Hi,
I currently have some systems on AsteriskNOW 1.7 have been happy with its
clean simplicity reliability. Are many people here using AsteriskNOW 2.0.x?
How do you feel about it? Did Digium stick with their previous philosophy of
keeping everything very vanilla making it clean simple for
and if that
fixes it then I will open a JIRA ticket with more details.
Luke
--
From:
asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com]
On Behalf Of Douglas
or earlier and if that
fixes it then I will open a JIRA ticket with more details.
Luke
--
From:
asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com]
On Behalf Of Douglas
: 505.327.7300
-Original Message-
From: Steve Edwards [mailto:asterisk@sedwards.com]
Sent: Monday, December 19, 2011 5:52 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Limit # of inbound calls on SIP trunk
On Mon, 19 Dec 2011, Douglas
Hello,
I have a SIP provider whom I may want to have multiple trunks with, rather than
just adding more channels to the individual trunk. I have discussed the matter
with them they have told me that the only way that they identify which trunk
should be used for each call is simply by the
Hello all,
I have a system with FreePBX, and as far as I can tell it does not provide a
means to limit the number of simultaneous inbound calls on a SIP trunk.
Therefore I suspect that I'll need to do some manual dialplan manipulation.
Essentially I will have 1 (or possibly 2) SIP trunk(s)
Any suggestions from people who have done this before?
Thanks,
-
Doug Mortensen
Network Consultant
Impala Networks Inc
CCNA, MCSA, Security+, A+
Linux+, Network+, Server+
A.A.S. Information Technology
.
www.impalanetworks.comhttp://www.impalanetworks.com/
P: (505) 327-7300
F: (505) 327-7545
--
Yes. That's exactly what 802.1q is. Technically 802.1q allows the network
devices to tag each Ethernet frame with a VLAN ID. This way if you have 3
vlans, they can all be trunked over 1 Ethernet port by means of tagging the
VLAN ID.
-
Doug Mortensen
Sent via DroidX2 on Verizon Wireless™
8 comes before 42.
-
Doug Mortensen
Network Consultant
Impala Networks
P: 505.327.7300
.
From: Todd Routhier [mailto:fonema...@gmail.com]
Sent: Monday, November 28, 2011 3:55 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Recommendations
Danny,
Hello,
Does anyone have any idea of how I can program a 100ms delay in between the
ringing of 2 subsequent calls in a queue configured with a ringall strategy? In
other words, our queue ringing strategy rings all queue agents with the first
caller in line in the queue. We only permit 1 ringing
Hi,
Does a parameter exist for a queue to delay ringing/sending a caller to all
agent phones after the previous call is answered by an agent? My queue ring
strategy is set to ringall. I am using Polycom KIRK wireless DECT SIP phones.
And it looks like the KIRK wireless server may need a split
. If this becomes the solution, I may need some assistance (although I'm
sure I'd eventually figure it out).
Again, your help is appreciated.
-
Doug Mortensen
Network Consultant
Impala Networks
P: 505.327.7300
.
From: Douglas Mortensen [mailto:d...@impalanetworks.com]
Sent: Monday, November 21, 2011 9:56 AM
Of Douglas Mortensen
Sent: Monday, November 14, 2011 3:52 PM
To: 'asterisk-users@lists.digium.com'
Subject: [asterisk-users] How do extensions stay registered
I know this is probably a very basic question for many on this list. But in
troubleshooting an issue, I wanted to take a step back ask the question
-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com]
On Behalf Of Douglas Mortensen
Sent: Monday, November 14, 2011 3:52 PM
To: 'asterisk-users@lists.digium.commailto:asterisk-users
I know this is probably a very basic question for many on this list. But in
troubleshooting an issue, I wanted to take a step back ask the question. In
Asterisk (or maybe all SIP), how do extensions stay registered with the SIP
server?
Do the extensions simply register repeatedly as a means
I think that you actually should be looking to your state. I'm pretty sure that
even if CLEC is an FCC designation, it is implemented either on a per-state or
per-LATA basis. Here in NM there's only 1 LATA, which is why I'm not completely
sure. But I believe that the CLEC qualifications
Asterisk-specific log parser utility so far.
Honestly, I never needed one.
On Thu, Oct 27, 2011 at 5:16 AM, Douglas Mortensen
d...@impalanetworks.commailto:d...@impalanetworks.com wrote:
Hello all,
I have been running asterisk systems since summer of 2008. I do not claim to be
an expert. But I
swatch, splunk, zabbix etc etc etc to parse the logs for you and generate
alerts.
I haven't came across any Asterisk-specific log parser utility so far.
Honestly, I never needed one.
On Thu, Oct 27, 2011 at 5:16 AM, Douglas Mortensen
d...@impalanetworks.commailto:d...@impalanetworks.com wrote
Hello all,
I have been running asterisk systems since summer of 2008. I do not claim to be
an expert. But I have worked through many issues during this period. I have
setup manage 5 systems, which serve 6 companies total (and of course process
calls for all of the people they do business
Any ideas on why callers who call into my customer's SIP trunk are not hearing
a ringback tone? I had this on one other asterisk system, and wound up needing
to set progressinband=yes in the SIP trunk config.
I have set this on the current system restarted asterisk, but to no avail.
I am
-users] No ringback even though progressinband=yes is set
On 4/7/2011 11:02 AM, Douglas Mortensen wrote:
Any ideas on why callers who call into my customer's SIP trunk are not
hearing a ringback tone? I had this on one other asterisk system, and wound
up needing to set progressinband=yes
]
Sent: Thursday, April 07, 2011 10:49 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] No ringback even though progressinband=yes is set
On 7 April 2011 17:02, Douglas Mortensen d...@impalanetworks.com wrote:
Any ideas on why callers who call into my
is the most stable version of asterisk?
On Wed, Mar 23, 2011 at 12:58 PM, Douglas Mortensen d...@impalanetworks.com
wrote:
1.2? 1.4? 1.6? 1.8?
Thanks,
-
Doug Mortensen
Network Consultant
Impala Networks Inc
CCNA, MCSA, Security+, A+
Linux+, Network+, Server+
.
www.impalanetworks.com
P: (505) 327-7300
PM, Douglas Mortensen
d...@impalanetworks.commailto:d...@impalanetworks.com wrote:
1.2? 1.4? 1.6? 1.8?
Thanks,
-
Doug Mortensen
Network Consultant
Impala Networks Inc
CCNA, MCSA, Security+, A+
Linux+, Network+, Server+
.
www.impalanetworks.comhttp://www.impalanetworks.com
P: (505) 327-7300
F: (505
Do you have the same ratio of deployments using 1.4 as you do with 1.2? What
about 1.6 or 1.8? I simply question how accurate a comparison can be made when
one is comparing 50 1.2 deployments to 5 1.4 deployments? I'm sure it says
something, and I do appreciate the feedback.
-
Doug Mortensen
Now that we've hashed out some thoughts on the most stable version of asterisk,
I'd like to ask the question as to why I should NOT use 1.8? What are specific
reasons? For instance a few days back I was speaking with James at Rhino
Equipment. He said that he has no real data on why I shouldn't
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Why shouldn't I use 1.8?
On Fri, Mar 25, 2011 at 7:36 AM, Douglas Mortensen
d...@impalanetworks.commailto:d...@impalanetworks.com wrote:
But I would like specific reasons why I
A quick question. When looking at issues.asterisk.org, It allows issues/bugs to
be filtered by Asterisk Version. The 1.8.x options for the filter are:
1.8.2.3
1.8.2.4
1.8.3.2
1.8.4-rc2
Do you guys know whether bugs from the older version should still show up as
issues in the newer versions
1.2? 1.4? 1.6? 1.8?
Thanks,
-
Doug Mortensen
Network Consultant
Impala Networks Inc
CCNA, MCSA, Security+, A+
Linux+, Network+, Server+
.
www.impalanetworks.com
P: (505) 327-7300
F: (505) 327-7545
--
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Every so often, I have an asterisk 1.4.22-4 system that goes into an endless
loop with the following:
[Jun 1 13:30:44] VERBOSE[13160] logger.c: -- Playing 'dir-nomatch'
(escape_digits=) (sample_offset 0)
[Jun 1 13:30:44] WARNING[13160] file.c: Failed to write frame
[Jun 1 13:30:44]
Hello everyone.
I am wondering whether there is a certain technique I should use to identify
all log lines in the asterisk/full logfile that are related to a single call.
If a user reports that something strange happened with a certain call, I'd like
to be able to easily go back and look at
I just had a user report that they called out to someone on a cell phone this
morning, and after a short conversation, the call was dropped/lost. The person
on the cell phone says that this is very rare, and would not suspect the
dropped/lost call to be on their side. I have looked at the
I am wondering if anyone knows of a way to do this, as it would be much
more meaningful for our CDR reports. We use FreePBX under the Elastix
distro. We are able to set the CALLER's CID on inbound calls by using
the Asterisk Phonebook module in FreePBX, then configure the Inbound
Route settings to
Steve:
=
Thanks for the info on the agi debug command. We'll see what information
we can garner with that. Thanks also for the advanced logging info.
Unfortunately, we are pretty aware of how AGI works (at least at the
level that you explained it). Thanks for the
An update here. Yesterday's problem has been solved (partly). After
looking closer at the results of my perl script, as well as looking at
the documentation, I realized that we were leaving the escape character
argument off of the STREAM FILE command in the mono application. After
appending to
Hello.
I have a software developer creating a .Net / mono program to use as an
AGI script. We are having problems getting it to stream files. From what
we can tell, it is talking to asterisk correctly when called from the
dial plan. Its stderr output goes to the asterisk console. But asterisk
@lists.digium.com
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On Mon, 2008-11-10 at 10:50 -0700, Steve Murphy wrote:
On Mon, 2008-11-10 at 10:10 -0700, Douglas Mortensen wrote:
I have an asterisk system where the asterisk daemon dies typically at
least once per day
I have an asterisk system where the asterisk daemon dies typically at least
once per day. It is running in the wrapper safe_asterisk, which automatically
starts the daemon back up. But we find this unacceptable because when the
daemon dies, we usually have active calls drop, and sometimes we
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