Re: [asterisk-users] Any thoughts on Asterisk 16.17.0 outputting FRACK refcount related messages

[Telium Technical Support]

I don't *think* it would purely volume related.  We have 16.17 deployments
with very large loads running without issue, and we also run 16.17 against
load simulators without issue.

 

In each case you have to traceback to find the cause of the problem.  For
example, a bad SBC which does not fully adhere to the SIP protocol could be
confusing PJSIP, causing timeouts, etc.  Our engineers spent a few days
tracing such a problem before we shipped a high volume system to a customer
earlier this year.  (I have NOT traced through your logs below, so I'm not
saying that is your problem)

 

There are some helpful posts here to help you trace, in the archives of this
list.  (In fact I see one responding to a question you asked 2 months ago).
Did that yield nothing?  (The response was dead on in terms of how to
diagnose).  If you share the result of that diagnosis you might get some
helpful answers.

 

Dave

 

 

From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On
Behalf Of Dan Cropp
Sent: Thursday, September 23, 2021 12:59 PM
To: 'asterisk-users@lists.digium.com' 
Subject: [asterisk-users] Any thoughts on Asterisk 16.17.0 outputting FRACK
refcount related messages

 

We have an extremely busy/large customer.  They run fine most of the time,
but periodically asterisk will output FRACK refcount related messages.  It
doesn't seem to be related to the volume, because it's not breaking during
their peak times.

 

When this happens, the system becomes unstable and they have to restart to
get things resolved.

To give an idea of the instability, we have seen INVITE/Trying responses in
SIP messaging logs.

We tell Asterisk to answer via AMI, but Asterisk never sends the OK (even 24
seconds later it hasn't sent).

Eventually the other send CANCEL of the call.

 

 

We've now captured 4 different days where something like the following
occurs.

1) Is there a good way to tell if this may be fixed in Asterisk 16.20.0
(short of upgrading)?

2) Would this be something I should submit as an asterisk issue?
Unfortunately, site is so busy capturing the debug will be very difficult
(if not impossible) due to amount of data.

 

 

[09/23 14:43:45.095] ERROR[34763][C-1d7f] frame.c: Excessive refcount
10 reached on ao2 object 0x5637ed3742b8

[09/23 14:43:45.095] ERROR[34763][C-1d7f] frame.c: FRACK!, Failed
assertion Excessive refcount 10 reached on ao2 object 0x5637ed3742b8 (0)

[09/23 14:43:45.095] ERROR[29830] frame.c: Excessive refcount 10 reached
on ao2 object 0x5637ed3742b8

[09/23 14:43:45.095] ERROR[29832][C-1920] frame.c: Excessive refcount
10 reached on ao2 object 0x5637ed3742b8

[09/23 14:43:45.095] ERROR[29830] frame.c: FRACK!, Failed assertion
Excessive refcount 10 reached on ao2 object 0x5637ed3742b8 (0)

[09/23 14:43:45.095] ERROR[29832][C-1920] frame.c: FRACK!, Failed
assertion Excessive refcount 10 reached on ao2 object 0x5637ed3742b8 (0)

[09/23 14:43:45.095] ERROR[32973] frame.c: Excessive refcount 10 reached
on ao2 object 0x5637ed3742b8

[09/23 14:43:45.095] ERROR[32973] frame.c: FRACK!, Failed assertion
Excessive refcount 10 reached on ao2 object 0x5637ed3742b8 (0)

[09/23 14:43:45.095] ERROR[3248] frame.c: Excessive refcount 10 reached
on ao2 object 0x5637ed3742b8

[09/23 14:43:45.095] ERROR[3248] frame.c: FRACK!, Failed assertion Excessive
refcount 10 reached on ao2 object 0x5637ed3742b8 (0)

[09/23 14:43:45.095] WARNING[2798][C-0093] channel.c: Exceptionally long
queue length queuing to
CBAnn/IS__8b9c6719-ca29-4c1b-ac87-75e8c6fe7074-0062;1

[09/23 14:43:45.095] ERROR[12979] frame.c: Excessive refcount 10 reached
on ao2 object 0x5637ed3742b8

[09/23 14:43:45.095] ERROR[12979] frame.c: FRACK!, Failed assertion
Excessive refcount 10 reached on ao2 object 0x5637ed3742b8 (0)

[09/23 14:43:45.095] WARNING[21123][C-1157] channel.c: Exceptionally
long queue length queuing to
CBAnn/IS__a652155f-b1fb-4c31-83e5-09ffa2107979-10de;1

[09/23 14:43:45.096] ERROR[29830] : Got 8 backtrace records

# 0: /usr/sbin/asterisk(__ao2_ref+0x209) [0x5637ebef1519]

# 1: /usr/sbin/asterisk(ast_frdup+0x1e2) [0x5637ebf96612]

# 2: /usr/sbin/asterisk(ast_bridge_channel_queue_frame+0x61)
[0x5637ebf1cfe1]

# 3: /usr/lib/asterisk/modules/bridge_softmix.so(+0x40af) [0x7fc15362f0af]

# 4: /usr/lib/asterisk/modules/bridge_softmix.so(+0x560a) [0x7fc15363060a]

# 5: /usr/sbin/asterisk(+0x1db41f) [0x5637ec06041f]

# 6: /lib/x86_64-linux-gnu/libpthread.so.0(+0x76db) [0x7fc1e9ebf6db]

# 7: /lib/x86_64-linux-gnu/libc.so.6(clone+0x3f) [0x7fc1e93f971f]

 

[09/23 14:43:45.097] WARNING[36475][C-0172] channel.c: Exceptionally
long voice queue length queuing to
CBAnn/IS__64bc075a-1ba4-4ad8-ba48-f0aea6ca6bab-1ed3;1

[09/23 14:43:45.098] ERROR[12979] : Got 8 backtrace records

# 0: /usr/sbin/asterisk(__ao2_ref+0x209) [0x5637ebef1519]

# 1: /usr/sbin/asterisk(ast_frdup+0x1e2) [0x5637ebf96612]

# 2: 

Re: [asterisk-users] Asterisk bring in RTP audio

[Telium Technical Support]

Turn you 16 RTP port device into a SIP UA.  Use one of the open source SIP 
phones as starting point, setup as autoanswer, and start streaming the RTP.  

High level answer for high level question…but that should point you In the 
right direction

 

From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On Behalf 
Of Jerry Geis
Sent: Sunday, November 7, 2021 8:29 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Subject: [asterisk-users] Asterisk bring in RTP audio

 

Hi -

 

I have a device that has 16 RTP ports.  I desire to bring that audio into 
Asterisk... is that possible ?

The device does not run SIP at all just RTP audio. I am using Asterisk 18.

How might I do that ?

 

Thanks,

 

Jerry

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Re: [asterisk-users] recording not working to NFS

[Telium Technical Support]

Just adding my 2c

I don't think permissions which cause one process to see the mounted file 
system and another to see the directory underneath.  I think using automount 
could cause this but there is still some other factor contributing to the 
problem.

-Original Message-
From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On Behalf 
Of Dave Platt
Sent: Saturday, October 16, 2021 1:35 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] recording not working to NFS


> 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 the other 
> side, so NFS is fine.
> Any idea?

That's a bit bizarre.  I had first though that this might be a problem if you 
were to start Asterisk before mounting the share... Asterisk might have opened 
the message directory when it started, and then doing directory-relative file 
creation and moves.  But, you say that restarting Asterisk doesn't change the 
behavior.

On your system, are you using containers, or namespaces, or etc.?  You might be 
accidentally setting up an environment in which the NFS mount isn't being 
"seen" by the environment in which Asterisk is running.

It might also be worth checking if you can manually create files in the shared 
location when running as the same user-ID/group-ID as Asterisk is configured to 
use.  You might be seeing some sort of odd permissions-based problem.



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Re: [asterisk-users] recording not working to NFS

[Telium Technical Support]

If Asterisk is writing files into the local directory that is the mount
point for a remote NFS connection, then this is not an asterisk problem.
It's a local config/network issue.

No application should be able to write to the local disk dir used as a mount
point .  So if that is what's happening, your NFS mount is not active.  Are
you using automount?

You need to dig deeper into NFS mount...it's not working the way you think
it is.

-Original Message-
From: cio-al...@playerschool.edu [mailto:cio-al...@playerschool.edu] 
Sent: Friday, October 15, 2021 4:18 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion

Cc: Telium Technical Support 
Subject: Re: [asterisk-users] recording not working to NFS

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 the other side,
so NFS is fine.
Any idea?


On 2021-10-13 12:04, Telium Technical Support wrote:


> If unmounting makes your files appear on the NFS mount, then there may
> be some caching going on, or files not being closed (by Asterisk).
> Unmounting will force files to close and could make them appear.
> 
> Try restarting Asterisk (with NFS still mounted).  Do the files then 
> appear?
> 
> -Original Message-
> From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com]
> On Behalf Of cio-al...@playerschool.edu
> Sent: Wednesday, October 13, 2021 1:37 PM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] recording not working to NFS
> 
> I have an NFS mount and I am trying to record to it. The mount works
> fine, I create a directory and it shows on the server, I delete it and
> it gets deleted at the server, but Asterisk 16-latest is always
> recording to the local drive, it ignores the NFS mount.
> Once I unmount the directory, the recordings show up in the drive.
> Is this by design?
> 
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> https://community.asterisk.org/
> 
> New to Asterisk? Start here:
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> 
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Re: [asterisk-users] recording not working to NFS

[Telium Technical Support]

If unmounting makes your files appear on the NFS mount, then there may be some 
caching going on, or files not being closed (by Asterisk).  Unmounting will 
force files to close and could make them appear.

Try restarting Asterisk (with NFS still mounted).  Do the files then appear?  

-Original Message-
From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On Behalf 
Of cio-al...@playerschool.edu
Sent: Wednesday, October 13, 2021 1:37 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] recording not working to NFS

I have an NFS mount and I am trying to record to it. The mount works fine, I 
create a directory and it shows on the server, I delete it and it gets deleted 
at the server, but Asterisk 16-latest is always recording to the local drive, 
it ignores the NFS mount.
Once I unmount the directory, the recordings show up in the drive.
Is this by design?

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Re: [asterisk-users] SIP Source Port

[Telium Technical Support]

I don’t think I’ve seen that requirement before, so someone else may have to 
answer if there is a PJSIP specific setting

 

However, if not then it may be simple to achieve the same result by using your 
firewall NAT rules.

 

From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On Behalf 
Of Alexander Perkins
Sent: Saturday, July 10, 2021 1:39 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] SIP Source Port

 

Hi All.  We have a provider that requires us to SOURCE the SIP connection on 
TCP 5061.  I honestly have no clue how to force Asterisk to always SOURCE the 
SIP connection on a certain port.  

 

Can anybody point me in the right direction?  I am using PJSIP.

 

Thank you,

Alex

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Re: [asterisk-users] Hook Flash

[Telium Technical Support]

Since this function is handled by the ATA, you would have to look there (or 
post details) for something ATA specific.  In general I don’t think so, hook 
flash just puts one channel on hold a creates/answers another.  But, you may be 
able to script the functionality you need it in the Ast dialplan.

 

From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On Behalf 
Of Dovid Bender
Sent: Friday, June 25, 2021 3:26 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Subject: [asterisk-users] Hook Flash

 

Hi,

 

It's been a very long time since I dealt with a along lines. Does anyone know 
if there is a way to "pass though" a hook flash? I am working on a project 
where there will be one FXS and one FXO. I want if there is call waiting for 
the phone connected to the FXS to be able to hit the hook and have that sent 
back out on the FXO port.

 

TIA

 

 

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Re: [asterisk-users] Server loses sip registrations after converting to vm to mysql storage.

[Telium Technical Support]

How about starting a console with verbose turned up.  After a loss of
registrations review the console output to see if there is some event.

 

From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On
Behalf Of Mike Diehl
Sent: Tuesday, April 20, 2021 3:54 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion

Subject: [asterisk-users] Server loses sip registrations after converting to
vm to mysql storage.

 

Hi all,

 

I've got an old server (Asterisk 13.28.0) that I'm trying to configure to
store voicemail in a mysql database. 

 

I have sip realtime working via odbc and it's been working well for years.

 

However, when I recompile Asterisk in order to store voicemail in the
database, I have problems. (That is the ONLY thing I change.)

 

The server seems to run for a while and voicemail seems to work. Then, the
server loses ALL of it's sip registrations. I have a script that I can run
to reload the registrations, but the server eventually loses them again.

 

Any ideas as to where I should start looking?

 

Thanks in advance,

 

-- 

Mike Diehl

 

 

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Re: [asterisk-users] STIR/SHAKEN

[Telium Technical Support]

If you operate a small PBX for a business your approach is fine.

 

If you operate a large PBX, or just have lots of high toll rate calls, the 
price difference between carriers can add up to a lot money every day.  These 
operators will route their calls to whomever offers the best rate for that 
route.  

 

And that’s the problem being solved.  STIR/SHAKEN makes it tough for spoofers, 
but also tough for businesses doing LCR.  Sadly, the easier it becomes to 
implement STIR/SHAKEN (telling the next hop along the route to trust your 
identity), the easier it will be for spoofers to do the same.  I suspect it 
won’t be long until unscrupulous service providers undermine STIR/SHAKEN 

 

From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On Behalf 
Of Sebastian Nielsen
Sent: Thursday, March 11, 2021 7:34 PM
To: 'Mailing List' 
Subject: Re: [asterisk-users] STIR/SHAKEN

 

I reallt don’t understand why people simply use the same operator to terminate 
your calls, which also provide DIDs for you.

 

Then you don’t need to touch this at all, your carrier will do all the 
STIR/SHAKEN handling for you, you are just a PBX customer.

And then the operator then simply limits your account to only present your DID 
as outgoing number.

 

Seems to be a unneccesary complicated solution just to have your numbers at 
company 1 and have your call termination at company 2.

So fricking unneccessary.

 

What I know there is requirements of number portability, so as long as company 
2 can handle DIDs (ergo ”own” DIDs) you should be able to move your DIDs from 
company 1 to company 2 – then company 2 owns your DIDs.

 

Best regards, Sebastian Nielsen

 

Från: asterisk-users-boun...@lists.digium.com 
  
mailto:asterisk-users-boun...@lists.digium.com> > För Alexander Perkins
Skickat: den 12 mars 2021 01:23
Till: asterisk-users@lists.digium.com  
Ämne: Re: [asterisk-users] STIR/SHAKEN

 

Hi Jeff.  What exactly do you mean by the 'inbound piece'?  I've spent quite a 
lot of time with the folks at TILTX understanding the framework; but I am not 
exactly sure what you mean by the 'inbound piece.

 

Greg/Doug, like many folks here, we use LCR.  So, the terminating carrier is 
not necessarily the one that issued us the telephone numbers.  So, they will 
not sign it or simply cannot sign it.  Remember that a very limited number of 
companies can actually sign the calls; the rest have to buy it from these 
'Service Providers'.  

 

And there is another situation - the company you purchase your numbers from and 
the company you place your calls through may be different and both may not be 
able to sign your calls.  Again, a very limited number of service providers 
that can actually sign your calls.  So what do you do in that scenario?  You 
have to find a Service Provider that can:

 

1.  Verify you own that telephone number(s).

2.  Sign your calls.

3.  Provide you with the technical means to do so.

 

So, that's that...  I hope this makes sense.  

 

Alex

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Re: [asterisk-users] Failed to authenticate device message

[Telium Technical Support]

You didn’t post the Asterisk version, but if this is an OLD asterisk version 
then the source IP may be missing from messages/logs.

 

If you have low traffic in general then using something like Wireshark may help 
you examine any suspicious SIP packet on the PBX.  For higher volumes it’s like 
drinking from a fire hydrant, so not suitable.

 

If this is a small PBX, have a look at the SecAst product 
(https://teium.io/secast).  It’s free for small installations.  It’s an 
Asterisk security product that monitors network traffic at a the adapter level 
so it can sniff the source.  It also talks to Asterisk through the AMI so it 
can get more details of the connection/session that way.  If this is for a 
larger PBX then you would have to move the discussion to the biz list for more 
info on SecAst.  (Or email me off list)

 

 

From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On Behalf 
Of Jerry Geis
Sent: Wednesday, July 22, 2020 11:37 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Subject: Re: [asterisk-users] Failed to authenticate device message

 

>Did you check your security log?
 
>There is usually a wealth of info there about who, what, where when and why
 
I also checked /var/log/asterisk/messages and it just has the same line. 
Nothing additional.
 
Jerry
 
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Re: [asterisk-users] Stir Shaken

[Telium Technical Support]

This sounds like the kind of business I can trust with my calls, and am eager 
to buy from.  

 

Oozing with professionalism.  Well done sir!

 

:)

 

From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On Behalf 
Of d...@donkelly.biz
Sent: Tuesday, July 14, 2020 4:48 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion' 

Subject: Re: [asterisk-users] Stir Shaken

 

 

 

From: asterisk-users mailto:asterisk-users-boun...@lists.digium.com> > On Behalf Of Saint Michael
Sent: Tuesday, July 14, 2020 2:35 PM
To: asterisk-users@lists.digium.com  
Subject: [asterisk-users] Stir Shaken

 

I need to point out the this is factually misleading and materially false:

"I think this, being the basis of your whole argument, is the fallacy. 

S/S is forcing people to take responsibility, for sure, but carriers
won't just let their customers leave because they don't want to sign
calls.  It will force them to make sure they know who their customers
are, and make it impossible for those customers to escape consequences if they 
misbehave."

 

There is Law of The Land that is about to take effect. Use google and search 
"stir shaken" Whoever thinks I am exaggerated is dreaming. Also: it is true 
that my service is the only one for asterisk --worldwide. The model proposed by 
Transexus (302 redirect with a new header) can't be used by Asterisk. 

But don't take my word for it:

https://issues.asterisk.org/jira/browse/ASTERISK-28924 

 

 

 

I need to point out again that this is not the forum for your business 
proposition. Please take it to the business list.

 

  --Don

 

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Re: [asterisk-users] Voice broken during calls (again...)

[Telium Technical Support]

Still lots of detail missing, butlikely causes include:
1.  Egress latency (does your router/firewall support QoS, are you leaving 
headroom )
2. Ingress latency - does your ITSP support it
3. Router/firewall latency - can it keep up with the traffic and packet size.  
Do you have way too many iptables rules in your Debian box?

Between ping and traceroute you can probably get some basic stats.  Some speed 
test websites even report latency, other sites will should tracert/ping from 
outside in to you.

How about putting a phone on the DSL/cable modem directly and calling 
out...same problem?

-Original Message-
From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On Behalf 
Of Luca Bertoncello
Sent: Monday, June 22, 2020 11:19 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Subject: Re: [asterisk-users] Voice broken during calls (again...)

Am 22.06.2020 um 17:01 schrieb Telium Technical Support:
> I don't know if there was a prior email with more details, but
> 
> Latency is as important as speed.  Have you checked latency between your 
> device and pop?  What about QoS at your location, and does your ITSP 
> support/respect QoS?

That's a very good idea...
Could you suggest me how can I check it?
The Gateway is a Linux with Debian 9.

> Could problem be inside your network?  Have you tested/optimized internal?

Really difficult to believe... If I call another VoIP-phone in my network 
(using the "internal number") the quality is excellent.

If I call my wife using the "external number", the quality is very bad...

Thanks
Luca Bertoncello
(lucab...@lucabert.de)

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Re: [asterisk-users] Voice broken during calls (again...)

[Telium Technical Support]

I don't know if there was a prior email with more details, but

Latency is as important as speed.  Have you checked latency between your device 
and pop?  What about QoS at your location, and does your ITSP support/respect 
QoS?

Could problem be inside your network?  Have you tested/optimized internal?

-Original Message-
From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On Behalf 
Of Luca Bertoncello
Sent: Monday, June 22, 2020 10:49 AM
To: Asterisk Users 
Subject: [asterisk-users] Voice broken during calls (again...)

Hi list!

So, now I have a business contract and a technician was here to check the DSL...
Nothing found, except that for 50Mbps I need now vectoring. Really nice... A 
couple of years ago I could get 50Mbps without vectoring.
Of course, Deutsche Telekom said nothing about this change...

Well, I got it working, and now I have 48Mbps down and 10Mbps up.
I _REALLY CAN'T_ believe, that this is not enough...

The problem with many little disruptions during calls is always here.

I tried changing the codecs and changing some settings in the SIP configuration 
of the peers.
No changes...

On the Gateway (Banana PI), where the Asterisk server also runs, the load is 
about 0.50 during calls and it has a Gbps LAN.
I can't believe, the problem is here...

@all german users using Telekom: how did you configured your Asterisk?
@all: thank you for all your suggestion, I really don't know anymore what I can 
do...

Thanks
Luca Bertoncello
(lucab...@lucabert.de)

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Re: [asterisk-users] Any api (agi/ari/ami) equivalent of "core show calls"?

[Telium Technical Support]

Just run ‘core show calls’ as a command  from the AMI, and parse the results.  
I don’t think there is an equivalent pure AMI command.

 

From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On Behalf 
Of Jonathan H
Sent: Sunday, June 14, 2020 5:45 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Subject: Re: [asterisk-users] Any api (agi/ari/ami) equivalent of "core show 
calls"?

 

Wow! I've been *-ing for about 6 years and had literally no idea about that! 

 

I can see a way I could put it to a different use, but it seems to be a bit of 
a sledgehammer to crack the walnut of "how many current callers" compared to 
one line of (albeit hacky) dialplan. 

 

That's making me sound ungrateful. I don't mean to be!

 

On Sun, 14 Jun 2020, 22:39 Steve Edwards, mailto:asterisk@sedwards.com> > wrote:

On Sun, 14 Jun 2020, Jonathan H wrote:

> Thank you... but "just update the database" - hmm, what database?

I used MySQL.

> Did you mean ARI? I still can't find the command! The asterisk wiki is 
> somewhat, um... spread around!

ARA as in Asterisk RealTime Architecture

https://wiki.asterisk.org/wiki/display/AST/Realtime+Database+Configuration
https://www.voip-info.org/asterisk-realtime/

As I recall (back from 2015), you tell Asterisk which 'configuration file' 
you want to read from MySQL like this:

# /etc/asterisk/extconfig.conf

[settings]
 musiconhold.conf= mysql,vchat,static
;   musiconhold.conf= mysql,vchat,musiconhold

I have no idea if this will help, but here are the tables as I defined them 
back in 2015.

 create  table   if not exists   static
 (
   idint(11) not null auto_increment
 , cat_metricint(11) not null default '0'
 , var_metricint(11) not null default '0'
 , commented int(11) not null default '0'
 , filename  varchar(128) not null default ''
 , category  varchar(128) not null default 'default'
 , var_name  varchar(128) not null default ''
 , var_val   varchar(128) not null default ''
 , primary key   (id)
 )
 ;

-- defaults
 set @CAT_METRIC = 0;
 set @FILENAME   = 'musiconhold.conf';
 set @VAR_METRIC = 0;

-- Funk Dance
 set @COMMENTED  = 0;
 set @NAME   = 'Funk Dance';
 insert into static set
   cat_metric= @CAT_METRIC
 , category  = @NAME
 , commented = @COMMENTED
 , filename  = @FILENAME
 , var_metric= @VAR_METRIC
 , var_name  = 'directory'
 , var_val   = concat('/source/src/tmp/T2/moh/', 
@NAME, '/')
 ;
 insert into static set
   cat_metric= @CAT_METRIC
 , category  = @NAME
 , commented = @COMMENTED
 , filename  = @FILENAME
 , var_metric= @VAR_METRIC
 , var_name  = 'mode'
 , var_val   = 'files'
 ;
 insert into static set
   cat_metric= @CAT_METRIC
 , category  = @NAME
 , commented = @COMMENTED
 , filename  = @FILENAME
 , var_metric= @VAR_METRIC
 , var_name  = 'sort'
 , var_val   = 'random'
 ;
 insert into static set
   cat_metric= @CAT_METRIC
 , category  = @NAME
 , commented = @COMMENTED
 , filename  = @FILENAME
 , var_metric= @VAR_METRIC
 , var_name  = 'type'
 , var_val   = 'preset'
 ;
--  insert into static set
--cat_metric= @CAT_METRIC
--  , category  = @NAME
--  , commented = @COMMENTED
--  , filename  = @FILENAME
--  , var_metric= @VAR_METRIC
--  , var_name  = 'application'
--  , var_val   = '/usr/bin/mpg123 --mono -b 0 -f 8192 
-q -r 8000 -s -@ http://206.190.136.141:5022/Live'
--  ;

-- FILES
--  set @COMMENTED  = 0;
--  insert into static set
--cat_metric= 

[asterisk-users] Send message to AMI from dialplan

[Telium Technical Support]

Is it possible to simply send a message to appear as an AMI message/event,
from the dialplan?  For example

 

exten =>123,1,ami(myEvent, param1, param2)

 

and in the AMI a message appears like:

 

Event: myEvent

Privilege: call,all

Channel: PJSIP/misspiggy-0001

Uniqueid: 1368479157.3

ChannelState: 3

ChannelStateDesc: Up

CallerIDNum: 657-5309

CallerIDName: Miss Piggy

ConnectedLineName:

ConnectedLineNum:

AccountCode: Pork

Priority: 1

Exten: 123

Context: inbound

Parameter1: param1

Parameter2: param2

 

 

I'm thinking about ways that I can send messages from the dialplan to my own
application which listens to AMI events.

 

Thanks

Andrea

Trainee

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Re: [asterisk-users] CLI color prompt

[Telium Technical Support]

That means that Asterisk is not echoing the escape character (27) to your 
terminal.

Try different escape formats (octal, slash prefix, etc)

-Original Message-
From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On Behalf 
Of Fourhundred Thecat
Sent: Sunday, May 31, 2020 10:31 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Subject: Re: [asterisk-users] CLI color prompt

 > On 2020-05-31 16:25, Jeff LaCoursiere wrote:
> I'm pretty sure that means your are using a non-color capable 
> terminal, or your termtype variable is incorrect.  What are you using 
> for a terminal emulator?

my terminal supports colors, I am using colored prompt in bash/zsh already. I 
made a screenshot:

https://paste.pics/d1eb46bac0a8d06d645230225191615e


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Re: [asterisk-users] CLI color prompt

[Telium Technical Support]

Have you tried adding ANSI color escape codes?

There's lots of documentation for BASH prompt color using escape codes.  Give 
those a try.

(I haven't tried it, but would make sense)

-Original Message-
From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On Behalf 
Of Fourhundred Thecat
Sent: Sunday, May 31, 2020 9:45 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] CLI color prompt

Hello,

how can I change the color of the asterisk prompt to red ?

I read in the wiki that I can use %Cn[;n]

https://wiki.asterisk.org/wiki/display/AST/Asterisk+CLI+Configuration

But what does this mean ?
There is no example how to actually use it.
where do I put it?
What syntax is that anyway?
How do I specify red ?

I currently have this in my environment:

export ASTERISK_PROMPT="[%H]: "

which changes the prompt to hostname

Ho can I make this prompt red ?


thanks,

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Re: [asterisk-users] Troubleshooting load issues

[Telium Technical Support]

I assumed the spikes were within the Asterisk process.   If the spikes last 
long enough use htop and iotop to see if the spikes are outside of your process.

 

If outside the Asterisk process then there are lots of generic troubleshooting 
guides.  If within the Asterisk process (and no transcoding) then turn verbose 
way up and watch for clues on CLI when a spike occurs.

 

 

From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On Behalf 
Of Dovid Bender
Sent: Wednesday, April 22, 2020 2:55 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Subject: Re: [asterisk-users] Troubleshooting load issues

 

All the calls are using ulaw. The files that I am playing are gsm. I suppose 
doing a file convert with sox to .ulaw may help but it should be able to do 500 
calls without an issue. Can it possibly be a bug? if not how do I profile which 
call(s) can be causing the spike? 

 

 

On Wed, Apr 22, 2020 at 2:21 PM Telium Technical Support mailto:supp...@telium.io> > wrote:

Could some calls be arriving with a different codec?  (Is transcoding causing 
the spikes)?  Are you limiting codecs to match your audio files?

 

From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com 
<mailto:asterisk-users-boun...@lists.digium.com> ] On Behalf Of Dovid Bender
Sent: Wednesday, April 22, 2020 2:01 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion 
mailto:asterisk-users@lists.digium.com> >
Subject: [asterisk-users] Troubleshooting load issues

 

Hi,

 

I have an Asterisk box which has an IVR that plays random gsm files. The box 
has SSD's and two CPU E5-2695 v2 cpus with 64GB ram. The Asterisk CPU usage 
along with the load seems to jump around. With about 500 callers it hovers 
between 250-400% CPU (so 2.5 to 4 cores) which seems reasonable. Every so often 
the load average spikes. The idle never drops below 85%. When the load average 
spikes I see a lot of kworker threads and the CPU usage tends to (not not 
always) go up as well. How would I go about seeing what in Asterisk is causing 
the spike? The box is locked down and only takes calls from an OpenSiPS box. 
There is nothing else running on the box.

 

TIA.

 

Dovid

 

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Re: [asterisk-users] Troubleshooting load issues

[Telium Technical Support]

Could some calls be arriving with a different codec?  (Is transcoding causing 
the spikes)?  Are you limiting codecs to match your audio files?

 

From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On Behalf 
Of Dovid Bender
Sent: Wednesday, April 22, 2020 2:01 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Subject: [asterisk-users] Troubleshooting load issues

 

Hi,

 

I have an Asterisk box which has an IVR that plays random gsm files. The box 
has SSD's and two CPU E5-2695 v2 cpus with 64GB ram. The Asterisk CPU usage 
along with the load seems to jump around. With about 500 callers it hovers 
between 250-400% CPU (so 2.5 to 4 cores) which seems reasonable. Every so often 
the load average spikes. The idle never drops below 85%. When the load average 
spikes I see a lot of kworker threads and the CPU usage tends to (not not 
always) go up as well. How would I go about seeing what in Asterisk is causing 
the spike? The box is locked down and only takes calls from an OpenSiPS box. 
There is nothing else running on the box.

 

TIA.

 

Dovid

 

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[asterisk-users] Compile Asterisk without CPU specific extensions/optimizations

[Telium Technical Support]

I'm compiling an Asterisk system on a ESXi VM with recent CPU, but will
deploy onto an old ESXi VM with older CPU.  

 

Is it possible to configure Asterisk to NOT use CPU specific
instructions/optimizations so that the executable is portable?

 

Thanks

Dan

(in learning mode)

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Re: [asterisk-users] Lightweight ODBC DB

[Telium Technical Support]

IF you use the HAAst or PBXSync solution, you can include/exclude at the table 
and database levels.  You can also use SQLite if the data is suitable (and 
these products can sync SQLite too).

 

If you want a non-commercial solution, MySQL’s log rolling may be most suitable.

 

 

 

From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On Behalf 
Of Doug Lytle
Sent: Thursday, August 1, 2019 6:06 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Lightweight ODBC DB

 

On 8/1/19 5:08 PM, Dovid Bender wrote:

Glenn,

 

I can't use MySQL as each node currently has MySQL however there is a lot of 
data that is stored locally on each box. I may have to take this route if I 
can't find something else but that would mean syncing all sorts of data that 
does not need to be synced.


If I recall correctly, you can exclude databases.

Doug

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Re: [asterisk-users] Lightweight ODBC DB

[Telium Technical Support]

Have you looked at PBXSync (or HAAst) from Telium?  (https://telium.io)

 

These products will sync MySQL, SQLite, plus files, directories, etc. 
intelligently.  (Differentials only) between PBX’s, reload configurations on 
the fly, etc.  No need roll logs or recover from a base in case they get too 
far out of sync.

 

HAAst will also prevent synchronizing if a node is in poor health (to avoiding 
sync’ing in corrupted data).

 

I’m not sure what you are building but this might help.  Aside from this, avoid 
block based synchronization of databases (eg: DRBD) for the obvious reasons.

 

There are Master-Master sync tools out there, but if you are trying to wrap 
some intelligence around that then you are basically building your own sync 
product.

 

-Raj-

 

From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On Behalf 
Of Dovid Bender
Sent: Tuesday, July 30, 2019 9:55 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Subject: [asterisk-users] Lightweight ODBC DB

 

Hi,

 

I am running several Asterisk boxes with realtime around the world. Does anyone 
have a recommendation for a "light" db that would work with Asterisk that would 
also allow replication between all sites (so if I add an entry to one box it 
will work with the rest)?

 

TIA.

 

Dovid

 

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Re: [asterisk-users] Does Asterisk cache AstDB?

[Telium Technical Support]

Great - thank you!

-Original Message-
From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On Behalf 
Of Joshua C. Colp
Sent: Monday, May 6, 2019 2:41 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Does Asterisk cache AstDB?

On Mon, May 6, 2019, at 3:34 PM, Telium Technical Support wrote:
> Is the Asterisk internal database cached by Asterisk? Or is it always 
> reading/writing to the SQLite database? (If I read from the SQLite DB 
> is it sure to match what Asterisk is using)

There is no additional caching built into Asterisk itself for it. The sqlite 
library calls are directly used and their results provided.

--
Joshua C. Colp
Digium - A Sangoma Company | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: 
www.digium.com & www.asterisk.org

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[asterisk-users] Does Asterisk cache AstDB?

[Telium Technical Support]

Is the Asterisk internal database cached by Asterisk?  Or is it always
reading/writing to the SQLite database?  (If I read from the SQLite DB is it
sure to match what Asterisk is using)

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[asterisk-users] using CIDR for hosts entry in sip.conf

[Telium Technical Support]

I am setting up a system with a large number of trusted trunks (by IP).  I
find that I have to make one entry sip.conf for each trunk becauses the
host= line requires a single IP.

 

Does asterisk support a CIDR or wildcard or multi-ip format for the host=
line in sip.conf?

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Re: [asterisk-users] Dialplan reload from AMI

[Telium Technical Support]

Does reloading pbx_config ONLY reload the dialplan?  Or is something else 
reloaded too?

 

This sounds like a preferable way to do it

 

From: Ian McMaster [mailto:ian.mcmas...@gmail.com] 
Sent: Saturday, April 20, 2019 1:19 PM
Subject: Dialplan reload from AMI

 

Rather than

Action: Command

Command: dialplan reload

 

Prefer this:

 

Action: Reload

Module: pbx_config

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[asterisk-users] Reload dialplan from AMI

[Telium Technical Support]

I see there is a modulereload function available from the AMI, but none of
the listed modules (on the wiki) seem to reload the dialplan.  Is there a
way to reload the dialplan through this function?  Or do I have to use the
'command' action?

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Re: [asterisk-users] Asterisk - can't hear other side. Or other side does not hear us

[Telium Technical Support]

This is usually a symptom of something on the call path mishandling the session 
setup.  Check routers/firewall/SIP proxy, etc.  Likely a firmware bug is 
causing it to use the wrong IP address and passing that to the other end.

 

Even if you disabled these devices, REMOVE them from the call path (or replace) 
for testing.  Add them back one at a time to confirm source of problem.

 

Sue

 

From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On Behalf 
Of Ivan Demkovitch
Sent: Wednesday, February 27, 2019 5:11 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Asterisk - can't hear other side. Or other side does 
not hear us

 

Hello,

 

This is not technical post, just looking for suggestions on what to check.

I have asterisk for long time, no updates, just maintain OS updates.

 

I use SPA504G phones

 

Very rarely and randomly when we pickup a phone - other side does not hear us. 
Call them back and all works.

 

Now I have couple people I'm talking to and it seems like very call like this. 
Someone can't hear someone.

 

Don't know where to start to troubleshoot and what to look for.

 

Thanks!

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Re: [asterisk-users] Pass through registration / proxy

[Telium Technical Support]

OK - I'll have to rethink how to solve this problem.  Maybe I made some
assumptions...here's what I'm trying to accomplish:

I've been given a legacy PBX with SIP capabilities.  I need to have SIP
phones connect to Asterisk (for other features, part of the next step) which
passes the calls through to the legacy PBX.  And conversely, calls to that
extension number on the legacy PBX have to ring the SIP phone connected to
Asterisk.

Maybe proxy is the wrong word I chose.  Asterisk is something like a peer to
the legacy PBX.  I thought about setting up individual SIP accounts on the
Asterisk box to connect to the legacy PBX, or maybe a SIP trunk to the
legacy PBX (assuming it can route calls through the SIP trunk to a peer to
reach a phone).  The legacy PBX is a Nortel in case that matters.

I'm supposed to figure this out and present options but having trouble
figuring out if Asterisk would be a peer, or pretend to be many sip agents
registering on the legacy Sip pbx, etc.  I think I'm stuck at the conceptual
level.  (Still a beginner in training - but having fun learning Asterisk)


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Daniel Tryba
Sent: Wednesday, April 11, 2018 9:27 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com>
Subject: Re: [asterisk-users] Pass through registration / proxy

On Tue, Apr 10, 2018 at 09:22:02PM -0400, Telium Technical Support wrote:
> I need to create a SIP proxy to be placed in front of a legacy PBX.  
> When a phone registers with the proxy, I would like Asterisk to 
> register with the PBX behind it.  (To tell the PBX to send calls to 
> the proxy and then to the SIP phone).
> 
> Can I use Asterisk to create a proxy like this?  Is there a way to 
> cause the Asterisk to register with another host when it receives a 
> successfully registration?

You can, but maybe you should use a sip proxy (like kamailio) for this task
instead of a back to back user agent like asterisk.

You can listen to events triggered on registration to asterisk and with
realtime intergration add the register to the PBX (or manipulate sip.conf).
This still might be easier to implement compared to (for
example) kamailio if you are new to that.

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Re: [asterisk-users] Pass through registration / proxy

[Telium Technical Support]

I’ve been tasked with building the whole thing in just Asterisk (as an 
exercise).  Trying to figure out how/if Asterisk alone can do thi.

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Carlos Rojas
Sent: Tuesday, April 10, 2018 9:30 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion 
<asterisk-users@lists.digium.com>
Subject: Re: [asterisk-users] Pass through registration / proxy

 

Hi

 

You could use kamailio +asterisk

 

On Tue, Apr 10, 2018, 9:25 PM Telium Technical Support <supp...@telium.ca 
<mailto:supp...@telium.ca> > wrote:

I need to create a SIP proxy to be placed in front of a legacy PBX.  When a 
phone registers with the proxy, I would like Asterisk to register with the PBX 
behind it.  (To tell the PBX to send calls to the proxy and then to the SIP 
phone).

 

Can I use Asterisk to create a proxy like this?  Is there a way to cause the 
Asterisk to register with another host when it receives a successfully 
registration?

 

Thanks!

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[asterisk-users] Pass through registration / proxy

[Telium Technical Support]

I need to create a SIP proxy to be placed in front of a legacy PBX.  When a
phone registers with the proxy, I would like Asterisk to register with the
PBX behind it.  (To tell the PBX to send calls to the proxy and then to the
SIP phone).

 

Can I use Asterisk to create a proxy like this?  Is there a way to cause the
Asterisk to register with another host when it receives a successfully
registration?

 

Thanks!

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Re: [asterisk-users] Blacklist failed attempts

[Telium Technical Support]

If this is a home system, try the free edition of SecAst (www.telium.ca/?secast 
 ).  If allows you to set thresholds for the 
number of attempts, and specify the period in which they occur.  The Free 
edition of SecAst is a drop-in replacement for fail2ban (but with a lot more 
intelligence included for free).

 

If this is for a business / you are looking for a commercial product 
recommendation then post on the commercial list :)

 

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Atux Atux
Sent: Thursday, March 1, 2018 8:03 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Subject: [asterisk-users] Blacklist failed attempts

 

Hi. I would like to protect my system from failed attempts. I would like to ask 
if there is a way to do a blacklist for certain amount of time consecutive 
attempts from the same IP. For example if we have an IP that gets a wrong 
passwd an it had tried more than 3 times the last 5 minutes, blacklist it for 
an hour. I have tried to implement it through fail2ban, but it doe snot seem to 
work for my asterisk implementation.

Is there any other way?



 

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Re: [asterisk-users] Detecting DoS attacks via SIP

[Telium Technical Support]

If this is a home system, try the free edition of SecAst (www.telium.ca/?secast 
<http://www.telium.ca/?secast> ).  It uses the AMI for detecting simple failed 
events , but can do more than fail2ban.  More importantly it can block at the 
network edge by talking to you firewall (don’t let the script kiddies onto you 
LAN).

 

If decide to try geofencing using just IP rules than you will really slow your 
system (as the number of rules and exceptions is massive in order to be 
useful).  There are some open source IP to location services (SaaS) which are 
free if it’s not for commercial use.

 

-Raj-

 

All opinions expressed on the boards/chat groups are my own.  As an employee of 
Telium my views may appear seriously biased – but I hope there’s some helpful 
info in there for you :)

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Diehl
Sent: Saturday, August 19, 2017 11:54 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion 
<asterisk-users@lists.digium.com>
Subject: Re: [asterisk-users] Detecting DoS attacks via SIP

 

I appreciate the discussion on the question I asked.

I currently listen for failed registration attempts via AMI and automatically 
block the offending IP address at the firewall.  I was hoping to find another 
AMI event that would be the magic bullet I need, but it doesn't sound like 
that's going to happen.

I understand that fail2ban is probably not what I want and probably wouldn't 
detect the attacks I'm seeing.

It turns out that not all of the attacks are from the "friendly scanner," but 
enough of them are that it's a good start.

So, I really like the idea of the IP geo location firewall rules coupled with 
the "friendly scanner" filter, as provided by a few of you guys.  It was 
mentioned that this is a broad hammer, but I'm kinda looking for a broad 
hammer! ;^)

Looks like I need to do some research, but I think I have what I need.

Thanks again,

Mike Diehl.

 

On Sat, Aug 19, 2017 at 4:36 PM, Telium Technical Support <supp...@telium.ca 
<mailto:supp...@telium.ca> > wrote:

I think you missed the point of the Digium post.  Fail2ban can ONLY ban IP’s if 
Asterisk records a failure to register.  Asterisk does not detect malformed SIP 
packets, buffer overflow attacks, suspicious dialing patterns, connection 
attempts outside geofenced areas, use of stolen credentials (rapid  ramp of 
calls using one set of credentials), etc.

 

Asterisk only gives you a rudimentary “failed” message for a failure to 
register / wrong credentials.  And of course fail2ban only responds to Asterisk 
log messages, so it does little more than ban the annoying script kiddies.

 

Have a good look at that Voip-Info page and read what actual SIP security 
systems do.  Then compare that to fail2ban and it’s night & day difference.  
People still think fail2ban is a security system, and Digium is very clear that 
it is NOT.

 

 

From: asterisk-users-boun...@lists.digium.com 
<mailto:asterisk-users-boun...@lists.digium.com>  
[mailto:asterisk-users-boun...@lists.digium.com 
<mailto:asterisk-users-boun...@lists.digium.com> ] On Behalf Of Kseniya 
Blashchuk
Sent: Thursday, August 17, 2017 12:41 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion 
<asterisk-users@lists.digium.com <mailto:asterisk-users@lists.digium.com> >
Subject: Re: [asterisk-users] Detecting DoS attacks via SIP

 

Well, correct me if I'm wrong, but I would say this conversation you have 
posted is a bit outdated, now fail2ban can be used with asterisk security log 
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Security+Event+Logger.

 

On Thu, Aug 17, 2017, 4:53 AM Telium Technical Support <supp...@telium.ca 
<mailto:supp...@telium.ca> > wrote:

Keep in mind that the attacks you are seeing in the log are ONLY the ones
that Asterisk is detecting and rejecting.  All other attacks aren't even
showing up!

There's a good discussion of how to secure your PBX here:
https://www.voip-info.org/wiki/view/asterisk+security

In general, don't let the malevolent traffic get as far as the PBX (block at
the firewall).  Also, Digium regularly warns users that fail2ban is NOT a
security system: http://forums.asterisk.org/viewtopic.php?p=159984

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
<mailto:asterisk-users-boun...@lists.digium.com> 
[mailto:asterisk-users-boun...@lists.digium.com 
<mailto:asterisk-users-boun...@lists.digium.com> ] On Behalf Of mdiehl
Sent: Tuesday, August 15, 2017 3:38 PM
To: asterisk-users@lists.digium.com <mailto:asterisk-users@lists.digium.com> 
Subject: [asterisk-users] Detecting DoS attacks via SIP

Hi all,

Lately, I've seen an increase in the number of attacks against my system
from the so-called "Friendly Scanner."  When one of these script kiddies
targets my server, all I

Re: [asterisk-users] Detecting DoS attacks via SIP

[Telium Technical Support]

I think you missed the point of the Digium post.  Fail2ban can ONLY ban IP’s if 
Asterisk records a failure to register.  Asterisk does not detect malformed SIP 
packets, buffer overflow attacks, suspicious dialing patterns, connection 
attempts outside geofenced areas, use of stolen credentials (rapid  ramp of 
calls using one set of credentials), etc.

 

Asterisk only gives you a rudimentary “failed” message for a failure to 
register / wrong credentials.  And of course fail2ban only responds to Asterisk 
log messages, so it does little more than ban the annoying script kiddies.

 

Have a good look at that Voip-Info page and read what actual SIP security 
systems do.  Then compare that to fail2ban and it’s night & day difference.  
People still think fail2ban is a security system, and Digium is very clear that 
it is NOT.

 

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kseniya Blashchuk
Sent: Thursday, August 17, 2017 12:41 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion 
<asterisk-users@lists.digium.com>
Subject: Re: [asterisk-users] Detecting DoS attacks via SIP

 

Well, correct me if I'm wrong, but I would say this conversation you have 
posted is a bit outdated, now fail2ban can be used with asterisk security log 
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Security+Event+Logger.

 

On Thu, Aug 17, 2017, 4:53 AM Telium Technical Support <supp...@telium.ca 
<mailto:supp...@telium.ca> > wrote:

Keep in mind that the attacks you are seeing in the log are ONLY the ones
that Asterisk is detecting and rejecting.  All other attacks aren't even
showing up!

There's a good discussion of how to secure your PBX here:
https://www.voip-info.org/wiki/view/asterisk+security

In general, don't let the malevolent traffic get as far as the PBX (block at
the firewall).  Also, Digium regularly warns users that fail2ban is NOT a
security system: http://forums.asterisk.org/viewtopic.php?p=159984

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
<mailto:asterisk-users-boun...@lists.digium.com> 
[mailto:asterisk-users-boun...@lists.digium.com 
<mailto:asterisk-users-boun...@lists.digium.com> ] On Behalf Of mdiehl
Sent: Tuesday, August 15, 2017 3:38 PM
To: asterisk-users@lists.digium.com <mailto:asterisk-users@lists.digium.com> 
Subject: [asterisk-users] Detecting DoS attacks via SIP

Hi all,

Lately, I've seen an increase in the number of attacks against my system
from the so-called "Friendly Scanner."  When one of these script kiddies
targets my server, all I see for symptoms is a few of my trunks become
lagged due to server load and a stream of messages on the console that
resemble this:

[Aug  2 20:27:50]   == Using SIP VIDEO CoS mark 6
[Aug  2 20:27:50]   == Using SIP RTP TOS bits 24
[Aug  2 20:27:50]   == Using SIP RTP CoS mark 5
[Aug  2 20:32:47]   == Using SIP VIDEO TOS bits 24
[Aug  2 20:32:47]   == Using SIP VIDEO CoS mark 6
[Aug  2 20:32:47]   == Using SIP RTP TOS bits 24
[Aug  2 20:32:47]   == Using SIP RTP CoS mark 5
[Aug  2 20:34:26]   == Using SIP VIDEO TOS bits 24
[Aug  2 20:34:26]   == Using SIP VIDEO CoS mark 6


I have to turn on sip debugging to find out who's hitting me.  However, I
can't just leave it on because it would kill my logging system.

So, how are other people handling this?  Is there an AMI event I want watch
for?  I watch for PeerStatus, but since there's no actual peer in the
attack, I don't seem to get an event from AMI.

Any ideas?

Mike Diehl.

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Re: [asterisk-users] Detecting DoS attacks via SIP

[Telium Technical Support]

Keep in mind that the attacks you are seeing in the log are ONLY the ones
that Asterisk is detecting and rejecting.  All other attacks aren't even
showing up!

There's a good discussion of how to secure your PBX here:
https://www.voip-info.org/wiki/view/asterisk+security

In general, don't let the malevolent traffic get as far as the PBX (block at
the firewall).  Also, Digium regularly warns users that fail2ban is NOT a
security system: http://forums.asterisk.org/viewtopic.php?p=159984

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of mdiehl
Sent: Tuesday, August 15, 2017 3:38 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Detecting DoS attacks via SIP

Hi all,

Lately, I've seen an increase in the number of attacks against my system
from the so-called "Friendly Scanner."  When one of these script kiddies
targets my server, all I see for symptoms is a few of my trunks become
lagged due to server load and a stream of messages on the console that
resemble this:

[Aug  2 20:27:50]   == Using SIP VIDEO CoS mark 6
[Aug  2 20:27:50]   == Using SIP RTP TOS bits 24
[Aug  2 20:27:50]   == Using SIP RTP CoS mark 5
[Aug  2 20:32:47]   == Using SIP VIDEO TOS bits 24
[Aug  2 20:32:47]   == Using SIP VIDEO CoS mark 6
[Aug  2 20:32:47]   == Using SIP RTP TOS bits 24
[Aug  2 20:32:47]   == Using SIP RTP CoS mark 5
[Aug  2 20:34:26]   == Using SIP VIDEO TOS bits 24
[Aug  2 20:34:26]   == Using SIP VIDEO CoS mark 6


I have to turn on sip debugging to find out who's hitting me.  However, I
can't just leave it on because it would kill my logging system.

So, how are other people handling this?  Is there an AMI event I want watch
for?  I watch for PeerStatus, but since there's no actual peer in the
attack, I don't seem to get an event from AMI.

Any ideas?

Mike Diehl.

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Re: [asterisk-users] asterisk 13.16 / pjsip / t.38: res_pjsip_t38.c:207 t38_automatic_reject: Automatically rejecting T.38 request on channel 'PJSIP/91-00000007'

[Telium Technical Support]

Just a guess (without knowing about your network), but are the two ends
points on public networks and visible to one another?  If not the reinvite
may be passing an internal (nat'ed) address to the other and the connection
will fail...just a though

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michael Maier
Sent: Sunday, June 4, 2017 3:39 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion

Subject: [asterisk-users] asterisk 13.16 / pjsip / t.38: res_pjsip_t38.c:207
t38_automatic_reject: Automatically rejecting T.38 request on channel
'PJSIP/91-0007'

Hello!

I'm still trying to get a working t.38 configuration w/ pjsip.

I'm now able to send t.38 faxes to my own extension:


hylafax -> t38modem -> extension -> extension -> t38modem -> hylafax.


The fax is sent by t38modem. The receiving part of t38modem accepts the
call, sends ReInvite for t.38 and things are working as expected.



Now, let's do the nearly same thing, but use an ISP:

hylafax -> t38modem -> extension -> trunk -> ISP -> trunk -> extension ->
t39modem(2) -> hylafax


Same procedure: the receiving t38modem(2) sends ReInvite for t.38 - but this
time, the extension / asterisk just ignores it. After the 5. retry to switch
to T.38, asterisk tells:

res_pjsip_t38.c:207 t38_automatic_reject: Automatically rejecting T.38
request on channel 'PJSIP/91-0007'

=> Why does asterisk reject the switch / ReInvite to T.38 this time? It
never even tried to send it to the ISP!


Thanks for any hint!

Regards,
Michael

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Re: [asterisk-users] iaxModem pickup problem

[Telium Technical Support]

The file astdb.sqlite is a SQLite 3 file (all tables and indices rolled into
one file).  While Asterisk is running the astdb file is always open for r/w.
FreePBX regularly updates rows ("keys") in this database, so writes are
often in progress.

In the event of a power failure the file will not be properly closed, or
worse be left in an invalid state.  Once Asterisk starts it is may refuse to
read some astdb rows, or potentially the whole file.

Asterisk natively does not need the astdb file but FreePBX makes extensive
use of it.  In particular, FreePBX dialplans check device/user information
in the astdb for call handling.  So a missing/corrupt user/device will cause
the dialplan to fail.  (That's why I suggested to you watch the dialplan
from the Asterisk CLI when a fax comes in).

This design (FreePBX) makes Asterisk much more fragile than it has to be.
It's a good idea to keep a backup astdb on the PBX in case of corruption.

-Original Message-
From: James B. Byrne [mailto:byrn...@harte-lyne.ca] 
Sent: Thursday, May 4, 2017 12:29 PM
To: Telium Technical Support <supp...@telium.ca>
Cc: asterisk-users@lists.digium.com
Subject: RE: [asterisk-users] iaxModem pickup problem


On Thu, May 4, 2017 11:38, Telium Technical Support wrote:
> It depends a bit on your version of FreePBX, but here's a link to show 
> you how:
>
> http://telium.ca/pages/forums/viewtopic.php?f=7=19
>
> Hopefully option 1 works for you (quick and easy).  If not, you'll 
> have to try option 2.  Ignore option 3 since that's only for users of 
> High Availability for Asterisk (HAAst).
>
> (I assume that if you had a full backup you would have already tried 
> to restore it)
>

No, I did not try to restore from backups; and yes I have daily backups to
recover from if that is necessary.  However, I have since corrected the
damaged rows in astdb.sqlite and the fax service is now working again.

If someone could explain what likely happens to damage astdb.sqlite when the
system is suddenly powered off I would appreciate it.

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James B. Byrnemailto:byrn...@harte-lyne.ca
Harte & Lyne Limited  http://www.harte-lyne.ca
9 Brockley Drive  vox: +1 905 561 1241
Hamilton, Ontario fax: +1 905 561 0757
Canada  L8E 3C3



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Re: [asterisk-users] iaxModem pickup problem

[Telium Technical Support]

It depends a bit on your version of FreePBX, but here's a link to show you
how:

http://telium.ca/pages/forums/viewtopic.php?f=7=19

Hopefully option 1 works for you (quick and easy).  If not, you'll have to
try option 2.  Ignore option 3 since that's only for users of High
Availability for Asterisk (HAAst).

(I assume that if you had a full backup you would have already tried to
restore it)

-Raj-

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of James B. Byrne
Sent: Thursday, May 4, 2017 11:00 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] iaxModem pickup problem


On Thu, May 4, 2017 10:22, James B. Byrne wrote:

I am advised that it may be possible thast the astdb.sqlite3 database may be
corrupted.  Are there procedures to rebuild or repair this? 
Where are they documented?  If not then how does one repair such?

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James B. Byrnemailto:byrn...@harte-lyne.ca
Harte & Lyne Limited  http://www.harte-lyne.ca
9 Brockley Drive  vox: +1 905 561 1241
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Re: [asterisk-users] PBX selection

[Telium Technical Support]

Have a look at xCally from Xenialabs too – they are particularly popular with 
call centers (and still asterisk based).

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Roamer2998
Sent: Tuesday, April 18, 2017 11:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Subject: Re: [asterisk-users] PBX selection

 

Thanks All.

 

Thanks Alex, we also tested thirdlane PBX, and comparing it with PortSIP PBX, 
Vodia PBX, we hope we can make decision next week.

 

Best regards,

 

On Wed, Apr 19, 2017 at 10:05 AM, Alex Epshteyn  > wrote:

The solution you choose should be based on many factors which should include 
your business requirements, team's experience, your budget, growth expectations 
and more.

You can choose Asterisk or Freeswitch as a platform and start building on that 
- but it is not simple and being new to VoIP you are likely to make mistakes. 
The "do-it-yourself" approach will some money initially, but will be the most 
expensive option long term - as you will be denying the economy of scale. 
Bringing a "smart programmer" won't help much as you will also create a 
"lock-in". In fact, this could be worse than a dependency created when you use 
a commercial or a known open source solution as while you would still be able 
to get help from the community for the "base" part of your pbx, your custom 
part will be much harder to deal with.

Our company started building Asterisk based PBX in 2002 and Multi Tenant PBX in 
2005 - we do this as our core business and are still finding areas for 
improvement :). As your experience with VoIP is minimal I would side with your 
CTO - you should find a solution high enough in the stack to avoid the 
complexity of building it all yourself.

Good luck,

Alex

--

Alex Epshteyn
email: a...@thirdlane.com  
web: www.thirdlane.com  
phone +1 415.261.6601  



- Original Message -
> From: "J Montoya or A J Stiles"   >
> To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
>  >
> Sent: Tuesday, April 18, 2017 1:40:47 AM
> Subject: Re: [asterisk-users] PBX selection
>
> On Monday 17 Apr 2017, Speed Boy wrote:
> >  Hi all, I'm new to VoIP, now we have a project that needs a
> >  PBX with client APPs.
> > In our team we have argument for choosing PBX. By so far, we
> >  have following candidates:
> >
> > A: Open source
> >
> >  1) Asterisk PBX (http://www.asterisk.org) (with longest
> >  history that almost every one knows it, now the last version using
> >  the
> > PJSIP stack)
> >  2) FreeSwitch (http://www.freeswitch.org) (A lot people
> >  recommended it to us)
> >
> >
> > B: Commercial
> >
> > 1) Vodia PBX (http://www.vodia.com). It comes from SNOM, now
> > acquired by a HongKong company now
> > 2) PortSIP PBX (http://www.portsip.com/portsip-pbx). It
> > also includes VoIP SDK, WebRTC and offer rebranding app for free.
> >
> > My boss prefers the Open Source PBX since they are free,
> > but our CTO prefers the commercial editions, according to
> > whom the business PBX has better support, and the
> > performance is good, and easy to use - considering our team
> > all are new to VoIP/PBX.
>
> Proponents of proprietary solutions always like to say "If an Open
> Source
> solution breaks, who can you call?"  The answer is, "Any
> sufficiently-competent
> programmer -- it may be broken, but we have all the pieces".  Whereas
> if you
> spend money on proprietary software and it breaks, then there is only
> *one*
> place you can call -- and you'd better hope they are interested to
> fix your
> problem.
>
> On the other hand, if you could get full Source Code and Modification
> Rights
> (basically, "everything we could do with a GPL program except
> distribute
> copies"),  a proprietary solution might not be so bad after all.  But
> since
> the goal of most proprietary software vendors is to extract money
> from you and
> maintaining you in a state of perpetual helplessness is highly
> desirable in
> the course of this, do not expect to get such a deal in real life.
>
> > We have did some searching of Asterisk, here are my questions:
> >
> > 1. Does the last Asterisk using PJSIP stack ?
>
> Yes.
>
> > 2. Does there has the comparison of PJSIP and reSIProcate,
> > sofia(using by
> > FreeSwicth) ?
>
> Not sure about this.  We're still using the original chan_sip driver.
>
> > 3. Is it easy to compile and setup Asterisk?
>
> It's about as easy as compiling anything from Source Code.  Harder
> than LAME
> MP3 encoder, but easier than the Linux kernel.  If you altered
> `monop` from
> the BSDgames package to make the streets match your local edition of
> the game,
> 

Re: [asterisk-users] AGI Exec Voicemail

[Telium Technical Support]

Why not use an ALIAS and let sendmail send the email to a distribution
group?

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff
LaCoursiere
Sent: Wednesday, April 12, 2017 1:09 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion

Subject: [asterisk-users] AGI Exec Voicemail


Hi,

I have a voicemail broadcast AGI that has been running fine for years - it
collects extensions and then EXECs the Voicemail app, like this:

EXEC Voicemail \"%s\"

(%s is the extension list like AAA etc)

This works fine, but after leaving the message and pressing "#", I just get
"Thank you" and a hangup.  I would like to have the option to review,
re-record, or cancel.  It isn't clear how to enable this option via EXEC.  I
tried:

EXEC Voicemail \"%s,review=yes\"

but there is no effect at all.

Any clues?

Thanks,

j

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Re: [asterisk-users] Manager events showing in CLI

[Telium Technical Support]

I did that too – no debug related settings in there!  That’s why I’m stumped.

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Victor Villarreal
Sent: Sunday, March 26, 2017 2:06 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion 
<asterisk-users@lists.digium.com>
Subject: Re: [asterisk-users] Manager events showing in CLI

 

Ok,

 

Please, check your manager.conf and logger.conf for any clue about debugging 
options, into the Asterisk configuration directory. 

 

El 26 mar. 2017 14:52, "Telium Technical Support" <supp...@telium.ca 
<mailto:supp...@telium.ca> > escribió:

I tried that but it had no effect.  Still see things like:

 

[2017-03-26 13:49:39] DEBUG[2088]: manager.c:5693 match_filter: Examining AMI 
event:

Event: SuccessfulAuth

Privilege: security,all

EventTV: 2017-03-26T13:49:39.407-0400

Severity: Informational

Service: SIP

EventVersion: 1

AccountID: 221essionID: 0x7fa0cc005cc8

LocalAddress: IPV4/UDP/192.168.67.4/5060 <http://192.168.67.4/5060> 

RemoteAddress: IPV4/UDP/192.168.67.26/5060 <http://192.168.67.26/5060> 

UsingPassword: 1

 

 

[2017-03-26 13:49:39] DEBUG[1882]: chan_sip.c:9196 __find_call: = Looking for  
Call ID: 280f68000ff289291b366a1242530ce8@192.168.67.4:5060 
<http://280f68000ff289291b366a1242530ce8@192.168.67.4:5060>  (Checking To) 
--From tag as494dfc4b --To-tag 4155795028  

[2017-03-26 13:49:39] DEBUG[1882]: chan_sip.c:4419 __sip_ack: Stopping 
retransmission on '280f68000ff289291b366a1242530ce8@192.168.67.4:5060 
<http://280f68000ff289291b366a1242530ce8@192.168.67.4:5060> ' of Request 102: 
Match Found

[2017-03-26 13:49:39] DEBUG[1882]: chan_sip.c:6725 sip_destroy: Destroying SIP 
dialog 280f68000ff289291b366a1242530ce8@192.168.67.4:5060 
<http://280f68000ff289291b366a1242530ce8@192.168.67.4:5060> 

[2017-03-26 13:49:39] DEBUG[1882]: chan_sip.c:4275 __sip_autodestruct: Auto 
destroying SIP dialog 'cbf5d92f6844702b'

[2017-03-26 13:49:39] DEBUG[1882]: chan_sip.c:6725 sip_destroy: Destroying SIP 
dialog cbf5d92f6844702b

[2017-03-26 13:49:39] DEBUG[2088]: manager.c:6138 process_message: Running 
action 'Command'

[2017-03-26 13:49:39] DEBUG[1951]: manager.c:6138 process_message: Running 
action 'Command'

 

cli> manager set debug off

 


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Re: [asterisk-users] Manager events showing in CLI

[Telium Technical Support]

I tried that but it had no effect.  Still see things like:

 

[2017-03-26 13:49:39] DEBUG[2088]: manager.c:5693 match_filter: Examining AMI 
event:

Event: SuccessfulAuth

Privilege: security,all

EventTV: 2017-03-26T13:49:39.407-0400

Severity: Informational

Service: SIP

EventVersion: 1

AccountID: 221essionID: 0x7fa0cc005cc8

LocalAddress: IPV4/UDP/192.168.67.4/5060

RemoteAddress: IPV4/UDP/192.168.67.26/5060

UsingPassword: 1

 

 

[2017-03-26 13:49:39] DEBUG[1882]: chan_sip.c:9196 __find_call: = Looking for  
Call ID: 280f68000ff289291b366a1242530ce8@192.168.67.4:5060 (Checking To) 
--From tag as494dfc4b --To-tag 4155795028  

[2017-03-26 13:49:39] DEBUG[1882]: chan_sip.c:4419 __sip_ack: Stopping 
retransmission on '280f68000ff289291b366a1242530ce8@192.168.67.4:5060' of 
Request 102: Match Found

[2017-03-26 13:49:39] DEBUG[1882]: chan_sip.c:6725 sip_destroy: Destroying SIP 
dialog 280f68000ff289291b366a1242530ce8@192.168.67.4:5060

[2017-03-26 13:49:39] DEBUG[1882]: chan_sip.c:4275 __sip_autodestruct: Auto 
destroying SIP dialog 'cbf5d92f6844702b'

[2017-03-26 13:49:39] DEBUG[1882]: chan_sip.c:6725 sip_destroy: Destroying SIP 
dialog cbf5d92f6844702b

[2017-03-26 13:49:39] DEBUG[2088]: manager.c:6138 process_message: Running 
action 'Command'

[2017-03-26 13:49:39] DEBUG[1951]: manager.c:6138 process_message: Running 
action 'Command'

 

cli> manager set debug off

 

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[asterisk-users] Manager events showing in CLI

[Telium Technical Support]

I somehow cause AMI events to appear as output in the CLI, and I can't
figure out how to turn them off.  Can someone offer a command which will
suppress AMI events/commands from showing in the CLI?

 

Ron

 

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Re: [asterisk-users] Large astDB - millions of tuples - issues?

[Telium Technical Support]

We wrote a call screening (and CID rewrite) app for an ITSP a few years ago.  
We had to use MySQL as the astDB could not keep up (* was choking – we did dig 
deeper we just switched to MySQL).  I don’t think astDB is the right way to go. 
 If you’re comfortable writing a * func then you might as well go with MySQL.

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gabriel Ortiz Lour
Sent: Wednesday, March 22, 2017 9:51 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Subject: Re: [asterisk-users] Large astDB - millions of tuples - issues?

 

Hi Dovid,

  I'm trying to get rid of my AGIs. I wrote an * func to check directly my PG 
database, and then I saw that * already has a func_blacklist that will check 
astDB.

  I was thinking the issues I might have using astDB for it. If I do some 
performance tests I get back here with the results.

Thanks,

Gabriel

 

2017-03-22 10:40 GMT-03:00 Dovid Bender  >:

I  have never tested something that large but I would think it would be slow. 
Why not use an age with reddis or mysql?

 

On Mar 22, 2017 9:32 AM, "Gabriel Ortiz Lour"  > wrote:

Hi all,

  Does anyone uses astDB for a large amount of data, in special for 
implementing black lists with millions of numbers (i'd like about 2 or 3 
million)?

 

  That would be held in memory right? Is this (memory consumption) the only 
problem I could face?

Att.

Gabriel

 

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Re: [asterisk-users] Having problem getting Asterisk to work on CentOS 7

[Telium Technical Support]

Dan - you probably installed the init script (look in /etc/init.d for an
'asterisk' file).  Asterisk includes the older init style scripts which are
*compatible* with systemd but you don't have as much control compared to
creating an Asterisk systemd file.  (SystemD service files replace InitD
scripts).  So that might be part of the solution, but first.

 

If disabling Selinux allows Asterisk to run as you expect then you can
create an selinux policy exception for Asterisk - BUT, ignore that for now.
Just keep SElinux disabled (edit /etc/sysconfig/selinux and set to disabled)
and come back to that later.

 

So in preparation to diagnose further:

1.  Disable asterisk service (systemctl disable asterisk)

2.  Disable selinux (as described above)

3.  Reboot.

 

Next, try to start asterisk with 'systemctl start asterisk'.  Does it work
as expected?  If no, what user have you logged in with?

If not root, su to root and try again.  Did it asterisk service start
properly?

If yes, you should create a systemd service file and use the 'user=root'
parameter (and remove the initd service script).

Does Asterisk start properly now every time?  If yes re-enable to your
systemd Asterisk service to start with the system.

 

I don't see any attachment (probably stripped by the list manager) but that
shouldn't matter - if your Asterisk service is not running as root that
would explain a range of strange behaviours.

 

*Jason*

 

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Cropp
Sent: Wednesday, March 15, 2017 12:41 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com>
Subject: Re: [asterisk-users] Having problem getting Asterisk to work on
CentOS 7

 

Thanks Jason.

 

I will try to explain what I'm seeing for this issue.

 

I did a fresh install of CentOS 7 Minimal into a VM with VMWare Workstation.
Followed the Asterisk from Source instructions using pjproject 2.6 and
asterisk 13.14.0 for the configure, install, .   At the end of the asterisk
portion, I ran the make config which I understand installs the
Initialization scripts.

 

After this, when I restart my CentOS 7 Minimal, I was seeing the
safe_asterisk process, but asterisk would not start.  I could run it from
the command line and it would run.

 

It was suggested that it's an selinux problem.  They had me try 'setenforce
0'.  After this, asterisk process starts running.

As I understand it, there was mention of using systemd instead of using
safe_asterisk.

Other e-mails indicated I should look at the audit.log, so I included that
information.  This audit.log mentioned astdb.sqlite3, so I wasn't sure if
that's the problem.

 

I also just tried a restart and ran 'systemctl start asterisk'.  This did
not start the asterisk process.

 

Through the various recommendations, I've become confused on what the
correct path would be.  I have had zero problems with Debian and Asterisk
for many years.  Making the change to CentOS.  Followed the instructions
from asterisk.org, but for some reason I hit a problem with this on my
CentOS VM.  

 
<https://wiki.asterisk.org/wiki/display/AST/Installing+Asterisk+From+Source>
https://wiki.asterisk.org/wiki/display/AST/Installing+Asterisk+From+Source

 

Simply looking for guidance on what the correct approach to solve this
problem is.

 

Have a great day!

 

Dan

 

 

From:  <mailto:asterisk-users-boun...@lists.digium.com>
asterisk-users-boun...@lists.digium.com [
<mailto:asterisk-users-boun...@lists.digium.com>
mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Telium
Technical Support
Sent: Wednesday, March 15, 2017 11:08 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Having problem getting Asterisk to work on
CentOS 7

 

The history of the question is lost (in the mail thread) so I'll jump in
based on what I could see in my recent mail and the subject line:

-The ASTDB should have no impact on Asterisk service start (which I
assume is the problem given the subject line)

-If you disabled SElinux then that's not the problem in starting
asterisk

 

>From another posting it appears that you can start Asterisk from the binary,
and from safe_asterisk.  If that's correct, then are you able to start/stop
Asterisk from the service file?  With CentOS7 that would be:

 

systemctl start asterisk

 

Is your asterisk service file present?  (You can create one easily based on
samples on the internet).  If you have an asterisk service file but startup
fails post the relevant portion of your syslog (journalctl).

 

If your question has changed (you mentioned 'the first problem') then ignore
the above; jumping in late.

 
 
*Jason*
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Re: [asterisk-users] Having problem getting Asterisk to work on CentOS 7

[Telium Technical Support]

The history of the question is lost (in the mail thread) so I'll jump in
based on what I could see in my recent mail and the subject line:

-The ASTDB should have no impact on Asterisk service start (which I
assume is the problem given the subject line)

-If you disabled SElinux then that's not the problem in starting
asterisk

 

>From another posting it appears that you can start Asterisk from the binary,
and from safe_asterisk.  If that's correct, then are you able to start/stop
Asterisk from the service file?  With CentOS7 that would be:

 

systemctl start asterisk

 

Is your asterisk service file present?  (You can create one easily based on
samples on the internet).  If you have an asterisk service file but startup
fails post the relevant portion of your syslog (journalctl).

 

If your question has changed (you mentioned 'the first problem') then ignore
the above; jumping in late.

 
 
*Jason*
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Re: [asterisk-users] fail2ban Asterisk 13.13.1

[Telium Technical Support]

If this is a small site, I recommend you download the free version of SecAst
(www.telium.ca  ) and replace fail2ban.  SecAst does
NOT use the log file, or regexes, to match etc.instead it talks to Asterisk
through the AMI to extract security information.  Messing with regexes is a
losing battle, and the lag in reading logs can allow an attacker 100+
registration attempts before fail2ban even does anything (assuming the IP is
exposed in the Asterisk log).

 

If this is a large install then post in the commercial list for more
information.

 

-Raj-

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tech Support
Sent: Wednesday, March 1, 2017 2:37 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'

Subject: Re: [asterisk-users] fail2ban Asterisk 13.13.1

 

It's possible that you need to increase the value of 'findtime' to
something greater than 300 secs. You also may want to set "timestamp = yes"
in asterisk.conf so each line in the CLI will be time stamped. Time stamping
it will be the definitive determination on whether or not the 'findtime' is
the culprit.

Regards;

John V.  

 

From: asterisk-users-boun...@lists.digium.com

[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Motty Cruz
Sent: Wednesday, March 01, 2017 01:29 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] fail2ban Asterisk 13.13.1

 

Hello, fail2ban does not ban offending IP. 

 

NOTICE[29784] chan_sip.c: Registration from
'"user3"' failed for 'offending-IP:53417' - Wrong
password

NOTICE[29784] chan_sip.c: Registration from
'"user3"' failed for 'offending-IP:53911' - Wrong
password

 

 

# A host is banned if it has generated "maxretry" during the last "findtime"

# seconds.

findtime  = 300

 

[asterisk-iptables]

enable = true

port = 5060,5061

filter   = asterisk

action   = iptables-allports[name=ASTERISK, protocol=all]

  sendmail[name=ASTERISK, dest=mo...@email.com
 , sender=fail2...@asterisk-ip.com
 ]

#action   = %(banaction)s[name=%(__name__)s-tcp, port="%(port)s",
protocol="tcp", chain="%(chain)s", actname=%(banaction)s-tcp]

   %(banaction)s[name=%(__name__)s-udp, port="%(port)s",
protocol="udp", chain="%(chain)s", actname=%(banaction)s-udp]

   %(mta)s-whois[name=%(__name__)s, dest="%(destemail)s"]

logpath  = /var/log/asterisk/messages

maxretry = 3

findtime  = 300

bantime  = -1

 

 

in filter.d

asterisk.conf

failregex = ^%(__prefix_line)s%(log_prefix)s Registration from '[^']*'
failed for '(:\d+)?' - (Wrong password|Username/auth name mismatch|No
matching peer found|Not a local domain|Device does not match ACL|Peer is not
supposed to register|ACL error \(permit/deny\)|Not a local domain)$

^%(__prefix_line)s%(log_prefix)s Call from '[^']*'
\(:\d+\) to extension '[^']*' rejected because extension not found in
context

^%(__prefix_line)s%(log_prefix)s Host  failed to
authenticate as '[^']*'$

^%(__prefix_line)s%(log_prefix)s No registration for peer
'[^']*' \(from \)$

^%(__prefix_line)s%(log_prefix)s Host  failed MD5
authentication for '[^']*' \([^)]+\)$

^%(__prefix_line)s%(log_prefix)s Failed to authenticate
(user|device) [^@]+@\S*$

^%(__prefix_line)s%(log_prefix)s hacking attempt detected
''$

^%(__prefix_line)s%(log_prefix)s
SecurityEvent="(FailedACL|InvalidAccountID|ChallengeResponseFailed|InvalidPa
ssword)",EventTV="([\d-]+|%(iso8601)s)",Severity="[\w]+",Service="[\w]+",Eve
ntVersion="\d+",AccountID="(\d*|)",SessionID=".+",LocalAddress="IPV
[46]/(UDP|TCP|WS)/[\da-fA-F:.]+/\d+",RemoteAddress="IPV[46]/(UDP|TCP|WS)//\d+"(,Challenge="[\w/]+")?(,ReceivedChallenge="\w+")?(,Response="\w+",Ex
pectedResponse="\w*")?(,ReceivedHash="[\da-f]+")?(,ACLName="\w+")?$

^%(__prefix_line)s%(log_prefix)s "Rejecting unknown SIP
connection from "$

^%(__prefix_line)s%(log_prefix)s Request (?:'[^']*' )?from
'[^']*' failed for '(?::\d+)?'\s\(callid: [^\)]*\) - (?:No matching
endpoint found|Not match Endpoint(?: Contact)? ACL|(?:Failed|Error) to
authenticate)\s*$

 

failregex = NOTICE.* .*: Registration from '.*' failed for '' - Wrong
password

NOTICE.* .*: Registration from '.*' failed for ':.*' - No
matching peer found

NOTICE.* .*: Registration from '.*' failed for '' - No
matching peer found

NOTICE.* .*: Registration from '.*' failed for '' -
Username/auth name mismatch

NOTICE.* .*: Registration from '.*' failed for '' - Device
does not match ACL

NOTICE.* .*: Registration from '.*' failed for '' - Peer
is not supposed to register

NOTICE.* .*: Registration from 

[asterisk-users] Replacing PBX during a call in progress

[Telium Technical Support]

This was asked many years ago but I thought I would check to see if things
have changed.  Is it possible to take over a call in progress - using a
replacement Asterisk server?  

 

In other words, if 2 user agents are connected through an Asterisk PBX, and
I tracked the call ID, IP of each UA (and anything else needed), could I
remove the PBX and put a new one in its place (at the same IP address) and
resume the call?  Somehow keeping the call up on the UA's and telling
Asterisk to just resume a call given specified parameters (so the UA's
wouldn't notice the change)?

 

 

 

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Re: [asterisk-users] Issue command to force SIP client to re-register

[Telium Technical Support]

Thanks - I actually found the SIP notify command before, but the options
seem to force checking for new config (in which case reboot), or cold/warm
restart.

 

I was hoping to just force a re-register, not reboot.  (Which in this
particular case is a long interval which I cannot change, so need to force
re-register on demand).

 

-Raj-

 

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[asterisk-users] Issue command to force SIP client to re-register

[Telium Technical Support]

Is there a way to force a SIP client to re-register using a SIP command (or
an AMI command)?

 

If not, is there some other standard way to do so - or would I have to
post/get to a web GUI of the phone (unique to each model) to force a reset,
etc.

 

-Raj-

 

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[asterisk-users] AMI version in Asterisk 14

[Telium Technical Support]

I noticed that Asterisk 14 has changed the output format for some commands
(eg: "Output: ").  However, the AMI reports version 2.8.0 which is the same
as Asterisk 13

 

Is that considered a bug?  Since the AMI output format has changed,
shouldn't the AMI version be incremented?  (Makes is hard for developers to
maintain compatability)

 

Hans

 

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Re: [asterisk-users] RS485 Audio device

[Telium Technical Support]

Forget RS485 at that distance (your throughput will be too low).  I would 
suggest you pull a fiber and just create an LAN connection on the end.

 

I’m sure you would have had fun getting some of the old IP over Serial drivers 
working J

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis
Sent: Wednesday, November 02, 2016 3:46 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] RS485 Audio device

 

Hi All,

 

The reason for the question was simply that the customer desired some solution

called an "AOR" or Area of refuge - I think it was. Basically a call button, 
microphone and speaker to hear back

with the kicker being "a long distance" the solution has to run.  RS485 is like 
4000 feet.

 

There are solutions our there apparently that are not built on asterisk - so I 
was just trying to find

a solution that potentially worked with asterisk. 

 

Thanks! 

 

Jerry

 

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Re: [asterisk-users] RS485 Audio device

[Telium Technical Support]

This one caught my interest too...more out of curiosity!  Keep in mind that
RS485 max speed drops to 100kbps after a relatively short distance.  And,
100kbps is RAW speed.  If you encapsulated your audio stream in that you'd
lose another 10%.

So why are you doing this?  If you are running a 100m cable (4 wire +
shield) why not just pull at cat5/6 cable instead?  Or just send analog
audio over 2 of the wires with the shield to keep out hum.   If there is a
need to use rs485 you could stream your audio over that connection - but I'm
curious why first.

We did some work for broadcast (radio station) doing AoIP and converting
some analog feeds but this seems unusual.

Jason

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matt
Fredrickson
Sent: Wednesday, November 02, 2016 10:33 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] RS485 Audio device

On Fri, Oct 28, 2016 at 7:09 PM, Jerry Geis  wrote:
> Hi All,
>
> Is there any devices or pair of devices that do audio over RS485
> and then convert to SIP for us in asterisk?
> Of course a speaker and push button at the other end.
>
> Is there anything like that out there?

Ok, I'll bite.  How does one do audio over RS485?

I've never worked with RS485, but from some brief googling it looks
like it's a fancy version of RS232.  I'm not sure where you'd get
(analog) audio from on RS232.

-- 
Matthew Fredrickson
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

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[asterisk-users] SIP show peers content

[Telium Technical Support]

When I issue a 'sip show peers' command the left most column is titled
'name/username'.  Some lines show one item in the column like 123, others
show bob/123.  Can someone explain the difference? (What does does name vs
username mean)

 

And why does 'sip show users' not show a name column title?

 

Thanks!

 

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Re: [asterisk-users] IAX UNREACHABLE : Ignoring bindport/bindaddr on reload

[Telium Technical Support]

Possibly - I noticed this thread only in the context of an IAX problem.  I 
can't speak to UCARP 

If you're trying to my a high availability cluster out of Asterisk servers have 
a look at 
http://serverfault.com/questions/733403/high-availability-asterisk-options/733441#733441


-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vitor Mazuco
Sent: Monday, August 29, 2016 11:38 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] IAX UNREACHABLE : Ignoring bindport/bindaddr on 
reload

Hummm, but why It is with that problem?

I use UCARP, maybe is this the problem?


2016-08-29 12:17 GMT-03:00, Telium Technical Support <supp...@telium.ca>:
> Oh!  In that case ignore it.
>
> Asterisk won't rebind the adapter if you've only changed parameters.  The
> message is misleading
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vitor Mazuco
> Sent: Monday, August 29, 2016 10:41 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] IAX UNREACHABLE : Ignoring bindport/bindaddr
> on reload
>
> Sorry,
>
> I just see warning.
>
>
>
> 2016-08-29 11:40 GMT-03:00, Vitor Mazuco <vitor.maz...@gmail.com>:
>> I just see  warning?
>>
>>
>> 2016-08-29 11:30 GMT-03:00, Telium Technical Support <supp...@telium.ca>:
>>> This shows that asterisk's IAX is already bound to all adapters - so
>>> that's
>>> good.  Symptomatically does your IAX stop working?  Or do you just see a
>>> warning?
>>>
>>> -Original Message-
>>> From: asterisk-users-boun...@lists.digium.com
>>> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vitor
>>> Mazuco
>>> Sent: Monday, August 29, 2016 8:46 AM
>>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>>> Subject: Re: [asterisk-users] IAX UNREACHABLE : Ignoring
>>> bindport/bindaddr
>>> on reload
>>>
>>> Hi, see the log below
>>>
>>> root@AsteriskSlave:~# ip addr
>>> 1: lo: <LOOPBACK,UP,LOWER_UP> mtu 65536 qdisc noqueue state UNKNOWN
>>> group default
>>> link/loopback 00:00:00:00:00:00 brd 00:00:00:00:00:00
>>> inet 127.0.0.1/8 scope host lo
>>>valid_lft forever preferred_lft forever
>>> inet6 ::1/128 scope host
>>>valid_lft forever preferred_lft forever
>>> 2: p3p1: <BROADCAST,MULTICAST> mtu 1500 qdisc noop state DOWN group
>>> default qlen 1000
>>> link/ether 00:13:3b:12:02:21 brd ff:ff:ff:ff:ff:ff
>>> 3: p4p1: <BROADCAST,MULTICAST,UP,LOWER_UP> mtu 1500 qdisc pfifo_fast
>>> state UP group default qlen 1000
>>> link/ether 00:e0:4c:44:01:95 brd ff:ff:ff:ff:ff:ff
>>> inet 192.168.25.25/24 brd 192.168.25.255 scope global p4p1
>>>valid_lft forever preferred_lft forever
>>> inet6 2804:7f1:4080:fe45:2e0:4cff:fe44:195/64 scope global dynamic
>>>valid_lft 86398sec preferred_lft 43198sec
>>> inet6 fe80::2e0:4cff:fe44:195/64 scope link
>>>valid_lft forever preferred_lft forever
>>> 4: p5p1: <BROADCAST,MULTICAST> mtu 1500 qdisc noop state DOWN group
>>> default qlen 1000
>>> link/ether 14:dd:a9:82:38:ff brd ff:ff:ff:ff:ff:ff
>>>
>>> and
>>>
>>> root@AsteriskSlave:~# netstat -anp | grep ast
>>> tcp0  0 0.0.0.0:20000.0.0.0:*
>>> OUÇA   2050/asterisk
>>> tcp0  0 0.0.0.0:53380.0.0.0:*
>>> OUÇA   2050/asterisk
>>> udp0  0 0.0.0.0:38180   0.0.0.0:*
>>>  2050/asterisk
>>> udp0  0 0.0.0.0:45200.0.0.0:*
>>>      2050/asterisk
>>> udp0  0 0.0.0.0:46590.0.0.0:*
>>>  2050/asterisk
>>> udp0  0 0.0.0.0:27270.0.0.0:*
>>>  2050/asterisk
>>> udp0  0 0.0.0.0:50000.0.0.0:*
>>>  2050/asterisk
>>> udp0  0 0.0.0.0:50890.0.0.0:*
>>>  2050/asterisk
>>> unix  2  [ ACC ] STREAM OUVINDO   484
>>> 2050/asterisk   /var/run/asterisk/asterisk.ctl
>>> unix  2  [ ] DGRAM116862050/asterisk
>>>
>>> 2016-08-26 19:21 GMT-03:00, Telium Technical Support <supp...@telium.ca>:
>>>> Could y

Re: [asterisk-users] IAX UNREACHABLE : Ignoring bindport/bindaddr on reload

[Telium Technical Support]

Oh!  In that case ignore it.

Asterisk won't rebind the adapter if you've only changed parameters.  The 
message is misleading

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vitor Mazuco
Sent: Monday, August 29, 2016 10:41 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] IAX UNREACHABLE : Ignoring bindport/bindaddr on 
reload

Sorry,

I just see warning.



2016-08-29 11:40 GMT-03:00, Vitor Mazuco <vitor.maz...@gmail.com>:
> I just see  warning?
>
>
> 2016-08-29 11:30 GMT-03:00, Telium Technical Support <supp...@telium.ca>:
>> This shows that asterisk's IAX is already bound to all adapters - so
>> that's
>> good.  Symptomatically does your IAX stop working?  Or do you just see a
>> warning?
>>
>> -Original Message-
>> From: asterisk-users-boun...@lists.digium.com
>> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vitor
>> Mazuco
>> Sent: Monday, August 29, 2016 8:46 AM
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> Subject: Re: [asterisk-users] IAX UNREACHABLE : Ignoring
>> bindport/bindaddr
>> on reload
>>
>> Hi, see the log below
>>
>> root@AsteriskSlave:~# ip addr
>> 1: lo: <LOOPBACK,UP,LOWER_UP> mtu 65536 qdisc noqueue state UNKNOWN
>> group default
>> link/loopback 00:00:00:00:00:00 brd 00:00:00:00:00:00
>> inet 127.0.0.1/8 scope host lo
>>valid_lft forever preferred_lft forever
>> inet6 ::1/128 scope host
>>valid_lft forever preferred_lft forever
>> 2: p3p1: <BROADCAST,MULTICAST> mtu 1500 qdisc noop state DOWN group
>> default qlen 1000
>> link/ether 00:13:3b:12:02:21 brd ff:ff:ff:ff:ff:ff
>> 3: p4p1: <BROADCAST,MULTICAST,UP,LOWER_UP> mtu 1500 qdisc pfifo_fast
>> state UP group default qlen 1000
>> link/ether 00:e0:4c:44:01:95 brd ff:ff:ff:ff:ff:ff
>> inet 192.168.25.25/24 brd 192.168.25.255 scope global p4p1
>>valid_lft forever preferred_lft forever
>> inet6 2804:7f1:4080:fe45:2e0:4cff:fe44:195/64 scope global dynamic
>>valid_lft 86398sec preferred_lft 43198sec
>> inet6 fe80::2e0:4cff:fe44:195/64 scope link
>>valid_lft forever preferred_lft forever
>> 4: p5p1: <BROADCAST,MULTICAST> mtu 1500 qdisc noop state DOWN group
>> default qlen 1000
>> link/ether 14:dd:a9:82:38:ff brd ff:ff:ff:ff:ff:ff
>>
>> and
>>
>> root@AsteriskSlave:~# netstat -anp | grep ast
>> tcp0  0 0.0.0.0:20000.0.0.0:*
>> OUÇA   2050/asterisk
>> tcp0  0 0.0.0.0:53380.0.0.0:*
>> OUÇA   2050/asterisk
>> udp0  0 0.0.0.0:38180   0.0.0.0:*
>>  2050/asterisk
>> udp0  0 0.0.0.0:45200.0.0.0:*
>>  2050/asterisk
>> udp0  0 0.0.0.0:46590.0.0.0:*
>>  2050/asterisk
>> udp0  0 0.0.0.0:27270.0.0.0:*
>>  2050/asterisk
>> udp    0  0 0.0.0.0:50000.0.0.0:*
>>  2050/asterisk
>> udp0  0 0.0.0.0:50890.0.0.0:*
>>  2050/asterisk
>> unix  2  [ ACC ] STREAM OUVINDO   484
>> 2050/asterisk   /var/run/asterisk/asterisk.ctl
>> unix  2  [ ] DGRAM116862050/asterisk
>>
>> 2016-08-26 19:21 GMT-03:00, Telium Technical Support <supp...@telium.ca>:
>>> Could you post the result of "ip addr" command, and "netstat -anp | grep
>>> ast" after the reload?
>>>
>>> I suspect something else is going on here...
>>>
>>>
>>>
>>> --
>>> _
>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>>
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>>>   http://www.asterisk.org/community/astricon-user-conference
>>>
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>>>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>>
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>>> To UNSUBSCRIBE or update options visit:
>>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
>>
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>>
>> Join the Asterisk Community at

Re: [asterisk-users] IAX UNREACHABLE : Ignoring bindport/bindaddr on reload

[Telium Technical Support]

This shows that asterisk's IAX is already bound to all adapters - so that's 
good.  Symptomatically does your IAX stop working?  Or do you just see a 
warning?

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vitor Mazuco
Sent: Monday, August 29, 2016 8:46 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] IAX UNREACHABLE : Ignoring bindport/bindaddr on 
reload

Hi, see the log below

root@AsteriskSlave:~# ip addr
1: lo: <LOOPBACK,UP,LOWER_UP> mtu 65536 qdisc noqueue state UNKNOWN
group default
link/loopback 00:00:00:00:00:00 brd 00:00:00:00:00:00
inet 127.0.0.1/8 scope host lo
   valid_lft forever preferred_lft forever
inet6 ::1/128 scope host
   valid_lft forever preferred_lft forever
2: p3p1: <BROADCAST,MULTICAST> mtu 1500 qdisc noop state DOWN group
default qlen 1000
link/ether 00:13:3b:12:02:21 brd ff:ff:ff:ff:ff:ff
3: p4p1: <BROADCAST,MULTICAST,UP,LOWER_UP> mtu 1500 qdisc pfifo_fast
state UP group default qlen 1000
link/ether 00:e0:4c:44:01:95 brd ff:ff:ff:ff:ff:ff
inet 192.168.25.25/24 brd 192.168.25.255 scope global p4p1
   valid_lft forever preferred_lft forever
inet6 2804:7f1:4080:fe45:2e0:4cff:fe44:195/64 scope global dynamic
   valid_lft 86398sec preferred_lft 43198sec
inet6 fe80::2e0:4cff:fe44:195/64 scope link
   valid_lft forever preferred_lft forever
4: p5p1: <BROADCAST,MULTICAST> mtu 1500 qdisc noop state DOWN group
default qlen 1000
link/ether 14:dd:a9:82:38:ff brd ff:ff:ff:ff:ff:ff

and

root@AsteriskSlave:~# netstat -anp | grep ast
tcp0  0 0.0.0.0:20000.0.0.0:*
OUÇA   2050/asterisk
tcp0  0 0.0.0.0:53380.0.0.0:*
OUÇA   2050/asterisk
udp0  0 0.0.0.0:38180   0.0.0.0:*
 2050/asterisk
udp0  0 0.0.0.0:45200.0.0.0:*
 2050/asterisk
udp0  0 0.0.0.0:46590.0.0.0:*
 2050/asterisk
udp0  0 0.0.0.0:27270.0.0.0:*
 2050/asterisk
udp0  0 0.0.0.0:50000.0.0.0:*
 2050/asterisk
udp0  0 0.0.0.0:50890.0.0.0:*
 2050/asterisk
unix  2  [ ACC ] STREAM OUVINDO   484
2050/asterisk   /var/run/asterisk/asterisk.ctl
unix  2  [ ] DGRAM116862050/asterisk

2016-08-26 19:21 GMT-03:00, Telium Technical Support <supp...@telium.ca>:
> Could you post the result of "ip addr" command, and "netstat -anp | grep
> ast" after the reload?
>
> I suspect something else is going on here...
>
>
>
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Re: [asterisk-users] IAX UNREACHABLE : Ignoring bindport/bindaddr on reload

[Telium Technical Support]

Could you post the result of "ip addr" command, and "netstat -anp | grep
ast" after the reload?

I suspect something else is going on here... 



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[asterisk-users] How does Ast use IP vs FQDN for SIP header fields

[Telium Technical Support]

We are working with an ISP that needs Asterisk to place a FQDN name in the
SIP 'FROM' and 'INVITE' fields - where Asterisk is currently using an IP
address.  A SIP trace shows the following from my Asterisk box:

 

INVITE sip:62351155@1.1.1.1 SIP/2.0

Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK49f1d30e

From: "MYNAM" ;tag=as3d9596b0

 

We tried adding hosts file entries mapping these IP's to hostname's but
Asterisk didn't use them.  Can someone suggest how to do this?

 

 

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Re: [asterisk-users] SPA112 flapping

[Telium Technical Support]

I would guess conflicting IP addresses.  It comes back up at regular
intervals, detects the conflict, and shuts down..

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Diehl
Sent: Friday, June 17, 2016 5:57 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] SPA112 flapping

Hi all,

I've got a device that seems to become unreachable for about 2 minutes,
every 
hour.  From what I can tell, it isn't due to network or server issues.  Any 
ideas?

TIA.


-- 
Mike Diehl
Diehlnet Communications, LLC.
Voice: (505) 903-5700
Fax: (505) 903-5701


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Re: [asterisk-users] asterisk admin interface

[Telium Technical Support]

I think only PJSIP and MWI support Sorcery – so that likely won’t do what’s 
being asked for…

 

And reading/writing a flat file should be even easier than learning the ARI

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Re: [asterisk-users] asterisk admin interface

[Telium Technical Support]

>Thanks Raj
>You are correct. Is there any open source application in that? 

Not that I know of – I think it’s getting too simplistic J  We created some C++ 
functions for our High Availability for Asterisk product (HAAst) which modify 
config files and extensions files, but it’s more work to adapt them than just 
write your own.  In a few hours of coding you should be able to have it done…

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Re: [asterisk-users] asterisk admin interface

[Telium Technical Support]

You don't mention a configuration generator (like Elastix/FreePBX) so I'll
assume you are using a plain old vanilla Asterisk installation.  In which
case all user/endpoint information is kept in config (ini) files, and no
user/endpoint manipulation is done through the CLI or AMI.

In this case a very simple solution is to modify the Asterisk config files
to add/remove users, then tell Asterisk to reload from the CLI/AMI.  And
that's it!

-Raj-


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Re: [asterisk-users] AMI: check if the user has a Mailbox

[Telium Technical Support]

I don't think the directory check method is reliable; a user can have a
mailbox but if no messages have been left then the directory structure may
not exist.  Through the AMI you can show peer/user information and I believe
it shows you the mailbox associated with user/peer.

-Raj-

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tech Support
Sent: Thursday, April 21, 2016 4:23 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] AMI: check if the user has a Mailbox

I don't think you are going to be able to get that information using the
AMI. You should be able to figure it out though by looking at the voicemail
directory structure in /var/spool/asterisk/voicemail// or
in your database if you're using real time. It's probably just as easy to
write a script that will look for the proper directory as it is to write a
script to query the AMI. 
Regards;
John V. 

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Luca
Bertoncello
Sent: Thursday, April 21, 2016 3:35 PM
To: Asterisk Users
Subject: [asterisk-users] AMI: check if the user has a Mailbox

Hi list!

On an Asterisk-Server I have some users. Just two of them have a Mailbox.
I want to write a little Web interface to manage many things and I'd like to
have a menu point for the voicemail, but just if the user has a Mailbox.

I found the AMI-Command MailboxStatus, but it does not return what I need,
since it returns 0 if the user has a Mailbox but no messages and if the user
has no Mailbox...

Could someone suggest me a way to get this information?

Thanks a lot!
Luca Bertoncello
(lucab...@lucabert.de)

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Re: [asterisk-users] Asterisk load balancing for TCP/SIP and RTP?

[Telium Technical Support]

Have a look at this page for HA ideas:

http://www.voip-info.org/wiki/view/Asterisk+High+Availability+Design

 

There are a lot of tradeoffs in design, and easy to confuse load balancing with 
HA

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tickling Contest
Sent: Tuesday, March 29, 2016 7:02 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Asterisk load balancing for TCP/SIP and RTP?

 

Has anyone fronted Asterisk with HAProxy? If so, what is a good production 
configuration for Asterisk? I need direct_media=yes (and so I have to LB RTP to 
the same Asterisk server).

 

If HAProxy is not a good solution, what other solutions do you propose?

 

Any insight appreciated.

 

 

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Re: [asterisk-users] Pass variable to voicemail script

[Telium Technical Support]

>If you are talking about the 'externnotify' parameter in
voicemail.conf, the variables are passed simply as @ARGV.

 

I'm referring to the mailcmd= setting in voicemail.conf.  Asterisk runs this
when emailing a voicemail (with attachment)

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Re: [asterisk-users] FAX Detection.

[Telium Technical Support]

Perhaps use T38 instead?  Would make your life a lot easier.  (And you can use 
a T38modem software).  

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Re: [asterisk-users] Panic Button SMS Asterisk Integration

[Telium Technical Support]

We integrated a variety of USB devices with Asterisk.  A number of ways to do 
so…the device interface has a lot to do with the how.

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Chris Nighswonger
Sent: Friday, February 05, 2016 7:51 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Panic Button SMS Asterisk Integration

 

Has anyone done any integration of USB, etc. panic buttons and Asterisk?

The basic idea would be to have a USB based panic button[1] along with a bit of 
code which would trigger a group SMS or perhaps a pre-recorded call to a group.

Kind regards,

Chris




[1]http://www.amazon.com/StealthSwitch-Pro-USB-Foot-Pedal/dp/B00MI6K77K

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Re: [asterisk-users] Failed to authenticate device 100

[Telium Technical Support]

The details of the source IP are available in the asterisk security log (if you 
have that enabled) – but that particular attack hides its address from the 
messages file.

 

It’s essential that you secure your PBX; there are options ranging from free to 
commercial.  Have a look at:

 

http://www.voip-info.org/wiki/view/Asterisk+security

 

It’s easy to get a $20,000 phone bill, so take securing your PBX seriously.

 

-M-

 

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Motty
Sent: Wednesday, December 02, 2015 1:12 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion; 
motty.c...@gmail.com
Subject: [asterisk-users] Failed to authenticate device 100

 

Hello, I continued to see this errors in the logs: 

[2015-12-02 10:05:57] NOTICE[19949]: chan_sip.c:23277 handle_request_invite: 
Failed to authenticate device 100  
;tag=10cdeaf7

how do I guard against this kinds of attacks? Also, to get the IP address from 
where this attack come from I use the following command "tcpdump -lni eth0 -f 
"udp port 5060" is there an easy way to get the attacker's IP? 

Thanks, 
Motty

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Re: [asterisk-users] Which router/firewall would you use for a virtual-PBX Asterisk installation?

[Telium Technical Support]

If you are focused on routing, we've used 4 Cisco SG300-28p in router mode -
economical way to handle vlans etc for ~100 POE phone sets,  (with GB
interconnects).  At the edge Cisco ASA-55xx work well, and we've done a few
deployments with Mikrotik routers that are quite inexpensive and performed
impresively for their cost.

>From a security standpoint, consider what happened last summer when hackers
found an exploit in the FreePBX web interface.  They rewrote the PBX
dialplan, disabled CDR's, and made unlimited calls to premium rate numbers.
This was a real wakeup call for FreePBX users who though Fail2Ban was a
security system, or CDR's could be used to catch compromised accounts.
Digium warns everyone that fail2ban is not a security system:
http://forums.asterisk.org/viewtopic.php?p=159984

If you don't want a security system on your PBX, see if your ITSP will limit
your account to $X/day, restrict routes, etc.  

There are also some great Astricon videos online where they invite speakers
to talk about security.  You'll see that fail2ban + A2Billing doesn't keep
out anyone except the script kiddies.



-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ernie Dunbar
Sent: Monday, November 23, 2015 2:17 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Which router/firewall would you use for a
virtual-PBX Asterisk installation?

Oh, don't worry about us going cheap on security. We use A2Billing 
(along with some Fail2Ban configuration for bad logins) to limit the 
number and cost of calls that can go out through a compromised SIP 
account, so that when, not *if*,  a customer's SIP account gets 
compromised, the attacker gets cut off at the knees before they can even 
get out the door. We've even added bogus connection charges on 
international calls that get removed before we bill our customers, to 
speed up the process and reduce our losses even further. Our customers 
are even happy that these billing limits are in place.

No, this is all about playing nice with our load balancing software and 
protecting databases and backend servers that have no business being 
available to the public. But mostly it's about the load balancer 
(IPTables on said servers can take care of "visible to the public). I 
just want to make sure that the router we use will play nice with 
Asterisk, since we've already seen network hardware that looks good on 
paper, but fails miserably in practice. In fact, we see it so often with 
individual customers' crap routers causing voice quality issues, that by 
default we don't trust simple math.

So here I am, asking everyone what router they use, and whether you're 
happy with the results when there's 100 simultaneous SIP calls in 
progress. I want to know what happens when the rubber hits the road.

On 2015-11-20 14:22, Telium Technical Support wrote:
> Well router and firewall are very different...it depends on what you 
> are
> trying to accomplish.
> 
> If you are trying to secure an Asterisk-based call center, get a real
> security product.  Look here for details:
> http://www.voip-info.org/wiki/view/Asterisk+security
> 
> This covers firewall, Asterisk lock-down, and Asterisk specific 
> security.
> The average break-in/fraud cost is $25,000 per day.  (watch the 
> Astricon
> videos for more details).  So going cheap on security isn't a smart 
> move for
> a commercial installation.
> 
> If you just want a router/switch, figure out the simultaneous call 
> capacity
> x codec demands in bps, and there is your backplane switching speed
> requirements.  Even with 100 simultaneous calls at g711, a lower end 
> Cisco
> (3xx) router/switch will have no problem.
> 
> -M-
> 
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ernie 
> Dunbar
> Sent: Friday, November 20, 2015 3:25 PM
> To: Asterisk Users
> Subject: [asterisk-users] Which router/firewall would you use for a
> virtual-PBX Asterisk installation?
> 
> Hi everyone.
> 
> We've got a fairly large base of customers who use our Asterisk server
> for phone service in a virtual PBX kind of way, where the server is
> security hardened and exposed to the internet for them to connect to
> remotely with SIP and IAX. It's certainly not the sort of affair where
> we're running it as a PBX just within the building. As a result, we see
> network traffic coming through eth0 between 512 Kbps and about 3.0 
> Mbps,
> depending on the time of day.
> 
> We haven't so far been using a hardware firewall/router on our server
> network, but it's becoming increasingly clear that we need to. We have
> enough experience to know that Asterisk is pretty sensitive when it
> comes to

Re: [asterisk-users] Which router/firewall would you use for a virtual-PBX Asterisk installation?

[Telium Technical Support]

Well router and firewall are very different...it depends on what you are
trying to accomplish.

If you are trying to secure an Asterisk-based call center, get a real
security product.  Look here for details:
http://www.voip-info.org/wiki/view/Asterisk+security

This covers firewall, Asterisk lock-down, and Asterisk specific security.
The average break-in/fraud cost is $25,000 per day.  (watch the Astricon
videos for more details).  So going cheap on security isn't a smart move for
a commercial installation.

If you just want a router/switch, figure out the simultaneous call capacity
x codec demands in bps, and there is your backplane switching speed
requirements.  Even with 100 simultaneous calls at g711, a lower end Cisco
(3xx) router/switch will have no problem.

-M-

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ernie Dunbar
Sent: Friday, November 20, 2015 3:25 PM
To: Asterisk Users
Subject: [asterisk-users] Which router/firewall would you use for a
virtual-PBX Asterisk installation?

Hi everyone.

We've got a fairly large base of customers who use our Asterisk server 
for phone service in a virtual PBX kind of way, where the server is 
security hardened and exposed to the internet for them to connect to 
remotely with SIP and IAX. It's certainly not the sort of affair where 
we're running it as a PBX just within the building. As a result, we see 
network traffic coming through eth0 between 512 Kbps and about 3.0 Mbps, 
depending on the time of day.

We haven't so far been using a hardware firewall/router on our server 
network, but it's becoming increasingly clear that we need to. We have 
enough experience to know that Asterisk is pretty sensitive when it 
comes to network hardware in our situation - we've had to replace one 
otherwise perfectly good 100 Mbps network switch because it simply 
wasn't able to keep up with the amount of streaming audio we put through 
it, and it badly affected voice quality. We have other traffic flowing 
through our server network too, including a significant amount of e-mail 
and web traffic, although that's not quite as sensitive to the quality 
of our network hardware.

If you've got these large requirements for Asterisk, I'd love to hear 
what you use for a router, and whether that router has met your needs. 
It would also be nice to hear about what kinds of routers to avoid that 
you may have tried in the past and found lacking.

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Re: [asterisk-users] Asterisk HA with heartbeat and systemd

[Telium Technical Support]

If you’re still in the planning stage, there’s a lot more to think about.  Your 
Asterisk failure detection will be very simplistic (is the process dead).  
Synchronization of data – without risking synchronization of corrupt data to a 
peer.  Prevent a deteriorating/failing peer corruption from corrupted the other 
peer (i.e. now shared resources).  Awareness of upstream (e.g.: route/network) 
failures making the peer unavailable – and how to detect that.  Etc. etc.

 

Here’s a good checklist of things to consider in your design:  
http://www.voip-info.org/wiki/view/Asterisk+High+Availability+Design

 

If you are building a small/home office HA then the free version of commercial 
tools may be the way to go.  If have a $0 budget but for a larger installation, 
use the design guide above to help figure out which compromises to make.  
(Heartbeat / Linux HA is better than nothing).

 

-M-

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Re: [asterisk-users] HA

[Technical Support]

Nicko,

 

Are you using HAAst (from www.telium.ca) for your Asterisk high availability?  
Or is this a DIY type HA solution?

 

If you are using HAAst then you might want to re-post on the commercial list.  
If you are just exploring HA solutions check out 
http://www.voip-info.org/wiki/view/Asterisk+High+Availability+Solutions

 

-M-

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Re: [asterisk-users] Fail2ban

[Technical Support]

I'm using the Fail2ban.  I configuration below. I want to try to
prevent the continuous password. Fail2ban password that does not
prevent this form. (Asterisk 1.8 / Elastix interface)

Is this a home/small installation?  If so try SecAst (from 
www.telium.ca) as a free drop in replacement for fail2ban.  You won't 
have to mess with regexes etc...and it should address the continuous 
password issue.


-Raj-

P.S. My opinions are my own and may not represent those of my employer.  
As an empolyee of Telium you can bet however that my opinions are biased :)


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[asterisk-users] Incoming calls get 488 error

[Technical Support]

I got a new SNOM M65 which works fine for outgoing calls, but incoming 
calls never ring at the handset.  I captured the SIP traffic and see 
that my M65 is replying with an 488 not acceptable here.  From what I 
read this is usually codec related but both asterisk and the M65 are set 
for ulaw as first choice.


I have a SIP trace below.  Can someone suggest why the 488 is being 
generated?


---

Received from udp:192.168.253.4:5060 at 21/08/2015 18:37:00  (1198 bytes)

INVITE sip:290006@192.168.253.20;line=14994 SIP/2.0
Via: SIP/2.0/UDP 192.168.253.4:5060;branch=z9hG4bK4990095a
Max-Forwards: 70
From: test user sip:230@192.168.253.4;tag=as7b616c8d
To: sip:290006@192.168.253.20;line=14994
Contact: sip:230@192.168.253.4:5060
Call-ID: 36334383058109cd2325341a0f18ac79@192.168.253.4:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.11.0(11.10.2)
Date: Fri, 21 Aug 2015 22:37:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, 
INFO, PUBLISH, MESSAGE

Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 606

v=0
o=root 1678280845 1678280845 IN IP4 192.168.253.4
s=Asterisk PBX 11.10.2
c=IN IP4 192.168.253.4
b=CT:384
t=0 0
m=audio 18090 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
m=video 12226 RTP/AVP 99 98 34 31
a=rtpmap:99 H264/9
a=fmtp:99 
redundant-pic-cap=0;parameter-add=0;packetization-mode=0;level-asymmetry-allowed=0

a=rtpmap:98 H263-1998/9
a=fmtp:98 F=0;I=0;J=0;T=0;K=0;N=0;BPP=0;HRD=0
a=rtpmap:34 H263/9
a=fmtp:34 F=0;I=0;J=0;T=0;K=0;N=0;BPP=0;HRD=0
a=rtpmap:31 H261/9
a=sendrecv


Sent to udp:192.168.253.4:5060 at 21/08/2015 18:37:00  (280 bytes)

SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.253.4:5060;branch=z9hG4bK4990095a
From: test user sip:230@192.168.253.4;tag=as7b616c8d
To: sip:290006@192.168.253.20;line=14994
Call-ID: 36334383058109cd2325341a0f18ac79@192.168.253.4:5060
CSeq: 102 INVITE
Content-Length: 0



Sent to udp:192.168.253.4:5060 at 21/08/2015 18:37:00  (441 bytes)

SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/UDP 192.168.253.4:5060;branch=z9hG4bK4990095a
Max-Forwards: 70
From: test user sip:230@192.168.253.4;tag=as7b616c8d
To: sip:290006@192.168.253.20;line=14994;tag=ld65q
Call-ID: 36334383058109cd2325341a0f18ac79@192.168.253.4:5060
CSeq: 102 INVITE
Contact: sip:290006@192.168.253.20;line=14994
User-Agent: snomM700/03.24.0007 (MAC=0004136103FB; SER= 0; HW=255)
Content-Length: 0



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Re: [asterisk-users] Looking for PRI Card with automatic fail over

[Technical Support]

: 201508040847.02341.asterisk_l...@earthshod.co.uk
mailto:201508040847.02341.asterisk_l...@earthshod.co.uk
Content-Type: Text/Plain;  charset=utf-8

On Monday 03 Aug 2015, Eric Klein wrote:
 Hi all,

 Strange request, I have a customer where we are putting an
Asterisk PBX in
 front of a legacy (non-VoIP) PBX. One of the requirements it
that the
 Asterisk PBX have 2 PRI ports (on towards the legacy PBX and one
towards
 the carrier) with the ability to go to pass through should the
Asterisk PBX
 (software or hardware level) fail.

 I did not see this feature in the Digium, Sangoma, Allo, or
OpenVox cards.

 Does anyone know of a card that will do this? I know that Digium
has an
 external box (the r850) that does something similar for 2 PBXs
making them
 high availability, but in this case I only have the 1 Asterisk
box acting
 as a gateway and passing some calls out over SIP and IAX2.

 Any suggestions would be appreciated.

Use a 4-pole change-over relay to switch the PRI connection either
to the
Asterisk box if it gets some sort of heartbeat signal from the
computer
(say, toggling one of the lines of a printer port, if the
motherboard still
has one),  or the old PABX?

You might have to do some mean, down and dirty low-level
programming, to embed
your heartbeat-generating code in Asterisk's idle loop; but the
Source Code is
out there, if you fancy the challenge .

--
AJS

Note:  Originating address only accepts e-mail from list! If
replying off-
list, change address to asterisk1list at earthshod dot co dot uk .



--

Message: 4
Date: Tue, 4 Aug 2015 08:23:47 -0400
From: Technical Support supp...@telium.ca mailto:supp...@telium.ca
To: asterisk-users@lists.digium.com
mailto:asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Looking for PRI Card with automatic fail
over
Message-ID: 55c0aed3.4030...@telium.ca
mailto:55c0aed3.4030...@telium.ca
Content-Type: text/plain; charset=windows-1252; format=flowed

On 8/4/2015 3:47 AM, A J Stiles wrote:
 On Monday 03 Aug 2015, Eric Klein wrote:
 Hi all,

 Strange request, I have a customer where we are putting an
Asterisk PBX in
 front of a legacy (non-VoIP) PBX. One of the requirements it
that the
 Asterisk PBX have 2 PRI ports (on towards the legacy PBX and
one towards
 the carrier) with the ability to go to pass through should the
Asterisk PBX
 (software or hardware level) fail.

 I did not see this feature in the Digium, Sangoma, Allo, or
OpenVox cards.

 Does anyone know of a card that will do this? I know that
Digium has an
 external box (the r850) that does something similar for 2 PBXs
making them
 high availability, but in this case I only have the 1 Asterisk
box acting
 as a gateway and passing some calls out over SIP and IAX2.

 Any suggestions would be appreciated.
 Use a 4-pole change-over relay to switch the PRI connection
either to the
 Asterisk box if it gets some sort of heartbeat signal from the
computer
 (say, toggling one of the lines of a printer port, if the
motherboard still
 has one),  or the old PABX?

 You might have to do some mean, down and dirty low-level
programming, to embed
 your heartbeat-generating code in Asterisk's idle loop; but the
Source Code is
 out there, if you fancy the challenge .

Building on the answer above, have a look at ESL labs - who make
such a
relay that can bypass the PRI to the Asterisk server.  As well, have a
look at HAAst (www.telium.ca http://www.telium.ca) which can
monitor Asterisk and then
control the ESL relay to bypass Asterisk in case of failure.

-Raj-

P.S. My opinions do not necessarily reflect those of my employer. As I
am employed by Telium you can bet that me opinions are biased!




If you place an ESL PRI A/B switch on either side of the Asterisk server 
you can easily bypass the Asterisk server, or place it inline with the 
PRI link.


You can also chose to TAP the lines as noted above, but you have to 
ensure that both PRI cards don't go live at once...(if I understand that 
product correctly).
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Re: [asterisk-users] Looking for PRI Card with automatic fail over

[Technical Support]

On 8/4/2015 3:47 AM, A J Stiles wrote:

On Monday 03 Aug 2015, Eric Klein wrote:

Hi all,

Strange request, I have a customer where we are putting an Asterisk PBX in
front of a legacy (non-VoIP) PBX. One of the requirements it that the
Asterisk PBX have 2 PRI ports (on towards the legacy PBX and one towards
the carrier) with the ability to go to pass through should the Asterisk PBX
(software or hardware level) fail.

I did not see this feature in the Digium, Sangoma, Allo, or OpenVox cards.

Does anyone know of a card that will do this? I know that Digium has an
external box (the r850) that does something similar for 2 PBXs making them
high availability, but in this case I only have the 1 Asterisk box acting
as a gateway and passing some calls out over SIP and IAX2.

Any suggestions would be appreciated.

Use a 4-pole change-over relay to switch the PRI connection either to the
Asterisk box if it gets some sort of heartbeat signal from the computer
(say, toggling one of the lines of a printer port, if the motherboard still
has one),  or the old PABX?

You might have to do some mean, down and dirty low-level programming, to embed
your heartbeat-generating code in Asterisk's idle loop; but the Source Code is
out there, if you fancy the challenge .

Building on the answer above, have a look at ESL labs - who make such a 
relay that can bypass the PRI to the Asterisk server.  As well, have a 
look at HAAst (www.telium.ca) which can monitor Asterisk and then 
control the ESL relay to bypass Asterisk in case of failure.


-Raj-

P.S. My opinions do not necessarily reflect those of my employer. As I 
am employed by Telium you can bet that me opinions are biased!


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Re: [asterisk-users] Fwd: What database should I use, for simple data storing? SQLite or the buitin one?

[Technical Support]

To some extent the answer depends on how you want to use it overall, and 
what you already have installed.



We did something similar on a project where we created a simple app 
accessible via AGI, and it stored/retrieved data to/from anXML file.  If 
your access frequency is low enough that might be a good solution.  On 
the other hand if you need complex query capability you should stay on 
the SQL side.



 If you already have MySQL installed for other Asterisk features (eg: 
CDR, or if you use FreePBX) then you might as well use that.


​


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[asterisk-users] hello

[Safarifone Technical Support Hassan Caynte]

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[asterisk-users] Error loading module 'cdr_radius.so'

[Safarifone Noc Technical Support s]

I have this Error   Please Help me
 
 loader.c: Error loading module 'cdr_radius.so': libradiusclient-ng.so.2: 
cannot open shared object file: No such file or directory
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[asterisk-users] ITSP's no longer supporting IAX?

[OCG Technical Support]

After a variety of connectivity problems, my itsp (Unlimitel.ca) blamed the
problem on the IAX protocol.  They told me that as of Asterisk 1.4 the IAX
protocol went downhill and many carriers (like VoicePulse) are discontinuing
support for IAX.

 

Is this correct?  We are all heading for SIP?

 

Thanks,

MD

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Re: [asterisk-users] ITSP's no longer supporting IAX?

[OCG Technical Support]

I use simple port forwarding on an Linux firewall (iptables)...so that's not
the issue.

I was referring to IAX2 of course (IAX has be gone a long time I think)...
Unlimitel is running * 1.4.x (and so am I)...

I just can't understand IAX2 connections suddenly dropping (on one day)
being protocol issues (if no one changed their * versions).  Or is this how
IAX2 fails?

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dr. Michael J.
Chudobiak
Sent: March 25, 2009 9:29 AM
To: Asterisk Users List
Subject: Re: [asterisk-users] ITSP's no longer supporting IAX?

 The choice of router/NAT is critical though. Unlimitel recommended the
 SnapGear 560 to me, and it eliminated all the issues I was having with
 IAX going through my Sonicwall devices.

 Just another datapoint for you...
 Just curious.
 
 Since IAX only uses ONE port, do you have any idea what the technical
 reason behind a specific router would be critical?

Well, with a Sonicwall TZ170, you had to enabled Firewall  VOIP  
Enable consistent NAT, which was not the default setting.

Then, you had to figure out that Firewall  Advanced  Default UDP 
Connection Timeout defaulted to 30 seconds, less than the normal 
Asterisk 60 second registration timeout.

Then, for some reason, the TZ170 would simply lose packets. A fraction 
of calls would end almost immediately after they started, with Asterisk 
reporting a raw hangup error and INVAL packets, suggesting that some 
IAX2 packets were being lost, mis-ordered, or mis-translated.

Anyway, the Sonicwall TZ170 was totally unreliable for IAX2 connections. 
They caused me a lot of grief. Avoid them like the plague.

The Snapgear 560 just works, which I appreciate very much!


- Mike

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Re: [asterisk-users] How to verify availability of the DID connection?

[OCG Technical Support]

Robert,

We've helped clients setup monitoring scripts for this type of situation - 2
different ways.  One is a ping script, the other monitors the asterisk peer
status of registration.  These were temporary until they could get to the
root cause however.  Since you have multiple providers going down, I would
dig into the cause on your end...

What diagnostics have you done so far?

Michelle Dupuis
www.generationd.com



-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Robert
Augustyn
Sent: March 7, 2009 9:36 AM
To: Asterisk Users List
Subject: Re: [asterisk-users] How to verify availability of the DID
connection?

All these questions are valid, though I want first to see that the DID does
not work then I will go and try to resolve it.  
I do not have a specific issue at this moment.

Sincerely,
Robert Augustyn

p:519.997.3106 ext:802
m:519.817.2503
www.linqone.com
 

-Original Message-
From: asterisk@sedwards.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Sent: Friday, March 06, 2009 9:45 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to verify availability of the DID
connection?

On Thu, 5 Mar 2009, Robert Augustyn wrote:

 Occasionally, DIDs from different providers stop working for some 
 reason.

 I would like to be able to monitor situations like that and react 
 before any of my clients start going ballistic on me.

Are you losing DIDs that terminate on your Asterisk box or your clients
Asterisk box?

Are these DIDs registering with Asterisk and are you re-registering often
enough?

Is it a problem within the providers? Can you port the DIDs to another
provider?

Why do the DIDs stop working? Is is a connectivity problem you could detect
with something like ping or Nagios?

Since you say different providers I'm thinking a general connectivity
problem or something generally out of whack with registrations.

Thanks in advance,

Steve Edwards  sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000

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Re: [asterisk-users] How to verify availability of the DID connection?

[OCG Technical Support]

There's a defaultexpirey setting in sip.conf but I wouldn't go there yet.  

Does your ping work sometimes and not other times?  Have you done
route/network diagnostics?

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Robert
Augustyn
Sent: March 7, 2009 3:39 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] How to verify availability of the DID
connection?
Importance: High

Thanks,
Well sometimes I have a situation that the trunk is registered but there is
no communication coming in.
So ping and looking for registration status does not work ...
When I run sip reload it starts working again ?
One difference is that I can see is the refresh on the registration is 585
and not the usual 105.
Can I adjust this down anywhere?


Sincerely,
Robert Augustyn

p:519.997.3106 ext:802
m:519.817.2503
www.linqone.com
 

-Original Message-
From: supp...@ocg.ca [mailto:asterisk-users-boun...@lists.digium.com] On
Behalf Of OCG Technical Support
Sent: Saturday, March 07, 2009 9:59 AM
To: 'Asterisk Users List'
Subject: Re: [asterisk-users] How to verify availability of the DID
connection?

Robert,

We've helped clients setup monitoring scripts for this type of situation - 2
different ways.  One is a ping script, the other monitors the asterisk peer
status of registration.  These were temporary until they could get to the
root cause however.  Since you have multiple providers going down, I would
dig into the cause on your end...

What diagnostics have you done so far?

Michelle Dupuis
www.generationd.com



-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Robert
Augustyn
Sent: March 7, 2009 9:36 AM
To: Asterisk Users List
Subject: Re: [asterisk-users] How to verify availability of the DID
connection?

All these questions are valid, though I want first to see that the DID does
not work then I will go and try to resolve it.  
I do not have a specific issue at this moment.

Sincerely,
Robert Augustyn

p:519.997.3106 ext:802
m:519.817.2503
www.linqone.com
 

-Original Message-
From: asterisk@sedwards.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Sent: Friday, March 06, 2009 9:45 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to verify availability of the DID
connection?

On Thu, 5 Mar 2009, Robert Augustyn wrote:

 Occasionally, DIDs from different providers stop working for some 
 reason.

 I would like to be able to monitor situations like that and react 
 before any of my clients start going ballistic on me.

Are you losing DIDs that terminate on your Asterisk box or your clients
Asterisk box?

Are these DIDs registering with Asterisk and are you re-registering often
enough?

Is it a problem within the providers? Can you port the DIDs to another
provider?

Why do the DIDs stop working? Is is a connectivity problem you could detect
with something like ping or Nagios?

Since you say different providers I'm thinking a general connectivity
problem or something generally out of whack with registrations.

Thanks in advance,

Steve Edwards  sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000

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Re: [asterisk-users] $20 Bounty

[OCG Technical Support]

Damn you for solving this before he upped the bounty by a pack of tictacs!!

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David
Backeberg
Sent: March 3, 2009 10:51 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] $20 Bounty

On Tue, Mar 3, 2009 at 10:25 PM, David Backeberg dbackeb...@gmail.com
wrote:

exten = 123,s,1 Playback(enterzipcode)
exten = 123,s,n Read(zip||5)
exten = 123,s,n System(wget http://pathtoyahooservice${zip} -o
forecast.txt)
exten = 123,s,n System(wget --post-file forecast.txt -o wav.url)
exten = 123,s,n System(wget --input-file wav.url -o voice.wav)
exten = 123,s,n Playback(voice)

exten = 123,h,1 Hangup

 On Tue, Mar 3, 2009 at 7:12 PM, Dean Collins d...@cognation.net wrote:
 I'll pay anyone a $20 bounty for someone to replicate the USA Asterisk
 Weather App on Tropo.

 All you have to do is violate the ToS on a few services:
 wget the weather from yahoo, for instance:
 http://weather.yahooapis.com/forecastrss?p=06513

 Conditions for New Haven, CT at 9:53 pm EST
 Current Conditions:
 Fair, 20 F
 Forecast:
 Tue - Clear. High: 25 Low: 13
 Wed - Mostly Sunny. High: 34 Low: 19

 do a wget post of that output from the previous wget to
 http://www.research.att.com/~ttsweb/tts/demo.php

 do a wget on the wav file that demo generates.

 It would be nicer if you record a prompt before asking for the
 zipcode, but it's not strictly necessary.

 You can paypal me the cash to my email. The legitimate license for
 ATT Natural Voices is more than $20, and nothing built into Asterisk
 for free is going to give you free-form text-to-speech.


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Re: [asterisk-users] How to generate core dump?

[OCG Technical Support]

Install a Microsoft product.

(Sorry I couldn't resist when I saw the subject)

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tzafrir Cohen
Sent: March 4, 2009 8:48 AM
To: Asterisk Users List
Subject: Re: [asterisk-users] How to generate core dump?

On Wed, Mar 04, 2009 at 09:32:43AM +0100, Klaus Darilion wrote:
 
 
 Mark Michelson schrieb:
  Ken D'Ambrosio wrote:
  Asterisk segfaulted on me the other day; how do I tell it to generate a
  core file so -- if it happens again -- I can attempt to debug?  I
looked
  in the obvious places in make menuconfig and didn't see anything
  appropriate.
 
  Thanks,
 
  -Ken
 
 
  
  Run Asterisk with the -g option and it will dump a core file if it
should crash.
 
 If you also want to specify the location/file name this can be useful 
 too (man core)
 
 echo /tmp/core.%p  /proc/sys/kernel/core_pattern

Hmm.. this way you can't tell which executable generated it . 

  echo /tmp/core.%e.%t  /proc/sys/kernel/core_pattern

Or maybe (untested)

 echo |/usr/local/sbin/core_handler '%e' '%s'

See the kernel documentation:

  http://kernel.org/doc/Documentation/sysctl/kernel.txt

This is handy for those of you with limited disk space. OTOH, it will
probably not work on legacy systems with kernel 2.6.18.

-- 
   Tzafrir Cohen
icq#16849755  jabber:tzafrir.co...@xorcom.com
+972-50-7952406   mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

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Re: [asterisk-users] $20 Bounty

[OCG Technical Support]

Perhaps if he threw in a paperclip and some tictacs people would respond...

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Sent: March 3, 2009 7:37 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] $20 Bounty

On Tue, 3 Mar 2009, Dean Collins wrote:

 I'll pay anyone a $20 bounty for someone to replicate the USA Asterisk
 Weather App on Tropo.

Wow. $20.

cricketcricketcricket :)

Thanks in advance,

Steve Edwards  sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000

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[asterisk-users] Windows Mobile MWI and asterisk

[OCG Technical Support]

Has anyone written a Windows Mobile app which gets the MWI info from a SIP
server, and updates the VM counter in the OS?

 

I'd like my PPC to show my voicemail count (and SIP MWI seems like the
easiest way)

 

 

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Re: [asterisk-users] strange text message:)

[OCG Technical Support]

Are you sure this is not just a standard SIP MWI message?

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Catalin S.
Sent: February 23, 2009 8:01 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] strange text message:)

is any chance to use this feature to send messages on this kind of phones?


On Tue, Feb 24, 2009 at 1:39 AM, David fire ddf...@gmail.com wrote:
 you are getting the info about the voicemail becausethe soft on your phone
 support it.
 in sip.conf you can find some parameters to send that info.
 in other soft phones like x-lite you will have the same info.
 David

 2009/2/23 Catalin S. jonsonpla...@gmail.com

 Hello guys,
 I recently observed that my asterisk sends me sms like messages on my
 phone (Nokia E71), I mean is SMS but is delivered some kind in-band
 though VoIP. Is strange because this messages contains informations
 about my voicemail and is sent by voicem...@mydomainxxx.com. I noticed
 that this messages appears every time when I logged in with my phone
 on my sip account. I'm interested about how can I send these messages
 with other information's or whatever I want to my terminals. Also I
 observed that works with Nokia E71 only. Maybe is because I updated
 some software on It , Not Firmware. Do you guys observed this too?
 Thank you for support.

 Catalin.

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 --
 (\__/)
 (='.'=)This is Bunny. Copy and paste bunny into your
 ()_()signature to help him gain world domination.


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Re: [asterisk-users] Pingable and Unreachable at the same time !

[OCG Technical Support]

Did you use the same screen name / name for the 2 SIP extensions you setup
on the one phone?  If so, some phones will confuse asterisk based on the SIP
header (in particular AASTRA phones).  If this is an Aastra phone, this is
probably the cause...

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
Sent: February 17, 2009 8:47 AM
To: Asterisk Users List
Subject: Re: [asterisk-users] Pingable and Unreachable at the same time !

 

 

2009/2/17 Marc STORCK msto...@voipgate.com

Asterisk doesn't use PING to check the STATUS, it uses a SIP OPTION message.


Yes.
I think that simply, in this case, the endpoint (SIP phone) is just broken :
it wouldn't reply to anything ...

I'm not 100% sure now, but wouldn't be surprised ...

 

Regards,

 

Marc

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
Sent: mardi 17 février 2009 14:06
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Pingable and Unreachable at the same time !

 

Hi,

Has anyone met something like this ?

dialor*CLI sip show peers
Name/username  HostDyn Nat ACL Port Status
7541/7541  (Unspecified)D  0UNKNOWN
7540/7540  (Unspecified)D  0UNKNOWN
7534/7534  (Unspecified)D  0UNKNOWN
7533/7533  (Unspecified)D  0UNKNOWN
7531/7531  192.168.100.199  D  5060 OK (10 ms)
7530/7530  192.168.100.196  D  5060 UNREACHABLE
patton/patton  192.168.100.52   D  5060 OK (33 ms)
trunk/trunk4ipbx   192.168.64.25060 OK (1 ms)
8 sip peers [Monitored: 3 online, 5 offline Unmonitored: 0 online, 0
offline]
dialor*CLI !ping 192.168.100.196
PING 192.168.100.196 (192.168.100.196) 56(84) bytes of data.
64 bytes from 192.168.100.196: icmp_seq=1 ttl=64 time=0.334 ms
64 bytes from 192.168.100.196: icmp_seq=2 ttl=64 time=0.305 ms
64 bytes from 192.168.100.196: icmp_seq=3 ttl=64 time=0.305 ms

Any explaination ?

Regards


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Re: [asterisk-users] Strange dialplan matching issue

[OCG Technical Support]

No, sorry, we match _XXX to jump to plant123

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Philipp
Kempgen
Sent: February 13, 2009 4:35 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] Strange dialplan matching issue
Importance: High

OCG Technical Support schrieb:
 We use extensions like plant201 and tunnel12 so it does work in 1.4

As a *pattern* (e.g. _plant2XX, _tunnel.)?

 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jose P.
 Espinal

 On extensions.conf.sample I see this:
 
 ; Extension names may be numbers, letters, or combinations
 ; thereof. If an extension name is prefixed by a '_'
 ; character, it is interpreted as a pattern rather than a
 ; literal.  In patterns, some characters have special meanings:
 ;
 ;   X - any digit from 0-9
 ;   Z - any digit from 1-9
 ;   N - any digit from 2-9
 ;   [1235-9] - any digit in the brackets (in this example,
1,2,3,5,6,7,8,9)
 ;   . - wildcard, matches anything remaining (e.g. _9011. matches
 ;   anything starting with 9011 excluding 9011 itself)
 ;   ! - wildcard, causes the matching process to complete as soon as
 ;   it can unambiguously determine that no other matches are possible
 
 Maybe after using '_' Asterisk is waiting for one of the above pattern 
 matching characters.
 
 a. The 'hilton-' part of your dialplan might not being considered valid, 
 and Asterisk *might* be trying to match the 'XX' part LITERALLY, and 
 would be trying to reach extension '2XX'
 
 exten = _hilton-2XX,1,Goto(hilton,${EXTEN:7},1)
 
 
 b. then, in:
 exten = hilton-203,1,Goto(hilton,${EXTEN:7},1)
 
 You provided the real extension number (after you take out the fist 7 
 digits).
 
 So, Asterisk reaches '203', etc.
 
 
 
 Try only using valid pattern matching characters in your dialplan to see 
 if it works.
 
 
 
 Chris Bagnall wrote:

 Wondering if anyone has come across this strange dialplan pattern
matching
 issue before:
 
 I have a context defined as follows (the plus simply implies it follows
on
 from an existing context in another #include - which, yes, has been
included
 first):
 [privatedundi](+)
 exten = _hilton-2XX,1,Goto(hilton,${EXTEN:7},1)
 
 When dialling hilton-202 from another box via IAX2, I get:
 NOTICE[3727]: chan_iax2.c:8085 socket_process: Rejected connect attempt
 from ip masked, request 'hilton-...@privatedundi' does not exist
 
 Changing the context to read as follows solves the problem immediately:
 [privatedundi](+)
 exten = hilton-201,1,Goto(hilton,${EXTEN:7},1)
 exten = hilton-202,1,Goto(hilton,${EXTEN:7},1)
 exten = hilton-203,1,Goto(hilton,${EXTEN:7},1)
 
 Dialling hilton-202 now works every time.
 
 The *really* strange thing is that I have a number of similar pattern
 matches, and all the others work fine, it's just this one that doesn't.
 
 The box in question is running 1.4.22, but I have had a similar issue in
 the past with a 1.2 box, so it does not appear to be version specific.


   Philipp Kempgen

-- 
AMOOCON 2009, May 4-5, Rostock / Germany   -  http://www.amoocon.de
Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied  -  http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
-- 

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Re: [asterisk-users] Strange dialplan matching issue

[OCG Technical Support]

We use extensions like plant201 and tunnel12 so it does work in 1.4

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jose P.
Espinal
Sent: February 11, 2009 10:16 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] Strange dialplan matching issue

On extensions.conf.sample I see this:

; Extension names may be numbers, letters, or combinations
; thereof. If an extension name is prefixed by a '_'
; character, it is interpreted as a pattern rather than a
; literal.  In patterns, some characters have special meanings:
;
;   X - any digit from 0-9
;   Z - any digit from 1-9
;   N - any digit from 2-9
;   [1235-9] - any digit in the brackets (in this example, 1,2,3,5,6,7,8,9)
;   . - wildcard, matches anything remaining (e.g. _9011. matches
;   anything starting with 9011 excluding 9011 itself)
;   ! - wildcard, causes the matching process to complete as soon as
;   it can unambiguously determine that no other matches are possible

Maybe after using '_' Asterisk is waiting for one of the above pattern 
matching characters.

a. The 'hilton-' part of your dialplan might not being considered valid, 
and Asterisk *might* be trying to match the 'XX' part LITERALLY, and 
would be trying to reach extension '2XX'

exten = _hilton-2XX,1,Goto(hilton,${EXTEN:7},1)


b. then, in:
exten = hilton-203,1,Goto(hilton,${EXTEN:7},1)

You provided the real extension number (after you take out the fist 7 
digits).

So, Asterisk reaches '203', etc.



Try only using valid pattern matching characters in your dialplan to see 
if it works.



Chris Bagnall wrote:
 Greetings list,
 
 Wondering if anyone has come across this strange dialplan pattern matching
issue before:
 
 I have a context defined as follows (the plus simply implies it follows on
from an existing context in another #include - which, yes, has been included
first):
 [privatedundi](+)
 exten = _hilton-2XX,1,Goto(hilton,${EXTEN:7},1)
 
 When dialling hilton-202 from another box via IAX2, I get:
 NOTICE[3727]: chan_iax2.c:8085 socket_process: Rejected connect attempt
from ip masked, request 'hilton-...@privatedundi' does not exist
 
 Changing the context to read as follows solves the problem immediately:
 [privatedundi](+)
 exten = hilton-201,1,Goto(hilton,${EXTEN:7},1)
 exten = hilton-202,1,Goto(hilton,${EXTEN:7},1)
 exten = hilton-203,1,Goto(hilton,${EXTEN:7},1)
 
 Dialling hilton-202 now works every time.
 
 The *really* strange thing is that I have a number of similar pattern
matches, and all the others work fine, it's just this one that doesn't.
 
 The box in question is running 1.4.22, but I have had a similar issue in
the past with a 1.2 box, so it does not appear to be version specific.
 
 Any thoughts?
 
 TIA.
 
 Regards,
 
 Chris
 
 
 
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-- 
Jose P. Espinal
http://www.eSlackware.com
IRC: [OFTC|FreeNode]
Khratos @ #slackware | #asterisk/-doc/-bugs


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Re: [asterisk-users] Aastra phone crashes with Asterisk 1.6

[OCG Technical Support]

Don't expect too much from Aastra.  In our previous dealings trying to
report serious bugs (like phone lockup/crash) to Aastra, they didn't want
the details, or they simply gave us canned answers which did no good.
(Superficial tech support)

We've moved away from Aastra for new installs, but we still have to support
old customers with Aastra


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Philipp
Kempgen
Sent: February 11, 2009 12:45 AM
To: Asterisk Users List
Subject: Re: [asterisk-users] Aastra phone crashes with Asterisk 1.6

Carlos Chavez schrieb:
   I upgraded my office server from 1.4.22 to 1.6.0.5 on the weekend
and
 after some testing there seems to be a compatibility problem when using
 Aastra phones.

 If I dial any of those phones the
 call will drop after a minute or so and the phone will crash.

I'm not saying it's not an Asterisk problem. Maybe something in
the SIP signaling/RTP is broken.

However it's definitely an Aastra problem. No matter how broken
the signaling -- that's no excuse for crashing. So make sure to
report the issue to Aastra as well.


   Philipp Kempgen

-- 
AMOOCON 2009, May 4-5, Rostock / Germany   -  http://www.amoocon.de
Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied  -  http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
-- 

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Re: [asterisk-users] Contact lookup

[OCG Technical Support]

Have a look at smartCID at www.generationd.com

Uses a simple mySQL database, allows for call blocking flag, reverse CID
lookup, etc.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gordon
Henderson
Sent: February 3, 2009 11:51 AM
To: Asterisk Users List
Subject: Re: [asterisk-users] Contact lookup

On Tue, 3 Feb 2009, Geoff Lane wrote:

 Hi All,

 Asterisk 1.4.12 on CentOS 5

 I'd like to be able to look up each incoming CLI to retrieve an
 associated name, if available, and then pass that to the extensions so
 that they can see both the name and number of the caller. I'm not
 after LDAP or anything else maintained externally, just a contact
 lookup for my system.

 I suspect that Astdb could be used for this, as could a relational
 database like MySQL or postgres (accessed via AGI?) Probably simpler
 would be to maintain a text configuration file since I'm only
 concerned about less than a hundred entries initially.

 I'd appreciate insight into which is the easiest way to do this, and
 also any pointers to tutorials etc.

AstDB:

At it's very simplest:

exten = s,n,Set(CALLERID(name)=Unknown)
exten = s,n,Set(name=${DB(cid/${CALLERID(number)})})
exten = s,n,GotoIf($[${name} = ]?endCID)
exten = s,n,Set(CALLERID(name)=${name})
exten = s,n(endCID),Noop(fixCallerID - End of processing - returning
${CALLERID(all)})

... somewhere in the incoming processing. (This is an extract from an 
overly complcated macro I use) Things to check for - a name already being 
present - eg. on an incoming SIP call. No name in the astDB - might want 
to substitute Unknown ..

All you need to do now is populate the astDB - I use a web interface and 
some php to drive the manager interface...

My biggest site has just under 300 lookup entries... (Which presents other 
issues with the web interface, but ...)

Gordon


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Re: [asterisk-users] Quiet 24 port POE gig switch

[OCG Technical Support]

Check out the HP ProCurve Switch 2610-24-PWR

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Chris Bagnall
Sent: February 1, 2009 6:58 AM
To: Asterisk Users List
Subject: Re: [asterisk-users] Quiet 24 port POE gig switch

 I can find FANLESS 24 port PoE 10/100

That's an achievement in itself. Can you post details - I have quite a few
locations where that might be useful...

TIA.

Regards,

Chris



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Re: [asterisk-users] Quiet 24 port POE gig switch

[OCG Technical Support]

My google search says fanless...

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paul Hales
Sent: February 1, 2009 6:49 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] Quiet 24 port POE gig switch
Importance: High


My memory of a HP procurve (a 2626 PWR from memory) was that it was
quite noisy - have they changed?

PaulH


OCG Technical Support wrote:
 Check out the HP ProCurve Switch 2610-24-PWR

 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Chris
Bagnall
 Sent: February 1, 2009 6:58 AM
 To: Asterisk Users List
 Subject: Re: [asterisk-users] Quiet 24 port POE gig switch

   
 I can find FANLESS 24 port PoE 10/100
 

 That's an achievement in itself. Can you post details - I have quite a few
 locations where that might be useful...

 TIA.

 Regards,

 Chris



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[asterisk-users] Quiet 24 port POE gig switch

[OCG Technical Support]

A little off topic but

 

I need to put a 24 port Gig PoE switch into a small office - no computer
room / rack etc.  All CAT5 terminates near the owners desk (smart huh?).

 

I want to put a PoE switch in place, with 24 ports and Gig speed.  Everyone
I've researched so far is LOUD...

 

Anyone know of a quiet one?

 

Thanks

 

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[asterisk-users] vmail.cgi - permissions error help

[OCG Technical Support]

We always install native Asterisk (not when over the other packaged
versions)

 

I tried setting the SUID bit on the vmail.cgi file but that didn't help...so
I must be missing something.  Can someone else suggest a fix?

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gondar Monn
Sent: January 31, 2009 2:17 AM
To: Asterisk Users List
Subject: Re: [asterisk-users] Asterisk VoiceMail: Is there a web interface
for checking voicemail?
Importance: High

 

Have you tried FreePBX ? It allows Asterisk administration via web interface
and has module just for that.

G.

On Fri, Jan 30, 2009 at 7:56 PM, OCG Technical Support supp...@ocg.ca
wrote:

No - the server generates the error:

 

Software error:

Hrm, can't seem to open
/var/spool/asterisk/voicemail/internalextensions/230/Old/msg.wav

For help, please send mail to th

 

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Soonthorn
Ativanichayaphong
Sent: January 30, 2009 10:14 PM


To: Asterisk Users List
Subject: Re: [asterisk-users] Asterisk VoiceMail: Is there a web interface
for checking voicemail?

Importance: High

 

It might be browser security issues? Have you tried with different browsers?


On Fri, Jan 30, 2009 at 9:53 PM, OCG Technical Support supp...@ocg.ca
wrote:

Strangely, I can DELETE the messages from the web interface...just playback
causes a permission error...

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of OCG Technical
Support
Sent: January 30, 2009 9:40 PM


To: Asterisk Users List
Subject: Re: [asterisk-users] Asterisk VoiceMail: Is there a web interface
for checking voicemail?

 

I just tried the vmail.cgi app.  Although working, there is clearly a
permissions problem preventing playing the wav files.  

 

I run Fedora 8, and the patch files (on  the wiki) are apparently broken.
Does anyone have a solution for fedora?

 

Thanks

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Soonthorn
Ativanichayaphong
Sent: January 30, 2009 7:09 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] Asterisk VoiceMail: Is there a web interface
for checking voicemail?

 

Thank you Lenz. That's exactly what I'm looking for.

On Fri, Jan 30, 2009 at 12:21 PM, Lenz Emilitri lenz.lo...@gmail.com
wrote:

We use the default web interface, it comes with a file called vmail.cgi in
the defaiult Asterisk tree.

Thanks

l.

 

2009/1/30 Soonthorn Ativanichayaphong soonth...@yapinc.com

Hi,

I'm very new to Asterisk. I tried VoiceMail() application. I'm able to
modify extensions.conf and voicemail.conf to send a voicemail audio file to
my email. It works great so far.

Now, I'm looking into publishing those voicemail files on a web page.
According to http://www.voip-info.org/wiki-Asterisk+VoiceMail, it mentions
that Asterisk VoiceMail supports Web interface for checking of voicemail.
Does anyone know where I can find more information about this Web interface
for checking of voicemail feature?

If there is no such thing, what an alternative is?  Do you think writing my
own AGI script is the right way to go? Is there already an existing AGI
scripts or other things that I should install on top of Asterisk that
already do something like this?

Basically, I try to avoid writing a bunch of codes to retrieve voicemail
from my email and build a brand new voicemail web page. If there are
something I can reuse, I'd like to use them.

Thank you for your help.

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Soonthorn Ativanichayaphong
Software Engineer
Yap Inc.
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Re: [asterisk-users] Quiet 24 port POE gig switch

[OCG Technical Support]

I can find FANLESS 24 port PoE 10/100, or FANLESS 24 port non-POE
10/100/1000

I guess I'll just have to wait for newer chips..till then dropping down to
10/100

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Darrick
Hartman
Sent: January 31, 2009 10:02 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] Quiet 24 port POE gig switch
Importance: High

OCG Technical Support wrote:
 A little off topic but
 
  
 
 I need to put a 24 port Gig PoE switch into a small office - no computer 
 room / rack etc.  All CAT5 terminates near the owners desk (smart huh?).
 
  
 
 I want to put a PoE switch in place, with 24 ports and Gig speed.  
 Everyone I've researched so far is LOUD...

Chances of finding a PoE switch that is quiet out of the box is about as 
good as finding a government 'worker'.  It's kind of an oxymoron.

Of the switches I've used, the Linksys/Cisco line was the loudest. 
Dlink's were quieter, but still not something you'd want sitting next to 
a desk.  About the only fanless PoE switches I've seen are the smaller 
Netgear's, but they are not Gigabit.

Darrick

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Re: [asterisk-users] Asterisk VoiceMail: Is there a web interface for checking voicemail?

[OCG Technical Support]

I just tried the vmail.cgi app.  Although working, there is clearly a
permissions problem preventing playing the wav files.  

 

I run Fedora 8, and the patch files (on  the wiki) are apparently broken.
Does anyone have a solution for fedora?

 

Thanks

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Soonthorn
Ativanichayaphong
Sent: January 30, 2009 7:09 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] Asterisk VoiceMail: Is there a web interface
for checking voicemail?

 

Thank you Lenz. That's exactly what I'm looking for.

On Fri, Jan 30, 2009 at 12:21 PM, Lenz Emilitri lenz.lo...@gmail.com
wrote:

We use the default web interface, it comes with a file called vmail.cgi in
the defaiult Asterisk tree.

Thanks

l.

 

2009/1/30 Soonthorn Ativanichayaphong soonth...@yapinc.com

Hi,

I'm very new to Asterisk. I tried VoiceMail() application. I'm able to
modify extensions.conf and voicemail.conf to send a voicemail audio file to
my email. It works great so far.

Now, I'm looking into publishing those voicemail files on a web page.
According to http://www.voip-info.org/wiki-Asterisk+VoiceMail, it mentions
that Asterisk VoiceMail supports Web interface for checking of voicemail.
Does anyone know where I can find more information about this Web interface
for checking of voicemail feature?

If there is no such thing, what an alternative is?  Do you think writing my
own AGI script is the right way to go? Is there already an existing AGI
scripts or other things that I should install on top of Asterisk that
already do something like this?

Basically, I try to avoid writing a bunch of codes to retrieve voicemail
from my email and build a brand new voicemail web page. If there are
something I can reuse, I'd like to use them.

Thank you for your help.




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-- 
Thanks,
Soonthorn Ativanichayaphong
Software Engineer
Yap Inc.
--
Confidential  Privileged: This email message is for the sole use of the
intended recipient(s) and may contain confidential and privileged
information. Any unauthorized review, use, disclosure or distribution is
prohibited. If you are not the intended recipient, please contact the sender
by reply email and destroy all copies of the original message.



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