ge...@riseup.net writes:
Any idea how to solve this?
You can control src address selection with with ip route command.
E.g. if you know that you want to reach 192.168.0.0/24 with a source
address 192.168.0.50, you can do:
ip route change 192.168.0.0/24 src 192.168.0.50 scope link dev eth0
On Wednesday 12 October 2011, Michael C. Robinson wrote:
[stuff deleted]
Seems to be working now, good eyes.
I have the O'Reilly book on Asterisk 1.8, though I'm wondering where to
look in it to learn how to allow incoming calls from the phone company
to ring SIP box and FXS connected
Hello,
I'm using asterisk with 84 extensions (aprox 45 always connected). When i
look to the opened channels i sow many channels opened without reason even i
don't have any active calls.
Is there someone else that en-counted the same problem? Is there any fix to
this bug? I have the following
On 10/12/2011 06:15 PM, ge...@riseup.net wrote:
I solved it by having two physical connections to my network.
Yes, I thought of this too.
I used the second nic for the drbd-communication, but I think I will have
to change this.
If your networking equipment supports VLAN, you could add a
Hi Dale,
Wow, thanks for the tip. I just start to change the network config, and
see how it works.
If your networking equipment supports VLAN, you could add a virtual lan
to the existing ethernet device.
Jeah, got a Cisco managed switch.
Then assign an address to eth0.42. You can use eth0
Opened the issue: ASTERISK-18707
On 10/12/11, Warren Selby wcse...@selbytech.com wrote:
Just a guess at this point, but I'd say because you had two agents
registered to the queue, but only one was available? If you dynamically
logout the Unavailable agent, it should not show up in the
On 10/12/2011 07:24 PM, Barry Miller wrote:
Up through 1.8, 'database show' returned results ordered by key. In 10,
the output is unordered (or maybe chronological?). Is this intentional?
It was not intentional, probably a side-effect of the switch to SQlite 3
from BDB. Unfortunately, that
Don't you think the problem will still occur that the answers from
asterisk seem to come from the main address assigned to the NIC? Or
isn't this possible because of the vlan?
Thanks,
Georg
It should work. As far as the OS and routing tables are concerned, eth0
and eth0.42 are different
If your networking equipment supports VLAN, you could add a virtual lan
to the existing ethernet device.
vconfig add eth0 42
Then assign an address to eth0.42. You can use eth0 for your endpoints
and eth0.42 for your provider or whichever way you want it.
You do then, of course, need to
Hi all,
I've got a bunch of Polycom 301's at a site that sits behind a Sonicwall
firewall.
What I'm seeing is that the phones are constantly becoming unavailable,
followed shortly by becoming available again.
The phones register just fine and sound great on out-bound calls. The phones
are
From time to time a similar subject pops up on the list. I'd like to repeat
it is extremely dangerous to ban IP based on a suspicious UDP activity. The
source IP of an UDP packet can be easily forged, so if you start using
fail2ban or other blacklist techniques, it can be very awesome to start
On 10/13/2011 10:53 AM, ge...@riseup.net wrote:
Just tested this, doesn't work. Asterisk ist still replying using the
main-address associated to the NIC.
In a previous posting, Jim Lucas proposed...
--- snip ---
I solved it by having two physical connections to my network.
PBX E0 IP
On 10/13/2011 06:21 PM, Mike Diehl wrote:
I'm thinking it's a firewall/NAT timeout issue. Has anyone seen this? Has
anyone fixed it? Any ideas, otherwise?
Did you try turning off the SIP ALG on the Sonicwall?
Regards,
Patrick
--
When you set bindaddr=0.0.0.0 Asterisk will not bind to any specific IP and the
OS will choose the source IP of the packet.Let me repeat this: THE OS PICKS
THE SOURCE IP.
If your OS routing tables are correct, then the packets will be sourced from
the correct IP.
-Original
This has two different subnets for eth0 and eth1.
Do you have different IP subnets for eth0 and eth0.42?
If you do, I do not know the reason for the problem.
Ah, sorry, overlooked this.
I will give it another try with different subnets.
Thanks,
Georg
--
On Thu, Oct 13, 2011 at 10:13:49AM -0500, Kevin P. Fleming wrote:
On 10/12/2011 07:24 PM, Barry Miller wrote:
Up through 1.8, 'database show' returned results ordered by key. In 10,
the output is unordered (or maybe chronological?). Is this intentional?
It was not intentional, probably a
Have you tried adding 'qualify=no' in the peer definition?
Are all the natted phones using port 5060 as their SIP port? I don't know
how many a bunch is but if it's not too many you might try having each
phone bind to a different port for its SIP signaling, sometimes that is
helpful with strict
While talking to Cisco about using the AS5400XM as a gateway for T1s
to SIP they recommended I also look at the Cisco 3925 ISR. Does
anyone have any experience using this device as a gateway to an
Asterisk server? I've heard really good things about the AS5400XM,
but nothing real world about the
Looks like we fixed it. The NAT session defaulted to 30 seconds...! When we
increased it to 2 minutes, the problem went away!
Thank you for your time!
Mike.
On Thursday 13 October 2011 1:00:49 pm Luke Hamburg wrote:
Have you tried adding 'qualify=no' in the peer definition?
Are all the
How do I specify command line arguments to the MEETME_AGI_BACKGROUND?
I'm using the 1.4.42
I am setting the value in a call file:
SetVar: MEETME_AGI_BACKGROUND=myagi,-myarg
also tried
SetVar: MEETME_AGI_BACKGROUND=myagi -myarg
In both cases the CLI said Failed to execute
Thanks,
Jerry
--
It was not intentional, probably a side-effect of the switch to SQlite 3
from BDB. Unfortunately, that command was not documented to produce the
database results ordered in any particular order, so this change isn't a
bug, just a side-effect.
Thanks. The only time it really matters to me
On 10/13/2011 03:25 PM, Jerry Geis wrote:
How do I specify command line arguments to the MEETME_AGI_BACKGROUND?
I'm using the 1.4.42
I am setting the value in a call file:
SetVar: MEETME_AGI_BACKGROUND=myagi,-myarg
also tried
SetVar: MEETME_AGI_BACKGROUND=myagi -myarg
In both cases the CLI
Can you post the call file with the pertinent info blacked out? I'm on
1.4.41 so I might be able to assist.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis
Sent: Thursday, October 13, 2011 3:31 PM
On Thu, Oct 13, 2011 at 02:52:58PM -0500, Terry Wilson wrote:
It was not intentional, probably a side-effect of the switch to SQlite 3
from BDB. Unfortunately, that command was not documented to produce the
database results ordered in any particular order, so this change isn't a
bug,
Can you post the call file with the pertinent info blacked out? I'm on
1.4.41 so I might be able to assist.
I am attempting to use a local call, start a meetme, bring others in the
conf, and run a background
agi to play the wav file in the conference. All automated.
Channel:
I would replace
SetVar: MEETME_PLAYFILE=/tmp/jerry.wav
With
SetVar: MEETME_PLAYFILE=/tmp/jerry
Also, you could replace your C AGI with a BASH AGI just to verify that the
data is getting there.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hello everybody,
I've got a ticket for Astricon but i can't go ... So i don't want to
lost it for nothing and i want to give it for free to the asterisk
community.
Just send me a tweet to @avencall with astricon in the tweet. I will
choose at random
Previously posted to the Users list (FYI).
We have a system running Asterisk-1.4.40. Queue calls are
distributed using rrmemory with a 20 second timeout. What we are
seeing is; when a call in the queue will call the first agent for 20
seconds, and subsequent attempts will call agents for random
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