Re: [asterisk-users] Bad Gateway

[Igor Hernandez]

Hey guys,

For future reference if it happens to anyone else. In this case the
problem was that for whatever reason sending a callerid with a + in it
caused the carrier to not connect our calls.



Igor Hernandez wrote:
> Hey guys,
> 
> I've been having a very odd problem that happens intermittently. I've
> had this happen with only a couple of providers and somewhat rarely but
> its to the point now that we need to fix it to be able to do business.
> 
> The scenario is as follows: We have a DID provider that routes calls to
> our asterisk boxes and we have an outbound provider to whom we send the
> calls of the person dialing in to the DID. So person dials into the
> asterisk server from a pstn number(DID) chooses where he wants to call
> and the call is sent over the outbound provider to terminate. Usually
> this works well without a glitch.
> 
> However, with trying to terminate to specific places we get "SIP 502 Bad
> Gateway" when dialing in like the user would(through the pstn to our
> asterisk box and then dialing out) but we connect just fine by using a
> sip phone registered directly to the asterisk box.
> 
> Searching around all I could find is:
> "The server, while acting as a gateway or proxy, received an invalid
>response from the downstream server it accessed in attempting to
>fulfill the request"
> from the rfc.
> 
> Below is the current configuration for the peers. If anyone knows
> anything about why this could be happening your help is very much
> appreciated.
> 
> [DID_Provider]
> type=peer
> canreinvite=no
> directrtpsetup=no
> context=incoming
> host=DID_PROVIDER_IP
> insecure=invite,port
> qualify=no
> 
> [outbound_provider]
> type=peer
> disallow=all
> allow=G729
> context=incoming
> host=outbound_provider_ip
> insecure=invite,port
> canreinvite=no
> directrtpsetup=no
> qualify=no
> nat=no
> 
> 

-- 
Igor Hernandez
Escape Communications
http://www.escapetel.com

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[asterisk-users] Bad Gateway

[Igor Hernandez]

Hey guys,

I've been having a very odd problem that happens intermittently. I've
had this happen with only a couple of providers and somewhat rarely but
its to the point now that we need to fix it to be able to do business.

The scenario is as follows: We have a DID provider that routes calls to
our asterisk boxes and we have an outbound provider to whom we send the
calls of the person dialing in to the DID. So person dials into the
asterisk server from a pstn number(DID) chooses where he wants to call
and the call is sent over the outbound provider to terminate. Usually
this works well without a glitch.

However, with trying to terminate to specific places we get "SIP 502 Bad
Gateway" when dialing in like the user would(through the pstn to our
asterisk box and then dialing out) but we connect just fine by using a
sip phone registered directly to the asterisk box.

Searching around all I could find is:
"The server, while acting as a gateway or proxy, received an invalid
   response from the downstream server it accessed in attempting to
   fulfill the request"
from the rfc.

Below is the current configuration for the peers. If anyone knows
anything about why this could be happening your help is very much
appreciated.

[DID_Provider]
type=peer
canreinvite=no
directrtpsetup=no
context=incoming
host=DID_PROVIDER_IP
insecure=invite,port
qualify=no

[outbound_provider]
type=peer
disallow=all
allow=G729
context=incoming
host=outbound_provider_ip
insecure=invite,port
canreinvite=no
directrtpsetup=no
qualify=no
nat=no


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Re: [asterisk-users] Providing Ringback

[Igor Hernandez]

Thanks a lot Grey. I'll look into it.

Regards,

-- 
Igor Hernandez
Escape Communications
http://www.escapetel.com


Grey Man wrote:
> Hi Igor,
> 
> We had an interconnect with a carrier that generated early media for
> progress indications but the carrier's switch, in this case a Cerpack,
> would only start sending the RTP for the early media AFTER it received
> an RTP packet from the Asterisk end. Completely stupid behaviour since
> early media is generally only one way but that's what it did.
> 
> We worked around it by recording 200ms of silence and playing that
> back to the carrier's Cerpack with the Background command whenever we
> received an incoming call. This got two way RTP set up and allowed the
> progress tones to be correctly passed through to the user.
> 
> [noringback]
> exten => _X.,1,Background(/var/lib/asterisk/custom-sounds/silence_200,n)
> exten => _X.,2,Goto(incoming, ${EXTEN}, 1)
> 
> Regards,
> 
> Greyman.
> 
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[asterisk-users] Providing Ringback

[Igor Hernandez]

Hello,

We've had this problem happen twice with retail customers already and
still have no solution. Basically there are times when customers can't
get any ring at all. It happens that they call our switch and even
though we are receiving ring from the carrier they hear no ring. We have
even put a fake-ring(with Rr) back at their request and they are unable
to get this ring either.

The first time it happened was with a customer running a Cisco switch,
now more recently we have a customer with VoipSwitch that gets no ring.
Our other customers receive the ring from the carrier fine.

Has anyone experienced this before and if so how did you solve it?


Regards,

Igor Hernandez
Escape Communications.

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Re: [asterisk-users] Phishing attempt

[Igor Hernandez]

I received it also.

Roderick A. Anderson wrote:
> John Todd wrote:
>> On Nov 5, 2008, at 7:26 AM, Roderick A. Anderson wrote:
>>
>>> FYI/Heads up,
>>>
>>> I /just/ received what looks like a phishing attempt for information
>>> about Open Source PBX usage.  It says it comes from Digium but all the
>>> links (including the one for digium.com) point elsewhere.
>>>
>>>
>>> Rod
>>
>> Rod -
>>It's a "legitimate" mail from Digium.   We're trying to use the  
>> tools that others have built instead of building our own systems, so  
>> this particular company that we've chosen for research collection has  
>> their own hosts outside of the Digium DNS and server structure.   
>> Myself, I'm a big fan of Google Docs but they do have some  
>> shortcomings (like also not being in our DNS or server structure) and  
>> not everyone uses the same tools here at Digium.   "There's more than  
>> one way to do it", as goes one of the more famous Asterisk sayings.
>>
>>Sorry if that message you received seemed unusual or unexpected  
>> because of the URL format.  You can opt-out at the bottom of the  
>> emailings in the future if you wish, though some of the data that  
>> we're collecting will be tangentally useful in promoting OSS Asterisk  
>> in various forms.  And to proactively answer the next question that  
>> may come up from someone:  We've _never_ used the asterisk-* mailing  
>> list for any direct marketing email input lists.  In other words:  
>> whatever email address you've used for the asterisk-* lists is  
>> private, and Digium never uses those for marketing purposes.
> 
> Thanks for the info John.  I figured the addresses used on the lists 
> were _private_.  And if memory serves my correct when I did a download 
> a few month ago I gave my email address and an OK to contact me.  The 
> breaking point was the digium.com link.
> 
> 
> Rod

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Re: [asterisk-users] Ultramonkey LVS + asterisk

[Igor Hernandez]

Hey Ron,

Did you get your ultramonkey setup working correctly?

I'm about to roll ultramonkey here, any tips?

Regards,

Igor H.

Nhadie wrote:
> hi,
> 
> has anyone implemented ultramonkey with asterisk? do i really need to 
> setup fwmark as discussed in the url below?  thanks!
> 
> http://www.gossamer-threads.com/lists/lvs/users/20871
> 
> regards,
> ron
> 
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Re: [asterisk-users] Maybe OT - routing calls in PSTN

[Igor Hernandez]

I've had issues with DID service from other providers. My experience has
been hit or miss. Some don't want to deal with any issues, they seem to
think that just because you can run an ITSP without having any lines you
should be exempt from providing any support on the issues that do come
up with the underlying infrastructure.

Others have extremely good tech support. For example, globalpops so far
has been excellent in this department. I've had problems with one of our
DID's and in a matter of around 30 minutes they had contacted the
underlying provider and resolved the issue.

Regards,

-- 
Igor Hernandez
Escape Communications
http://www.escapetel.com

Bill Michaelson wrote:
> That is my position, and I appreciate the affirmation, as well as the
> offer to determine the carrier. I might email you about that. But having
> no business relationship with the other carrier, it is at best awkward
> for me to initiate contact on this matter, and this should be obvious to
> Vitelity staff. Worse, they are now telling me to contact the user of
> the number to ask them what provider they use. I think this is apalling.
> 
> So I'm more concerned with the practicality of relying on Vitelity for
> service in general and in the future. Their tech support has been
> absolutely cavalier to the point of insulting in refusing to deal with
> this basic issue of connectivity. I'm wondering if my experience is unique.
>> From: Alex Balashov <[EMAIL PROTECTED]>
>> It is their responsibility to contact the underlying origination
>> carrier to resolve the issue.
>>
>>
>>  
>>> I have a Vitelity DID which generally works, but calls from a
>>> particular caller do not reach it.  Vitelity has thus far disavowed
>>> any responsibility for working through this problem.
>>> 
> 
> 
> 
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Re: [asterisk-users] New User with Calling Card Question

[Igor Hernandez]

Babcock, Michael Alex wrote:
> hi;
> I'm a new member to this list and have a question for you all. I'm  
> sure it's something simple but alas i must ask. I've wanted to offer  
> calling card features to my customers. For example someone buys a  
> calling card from me for for say 1000 minutes, i give them a phone  
> number/code to call in. However i would like the same number for all  
> callers, just a new card number for different clients. I've looked on  
> google and found a few different things but want to know what you all  
> suggest. I want to get something up maybe in the next 3-5 days.
> Please email me any ideas. Oh and bandwidth isn't a major issue. It's  
> on a dedicated box with a 100mbps connection to the internet. And  
> finally i will be using completely sip. also, i currently have  
> asterisk installed can i include this calling card in a context in  
> extensions.conf for example:
> [callingcard]
> but how would i get my sip file to go to that context when someone new  
> calls in?
> thanks, sorry for the newbeish questions
> 
> 
> thanks for reading
> Systems administrator and owner of http://gwhosting.net
> msn: [EMAIL PROTECTED]
> twitter: http://twitter.com/creepyblindy
> 
> 
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Hey,

You should look into a2billing, its easy to setup, free, and has a bunch
of decent features.

You can have all the customers call into the same number and just have
your extension in the dialplan run DeadAGI(a2billing) and it'll take
care of doing the auth/keeping track of how much balance is in the card,
etc. a2billing lets you do lcr, multiple ratecards, multiple trunks, etc.

Hope it helps,

-- 
Igor Hernandez
Escape Communications
http://www.escapetel.com

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Re: [asterisk-users] test call generator

[Igor Hernandez]

Sam Tam wrote:
> Hello everyone
> 
>  
> 
> I am trying to look for a free test call generator that will get me some
> stats like PDD, ASR and call quality etc on each route. As well as do
> test at every interval too
> 
> 
> If you know something like this please enlighten me.
> 
> Sam
> 
> 
> 
> 
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Hey Sam,

I've been looking for such a tool also. I can't seem to find a tool that
does those things.

If nothing comes up in the next couple of weeks I'm going to code
something up, I wouldn't mind letting you and anyone else who might be
interested have the source once its done.

Let me know if you find anything thats already out there in the
meantime, might just save me a few hours of work.

Regards,


-- 
Igor Hernandez
Escape Communications
http://www.escapetel.com

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Re: [asterisk-users] Server Dimensioning

[Igor Hernandez]

Jon Weisman wrote:
> All,
> 
> I'm planning on getting a Dell PowerEdge 1950. We want to use our Digium 
> TE410P card, calls will come in TDM and go out VoIP, we will require to 
> compress them using G729. What specs do I need to support for 4 E-1's with 
> cdr logging to mysql? We're thinking about getting two servers 4 E-1's each, 
> is it possible to fit both cards in one machine?
> 
> Thanks,
> Jon 
> 
> 
> 
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> 
I'd be guessing but I don't think you'll manage more than 70 channels on
it.

We are running dual clovertown systems(2.33ghz) and I don't think I
would want to throw more than 190 channels on it transcoding to g729.

-- 
Igor Hernandez
Escape Communications
http://www.escapetel.com

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Re: [asterisk-users] Terrible Experience Net2phone A-Z termination

[Igor Hernandez]

Bruno Castelo Branco wrote:
> you can try inphonex.com
> 
> Steve Totaro wrote:
>> Try Bandwidth.com or Junction Networks.  You get what you pay for.
>>
>> If you want a lower end provider, go with Vitelity, Gafachi, or even
>> VoicePulse.  I am not saying they are lower end on service
>> necessarily, but on reputation and corporate image.  Vitelity tested
>> very well in a very limited time frame.  VoicePulse was great too but
>> they kept making changes that resulted in outages, if engineered
>> properly, there should be no outage short of an act of God.
>>
>> Thanks,
>> Steve Totaro
>>
>> On Thu, Sep 25, 2008 at 3:31 AM, broadband Voice
>> <[EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]>> wrote:
>>
>> I'm using Net2phone termination and the experience has been
>> horrible for the past 2 weeks, I have put in several tickets and
>> nothing has been done. I get a lot of congestion, channel
>> unavailable and calls not going through. Does anyone use them? I
>> have been using SIP debug to try to resolve it but to no avail.
>> Are there any tier A-Z termination partners out there,
>>
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>>
>> 
>>
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> 
> 
> 
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Funny Net2Phone comes up. We talked to them when we were starting out
and they wanted to charge $500 "setup" fee because we had no volume. The
guy said "We have to charge this because we had many people coming to us
without volume, so we charge this setup fee in order to allow us to
still provide them service." Like that makes any sense to anyone. Either
way, they had the worst rates in the market and claimed extremely high
quality. I'm glad we didn't go with them.

Regards,

-- 
Igor Hernandez
Escape Communications
http://www.escapetel.com

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Re: [asterisk-users] g729 capacity

[Igor Hernandez]

Steve Totaro wrote:
> You urge and help Bret with his terribly intelligent G729 license
> sharing, clearing house plan.  I think he should register a domain name
> and have a PayPal Donation link.  I would certainly donate for the
> development and even share a few licenses.
> 
> Not sure of the legal ramifications but the idea could cause a
> revolution in licensing in general, not just G729.
> 
> Thanks,
> Steve Totaro
> 
> On Wed, Sep 24, 2008 at 9:40 PM, Igor H <[EMAIL PROTECTED]
> <mailto:[EMAIL PROTECTED]>> wrote:
> 
> Hey Robert,
> 
> In my experience you get dead silence and the call goes through. We
> run 1.4, it might be different for different setups.
> 
> On Wed, Sep 24, 2008 at 9:21 PM, Robert McNaught
> <[EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]>> wrote:
> > Hi,
> >
> > Does anyone know what happens if you exceed your G729 license
> > capacity?  Lets say you have 10 of 10 licenses being used by a PBX,
> > then an 11th call comes in set up to use G729.
> >
> > Does asterisk has the ability to stop offering that codec in the SDP
> > once the capacity is reached.
> >
> > Robert
> >
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> 
> 
> 
> 
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Hey Steve,

Can you elaborate on that?

Thanks,

-- 
Igor Hernandez
Escape Communications
http://www.escapetel.com

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Re: [asterisk-users] PRI Splitter

[Igor Hernandez]

Hy Craig,

Can you elaborate on that? In our setup we have it doing just that and
it works without a glitch.

Regards,

Igor H.

Craig Guy wrote:
> The FSV-4PFS as shipped will not switch Ethernet – it switches pins 1,2,4,5.
> 
>  
> 
> Craig
> 
>  
> 
> *From:* [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] *On Behalf Of *FailSafe
> Inc.
> *Sent:* Tuesday, 2 September 2008 11:27 PM
> *To:* asterisk-users@lists.digium.com
> *Subject:* Re: [asterisk-users] PRI Splitter
> 
>  
> 
> Although the original topic of this thread has changed quite a bit, I
> wanted to point out that the "SPF" Product that you are discussing is
> quite similar to our product, the FSV-4PFS.  Ours is a 4 port device
> which can switch 4 T1/E1/J1/Ethernet or as many as 16 analog lines from
> a primary to a backup server.  It uses similar logic (power outage =
> failover server, loss of hearbeat = failover server) and also has a
> physical mechanical switch on the front of it which allows manual
> override switching to main or secondary server.
> 
>  
> 
> We also have addressed the 'clean startup' that was discussed a few
> posts back.  The switch will start and remain in 'failover mode' until
> such time as it receives a hearbeat or the physical switch is moved to
> the "main' position.  A failed main server can be restarted/repowered
> without bothering the backup server operation one bit - until you are
> ready to switch back to the main server.
> 
>  
> 
> http://www.failsafevoip.com/index.php?main_page=product_info&products_id=1
> 
> 
>  
> 
> 
> -- 
> FailSafeVOIP, Inc.
> "Safe is always better than failed"
> http://www.failsafevoip.com
> [EMAIL PROTECTED] 
> 
>  
> 
> On Tue, 2 Sep 2008 00:22:45 +0200, "Christian Victor" said:
> 
>> that when both servers power fail you have a problem no matter if the
> 
>> failover switch ist still working or not.
> 
>  
> 
> You've got that right my friend! :-)
> 
>  
> 
> On Tue, 2 Sep 2008 00:22:45 +0200, "Christian Victor" said:
> 
>> http://store.variantdistribution.com/category-s/49.htmVariant - one of
> 
>> Rhinos distributors and the only source I was able to find
> 
>> - quotes the card for US$ 700.
> 
>  
> 
> Strange.  I've seen this happen before where retailers will list
> 
> outrageously high prices for soon-to-be-released products.   For example
> 
> the SNOM KlarVoice handset.  MSRP is $32, but I've seen it advertised
> for $200!
> 
>  
> 
> http://www.8774e4voip.com/SearchResults.asp?Search=klarvoice
> 
>  
> 
> I can say with confidence that the LIST price is US $350.  The street
> price will be considerably lower.  Frankly, if I were Snom or Rhino I'd
> be pretty cheezed off about this phenomenon.  After hearing the 'buzz'
> 
> about a new product such as this, I'd hate for customers to *decide*
> against it mistkenly believing this incorrect price.  I'd turn my nose
> at either of these two products for the incorrect prices I've seen
> advertised.
> 
>  
> 
> We're pretty stoked to have stumbled onto this product because it's
> brand new, and we've been looking for something like it for some time.
> 
>  
> 
> -Karl
> 
> 
> 
> 
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Re: [asterisk-users] DID number

[Igor Hernandez]

Hey,

Did you reload asterisk after changing the extensions.conf?

Also, if you try it with "sip set debug" on the console what do you see?


michel freiha wrote:
> Hello Air,
>  
> I did what you asked for but I got the following error:
>  
> extensions.conf:
> 
> [stations]
> exten => 442033553,1,Answer
> exten => 442033553,n,Playback(demo-nogo)
>  
> Error message:
> [Sep  3 20:43:02] NOTICE[14092]: chan_sip.c:14035 handle_request_invite:
> Call from '' to extension '442033553' rejected because extension not found.
> Regards
> On Wed, Sep 3, 2008 at 11:36 PM, Igor Hernandez <[EMAIL PROTECTED]
> <mailto:[EMAIL PROTECTED]>> wrote:
> 
> michel freiha wrote:
> > Hi All,
> > I bought a DID number from VOxbone...this number could be dialed from
> > any PSTN line and could be forwarded to any SIP server like asterisk
> > server...Now I need to forward this number to my asterisk server
> so when
> > a customer dial this number from his GSM or Land line PSTN number the
> > call will be forwarde to my asterisk server and I need to play a wav
> > file for example..
> > Can you please give me some tips about how to accomplish this task?
> >
> > Regards
> >
> >
> >
> 
> >
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> 
> Hello,
> 
> I have never used that provider but usually either the provider knows
> your switch's ip and routes the did traffic to it or you have asterisk
> register with the provider so that it knows where to route the calls.
> 
> Once thats done you can do something like
> 
> exten => XX,1,Answer
> exten => XX,n,Playback(file)
> 
> Where the x's are the number that you see coming in from your provider.
> If you're routed all your dids from what looks like one
> number(callcentric does this) then you might need to use the sip header
> to route your did to the particular extension you want. You shouldn't
> have to bother with this if you only have one did.
> 
> 
> Regards,
> 
> --
> Igor Hernandez
> Escape Communications
> http://www.escapetel.com <http://www.escapetel.com/>
> 
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> 
> 
> 
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Re: [asterisk-users] DID number

[Igor Hernandez]

michel freiha wrote:
> Hi All,
> I bought a DID number from VOxbone...this number could be dialed from
> any PSTN line and could be forwarded to any SIP server like asterisk
> server...Now I need to forward this number to my asterisk server so when
> a customer dial this number from his GSM or Land line PSTN number the
> call will be forwarde to my asterisk server and I need to play a wav
> file for example..
> Can you please give me some tips about how to accomplish this task?
>  
> Regards
> 
> 
> 
> 
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Hello,

I have never used that provider but usually either the provider knows
your switch's ip and routes the did traffic to it or you have asterisk
register with the provider so that it knows where to route the calls.

Once thats done you can do something like

exten => XX,1,Answer
exten => XX,n,Playback(file)

Where the x's are the number that you see coming in from your provider.
If you're routed all your dids from what looks like one
number(callcentric does this) then you might need to use the sip header
to route your did to the particular extension you want. You shouldn't
have to bother with this if you only have one did.


Regards,

-- 
Igor Hernandez
Escape Communications
http://www.escapetel.com

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Re: [asterisk-users] PRI Splitter

[Igor Hernandez]

We recently got a fsv-4fps failover switch from failsafevoip and it
seems to be pretty solid and affordable for the ability to switch 4
trunks. It switches on power failure or death of asterisk.

The only drawback being that its an external enclosure, if you're short
of space on the rack it might be hard finding a spot for it.




Karl Fife wrote:
> On Wed, 3 Sep 2008 06:53:11 "Olivier" <[EMAIL PROTECTED]> said:
>> and power it ...
>>
>> Maybe, an external USB port could be used to power the board but the
>> "enclosure" question remains ...
>>
> 
> You're right.  At this time there's no Rhino enclosure for the Single
> Port Failover card, but it definitely designed to be powered by the
> External USB port if you mount it externally.  If I ever need to mount
> it externally, I'll get a $5 project box at radio shack if there's not
> better available by then.  
> 
> ...in response to:
>> 2008/9/1 Karl Fife <[EMAIL PROTECTED]>
>>
 So this card has interesting price position, the main drawback being,
 IMHO,
 it's eating a slot, which can be a rare resource in rackable servers.

>>> You raise a very important point.  This device uses a BRACKET, but not a
>>> motherboard SLOT.
>>> In other words, it hangs free in one of the chassis slots that do not
>>> have a corresponding slot on the motherboard.
>>> If you do not have a bracket slot, you could mount it externally, but
>>> you'd have to engineer a way to hold it.
>>
> 
> 
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Re: [asterisk-users] Gateway errors

[Igor Hernandez]

Thats strange, have you checked that you're not having issues with your
router? Can you reach all the boxes in your lan while you are
experiencing this downtime?

voip crazy wrote:
> When I say extensions, I say extensions in the lan not in wan
> 
> Thanks.
> 
> VoipCrazy.
> 
> 2008/9/1 Igor Hernandez <[EMAIL PROTECTED]>:
>> Hello,
>>
>> By people do you mean people in the lan or external users?
>>
>> Regards,
>>
>> --
>> Igor Hernandez
>> Escape Communications
>> http://www.escapetel.com
>>
>>
>> voip crazy wrote:
>>> Hello list,
>>>
>>> I have an asterisk instalation with a bad internet connection cause
>>> this connection is down sometimes.
>>> When the connection is down and asterisk cannot get internet
>>> connection. All the extensions log out from the asterisk machine, and
>>> nobody can make any call.
>>>
>>> ¿Why if internet connection is down asterisk stops working correctly?
>>> ¿How could I solve that?
>>>
>>> Thansk.
>>>
>>> VoipCrazy
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>>
>>
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Re: [asterisk-users] Gateway errors

[Igor Hernandez]

Hello,

By people do you mean people in the lan or external users?

Regards,

-- 
Igor Hernandez
Escape Communications
http://www.escapetel.com


voip crazy wrote:
> Hello list,
> 
> I have an asterisk instalation with a bad internet connection cause
> this connection is down sometimes.
> When the connection is down and asterisk cannot get internet
> connection. All the extensions log out from the asterisk machine, and
> nobody can make any call.
> 
> ¿Why if internet connection is down asterisk stops working correctly?
> ¿How could I solve that?
> 
> Thansk.
> 
> VoipCrazy
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Re: [asterisk-users] Is there a way to encrypt passwords stored in the realtime database?

[Igor Hernandez]

Hey Eric,

That I really have no experience with. Never really played with security
modules. Although someone more experienced should be able to chime in.

Eric Chamberlain wrote:
> On Aug 20, 2008, at 12:34 PM, Igor Hernandez wrote:
> 
>> Hey SIP,
>>
>> I understand what you're saying but keeping the key in memory
>> permanently doesn't protect you for very long, it just makes the
>> attacker waste a bit more time scanning the memory to get at the key.
>>
>> In other words, if the key is available to asterisk it will be  
>> available
>> to anyone else in the system with sufficient privileges.
>>
> 
> Assume I'm using a FIPS 140-2 Level 4 HSM, now, how can I protect my  
> passwords when they are in the database?
> 
> --
> Eric Chamberlain
> Founder
> RF.com
> http://RF.com/
> 
> 
> 
> 
> 
> 
> 
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-- 
Igor Hernandez
Escape Communications
http://www.escapetel.com

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Re: [asterisk-users] Is there a way to encrypt passwords stored in the realtime database?

[Igor Hernandez]

I understand the advantage of md5 hashing, its been the standard for
years for day to day user auths. What we were discussing was the merits
of the proposed public key scheme for this application, where the
private key would always need to be available therefore not giving any
real security.

Regards,

-- 
Igor Hernandez
Escape Communications
http://www.escapetel.com

BJ Weschke wrote:
> Igor Hernandez wrote:
>> I was thinking the same thing I believe Tzafrir just alluded to. If the
>> passwords are encrypted in the DB with a public key then...asterisk
>> needs to have the private key stored somewhere to be able to decrypt the
>> values to authenticate the user. In this way there is nothing preventing
>> whoever intrudes your boxes from getting that key and decrypting the
>> values himself.
>>
>> I might be missing something though and if thats the case chime in, I'm
>> interested in this issue.
>>
>> Regards,
>>
>>   
> 
>  You are. md5secret simply stores the crypt hash. When it receives the 
> password attempt, it too, is crypted using MD5 algorithm and then the 
> two hashes are compared. Using MD5 crypt hash, there is no way to 
> "decrypt" the hash. It's a "brute force" methodology to get the password 
> back if you've lost it.
> 



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Re: [asterisk-users] Is there a way to encrypt passwords stored in the realtime database?

[Igor Hernandez]

Hey SIP,

I understand what you're saying but keeping the key in memory
permanently doesn't protect you for very long, it just makes the
attacker waste a bit more time scanning the memory to get at the key.

In other words, if the key is available to asterisk it will be available
to anyone else in the system with sufficient privileges.

-- 
Igor Hernandez
Escape Communications
http://www.escapetel.com


SIP wrote:
> Igor Hernandez wrote:
>> I was thinking the same thing I believe Tzafrir just alluded to. If the
>> passwords are encrypted in the DB with a public key then...asterisk
>> needs to have the private key stored somewhere to be able to decrypt the
>> values to authenticate the user. In this way there is nothing preventing
>> whoever intrudes your boxes from getting that key and decrypting the
>> values himself.
>>
>> I might be missing something though and if thats the case chime in, I'm
>> interested in this issue.
>>
>> Regards,
>>
>>   
> Absolutely. But if you can work it so that you have to key in the key 
> manually on startup, or store it on a removable flash drive and it 
> remains in memory during runtime, then you've achieved what you need. 
> Again... this is considerable complexity in the code -- not a simple 
> dialplan hack. BUT... it would add security.
> 
> I'm just tossing out ideas here.
> 
> 
> N.
> 
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Re: [asterisk-users] Is there a way to encrypt passwords stored in the realtime database?

[Igor Hernandez]

I was thinking the same thing I believe Tzafrir just alluded to. If the
passwords are encrypted in the DB with a public key then...asterisk
needs to have the private key stored somewhere to be able to decrypt the
values to authenticate the user. In this way there is nothing preventing
whoever intrudes your boxes from getting that key and decrypting the
values himself.

I might be missing something though and if thats the case chime in, I'm
interested in this issue.

Regards,

-- 
Igor Hernandez
Escape Communications
http://www.escapetel.com

SIP wrote:
> Tzafrir Cohen wrote:
>> On Wed, Aug 20, 2008 at 10:00:55AM -0700, Eric Chamberlain wrote:
>>   
>>> We are exploring using Asterisk for a project and we are looking for a  
>>> way to encrypt/decrypt the peer passwords stored in the realtime  
>>> database (postrges).
>>>
>>> Ideally, we want to use a public key to encrypt the passwords before  
>>> they go into the database and have Asterisk use a private key to  
>>> decrypt the password as part of the call out process.
>>>
>>> Has anyone developed something like this?
>>> 
>> What is the point in that? What threats does it help you to mitigate?
>>
>>   
> It helps you mitigate an incredible amount of headache if someone hacks 
> in and gains access to your DB. The user accounts are still rather 
> secure -- at least long enough to inform your users to change their 
> passwords.
> 
> And yes... you could just say, "Don't let that happen. Use better 
> security on the system."   However, that's not 100% effective, and most 
> hacks are done by disgruntled former employees who had legitimate access 
> to the system in the first place. As long as it CAN be done without 
> drastically affecting performance and/or user experience, any extra 
> security is a Good Thing.
> 
> N.
> 
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Re: [asterisk-users] RTCP Read too short error

[Igor Hernandez]

Its not an error, its a warning. I've been wondering what it is for a
long time now also. The most I could gather is maybe something the
upstream provider is using as a keepalive. But this is just speculation.
Regardless it seems to be harmless.

Mark Hamilton wrote:
> I've asked this question myself, numerous times. But just like a few things
> that no one likes to answer to, but are aware of - this gets lost in the
> mails without any answers.
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Jon Weisman
> Sent: August 20, 2008 10:33 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] RTCP Read too short error
> 
> [Aug 20 10:31:17] WARNING[13825]: rtp.c:892 ast_rtcp_read: RTCP Read too 
> short
> 
> I've got this message scrolling like crazy on my console when I have calls 
> up. The calls are from TDM to SIP. Did a google search, but wasn't able to 
> find anything that made sense to me. Any thoughts?
> 
> 
> Thanks,
> Jon
> 
> 
> 
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-- 
Igor Hernandez
Escape Communications
http://www.escapetel.com

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