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' and
'switch' in the same sentence seems confusing.
Do VOIP devices take more table entries than other Ethernet devices? I.e.
more than 1?
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Steve Edwards sedwa...@sedwards.com Voice: +1-760
= n,Dial(SIP/102,40,t)
same = n,Hangup()
Remember to set 'callcounter = yes' in sip.conf.
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Newline
,
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be a better tool.
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at the exact same time every day. The dialplan coding
is easier and no pesky codes to remember.
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,
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On Tue, 18 Nov 2014, Eric Wieling wrote:
Other than a few minor patches, we use stock phpagi.
Can you spare me a flat spot on my forehead and share the wealth?
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On Tue, 18 Nov 2014, Eric Wieling wrote:
diff at http://pastebin.com/wfDR6u0a
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.
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,
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Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
the off-list
exchanges. My apologies in advance if I've misinterpreted the situation.
I've copied the parties to the off-list exchanges in case they would care
to clarify my assumptions.
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,
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New
Asterisk does not need to care. Is it SIP all the way through?
Thanks,
Steve T
On Fri, Oct 3, 2014 at 3:12 PM, Todd R. tjrl...@live.com wrote:
OK, been messing with Asterisk for a long time and I have my opinion on
where the issues lies but sometimes it's just nice to see what others think
same = n, execif($[${DNIS:0:1} =
+]?set(DNIS=${DNIS:1}))
same = n, execif($[${DNIS:0:1} =
1]?set(DNIS=${DNIS:1}))
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a URL or from a memory buffer (either C or PHP)?
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On 09/23/2014 02:17 PM, Steve Edwards wrote:
For some applications, storing recorded audio (prompts and caller
recordings) as a BLOB in MySQL has advantages.
On Tue, 23 Sep 2014, Don Kelly wrote:
I'm curious about what the advantages are of storing audio in a blob.
Wouldn't it be more
4 digit extensions, all the better.
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On Sun, 7 Sep 2014, Steve Edwards wrote:
In specific, your ordering of '_xx' in the middle of 's' is odd. This would
disrupt the value of the priority in older versions of Asterisk, but it
appears that it does work in modern (I'm using 11) versions.
Disregard that. I can't even follow my own
,
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For future reference, a well chosen subject will yield more relevant
replies.
Better bait == better fish.
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On Mon, 25 Aug 2014, Patrick Laimbock wrote:
https://wiki.asterisk.org/wiki/displa/AST/Introduction+to+Local+Channels
s/displa/display/
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Steve Edwards sedwa...@sedwards.com Voice: +1-760
-paste into the 'dialplan show' CLI command.
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what callers are doing to reproduce the error
reliably yet.
Any clues or suggestions?
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PRI intense debug should show all you need to fix this.
On Wed, Aug 20, 2014 at 12:13 PM, Jeff LaCoursiere j...@jeff.net wrote:
Sadly none of these changes have made any difference. I'll report the
resolution for posterity once we find it.
Thanks,
j
On 08/20/2014 10:13 AM, Don Kelly
'get_lock/release_lock.'
If you're using some other database, see what locking features you have
available.
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Hi,
I am using Asterisk currently as a SIP proxy to a SIP provider. Inside my
real-time, I would set the SIP credential based on what the user has
provided.
For example
[name]
type=peer
defaultuser=USER_PROVIDED
secret=USER_PROVIDED
host=USER_PROVIDED
When I reset Asterisk, Asterisk will
Is there a way that I could set the configuration for reloading after ITSP
brute force timer expiration?
On Sun, Aug 17, 2014 at 3:51 AM, Paul Belanger paul.belan...@polybeacon.com
wrote:
On Sat, Aug 16, 2014 at 11:21 AM, Steve Ng steveng.1...@gmail.com wrote:
Hi,
I am using Asterisk
'slow' from time to time.
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connection while reading
response header from upstream, client: 116.15.31.xxx, server:
asteriskstage.xxx.yy, request: GET / HTTP/1.1, upstream:
http://127.0.0.1:8088/ws;, host: asteriskstage.xxx.yy
Anyone has any idea why?
Here's my nginx config:
http://pastebin.com/UU0G3YLh
Regards,
Steve
,
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Please don't top post.
Please keep the thread only on the list.
On Thursday, July 31, 2014 12:16 AM, Steve Edwards
asterisk@sedwards.com wrote:
I'm running Asterisk 11.11.0 on an HP ProLiant DL320e Gen8 E3-1240v2.
1,300 calls with no audio issues.
On Wed, 30 Jul 2014, babak wrote
as 729, do you have gobs of excess CPU capacity? If not, you'll
just be trading 1 resource limitation for another.
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at 460,000 blocks per second. I can remember when 10,000 was big :)
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On Tue, 22 Jul 2014, Steve Edwards wrote:
How about something like:
asterisk -rx core show channel SIP/spa841-0003\
| awk '/Call Identifer/ {gsub(/[][]/,); print $3}'
Or:
asterisk -rx core show channel SIP/spa841-0003\
| awk -F
, this is a warning, not an error.)
2) Please show a few more lines of console output (with verbose and
debug set high) to give us some context.
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Colin wrote:
Can you explain?
I'm guessing 'core show application MixMonitor' should give you a good
start.
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,${EXTEN},1)
So are the quotes now a requirement? (Quotes makes it look like you are
comparing strings instead of numbers to my old eyes.)
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. Most script-kiddies
won't bother to 'port-scan' to identify the new port number.
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Remember to always check your cables first.
Thanks,
Steve T
On Tue, Jun 24, 2014 at 1:47 PM, arun kumar arunvsadni...@gmail.com wrote:
Thank you Josh for your valuable reply. I will do try changing the server
and let you know what happening.
~Arun
On Tue, Jun 24, 2014 at 8:39 PM, Josh
Hi,
Have anyone tried using SIPML5 to connect to Asterisk over wss?
I'm having the error as shown below
Connecting to 'wss://54.xxx.xxx.xxx:8080/ws wss://54.254.228.251:8080/ws'
SIPml-api.js?svn=224:1
==stack event = starting SIPml-api.js?svn=224:1
__tsip_transport_ws_onerror
I am using Asterisk v12.3.
As far as DTLS, I understand that applying the following Javascript will
temporarily fix for SIPML5 to Asterisk:
https://gist.github.com/steve-ng/14b9b88af43f92db1e46
WS works for me, its just wss which I'm stuck currently.
On Thu, Jun 12, 2014 at 4:37 AM, Miguel
phone SIP acct passwords now, if you haven't
implemented the above precautions yet.
If I missed a previous post on this, forgive me.
Just thought you-all might appreciate a heads-up.
murf
--
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ParseTree Corporation
57 Lane 17
Cody, WY 82414
✉ murf at parsetree dot com
☎ 307
On Wed, 7 May 2014, Meriem Abid wrote:
salut,
je suis entrain de developper une application...
You will have better luck if you can post in English.
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a file system that isn't
maintaining atime/ctime/mtime/etc as expected, like NFS?
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it from happening, how about 'exten =
s,2,hangup()?'
If you're asking how to hang up the channel while it is in a loop, what
have you tried? Does 'channel request hangup' help?
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-
Steve Edwards sedwa
, Steve Edwards asterisk@sedwards.com wrote:
If you're asking how to prevent it from happening, how about 'exten =
s,2,hangup()?'
Note that you also could have added the 'missing' priority and reloaded
your dialplan and the hangup would have been executed on the next
iteration of the loop
output (with 'core agi set debug
on,' 'core set debug 99,' and 'core set verbose 99') of a successful and a
failed call.
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in the toolbox.' If C is not
in your 'wheelhouse,' use PHP or coughJava/cough. You (and the next
guy who gets to enhance and maintain this application) will be glad you
did.
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On Wed, 23 Apr 2014, Steve Edwards wrote:
I tried database access in the dialplan using the mysql() application
years ago, just to confirm I was right and I was :) What an ugly,
messy, fragile dialplan.
On Wed, 23 Apr 2014, Doug Lytle wrote:
With FuncODBC this is no longer an issue. All
,
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. I really
need to get my wtce43x card working and ISDN PRI setup on this platform.
Thanks for your assistance.
Steve
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New to Asterisk? Join us for a live
the file permissions are not set correctly on the
/dev/dahdi/channel:
Steve,
The default dahdi.rules file specifies the /dev/dahdi/ directory to be
owned by asterisk:asterisk. What's the contents of your
/etc/udev/rules.d/dahdi.rules file?
--
Russ Meyerriecks
Digium, Inc. | Linux Kernel
.
Thanks for the guidance, have a wonderful weekend!
Steve
Original Message
Subject: Re: [asterisk-users] DAHDI loading issue on Asterisk
From: Russ Meyerriecks rmeyerrie...@digium.com
Date: Fri, April 18, 2014 1:15 pm
To: Asterisk Users Mailing List - Non-Commercial Discussion
to rebuild the system with the username of
asterisk and go home!!!
Thanks again for the assist.
Steve
Oh I see, udev must have been setting the ownership to root because the
asterisk user/group doesn't exist on your system.
# DAHDI devices with ownership/permissions for running as non-root
SUBSYSTEM
Sean,
Yes, it is:
asteriskpbx@asteriskpbx:~$ lsmod | grep dahdi
dahdi 227741 2 oct612x,wcte43x
crc_ccitt 12707 1 dahdi
asteriskpbx@asteriskpbx:~$
Do you have the kernel module loaded?
lsmod | grep dahdi
sean
--
() or confbridge().
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oup = 1pickupgroup = 1immediate = noswitchtype = 5esssignalling = pri_cpecontext = incomingechocancel = yeschannel = 1-23
Any suggestions on what I am missing would be greatly appreciated.
Steve VanWambeck
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eudocrw-rw 1 root root 196, 253 Apr 15 11:30 timerroot@asteriskpbx:/dev/dahdi# cd
I compiled the dahdi package under "sudo su", perhaps that is what is wrong???
Steve
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(resend in plain text)...
Josh,
Yes, I only have one span currently connected, the other 3 are looped.
With the Asterisk process stopped I do see the OK on the dahdi_tool
screen.
I am not seeing any sort of errors in the /var/log/asterisk directory
but when I start asterisk manually with the -f
,
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On 04/09/2014 06:54 PM, Tzafrir Cohen wrote:
On Wed, Apr 09, 2014 at 10:19:59AM +0800, Steve Underwood wrote:
Hi Jeff,
On 04/08/2014 12:13 PM, Jeff Brower wrote:
Darrel- The G729 essential patents were *granted* in 1996, but
applied for prior to June 8 1995. That means their lifespan
I did this with SNOM phones and a special firmware a while ago. The trick
to get the VPN to extend to the PC port is bridge-utils. Worked very well.
On Apr 9, 2014 7:40 AM, Positively Optimistic
positivelyoptimis...@gmail.com wrote:
We are using vpn routers to connect home users back to our
,
Steve
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Wireshark.
On Fri, Apr 4, 2014 at 11:13 AM, Elliott W dig...@private-address.infowrote:
Ok, I think I am 90%+ there.
Note: the configuration or status is the same on both sides unless
otherwise noted.
I am using RSA keys for authentication and the calls are coming through as
Have you enabled IAX2 debugging and tried some test calls?
Thanks,
Steve T
On Fri, Apr 4, 2014 at 6:59 PM, Elliott W dig...@private-address.infowrote:
That answered my question as to whether it WAS encrypted, I think, and the
answer is no, the credentials are but all the rest
of 512 on an Intel E3-1240v3 with
'pre-packaged' Asterisk so I'm good for now.
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in advance,
-
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for the class.
Thanks,
Steve T
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asterisk-users
version. Currently we're connecting AMI
to a message bus and passing events across the bus.
What do you see as the advantages of a message bus (dbus?)?
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), but I'd lose some functionality and have to re-write parts of
my application.
Any clues of what limit I'm hitting and how to increase it?
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On Fri, 21 Mar 2014, Steve Edwards wrote:
The call file on box1 originates a call to box2 and then plays a 2 hour
WAV file.
The call file on box1 originates a SIP call to box2 and then plays a 2
hour WAV file.
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should be able to handle 512 conference participants on a modern server
system without problem. The current trunk of *DAHDI linux limits the number
of open pseudo channels to 512 for this reason*. [1]
Thanks,
Steve T
[1] http://svn.asterisk.org/view/dahdi?view=revisionrevision=9610
The new
On Fri, 21 Mar 2014, Steve Totaro wrote:
I found below here:
http://www.voip-info.org/wiki/view/Asterisk+cmd+MeetMe
If you have too many conferences, one CPU may not be able to mix all the
audio and you will have audio problems even if there are 7+ other CPUs
that are essentially idle
On Fri, Mar 21, 2014 at 2:26 PM, Steve Edwards asterisk@sedwards.comwrote:
On Fri, 21 Mar 2014, Steve Totaro wrote:
I found below here: http://www.voip-info.org/
wiki/view/Asterisk+cmd+MeetMe
If you have too many conferences, one CPU may not be able to mix all the
audio and you
,
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On Fri, 21 Mar 2014, Steve Edwards wrote:
Googling 'dahdi pseudo channel limit modprobe' showed the secret sauce.
Oops. Guess I should complete the thread...
You can set the DAHDI pseudo channel limit in /etc/modules.conf:
options dahdi max_pseudo_channels=x
or you can set it from
Is there any good documentation on that process?
On Fri, Mar 21, 2014 at 3:36 PM, John Novack
jnov...@stromberg-carlson.orgwrote:
Steve Edwards wrote:
On Fri, 21 Mar 2014, Adrian Serafini wrote:
Upgrade to 1.4? hehe, I thought you were the self proclaimed 1.2
luddite? I'm a big fan
On Fri, 21 Mar 2014, Steve Edwards wrote:
I'm trying to make the leap from 1.2 to 11.8.1
On Fri, 21 Mar 2014, Steve Totaro wrote:
Is there any good documentation on that process?
I haven't looked. I know they added a few of variables to the AGI
environment Asterisk passes to your AGI
Gateway computers rejects calls like this. I was informed that their
carrier rejects the calls because they cannot accurately bill.
It seems pretty silly with voip and number portability.
Thanks,
Steve T
On Mar 17, 2014 5:19 PM, Eric Wieling ewiel...@nyigc.com wrote:
Often it is P-Asserted
is top to bottom, left to right, answers
follow questions.
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, and then it dropped the call, is
that a complete FAX?
Steve
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Check here:
http://www.freepbx.org/forum/freepbx/installation/incoming-calls-drop-30-seconds-on-fpbx-2-9-0-1-8-7-0
Thanks,
Steve Totaro
On Mon, Mar 10, 2014 at 4:43 PM, alp...@gmail.com alp...@gmail.com wrote:
Yes, well, really is Elastix. Hmmm where I need to pt directmedia=no ?
Thanks
system should really wait for the final DCN message before
disconnecting, to ensure both sides have seen what they need. Spandsp
does that, but I am not sure about FFA.
Regards,
Steve
On 03/11/2014 03:03 AM, Mike Diehl wrote:
Steve,
I BELIEVE the fax is complete because the fax image is a form
participants need to have
privacy in the conference from a group of others and it all needs to be driven
from an AGI script).
Regards
Steve
The information contained in this email is intended for the personal and
confidential use
of the addressee only. It may also be privileged information
you to do what you need to do --
even if the next dialplan priority just says 'goto h.'
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hours') you are limiting your potential audience.
Posting late on a Friday afternoon can be an exercise in futility.
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,
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? I run tons of AGIs, but I write them in C.
Does 'vmstat 5' show swapping?
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