> Coming from outside the network, setting up for a couple rounds of
> NATting isn't going to work well. They are not seeing it between
> phones. Others, using the polycom phones have reported echo between two
> SIP on a 4ms ping trip.
Could this be due to a purely acoustic echo within the Polyc
>> I want to detect brute-force password hacking attacks - thus if there
>> are too many failed login attempts for a SIP account I want to "lock"
>> this account.
>
>> Does somebody have any ideas how this could be implemented?
The usual method (I think) is to monitor the log files, and
detect re
> I may be over simplifying but I would have a serial number object that
> gets incremented anytime it is called and will be set to 0 at start-up.
> I would then use it to generate a UUID like this:
> MAC.serialid.64bit timedate
I suggest reviewing RFC 4122, which discusses UUID formats in some
> One of my user was asking, can he use VPN to access asterisk ?
> What does it mean ?
>
> And its possible ?
>
> How ?VPN
Yes, it's possible.
As one example: I have the OpenVPN software installed on my Asterisk
server, and on my Nokia N810 wireless Internet tablet. The tablet is
configured to u
> Hi, all. My subject line says it all: is there a WiFi SIP phone with VPN
> abilities? Failing that, a WiFi phone that runs Linux? I already know
> one phone that does meet my requirements -- the iPhone. The new software
> comes with a Cisco VPN client, and a SIP client can be had from
> third
I'm looking for a way to delay the disconnection of a call to
a SIP extension (or pad it with silence) for a few seconds, after
an incoming call to that extension hangs up.
Rationale: I have an Asterisk PBX (current 1.4.20 codebase), with
a Leadtek BVP8051S ATA hooked to an analog phone which has
> I'm preparing for a client install of * by doing a fresh one in-house.
> Unlike my earlier installation that runs asterisk as superuser, my
> current experimental box runs without such privilege. This is causing
> it to moan that it can't set TOS. I absolutely don't want to install it
> on
> I just received an email notice from FWD about $30 membership fee.
> My question: Is the email genuine? Did anybody else receive it?
>
> I'm just trying to be sure that it is real and not a scam.
> The (FWD) does not do anything to authenticate such emails (implementing
> GPG/PGP signature etc.
> In Florida some new subdivision developers have sold the
> phone/cable/internet rights to a provider. They run fiber to each house
> and then have the uplink to provider which isn't a traditional telco.
> You can't get another provider as satellite dishes are limited in
> covenants and restrictio
> SIP was written in such a way that the hashes it sends for passwords
> could, with only a trivial rewrite of the server code, be SHA1 instead
> of MD5 -- which would increase security to the level that, currently, it
> would be far more trouble than it's worth to even bother to attempt to
> crack
> BTW, can someone explain to a libart major like me (;-)) where echo
> comes on in a telephone conversation? I seem to recall it's due to the
> length of the line between the CO and the local party, but I'm not
> sure.
I'll try.
Echo occurs when part of the signal traveling in one direction
on t
> I would think that VoIP over VPN is a bad idea as UDP packets need to be
> in realtime not corrected by the TCP of the VPN.
That depends very much on the VPN in use.
OpenVPN doesn't suffer from this problem. Although it's SSL-based
(and one might think it does everything through SSL-over-TCP)
> Could someone tell me how to set which IRQ the ISDN card picks up?
It's a multi-stage process.
Each PCI slot has four interrupt pins: INTA through INTD. A
PCI card can choose to use any of these four (or even more than
one of them, as some multi-port serial cards do). Most PCI cards
use only
> Searching their "support" forum, posted today is the fact they are
> discontinuing any VM
The message saying that they are discontinuing their offering of
voicemail was posted on August 24, 2007 - two years ago. That
doesn't seem to be a new issue.
___
>> Isn't an SSL based tunnel all TCP?
> Not in the case of OpenVPN. I'm not sure about the commercial
> offerings.
Correct. My recollection is that OpenSSL uses TCP for the setup
and management of the tunnel (e.g. authentication and key
exchange) and uses UDP to carry the actual payload... e
> Great idea !
> I didn't know it could be possible to run several instances of xinetd, each
> binded to a specific IP address.
> Is this specific to xinetd or does openbsd-inetd also support this feature ?
> Anyway, I'll check this in openbsd-inetd doc myself and (hopefully) report
> my findings
> It's set to bind to 0.0.0.0, which IIRC is nothing strange.
>
> The question remains: how can a remote Asterisk server be receiving
> SIP packets that still contain the private net IP address of a client?
It sounds to me as if the client hasn't been told to use its
gateway's public IP address
> [Dec 8 18:24:48] ERROR[10571]: chan_alsa.c:693 alsa_read: Read error:
Resource temporarily unavailable
I agree, this looks like some form of conflict for the sound device.
The first thing I'd suggest doing, is trying to reproduce the
error with a command-line tool, with asterisk out of the loo
>> See if it plays back properly.
>
> Running aplay as asterisk user seems to be no problem:
>
> aster...@puppy$ aplay /usr/share/sounds/alsa/Front_Center.wav
> Playing: WAVE '/usr/share/sounds/alsa/Front_Center.wav' : Signed 16 bit
> Little Endian, Rate: 48000 Hz, mono
> aster...@puppy:~$ aplay
> this i got from syslog:
>
> puppy:~# grep pulse /var/log/syslog | tail -3
> Dec 14 20:32:45 puppy pulseaudio[25967]: main.c: Unable to contact D-Bus:
> org.freedesktop.DBus.Error.Spawn.ExecFailed: /usr/bin/dbus-launch
> terminated abnormally without any error message
> Dec 14 20:32:46 puppy puls
> I've run into a odd issue where inbound calls to the SIP client work
> fine, but outbound from the SIP client do not.
>
> The path between the client and the server is as below.
>
> N900 SIP client <-- OpenVPN --> Asterisk
>
> The version of Asterisk in question is 1.6.0.18.
>
> Any suggestio
> Anyway - is there someone out there that know the behaviour of OpenVPN in
> regards of retransmits and such? A VPN that retransmits will at some point
> hurt you if you transmit media over it, especially if you scale it up.
OpenVPN is well-behaved in that way. It uses SSL over TCP for its
"ad
> Hello All,
>
> I have installed Asterisk 1.6 with openVPN in the same machine. I have set
> up a VPN connection between 2 SIP clients and Asterisk using x-lite.
>
> The 2 clients connects to Asterisk. SIP signaling goes ok over the vpn
> tunnel.
>
> When attempting to make a call between the c
> Thank you for your reply.
>
>
> The first proposed solution has resolved the problem for a test in the local
> network. Another test is planned today later with a client in the same NAT
> and another in the public internet with a public static ip address.
>
> Do you have any advice for that ca
>> > I need to keep out all connection from 5 countries, which originate
>> > most of the Denial of Service attacks. The entries are around 9000 if
>> > used as xx.xx.0.0/16. I heard that there is a smarter way to do this
>> > by using User Tables in iptables, that will keep the speed equal to
>>
> They've got a bunch of Grandstreams that seem to be rock solid... until
> 7:00pm. At 7:00, some of the phones become unavailable, and stay down. Call
> quality is solid almost all the time. But right at 7:00, things go bad.
> Only
> some of the phone lines go down and they stay down until
> Great discussion, all of it. Thanks, people.
>
> How much power does the home asterisk box need ?
>
> I'm using Asus Eee Box (1012Ps) as Myth front ends in another project.
> About $280 with 320 Gb hard drive and 2 GB RAM. Atom 510 processor. Built
> in Wifi. Nearly silent. Runs F15 nicel
> Is about outgoing calls from multiple devices with the same username at
> aprox same time. The overwritten is for incomming calls. I want to prevent
> using the same account in multiple devices at same time. The solution with
> IP will not apply because users may be behind nat or will change ever
> I've been trying to find a solution that would allow our sip phones to
> communication with walkie talkies. Our setup is that we have sip phones
> setup in 2 locations, headquarters and dome. We can communication from
> headquarters and dome through sip phones, but within the dome we have
>
> 5. Placing ferrite cores on the phone cables.
Do either of the phone lines in question have DSL on them?
If so, a ferrite core (which will block common-mode RF
signals) probably won't help much, if at all. DSL is a
differential-mode signal, and its frequency content starts
down in the tens of
> Ouch. That isn't going to be so easy to spot, then! You would have to guess
> a bunch of likely passwords, fake up a challenge with some known nonce, and
> compare the response against those you would expect with each of the various
> possible passwords. (You've already got the Source Cod
> In our app we do not forward packet immediately. After enough packet
> received to increase rtp packetization time (ptime) the we forward the
> message over raw socket and set dscp to be 10 so that this time
> packets can escape iptable rules.
>
>>From client side the RTP stream analysis shows
> I must be missing something. If a phone sends a UDP packet to
> 192.168.1.1, how does that get routed to (arrive at) the 10.0.2.1
> interface on the Asterisk server? The only way I can imagine that
> happening is if a router in between the phone and the server has been
> told that 192.168.1.
> Here is my IP-PBX setupmy setup is : sips softphone <-> asterisk <-> xorcom
> PSTN gateway <-> pstn line to telcoi'm using xlite for windows
> when I make a phone call (sip - outgoing channel),I can hear my own voice so
> clear. it's very annoying mewhen talking a little loud... any solution?
> Setting up a group of analog lines to use for outbound emergency calls
> (911). My current dial plan and debug output shown below. It appears
> that when the SoftHangup() is executed that the line does not really
> hang up. In the case shown, I had reduced the group to a single DAHDI
> (an
>> Here's where I am baffled and I am hoping someone with intricate
>> knowledge of this implementation may be able to explain it to me. What
>> we had to do to get this working was to set the host= parameter to the
>> respective endpoint IP's of the VPN tunnel, 172.10.1.1 in my case, and
>> 172.1
> I know understand the latency due to the resending .. But if the link was
> have a good speed internet, then resending will make a big latency?
>
> Maybe this latency better than having a cutting voice?
Fundamentally, TCP's congestion-avoidance and loss-recovery logic simply
won't work well w
> I'm having the following problem when using a headset on XP
> connected to an on-board Realtek soundcard on an AsusTek M2N68-AM Plus
> motherboard:
>
> - Using any sound recorder (Windows', Audacity, XLite), the level is
> just too low when speaking at a conversational level, even with the
>
> Different brand/model, but similar as they are both el cheapo,
> entry-level headsets. I tried using them on a laptop, and I get
> marginally better microphone output, even with its volume cranked all
> the way up + automatic gain control enabled.
>
> I guess those on-board soundcards by Real
> FATAL: Error inserting dahdi (/lib/modules/2.6.26-2-686/dahdi/dahdi.ko):
> Unknown symbol in module, or unknown parameter (see dmesg)
> [25991.968325] dahdi: no symbol version for crc_ccitt_table
> [25991.968330] dahdi: Unknown symbol crc_ccitt_table
I think the message "no symbol version for cr
> I may be wrong here, but I think you can only register once. The last
> registration received will overwrite the first one. You will need to
> specify a second entry and register that one separately. This is the
> same reason you cannot register two devices to the same extension.
Yes, that's
> In the meantime, does anyone have a nice way to update a stable/stock lenny
> installation with the updated glibc as well as the latest kernel
Scary and risky, as others have noted!
There is an official "backports" release kit associated with Debian,
which contains newer versions of many packa
> I know this is an {*} list but does anyone know if simply adding the Squeeze
> repository to my sources.lst and running an 'aptitude
> upgrade/safe-upgrade/full-upgrade" will just upgrade Lenny -> Squeeze
> without me having to rebuild the system from scratch?
In my experience: you're likely t
> How about encrypt the whole hard drive?
>
> If I built a server and give to other people, there is no easy way to
> stop them reset the root password or just mount my drive to read
> everything on it. But if build an encrypt OS then it will be secure.
It will be more secure. However, you (
> I'm using iaxagent on a Droid X to connect by IAX to 1.8.3 at the
> office. 1.8.3 has sip phones. The audio is fine on the Droid X side. On
> the office side, they hear an echo of _their_ speech, not mine.
>
> The office uses sip-providers generally without any echo problem.
>
> Where do I st
> If you leave your asterisk box open to the world with passwords like
> you deserve to be hacked..
Well, without making a moral judgment, I will agree that you are *going*
to be hacked if you do this!
The O.P. seems to have made two (fairly common) mistakes:
- Used a "secret" so obvious t
> As I mentioned, I'm not inclined to mess with the secrets, too much
> hassle for users.
I'm afraid that I have to consider that attitude to be a bit like
saying "It's too much hassle for us to insist that our employees
lock their desk drawers and the front door... or wash their
hands after goi
> I'm still trying to figure that out. Our SIP usernames are seven digit
> phone numbers, so not really difficult to guess, but the passwords are 7
> char alpha-numeric strings, auto generated. We don't at present restrict
> people to their addresses, as some are dynamic.
If they're randomly
> I'm still trying to figure that out. Our SIP usernames are seven digit
> phone numbers, so not really difficult to guess, but the passwords are 7
> char alpha-numeric strings, auto generated. We don't at present restrict
> people to their addresses, as some are dynamic.
If the extension in
> That would only be true if you used random characters in your 17-character
> passphrase. In fact, English text has somewhere between 0.6 and 1.5 bits of
> randomness per letter, whereas an SHA1sum has no more than 4 bits of
> randomness per letter. Let's assume the higher number of randomness f
> Is there a database of MAC address prefixes used the common VoIP
> devices. I see the Linksys Sipura devices state with 00:0E.
>
> Does the same apply to other Linksys VoIP equipment?
The Ethernet prefixes ("OUIs") are three octets long.
Linksys / Cisco has been assigned a number of OUIs,
one
> Yes. Just the lone integrated NIC that's always been there. NO hardware
> changes. Still eth0 with the same MAC address.
Do you have any additional, "soft" network interfaces defined? For
example, have you enabled OpenVPN, and thus loaded either the
"tap" or "tun" network-interface drivers?
> I don't think it's an endpoint issue. I think the SIP packet headers get
> over-written by the tunnel (openvpn) protocol.
I'd be rather astonished if OpenVPN itself were responsible for this.
As far as I know, OpenVPN doesn't do higher-level-protocol rewriting
of any sort. It just provides the
For what it's worth... I also observed an unusually high
level of CPU activity on my small (home-system) Asterisk
installation, shortly after I switched to 1.6.2.13.
The pattern in my case was (as best as I could tell) a
CPU load of 2% to 10%, pulsing upwards every few seconds.
I couldn't resolve
> Hi
>
> I wonder if anyone has any sugestions
>
>
> I have had a TDM400 for a couple of years, and I have always had problems
> with noise on the line, so tonight I have been doing some research and have
> found that if I load the CPU dahdi_test has almost perfect results and no
> noise
>
>
> Hi ,
> I am a newbie with Asterix and not sure if Asterix is a right tool for my
> needs.
>
> Let's suppose this scenario :
> I have a telephone line in one office( all calls are paid to telephone
> operator).
> In other offices I have only internet connections.
> Is it possible to use Asteri
> Is there any way to force this? I have several user agents and I want to
> achieve
> near 100% availability for all peers. I realise that the peer will be 'woken'
> up
> at my qualify intervals, but can I actually force registration from the CLI?
For those peers which are at known, fixed, pr
> In my case, I have good incoming quality and terrible quality going out.
> That is, I can hear people perfectly well but they complain that my
> voice drops out and is garbled regardless of who places the call.
This suggests to me that you may have congestion problems in your
"upstream" traffic
> Just checking the transcoding on our Asterisk boxes and I get the
> following results.
> I have the g726, ilbc and lpc10 formats and codecs enabled in 'make
> menuselect' so I dont understand why its showing as no translation path.
> Any ideas?
Are the modules actually loaded?
Try doing a "m
> Problem with this is client needs to listen to the call recordings and my
> interface will only display .wav or .mp3 so they will moan if they have to
> wait until the next day for today's recordings
If you're up to writing a bit of shell script, and are running
on Linux, you could automate t
> Is the destination Number like Country Code +972?
>
> +972 59 xx(x) mobile - Jawall [moving to 7-digit subscriber numbers]
>
> source - http://www.wtng.info/wtng-972-il.html
>
> My SIP Proxy logs all the unauth. INVITEs and I found the a lot calls go
> to the Country code +972 xx
> Hmm the calls are made during the day (and sometimes very early in the
> morning). Right now it looks like someone actually made these calls. If
> that is the case it's somewhat comforting to know the system wasn't
> compromised. However, the $25,000 phone bill still remains. Yikes. $6.25
> per m
> Someone on this list uses the address @sedwards.com
>
> I doubt this is their actual email address as there is no MX record for
> sedwards.com and I can't find registration for their domain either.
>
> Part of my mail servers reject these emails because they cannot be
> replied to, or are li
> 1 - My SIP server (Asterisk) will have some SIP clients registered in its SIP
> registrar. Let's say 6 SIP clients. In my project I have to implement a way
> of a SIP client making a call to a number and all others 5 SIP clients ring.
> That is, the others 5 SIP clients must receive the SIP IN
> Now I have the problem for my cellphone... I need to register from almost any
> IP (at least in Europe), so I can't restrict it.
> Well, the password is NOT simple and random.
>
> Now, I tried to register the user of my cellphone using a PC, as my cellphone
> was already registered.
> And Aster
> Thanks Jeff, just to confirm, password are not sent in plain text? I
> want to safeguard against man in the middle attacks, sniffing traffic of
> clients.
That's correct.
The way it works is:
- Both the client, and Asterisk, know what the password is.
- The client sends a SIP message whi
>OK. Maybe an echo canceller won't make any difference. But why does the
>remote side _always_ hear an echo if we use a local dahdi extension,
>and _never_ when we use a local SIP extension ??
The echo that the remote called hears, might be of either electrical or
acoustic origin.
If electrical,
> I am using ODBC realtime storage with Asterisk. Currently, with no password
> set, a user can dial the voicemail number to retrieve their own voicemail,
> without needing to enter a password (without hearing the password prompt).
> However, there is still a 'mailbox' prompt played, and if a diff
> So does the Dial command go directly to the registered device or does
> it use the extension?
If you've given the Dial() command the SIP/user1 format, it will attempt
to dial directly to the SIP device/phone/endpoint you specify. If you
specify SIP/user1&SIP/user2&... it attempts to dial dire
>>> so the main question is -- how to Disallow CALLS without registering
>>> on PBX
> In fact, I'm not sure that it's actually possible to disallow [authenticated]
> calls from a peer that hasn't registered!
>
> As far as I can tell, 'registration' was never intended to be part of the
> authen
> Not sure maybe there's a better solution but I thought about using another
> peer with type=user for incoming connections.
That's what I've done for my connection to the service provider
I use (Vitelity), as they have different inbound and outbound
hosts/proxies. This works fine.
--
_
I'm not sure of the precise specifics of how Digium runs the
list, but this sort of problem has been a "known issue" with
mailing list distributions ever since SPF and similar technologies
showed up, almost a decade ago. DomainKeys and DMARC makes it more of
an issue, but the overall problem is no
> Hey all
>
> I am trying to register a PJSIP server on our office to an Asterisk 11
> chan_sip server in a datacenter.
>
> I keep getting
> WARNING[18084]: res_pjsip_outbound_authenticator_digest.c:178
> digest_create_request_with_auth_from_old: Host: 'XXX.XXX.XXX.XXX:5060':
> Unable to creat
> I looked at your network diagram. Try checking the configuration of the
> Ethernet ports on the firewall and the Asterisk box. Make sure they are
> set to auto-negotiate and not set to a fixed speed and fixed duplex.
> I have found in the past that if one end of a link is expecting auto-
> negoti
>> A good Ethernet cable-pair tester can spot such things pretty quickly.
>
> I disagree.
>
> *Certainly*, incorrect pair terminations can cause the sort of problems
> described, however I haven't yet come across a cable tester which can
> identify
> that a cable correctly connected from end t
I did not explain myself well, for this I apologize.
The files never appear on the NFS mount, only in the local drive.
Restarting Asterisk with the mount on does not fix it.
Asterisk simply ignores the mount and writes to the local drive.
But the mount is fine, I can create a dir and it appears
On 10/17/21 12:59 PM, cio-al...@playerschool.edu wrote:
I did test manually and the NFS mount works fine. I do create a
directory and it shows at the server.
I am using containers, indeed. How can it be affecting Asterisk that I
am using LXC containers?
I'm by no means an expert in containers
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