C asterisk -o pid | head -1` | grep -Fc '/dev/zap/timer'
120
You have to open
Luki [EMAIL PROTECTED] wrote:
You
don't gain anything QOS-wise by going through a tunnel, except hiding
your traffic in case your ISP purposefully assigns lower priority to
VoIP traffic and doesn't do it to OpenVPN/GRE/insert your favorite
tunnel protocol traffic.
It's a pity that OpenVPN
I'm having trouble with Polycom 501 phones that asterisk forgets how
to reach them.
...
host=dynamic
We've found much better results with the static IP here.
Can you try this?
--
Henry J. Cobb
http://www.io.com/~hcobb/
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I ran across this article today:
http://economictimes.indiatimes.com/articleshow/726843.cms
Anybody know what the implications are for asterisk servers in and out
of the country used by people in the country?
Ummm
Anybody offering VPN IAX services yet?
--
Henry J. Cobb
http
Going to the other extreme, what would it take to create an untappable and
untraceable telephone service over the Internet?
Asterisk is a good start, especially because the code can be examined (as
long as G729 is avoided) and any law enforcement back doors removed.
Now instead of trying to
[EMAIL PROTECTED] wrote:
Can't help you on the licensing thing though. I guess no one wants to
touch it since Digium's stance seems to be that you should have a
license for each seat rather than a pool.
That's not enough.
You need one license per call, with no upper limit on the number of
Marco Mouta [EMAIL PROTECTED] wrote:
Hi all,
I'm developing a Click to call Website, but now i'm getting worried with
Click to Call fraud Imagine I just create one of this PhoneNumbers
(extra charged numbers: like games, erotic lines...) in a remote
country
Then i just go to a click
Is there any standard way to signal to an IAX provider that I want them to
conference in another Asterisk box located elsewhere and then hand off the
call to the remote center after a short period of three-way talk?
My problem is that I don't want to take a double hit for latency back and
forth
it to not resolv more
than once on each address.
Have you tried setting timeout, attempts and rotate in resolv.conf?
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Henry J. Cobb
http://www.io.com/~hcobb/
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asterisk-users mailing list
We did a setup of 70 sites connected back to a central Asterisk box, and
it worked very well over an MPLS VPN.
regards,
PaulH
AsteriskIT
www.asteriskit.com.au
The best part about VPN is that it makes it harder for the ISPs to track
and mess with. ;-)
--
Henry J. Cobb
http://www.io.com
[EMAIL PROTECTED] wrote:
On 7/12/06, Henry J. Cobb [EMAIL PROTECTED] wrote:
When I've tried it, app_conference always crashed within the hour.
that's strange. we've use app_conference for months and months on end
without incident.
are you building app_conference from the main svn trunk
of the out of bounds write or double free has
corrupted its memory structures.
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Henry J. Cobb
http://www.io.com/~hcobb/
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit
Brian Capouch [EMAIL PROTECTED] wrote:
Henry J. Cobb wrote:
I tried several different combinations of app_conference and Asterisk
versions and then I had to get back to actually providing phone service
that didn't crash.
I hate to me-too, but my experience was identical. Crash after crash
5
Does it show lots of processes that are blocked, waiting for the disk?
If I had to do this I would have a battery backed writeback RAID controller.
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Henry J. Cobb
http://www.io.com/~hcobb/
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jeff oconnell [EMAIL PROTECTED] wrote:
but while i'm in the code, i'll also take a look and see if i can
figure out what your memory issues are...
When I've tried it, app_conference always crashed within the hour.
I think that the entire Asterisk server, including app_conference, needs
to be
Hi,
Is it possible to encrypt the conversation between two parties on SIP,IAX
or
ZAP channels?
Sure, setup a VPN.
You can get a Linksys VPN router for less than $100 and run whatever
protocol you like over your VPN.
--
Henry J. Cobb
http://www.io.com/~hcobb
cards on each end (that
connect to cisco or other brand routers) or a voice route between two
Asterisk servers with voice T-1 cards.
The choice would be between capacity for say G729 trunks over a data link
or latency as voice T-1s.
--
Henry J. Cobb
http://www.io.com/~hcobb
plugged in but only a dozen calls going on at one
time or will you be running a predictive dialer with 80 agents talking to
contacts while another 150 calls are being placed predictively?
--
Henry J. Cobb
http://www.io.com/~hcobb/
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Mike Lynchfield [EMAIL PROTECTED] wrote:
could it be IPP VS digium implementation ?
Actually it seems to have been a NAT issue and adding canreinvite=no (as
suggested by another person offlist) fixed it up.
--
Henry J. Cobb
http://www.io.com/~hcobb
] chan_iax2.c: Ooh, voice format changed to 256
What I hear on the phone is one ring and then nothing.
This has only been in the past few days.
Has anybody else had a problem like this?
--
Henry J. Cobb
http://www.io.com/~hcobb/
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Do you have the g729 codec?
On 6/8/06, Henry J. Cobb [EMAIL PROTECTED] wrote:
Jun 8 01:27:26 VERBOSE[2798] logger.c: -- Format for call is g729
...
Jun 8 01:27:29 DEBUG[2798] chan_iax2.c: Ooh, voice format changed to
256
Yes, and that works fine when talking with the phone itself
Which DSP based boards does Asterisk support for G729 and are any of these
more cost effective than piling on Pentiums?
BTW: Can AMD CPUs handle a higher G729 load in 64 bit mode?
--
Henry J. Cobb
http://www.io.com/~hcobb/
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The CVS server for app_conference seems to be down.
Can somebody mail me a recent copy of the sources please?
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Henry J. Cobb
http://www.io.com/~hcobb/
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Asterisk-Users mailing list
We need to conference together a call center agent, a customer and a third
party IVR and send DTMF tones from the agent to the IVR.
MeetMe has been eating our DTMFs so we'd like to try app_conference.
Has anybody setup such a configuration in app_conference and how did you
configure it?
-HJC
to increase the security for remote extensions I would like to limit a
sip-peer to a specific MAC address. Is it possible to hook into the
authentication mechanism in asterisk and allow/deny incoming
registrations?
This would be only mildly useful on the same subnet and completely useless
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