Displaying 20 results from an estimated 1000 matches similar to: "iptables for Asterisk - Any good guides out there?"
2010 Oct 28
5
being bombarded with SIP packets
Over the last two weeks, we have had at least two "incidents" where our
asterisk server got flooded (a hundred or more per second) by SIP
packets. Once from 114.31.50.10, second time from 173.212.200.146. We
became aware of the problem when bandwidth started suffering because
asterisk got very busy sending back replies or rejects (dunno which, I
didn't investigate it any further).
2003 Apr 22
2
howto
I have this configuration:
UA1 ---> FW1 ---> Asterisk ----> FW2 --> Internet --> UA2
UA has provate address (192.168.x.x)
Asterisk has public address
I want to be reach somebody at the internet.
My idea was that asterisk works as a Proxy.
Then i would have a SIP/RTP connection between UA1 and Asterisk and an
other SIP/RTP connection between Asterisk and UA2. (asterisk is
2010 Jan 10
2
app_swift 1.6.2 DTMF issue
With app_swift 1.6.2 + asterisk 1.6.1.12, I've found that if you
enter DTMF during cepstral playback, the first digit of ${SWIFT_DTMF}
is [un]set in an odd way.
for example consider:
999,1,Swift(some long message that you dont want to wait for|5000|5)
999,n,NoOp(DTMF: ${SWIFT_DTMF})
if while I am listening to the playback, i interrupt and dial:
- "12345", SWIFT_DTMF is set to
2010 Nov 13
2
asterisk 1.8 fax woes
I upgraded from a perfectly working 1.6.2 asterisk installation
(including fax via app_fax_digium) to 1.8.0 this evening.
All my custom modules (including swift <thanks darren!>) are working
fine except for fax.
When a caller connects, asterisk switches to the fax context and hangs
up the call.
i've captured with:
core set verbose 10
core set debug 10
fax set debug on
sip
2012 Jan 23
1
ConfBridge details
running Asterisk 1.8.9.0-rc2, what are the ways to interface with
ConfBridge ?
I see the CLI command 'confbridge' documented for asterisk 10, but i
dont see how to interface with confbridge on 1.8
What I'm trying to do is keep track of conferences that are used.
I tried something like the below, but not only does Confbridge not
return, but i'd need something that erases the
2012 Sep 20
6
accept email and make phone call?
Any ideas on how asterisk could accept an email (such as an email to SMS or "number at mybox.org" sort of thing) and make a phone
call to a specific number and make an announcement?
I imagine the first part is the big question.
joe a.
2011 Apr 15
3
sip error logging
I recently noticed that asterisk is not logging unknown sip connections.
I'm not sure if I've broken something or if asterisk itself has been
broken.
the last entry I have is:
/var/log/asterisk/messages.9:[Feb 10 23:58:48] NOTICE[9868] chan_sip.c:
Registration from '<sip:22942 at 10.0.0.3>' failed for '10.0.0.228:5060' -
No matching peer found
my logger.conf
2012 Jul 18
4
asterisk 1.8 on Solaris/sparc
I've got the latest asterisk 1.8 running on a Netra X1 with Solaris 10 u10.
The system itself is happy and phone calls (between two parties) seem fine.
Unfortunately, when a caller listens to a Playback recording, there
seems to be moments of stutter - perhaps 1 second of stutter for every
10 seconds of Playback. The stutter is not consistent at the same point
of the playback file.
To
2010 Aug 29
1
evil disconnect of call with cisco 1760
I have asterisk 1.6.2.11 talking to a cisco 1760 running
ipvoicek9-mz.124-25b.
whenever a call goes through the 1760's FXO or FXS (in or out) there is
a 915 second maximum call time due to asterisk hanging up the call
because of a "critical packet" being missed.
I read doc/sip-retransmit.txt and I don't see anything there that is
helpful to my situation - the asterisk box is
2010 Nov 15
2
SIP calls destroyed after 1:20
After upgrading to Asterisk 1.8.0, I am finding that my outbound SIP
calls are being destroyed after 1 minute and 20 seconds (80 seconds).
Asterisk is sending a BYE message - I have no idea why.
http://jeremy.kister.net/tmp/20101115/sip.txt for a sip debug.
can anyone suggest how i can further deal with this?
--
Jeremy Kister
http://jeremy.kister.net./
2011 Mar 29
1
wrong from URI in options message
I recently configured a SIP peer which i must specify my fromuser as my
ten digit "DID". I send calls to this peer, but whenever Asterisk sends
an options message, the fromuser is "asterisk".
Is this a bug? Or is there some other config I must make ?
register = 2155551941:123456 at 10.0.138.226/2155551941~600
[peer](!)
type=peer
context=inbound
qualify=yes
2011 Jun 06
2
issues.asterisk.org
i'm trying to review issues that i'm monitoring and/or have reported at
http://issues.asterisk.org
when I click on 'My View' or 'View Issues' I get an error:
APPLICATION ERROR #401
Database query failed. Error received from database was #1142: DELETE
command denied to user 'mantisreadonly'@'localhost' for table
'mantis_tokens_table' for the
2013 Sep 10
3
Asterisk 1.8 drop calls after 15 minutes
Hi all,
I face the subject strange behavior: calls arre dropped after 15 minutes
on an asterisk 1.8.15.0. Only phones (SNOM300) connected to the Asterisk
through OpenVPN seems to have the problem.
From CDR, I see for 3 calls from this morning I'm aware of, that
asterisk hangup after respectively 899s 894s 898s
In logs I see
WARNING[8213] chan_sip.c: Retransmission timeout reached on
2013 Oct 10
1
asterisk 11.6 nat problem
using asterisk 11.6.0-rc1 i just converted my "nat=yes" to
"nat=auto_force_rport,auto_comedia"
I have my asterisk box on the same subnet as a cisco 1760 (vgw1).
a few times per day, Asterisk thinks vgw1 is dead (by qualify/options).
A 'sip reload' always fixes the problem.
i left 'sip set debug peer vgw1' on the console. but i dont see what's
2013 Feb 12
1
asterisk 11 AGI
I recently upgraded to asterisk 11 from 1.8.
I had VXML working via AGI in 1.8 - from extensions.conf:
[VXML]
exten => s,1,Answer
exten => s,n,Set(ENCODED=${URIENCODE(${ARG1})})
exten => s,n,AGI(agi://localhost/url=${ENCODED})
exten => s,n,Hangup
Using asterisk 11 on the same host with the same config in extensions.conf:
-- Executing [s at VXML:1]
2016 Apr 13
5
recreating extensions.conf from live dialplan ?
with the slip of a finger, i destroyed by extensions.conf (grep -i >
extensions.conf)
I have a backup that is dozens of hours of code old.
is there a way i can use the asterisk cli (or some other asterisky
method) to recreate that extensions.conf ?
2011 Jan 26
6
Asterisk 1.8.2.3 Now Available
The Asterisk Development Team has announced the release of Asterisk 1.8.2.3.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.8.2.3 resolves the following issue:
* Reimplemented fax session reservation to reverse the ABI breakage introduced
in r297486.
(Reported by Jeremy Kister on the asterisk-users
2011 Jan 26
6
Asterisk 1.8.2.3 Now Available
The Asterisk Development Team has announced the release of Asterisk 1.8.2.3.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.8.2.3 resolves the following issue:
* Reimplemented fax session reservation to reverse the ABI breakage introduced
in r297486.
(Reported by Jeremy Kister on the asterisk-users
2009 Dec 29
1
identifying channel for softhangup
When I place an outbound call from asterisk 1.6.1.12 to a FXO port on
my Cisco 1760V 12.4, the channel changes - seemingly incrementing:
e.g., in the first call, below, the channel name is
"SIP/vgw1-00000075" -- the second call (on the same FXO port after a
soft hangup on the CLI) is "SIP/vgw1-00000077"
How can I extract this information in the dialplan so that I can use
2011 May 13
1
asterisk 1.8 + google voice
somewhere along the way, i noticed incoming calls from google voice are
no longer working on my asterisk 1.8.3.2 system.
When the call comes in, asterisk immediately prints on the console:
== Spawn extension (google-in, s, 2) exited non-zero on
'Gtalk/+12153930924-f947'
[May 12 22:47:18] NOTICE[13835]: chan_gtalk.c:1977 gtalk_parser: Remote
peer reported an error, trying to