y-backing' may be slightly more efficient (in run time and
memory), ease of comprehension and maintenance are more important.
As an aside, why is strncasecmp() being used instead of strncmp()?
Wouldn't it be better to 'down-case' lang once instead of every time it is
use
when somebody says 'hey, my call didn't
connect yesterday' I have something to work with.
sngrep is a great tool for searching for calls and displaying decoded
dialogs.
--
Thanks in advance,
-----
Steve
patches to
ast 13 in the last 18 months
Steve
On Fri, 30 Nov 2018 at 09:18, Paddy Grice wrote:
> Thanks Leon
>
> I will implement and test but I knew there would be a fix for what I
> believe is a short coming in app_queue. How do I suggest this as a
Turn on PRI debugging and double check your cable.
On Mon, Nov 12, 2018 at 3:24 PM Jeff LaCoursiere
wrote:
>
> I've been struggling for a few weeks now with the local telco trying to
> bring up a trunk that has been down for a year (hurricanes in the
> caribbean). Box is a Dell R710, 16G RAM, U
80
5 8088
1 8443
1 873
1 8889
1 9124
2 9191
I hope those of you with internet accessible systems are following best
practices!
murf
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Steve Murphy
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7; commands.
--
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-----
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
https://www.linkedin.com/in/steve-edwards-4244281
--
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thing else?
A clever solution to a mobile user base is to use knockd to allow remote
access.
--
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-----
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
https://www.linkedin.com/i
On Fri, May 11, 2018, at 10:36 AM, Steve Edwards wrote:
So, Asterisk will defer it's choice of codec to match the codec it detects
in the incoming stream?
On Fri, 11 May 2018, Joshua Colp wrote:
It depends on the channel driver and configuration. The chan_sip module
always mat
anks in advance,
-----
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
https://www.linkedin.com/in/steve-edwards-4244281
--
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On Thu, May 10, 2018 at 11:44:14AM -0700, Steve Edwards wrote:
I receive an INVITE/SDP containing:
m=audio 11310 RTP/AVP 3 0 101
which I interpret as gsm, ulaw, rfc2833.
and I reply with an OK/SDP containing:
m=audio 15884 RTP/AVP 0 3 101
which I interpret as ulaw, gsm
?
--
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Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
https://www.linkedin.com/in/steve-edwards-4244281
/var/lib/asterisk/agi-bin/adddnc.php 7) Check the 'r' and 'x' bits on /var/, /var/lib/, /var/lib/asterisk/,
/var/lib/asterisk/agi-bin/.
8) cat /var/lib/asterisk/agi-bin/adddnc.php
--
Thanks in advance,
-
On Fri, 9 Feb 2018, Olivier wrote:
3. How do you capture an RTP flux with thark or tcpdump ?
Take a look at 'pcapsipdump.'
--
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-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-46
On Tue, 26 Dec 2017, Eric Wieling wrote:
Don't use an 'h' extension, use a hangup handler.
On 12/26/2017 04:43 PM, Steve Edwards wrote:
Why?
On Tue, 2 Jan 2018, Eric Wieling wrote:
From the hangup handler specification:
Hangup handlers are an
On Tue, 26 Dec 2017, Eric Wieling wrote:
Don't use an 'h' extension, use a hangup handler.
Why?
--
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-----
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,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
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;temp' directory on the same partition as
${astspooldir}/outgoing/ and then 'mv' the file to that directory.
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Steve Edwards sedwa...@sedwards.com Voice:
there are 'race condition' opportunities.
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-----
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
https://www.li
On Mon, 2 Oct 2017, Antony Stone wrote:
On Monday 02 October 2017 at 20:58:33, Steve Edwards wrote:
I recently received a GoIP-32 for a client project -- primarily outbound
calling.
How should a GoIP be configured for Asterisk?
Have you tried http://www.hybertone.com/en/solutionsClass.asp
.
How did you configure your GoIP and why?
What do your relevant sip.conf section(s) look like?
What does your dial command look like?
So far, all I've got out of it is a 503 Declined.
--
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-
Steve Ed
--
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-----
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
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Possibly the realm?
Thanks,
Steve
On Sat, Sep 2, 2017 at 3:58 AM, O. Hartmann wrote:
>
> It might sound stupid and a kind of "noobish", but I have serious trouble
> with
> registering one of my ITSP to Asterisk 13, running on a FreeBSD 12-CURRENT
> box.
>
> The f
in advance,
-
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https://www.linkedin.com/in/steve-edwards-4244281
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ough in my experience some jitter buffers
can handle it poorly.
Hope that helps,
Steve
On Tue, 29 Aug 2017 at 19:39 Mark Wiater wrote:
> Hi folks.
>
> I have a couple of questions regarding RTP.
>
> The background of my inquiry is that I have packet captures of SIP and RTP
>
x86_64 x86_64 x86_64
sedwards:~$ getconf LONG_BIT
64
--
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-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
https://www.linkedin.com/in/steve
I remember seeing something like this a long time ago. If memory serves me
correctly it was a problem at the physical layer and a couple of the PRI
cables got flipped and plugged into the wrong port. I had to change the
configs since I didn't have physical access to the box.
Thanks,
Stev
p?
--
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-----
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
https://www.linkedin.com/in/steve-edwards-4244281
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Based on the line number of that error in chan_sip.c, it looks like you're
running Asterisk 1.8 or earlier.
AFAIK, The issue you are seeing was fixed years ago, but not THAT many
years ago!
If I'm right, you should upgrade to fix that issue.
Cheers,
Steve
On Fri, 30 Jun 2017 at 13
I am also getting this, three or four times in the last month after years
of no problems.
I agree that Gmail is the likely common factor, but I would love to have
access to these bounce messages to know whether it is actually an
overly-paranoid list server!
Steve
On Mon, 12 Jun 2017 at 09:09
nks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
https://www.linkedin.com/in/steve-edwards-4244281
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On Thu, 1 Jun 2017, Pete Mundy wrote:
Heya Steve
I use the same Jeff recommended.
Eg this command would capture SIP traffic in capture files up to
100Mbytes each, with a maximum of 10 files in play and overwriting the
oldest automatically:
tcpdump -i eth0 -w rollingSIPtrace. -C
cript as
I change which hosts need monitoring.
I know... First world problems :)
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Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
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in advance,
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On Wed, 31 May 2017, Daniel Tryba wrote:
On Wed, May 31, 2017 at 01:39:25PM -0700, Steve Edwards wrote:
What bugs you about the output format?
It's been a while, but as I recollect, it included the date/timestamp in the
file name of the 'ring buffer' which meant that each ti
On Wed, 31 May 2017, Steve Edwards wrote:
I want to capture all SIP messages.
I have about 30 hosts in about 6 colos.
My first thought was dumpcap, but the output file name format bugs me.
What do you use for long term SIP capture?
A little more specificity...
I'd like the capture
On Wed, May 31, 2017 at 12:36:47PM -0700, Steve Edwards wrote:
I want to capture all SIP messages.
I have about 30 hosts in about 6 colos.
My first thought was dumpcap, but the output file name format bugs me.
What do you use for long term SIP capture?
On Wed, 31 May 2017, Daniel Tryba
I want to capture all SIP messages.
I have about 30 hosts in about 6 colos.
My first thought was dumpcap, but the output file name format bugs me.
What do you use for long term SIP capture?
--
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-
Steve
,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
https://www.linkedin.com/in/steve-edwards-4244281
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onist
member => SIP/101.2,4,Kelli
member => SIP/102.2,4,Traci
member => SIP/103.2,4,Debi
member => SIP/104.2,4,Debbie
member => SIP/105.2,4,Luci
...and so on...
:) See my problem?
Cheers,
Steve
On Thu, 11 May 2017 at 16:44 John Kiniston wrote:
> I have a real ugly queu
unless the whole thing is managed inside a single queue.
Cheers,
Steve
On Thu, 11 May 2017 at 16:36 Alexander Lopez wrote:
> If after 60 seconds you mean ’60 seconds of caller hold time’ then set up
> another queue as overflow,
>
>
>
> Set the first queue to timeout after 60
ad of blocking out agents.
Thanks,
Steve
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Check out the new Asterisk community forum at: https://community.asterisk.org/
New to Asterisk? Start her
use the 'site landline' to
confirm presence -- not their cell phone with the spoofed CID.
--
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-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
https://www.li
would that service be
called?
--
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-----
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
http
d. You could:
1) dial *..pause..* which will overcome that AFAIK.
2) Configure call.InternationalDialing.enabled="0" on the handset to stop
it.
3) Put a pattern of _[+],1,... into your dialplan.
That would be by guess anyway :)
Steve
--
___
an. As soon as I
press the first '*' my Poly sends the INVITE.
--
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-
Steve Edwards sedwa...@sedwards.com
same = n, answer()
I use an ancient Polycom IP 501 just fine.
Does 'dialplan show **@' yield any clues?
--
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-----
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vance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
https://www.linkedin.com/in/steve-edwards-4244281
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Chec
On Thu, 6 Apr 2017, Steve Edwards wrote:
You're welcome to the script at:
http://www.sedwards.com/recover-show-dialplan.php
Sorry about that...
Try:
http://www.sedwards.com/recover-show-dialplan.txt
--
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s' file for the asterisk user
that allows the single '/sbin/reboot' command. Do as I say, not as I've
done :)
--
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-
Steve Edwards sedwa...@s
' shell command and the Asterisk 'dialplan show' command
and then ran that through a PHP script that recovered enough.
You're welcome to the script at:
http://www.sedwards.com/recover-show-dialplan.php
--
Thanks in advance,
On Sat, 1 Apr 2017, Jonathan H wrote:
Any way of clearing ALL gosub stacks in dialplan?
1) rm -f /etc/asterisk/extensions.conf?
2) hangup()?
(It is April fools...)
--
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-
Steve Edwards sedwa
mats: (g729)
WriteFormat: slin
ReadFormat: g729
Why do I need to restart to get calls to actually use the new codec?
--
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-----
Steve Edwards sedwa...@sedwards.com Voice: +1-760-46
not the packetization.
How can I show the the configured and the actual (in use) packetization?
--
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-----
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
https://w
vance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
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Check out th
On Tue, Feb 7, 2017 at 4:47 PM, Steve Edwards wrote:
Now that the g729 patents have expired, how do we use g729 in
Asterisk?
Will Digium be releasing a g729 codec for 'free' use or do we
download the 'free' codec off the Internet now that we can use it
--
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-----
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https://www.linkedin.com/in/steve-edwards-4244281
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message
when it was most convenient for them. That way, they were in control and things
were done on
their terms.
On 6/02/2017, at 11:34 AM, Steve Edwards
wrote:
Love the idea. How?
On Mon, 6 Feb 2017, Matt Riddell wrote:
exten => _X.,1,Dial(SIP/0111${EXTEN}@myprovider&SIP/1
most convenient for them.
That way, they were in control and things were done on their terms.
Love the idea. How?
--
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-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
the CNAME in my DNS?
--
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-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
https://www.linkedin.com/in/steve-edwar
g
CALLER.
Yes, but the concept of 'answered' is vague if you are using analog
channels.
Visit http://www.voip-info.org and search for 'asterisk call back' for
examples of how others have approached this problem.
--
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-
Ste
in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
https://www.linkedin.com/in/steve-edwards-4244281
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DAHDI. The easiest way
would be to try compiling it.
Is 'DAHDI compiles without errors' the litmus test for acceptability?
Is the same true for Asterisk? If it compiles, it should work?
--
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-
2) What is the highest version of Asterisk I can run with kernel 2.6.26
and what would be the appropriate versions of DAHDI and libpri?
--
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-----
Steve Edwards sedwa...@sedwards.com Voice: +1-7
http://ip-address/admin/cfg.xml
To see the handset's current configuration.
Note that these phones are VERY fussy about being sent clean, compliant XML
for provisioning.
Hope that helps,
Steve
On Tue, 13 Dec 2016 at 12:37 Gopalakrishnan N
wrote:
> Thank you for the information. Actu
7;ve never had to do any path changes or anything for
asterisk on centos so I suspect it just isn't there...
Steve
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Check out the new Asterisk com
the Cisco build to the SIP build
on Cisco 7641 handsets, which have 2 similar builds, but none of the
techniques seemed to apply this time around.
Cheers,
Steve
On Sun, 4 Dec 2016 at 16:03 Gopalakrishnan N
wrote:
> Can't I upload the 3PCC firmware ? available from the Cisco website?
>
Hi,
You have to buy the 3PCC version for this to work. Once you have this, they
work very much like the Cisco SPA handsets.
I also ended up with a non-3PCC handset and it is useless, and as far as I
can tell they cannot be re-flashed.
Cheers,
Steve
On Fri, 2 Dec 2016 at 16:16 Gopalakrishnan
On Wed, 2 Nov 2016, Jerry Geis wrote:
"AOR" or Area of refuge
I have one of those. I call it my 'man cave.'
--
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-----
Steve Edwards sedwa...@sedwards.com Voice:
Rate
4487 Special Services Higher Rate
449 Premium rate
Having a correct rates table / normalising and validating your inputs
(as in FILTER) would both have potentially stopped the attack.
Steve
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e) so the 'telnet' needs to
match that IP address.
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-----
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
On Wed, 19 Oct 2016, Jerry Geis wrote:
I am playing with tcpenable... on 13.11.2
This may yield clues:
sudo netstat --all --numeric --program | grep asterisk
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-
Steve Edwards sedwa
ture-1.txt
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Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
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asswords, only allow needed
features to each class of users, etc.
--
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-----
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vance,
-
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GI protocol and lead to things breaking in weird
ways.
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Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
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I must have missed the memo, but the repos vanished on 2016-09-25 causing
all of my 'yum update's to fail.
Am I'm the only one using the repos?
--
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-----
Steve Edwards sedwa...@sedwards
SOLVED!
Many THANKS to George and Anthony! See at the very end, my comments...
On Thu, Sep 8, 2016 at 5:58 PM, Anthony Joseph Messina <
amess...@messinet.com> wrote:
> On Thursday, September 8, 2016 1:12:36 PM CDT Steve Murphy wrote:
> > Hello!
> >
> > Oh, wise on
(G.729 replaced with alaw)
direct_media=no
context=phone
rtp_timeout=120
set_var=__phoneid=12
set_var=__contacttypeid=4
set_var=__phonelineid=78
callerid="Steve Murphy" <101>
call_group=2
pickup_group=2
mailboxes=101@murftest
language=en
send_rpid=yes
send_pai=yes
OK, that complet
ekeeper (or maybe also systemd-coredump from
backports). I haven't tried any of those.
I think that should be a 'pipe,' not an exclamation mark to hand off the
core to an executable.
--
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--
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On 27/8/2016 10:48 μμ, Steve Edwards wrote:
Other alternatives involve modifying your dial plan. If you are
comfortable with that then you can consider alternatives like an AGI
that reads your text file and sets a channel variable.
On Sun, 28 Aug 2016, john
intain their own black list via a web page.
--
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Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
https://www.link
essing the '*'
key, while listening to /the other person's/ unavailable message?
So you can access your own voicemail remotely.
Steve
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New t
vance,
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detail.
Also on the 'SIP Flows' window is a 'Play Streams' button. Kind of scary
how easy this is...
--
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-
Steve Edwards sedwa...@sedwards.com Voice: +
Does a comparison of 'sip show channel' yield any clues?
3) Can you use 'sipdtmfmode()' to set a mode that works?
--
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-----
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-386
nt to Local dialplan, as you suggested
Does 'show channel' on a leg originated by a handset differ from a leg
originated by AMI?
--
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-----
Steve Edwards sedwa...@sedwards.com Voice: +1-7
On Mon, 6 Jun 2016, Frank Vanoni wrote:
On Sat, 2016-06-04 at 15:19 -0700, Steve Edwards wrote:
Using a 'goto' to exit from a gosub is a bad idea.
Why?
The purpose of a subroutine (code that is entered by a gosub and exited by
a return) is to allow the creation of easily reu
extension in the calling
context? Seems weird, but 'dialplan' is a weird language :)
--
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-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
ow the CLI output from 'dialplan show callthrough'
3) Please show the CLI output from a call matching the first test and from
a call matching the second test.
--
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-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
https:/
n source
application to execute "sudo asterisk -r -x 'reload'"?
However, if we use emacs instead of vi, we could write an 'asterisk' mode
and hook the 'reload' into the 'save-buffer' function...
--
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-
Please don't top post.
On Mon, 16 May 2016, Goke Aruna wrote:
can anyone give me a guide on any asterisk admin solution / interface
for config management, and monitoring? No database use is intended and I
prefer open source.
On May 16, 2016 20:50, "Steve Edwards"
wrote:
u be a bit more specific about what you want to accomplish?
I suspect at some point, most solutions will require a database of some
sorts.
--
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Steve Edwards sedwa...@sedwards.com Voice: +1-760-46
On Sat, 14 May 2016, Stefan Becker wrote:
On Sat, 14 May 2016, Steve Edwards wrote:
I think you need to make the outbound dial a single 'transaction'
either by using an extension pattern that includes the 0 like
'055' to dial 555-555- or eliminate the
numbering plan conflict with external
national numbering plan?
Dialing a prefix digit seems so 1970s to me.
--
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-----
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https://www.linkedin.com/in/steve-e
port 5060
This will display just the BYE messages.
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Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
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On Mon, 9 May 2016, Jonathan H wrote:
I realise mine is a bit of a niche case...
Have you looked at the externalivr() application?
--
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-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468
On Wed, 13 Apr 2016, Steve Edwards wrote:
// label
if ('[' == substr($line, 5, 1))
{
$line = str_replace(']', '[', $line);
$tokens = explode('[', $line);
t all that hot of a PHP programmer, so if anybody has some
constructive criticism, please jump it.)
--
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-----
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
https://ww
On Wed, 13 Apr 2016, Steve Edwards wrote:
Will 'dialplan save' help?
I just tried this one. It writes the dialplan, but without the application
arguements. Worthless.
Aside from just a great way to eff up your day, does 'dialplan save' have
any value?
sable
format.
--
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-----
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https://www.linkedin.com/in/steve-ed
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