Re: [asterisk-users] Asterisk 13 / chan_sip / registration after reject

[List Support]

Hello

Le 04/12/2023 à 10:56, Benoit Panizzon (by way of Benoit Panizzon 
) a écrit :

Hi List

We have some CPE which run an embedded asterisk 13 with chan_sip.

Unfortunately, when a registration is rejected, those stop trying.

I am familiar with pjsip which allows to configure:

auth_rejection_permanent=no

How do I achieve the same with chan_sip?
We run a cron script each 10min who will check the registration state 
and send a register if not registered.


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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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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] Load issues using AGI

[Tech Support]

One other thing. I use a package called AGISpeedy. Its available for both 
Perl and PHP and I’ve used it for years without any problems. Its supposedly an 
order of magnitude faster than regular FastAGI scripts. The only downside is 
that it hasn’t been maintained a while, but the package is solid and really 
doesn’t need much maintaining. I’ll probably write the author and see if I can 
keep it updated for him.

Regards;

John 

 

From: asterisk-users  On Behalf Of 
Tech Support
Sent: Friday, September 20, 2019 1:37 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion' 

Subject: Re: [asterisk-users] Load issues using AGI

 

Hello;

What programming language are you using? If you are using Perl, then I can 
suggest this.

(1)You are going to have to profile your scripts. This is a must. Without 
profiling, you don’t know what the problem is. In my opinion, Devel::NYTProf is 
the king of that hill. It could be as simple as optimizing a single subroutine. 
Then you will know exactly what the problem is.  

(2)See which version of Perl are you running and see if upgrading it solves 
your problems. The easiest way is to download the newest Perl they support from 
ActiveState.com. It creates a completely independent installation in /opt which 
will not interfere with your system Perl. You can also compile and install the 
newest Perl from source completely separate from your system Perl. That’s what 
I do. I have a couple of scripts to automate that process. If you first get a 
list of your installed modules using ‘perlmod’, you can clean up the output a 
bit and pipe that to cpanm to make sure that you have all the modules you need. 

 

If you are using PHP, then I’m sure that the above still applies, but PHP is 
not my area of expertise. 

 

Regards;

John V.

 

 

From: asterisk-users mailto:asterisk-users-boun...@lists.digium.com> > On Behalf Of Jöran Vinzens
Sent: Friday, September 20, 2019 12:47 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion 
mailto:asterisk-users@lists.digium.com> >
Subject: [asterisk-users] Load issues using AGI

 

Hi all,

 

we have just upgraded from Asterisk 11 to Asterisk 16.

After porting all the config to 16 we figured out some major load problems.

 

the majority running of our Asterisk instances is still having Asterisk 11 so 
we can compare load handling on both versions.

On the same hardware configuration we see load differences that Asterisk 16 
takes four times the load as Asterisk 11 (on 11 we see load 0.5, on 16 we see 
something around 2).

 

Our asterisk is only handling Calls, so there are no Subscription no 
Registration etc.

 

After some testing we figured out if we eliminate AGI Apps from Dialplan we 
reduce the load significantly.

At the moment we have 6 AGI calls for one single call. If we eliminate 3 of 
them we reduce the load by half.

I also tried to have the AGI replaced by some fake AGI which returns 
immediately to make sure it is not related to long running AGI Scripts but this 
made no difference.

I tried to tweaked a bit manipulating the thread limitations in stasis.conf but 
it had no significant effect on load.

Also I tried to "decline" all non AGI messages in stasis.conf. It seems it has 
very little effect.

 

Does anyone have similar issues or a solution?

Is there anyone who calls AGI several times during call establishment?

 

any hin and help would be very much appreciated!

 

I am happy to share more config and information if it helps to find a solution.

 

 

-- 

Jöran Vinzens - vinz...@sipgate.de <mailto:vinz...@sipgate.de> 


sipgate GmbH - Gladbacher Str. 74 - 40219 Düsseldorf
HRB Düsseldorf 39841 - Geschäftsführer: Thilo Salmon, Tim Mois
Steuernummer: 106/5724/7147, Umsatzsteuer-ID: DE219349391

www.sipgate.de <http://www.sipgate.de>  - www.sipgate.co.uk 
<http://www.sipgate.co.uk> 
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Re: [asterisk-users] Load issues using AGI

[Tech Support]

Hello;

What programming language are you using? If you are using Perl, then I can 
suggest this.

(1)You are going to have to profile your scripts. This is a must. Without 
profiling, you don’t know what the problem is. In my opinion, Devel::NYTProf is 
the king of that hill. It could be as simple as optimizing a single subroutine. 
Then you will know exactly what the problem is.  

(2)See which version of Perl are you running and see if upgrading it solves 
your problems. The easiest way is to download the newest Perl they support from 
ActiveState.com. It creates a completely independent installation in /opt which 
will not interfere with your system Perl. You can also compile and install the 
newest Perl from source completely separate from your system Perl. That’s what 
I do. I have a couple of scripts to automate that process. If you first get a 
list of your installed modules using ‘perlmod’, you can clean up the output a 
bit and pipe that to cpanm to make sure that you have all the modules you need. 

 

If you are using PHP, then I’m sure that the above still applies, but PHP is 
not my area of expertise. 

 

Regards;

John V.

 

 

From: asterisk-users  On Behalf Of 
Jöran Vinzens
Sent: Friday, September 20, 2019 12:47 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Subject: [asterisk-users] Load issues using AGI

 

Hi all,

 

we have just upgraded from Asterisk 11 to Asterisk 16.

After porting all the config to 16 we figured out some major load problems.

 

the majority running of our Asterisk instances is still having Asterisk 11 so 
we can compare load handling on both versions.

On the same hardware configuration we see load differences that Asterisk 16 
takes four times the load as Asterisk 11 (on 11 we see load 0.5, on 16 we see 
something around 2).

 

Our asterisk is only handling Calls, so there are no Subscription no 
Registration etc.

 

After some testing we figured out if we eliminate AGI Apps from Dialplan we 
reduce the load significantly.

At the moment we have 6 AGI calls for one single call. If we eliminate 3 of 
them we reduce the load by half.

I also tried to have the AGI replaced by some fake AGI which returns 
immediately to make sure it is not related to long running AGI Scripts but this 
made no difference.

I tried to tweaked a bit manipulating the thread limitations in stasis.conf but 
it had no significant effect on load.

Also I tried to "decline" all non AGI messages in stasis.conf. It seems it has 
very little effect.

 

Does anyone have similar issues or a solution?

Is there anyone who calls AGI several times during call establishment?

 

any hin and help would be very much appreciated!

 

I am happy to share more config and information if it helps to find a solution.

 

 

-- 

Jöran Vinzens - vinz...@sipgate.de <mailto:vinz...@sipgate.de> 


sipgate GmbH - Gladbacher Str. 74 - 40219 Düsseldorf
HRB Düsseldorf 39841 - Geschäftsführer: Thilo Salmon, Tim Mois
Steuernummer: 106/5724/7147, Umsatzsteuer-ID: DE219349391

www.sipgate.de <http://www.sipgate.de>  - www.sipgate.co.uk 
<http://www.sipgate.co.uk> 
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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] Sending SMS and SIM card

[Tech Support]

I use VoIP Innovations and ThinQ (formally SIPRoutes) and they both support 
SMS. That way it’s very easy to write it into the dial plan. 

Regards;

John 

 

 

From: asterisk-users  On Behalf Of 
bilal ghayyad
Sent: Thursday, April 25, 2019 7:56 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Sending SMS and SIM card

 

Thank you Steve.

Regarding to Goip32 that you used it before: how you were handling the received 
messages?

In other words: if you sent a message for someone and he replied for you, how 
you were able to see the reply? And was it possible to have any action based on 
his reply (for example, forward the message to email)?

 

Also regarding to the Goip32: how you were sending the SMS messages? >From CRM 
connected to it or it was having a web interface for sending the SMS messages?

 

Was you able to use Goip32 for GSM voice calls (sending and receiving)?

 

Regards

Bilal

 

> Is it possible to send SMS from asterisk? Using DAHDI or using what is 

> possible?

 

You can use an SMS provider like Twillio.

 

 

> And, is there a card that can be fixed in the machine and insert the SIM 

> card in this card to be used for GSM calls and sending SMS through 

> asterisk? Through which channel? Is it DAHDI or something else?

 

I've never used an internal card, but what you're looking for is a GSM 

gateway.

 

I used a Goip32 (32 SIMs, 32 channels) a couple of years ago. It is an 

external box you hang on your network via Ethernet.

 

 

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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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[asterisk-users] Problem with AudioCodes MP-114 ATA

[Tech Support]

All;

I have an AudioCodes MP-114 four FXS ATA that recently stopped
registering to my PBX. I'm pulling my hair out here trying to figure out the
root cause without much success. Does anyone have a sample config file that
I could use as a sample? Any insight at all would be greatly appreciated. 

Thanks Much;

John  

 

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[asterisk-users] Disabling a trunk at runtime

[Telium Support Group]

I have an Asterisk system with 2 trunks (as shown below).  I need to be able
to disable a trunk at runtime. I may not change the dialplan but I can
change sip.conf and reload.

 

Any attempt to dial in the dialplan uses trunk A and trunk B in that order.
Normally calls will route through trunk A, but if I disable A I want calls
to go to trunk B.

 

Is there a creative way to effectively disable a trunk at runtime given
these parameters?  I don't think there is an "enabled" key-value pair for
sip.conf stanzas.  If I change the host key value to 0.0.0.0 and reload will
that effectively cause the dialplan to use trunk B?

 

 

[trunk_A]

context=from-trunk-sip-trunk_A

 

[trunk_A_in]

type=peer

qualify=yes

host=1.2.3.4

context=from-trunk

 

[trunk_B]

context=from-trunk-sip-trunk_B

 

[trunk_B_in]

type=peer

qualify=yes

host=1.2.3.4

context=from-trunk

 

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[asterisk-users] Call Queue Data

[Tech Support]

All;

A few years back, we put a heck of a lot of effort into developing a
software package to analyze call queue data that we want to open source.
It's a pretty good package and I would like to dust it off and resurrect it.
What I need to do that is have sample call queue data to test with. If
anyone has queue data they would be able to send me, I would be very, very
grateful to them. And I give people my word that if they can send some data,
I would immediately anonymize it. If that's not possible, then how can I
generate sample queue test data to work with? Are there any packages out
there that can generate this?

Thanks in Advance;

John V.

 

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Re: [asterisk-users] AGI timeout option

[Tech Support]

Hello;

I’ve been using AGISpeedy for Perl for years. I just works, and it works 
really well. I’m a Perl programmer, not a PHP programmer, but it seems to me 
that you need the PHP equivalent of Perl’s ‘alarm’ command.  Just a thought.

Regards;

John V.

 

From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On Behalf 
Of Patrick Wakano
Sent: Thursday, September 13, 2018 09:05 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Subject: [asterisk-users] AGI timeout option

 

Hello list,

Hope you all doing  well!

 

Recently, I had an issue with a FastAGI PHP script, which under some specific 
situation would run into an infinity loop, consuming all CPU resources. This 
also was preventing Asterisk to terminated the call properly because it was 
waiting for the AGI to return... The application uses AGIspeedy to process the 
AGI calls, not sure if this can be affecting this situation somehow

Due to this problem I started looking for some option to timeout the AGI call 
and return to the dialplan after XYZ seconds and so this would protect Asterisk 
preventing the dialplan to get stuck due to some external script problem that 
is actually outside of Asterisk control. Does Asterisk provide some control of 
this sort? I searched around and could not find any. 

Any idea is appreciated!

 

Kind regards

Patrick Wakano

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[asterisk-users] Asterisk 16 AMI changes

[Telium Support Group]

Does anyone know if Asterisk 16 includes changes to the AMI?  (syntax /
commands / etc)

 

I see a release candidate is forthcoming.  Just curious

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Re: [asterisk-users] getting invites to rtp ports ??

[Telium Support Group]

Actually even the Security log (and AMI security event) is nothing more than 
failed dial/register attempts against Asterisk.  There is no awareness of 
corrupt SIP attacks, detection of polling for insecure extensions, goefencing 
based on source IP (why allow connections from Russia if all of your uses are 
in Texas), detection of rapid dialing rates once connected to an IVR, etc.

 

So your entire security system is based on Asterisk saying a dial/register 
failed.  That’s a small fraction of the attack types against, and attack 
surface offered by, PJSIP/SIP/Asterisk.  Even worse, if you run a configuration 
generator (eg FreePBX)..well…do a google search to see the exploits that are 
published regularly.  I realize FreePBX/Sangoma now owns Digium so that 
discussion should probably go no further.

 

So don’t get me wrong….fail2ban is way better than nothing.  But it may instill 
a false sense of security.  And that was Digium’s point in the post.  So if the 
OP needs a free and fast solution against simple script kiddie attacks then 
installing fail2ban is a big thumbs up in my opinion.

 

There have been similar discussions in other groups as to why even have a 
firewall, since you can close ports not needed by your services.  There are 
some people who are very passionate about their view that firewalls are a waste 
of time and money.  Far be it from me to say they’re wrong…but I’ve tried to 
point them to some interesting articles.

 

If you are a pure open source advocate there are still a lot more tools you can 
use to secure you PBX.  Think SNORT, I think pfsense offers a free database 
that’s accurate to a country level, etc.  If you want commercial then there are 
even more options.  But that’s the wrong forum for the biz stuff

 

I feel I tread on the edge of a holy war :)  So I’ll leave my thoughts here and 
go no further

 

 

 

From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On Behalf 
Of Matthew Jordan
Sent: Wednesday, August 29, 2018 10:52 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Subject: Re: [asterisk-users] getting invites to rtp ports ??

 

 

On Wed, Aug 29, 2018 at 6:20 PM Telium Support Group mailto:supp...@telium.ca> > wrote:

Depending on log trolling (Asterisk security log) misses a lot, and also 
depends on the SIP/PJSIP folks to not change message structure (which has 
already happened numerous time).  If  you are comfortable hacking chan_sip.c 
you may prefer to get the same messages from the AMI.  It still misses a lot 
but that approach is better than nothing.

Digium warns not to use fail2ban / log trolling as a security system: 
http://forums.asterisk.org/viewtopic.php?p=159984



 

That's some pretty old advice.

 

The rationale for *not* using general log messages with fail2ban still stands: 
the general WARNING/NOTICE/etc. log messages are subject to change between 
versions, and no one wants that to impact someone's security. So you should not 
use those messages as input into fail2ban.

 

That rationale did lead to the 'security' event type in log messages. Security 
Event Logging - as it is called - got added into Asterisk quite some time ago. 
So long ago I'm really not sure which version. At a minimum, Asterisk 11, but 
I'm pretty sure it was in 10 as well.

 

Documentation for it can be found here:

 

https://wiki.asterisk.org/wiki/display/AST/Asterisk+Security+Event+Logger

 

And here:

 

https://wiki.asterisk.org/wiki/display/AST/Logging+Configuration

 

Note that this also fires off AMI events (and ARI events, IIRC).

 

If, for whatever reason, you do not get a SECURITY log message or a 
corresponding event when something 'bad' happens, that would be worth some 
additional discussion. If anything, the events can be a bit chatty...

 

 

 

 

-Original Message-
From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com 
<mailto:asterisk-users-boun...@lists.digium.com> ] On Behalf Of sean darcy
Sent: Wednesday, August 29, 2018 6:33 PM
To: asterisk-users@lists.digium.com <mailto:asterisk-users@lists.digium.com> 
Subject: Re: [asterisk-users] getting invites to rtp ports ??

On 08/29/2018 11:59 AM, Telium Support Group wrote:
> Block a single IP is the wrong approach (whack-a-mole).  You should consider 
> a more comprehensive approach to securing your VoIP environment.  Have a look 
> at this wiki:
> 
> https://www.voip-info.org/asterisk-security/
> 
> 
> 
> -Original Message-
> From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com 
> <mailto:asterisk-users-boun...@lists.digium.com> ] 
> On Behalf Of sean darcy
> Sent: Wednesday, August 29, 2018 10:46 AM
> To: asterisk-users@lists.digium.com <mailto:asterisk-users@lists.digium.com> 
> Subject: Re: [asterisk-users] getting invites to rtp ports ??
> 
> On 08/29/2018 09:42 AM, Carlos Rojas wrote:
>> Hi
>>
>

Re: [asterisk-users] getting invites to rtp ports ??

[Telium Support Group]

Depending on log trolling (Asterisk security log) misses a lot, and also 
depends on the SIP/PJSIP folks to not change message structure (which has 
already happened numerous time).  If  you are comfortable hacking chan_sip.c 
you may prefer to get the same messages from the AMI.  It still misses a lot 
but that approach is better than nothing.

Digium warns not to use fail2ban / log trolling as a security system: 
http://forums.asterisk.org/viewtopic.php?p=159984


-Original Message-
From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On Behalf 
Of sean darcy
Sent: Wednesday, August 29, 2018 6:33 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] getting invites to rtp ports ??

On 08/29/2018 11:59 AM, Telium Support Group wrote:
> Block a single IP is the wrong approach (whack-a-mole).  You should consider 
> a more comprehensive approach to securing your VoIP environment.  Have a look 
> at this wiki:
> 
> https://www.voip-info.org/asterisk-security/
> 
> 
> 
> -Original Message-
> From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] 
> On Behalf Of sean darcy
> Sent: Wednesday, August 29, 2018 10:46 AM
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] getting invites to rtp ports ??
> 
> On 08/29/2018 09:42 AM, Carlos Rojas wrote:
>> Hi
>>
>> Probably somebody is trying to hack your system, you should block 
>> that ip on your firewall.
>>
>> Regards
>>
>> On Wed, Aug 29, 2018 at 9:34 AM, sean darcy > <mailto:seandar...@gmail.com>> wrote:
>>
>>  I'm getting invites to very high ports every 30 seconds from a
>>  particular ip address:
>>
>>  Retransmitting #10 (NAT) to 5.199.133.128:52734
>>  <http://5.199.133.128:52734>:
>>  SIP/2.0 401 Unauthorized
>>  Via: SIP/2.0/UDP
>>  
>> 0.0.0.0:52734;branch=z9hG4bK1207255353;received=5.199.133.128;rport=52734
>>  From: >  <mailto:sip%3A37120116780191250@67.80.191.250>>;tag=1872048972
>>  To: >  <mailto:sip%3A3712011972592181418@67.80.191.250>>;tag=as3a52e748
>>  Call-ID: 1504207870-295758084-609228182
>>  CSeq: 1 INVITE
>>  ...
>>  WARNING[150318]: chan_sip.c:4127 retrans_pkt: Timeout on
>>  1504207870-295758084-609228182...
>>
>>  I thought invites had to go to port 5060 or so. I don't understand
>>  why somebody (let's assume a bad guy) is trying ports above 5.
>>
>>  sean
>>
>>
> 
> Ok, so the high port is not the destination port but the source port.
> 
> So I hacked the log warning in chan_sip.c on non-critical invites to show the 
> source ip:
> 
> ast_log(LOG_WARNING, "Timeout on %s non-critic invite trans from 
> %s.\n",
> pkt->owner->callid,ast_sockaddr_stringify(sip_real_dst(pkt->owner)));
> 
> With that in the log, I'm now blocking the ip addresses.
> 
> Thanks,
> sean
> 
> 
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> 
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> https://community.asterisk.org/
> 

I agree. That's why I hacked chan_sip.c to get the addresses in the log.

I'm surprised they're not in the log by default. I must be the only person who 
gets these "non-critical invites".

sean



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Re: [asterisk-users] getting invites to rtp ports ??

[Telium Support Group]

Block a single IP is the wrong approach (whack-a-mole).  You should consider a 
more comprehensive approach to securing your VoIP environment.  Have a look at 
this wiki:

https://www.voip-info.org/asterisk-security/



-Original Message-
From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On Behalf 
Of sean darcy
Sent: Wednesday, August 29, 2018 10:46 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] getting invites to rtp ports ??

On 08/29/2018 09:42 AM, Carlos Rojas wrote:
> Hi
> 
> Probably somebody is trying to hack your system, you should block that 
> ip on your firewall.
> 
> Regards
> 
> On Wed, Aug 29, 2018 at 9:34 AM, sean darcy  > wrote:
> 
> I'm getting invites to very high ports every 30 seconds from a
> particular ip address:
> 
> Retransmitting #10 (NAT) to 5.199.133.128:52734
> :
> SIP/2.0 401 Unauthorized
> Via: SIP/2.0/UDP
> 0.0.0.0:52734;branch=z9hG4bK1207255353;received=5.199.133.128;rport=52734
> From:  >;tag=1872048972
> To:  >;tag=as3a52e748
> Call-ID: 1504207870-295758084-609228182
> CSeq: 1 INVITE
> ...
> WARNING[150318]: chan_sip.c:4127 retrans_pkt: Timeout on
> 1504207870-295758084-609228182...
> 
> I thought invites had to go to port 5060 or so. I don't understand
> why somebody (let's assume a bad guy) is trying ports above 5.
> 
> sean
> 
> 

Ok, so the high port is not the destination port but the source port.

So I hacked the log warning in chan_sip.c on non-critical invites to show the 
source ip:

ast_log(LOG_WARNING, "Timeout on %s non-critic invite trans from %s.\n", 
pkt->owner->callid,ast_sockaddr_stringify(sip_real_dst(pkt->owner)));

With that in the log, I'm now blocking the ip addresses.

Thanks,
sean


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[asterisk-users] Passing arguments to the 'mailcmd' option in voicemail.conf

[Tech Support]

All;

I'd like to change the default command that is used to send email when a
person has a new voicemail. I believe that's set in voicemail.conf as the
'mailcmd' option. The default is to use the /usr/sbin/sendmail -t command. I
wrote a quick test script to see what arguments are passed to the command
(@ARGV), but no arguments were sent. So my question is this, how are
arguments sent to the 'mailcmd' command?

Thanks in Advance;

John V.

 

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[asterisk-users] Core show channels concise = deprecated

[Telium Support Group]

I want to get a list of all active channels from the AM.  I've been using
'core show channels concise' (as a commands from the AMI) but I see in the
documentation that the command is deprecated and will be removed.

 

What's the best way to get the equivalent from the AMI?

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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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[asterisk-users] Looking for C library for the Asterisk AMI

[Tech Support]

All;

We do a lot of programming and customizations for Asterisk and normally,
we do everything in Perl. For that, we use the CPAN module Asterisk::AMI,
and it works great. However, we have several programs that would benefit
greatly if they were written in C. So my question is this: Does anyone know
of a C library used to access and communicate with the AMI? Any insight at
all would be a big help.

Thanks;

John V.

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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] Search for (multi tenant) fax to mail solution

[Tech Support]

Hey;

We've open-sourced our fax server software if you want to take a look at
it. It's written in Perl so hack away at it as much as you want. It's got a
web-based interface and installs as a Webmin module, so there is no limit to
the number of users you can have. You are also not limited to the number of
concurrent fax channels you can use. You are only limited by Asterisk. In a
controlled test environment with some badass hardware, we've broken 1,000
concurrent fax channels. Way back in the day, we've also done a proof of
concept where we've implemented a main fax server to control multiple slave
fax servers. It was definitely a pretty cool project. There is also an email
to fax server built in. Most of the ITSP's we deal with use the email to fax
server for all their users for some reason. You can write me below for more
information. 

Regards;

John

 

John V.

John V., Tech Support

VoIP Business Solutions

240-215-3479 x325

 <mailto:f...@voipbusiness.us> supp...@voipbusiness.us

 

 

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michael Maier
Sent: Friday, February 23, 2018 12:00 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Search for (multi tenant) fax to mail solution

 

Hello!

 

I'm just searching for a fax to email / email to fax open source based
complete solution which covers the following core features:

 

- high availability

- possibly multi tenant

- about 40,000 users

- about 100 lines parallel

- supports G.711 / T.38

- scriptable user management (reading, adding and removing of accounts)

  (via REST)

- personal and default smtp address for each user: each user becomes a

  personal address and an additional default fall back address (if

  personal address doesn't work any more).

- Additional web interface for user maintenance

- Support of user based faxcovers

- paid enterprise support

- Linux (RHEL or SLES) based

 

Does anybody have any idea or experience and may suggest to look at the one
or other solution?

 

 

Thanks in advance,

Michael

 

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https://wiki.asterisk.org/wiki/display/AST/Getting+Started

 

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[asterisk-users] Getting the number of parked calls in a parking lot

[Tech Support]

All;

With Asterisk 11, it was trivial to get the number of parked calls in
each parking lot simply by issuing the "parkedcalls show" command. However,
with Asterisk 13, things are done very differently and the "parkedcalls
show" command no longer exists. So my question is, how do I get that
information with Asterisk 13?

Thanks Again;

John V  

 

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Re: [asterisk-users] AMD and fax detection

[Tech Support]

I think that there is a much easier way to detect if the far end is a fax.
In your dialplan, include a section that uses the 'fax' extension. If the
call jumps to that extension, then the far end is a fax machine.
Regards;
John V.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Neil Youngman
Sent: Friday, February 09, 2018 09:10 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] AMD and fax detection

Is it feasible to enhance AMD to detect and report if the far end sends fax
tones?

I am guessing that, as it is using DSP to detect sounds and periods of
silence, the same DSP can also report if a CED tone is sent.

If it is feasible, is there a document describing the DSP interface that
someone who is not familiar with DSP can use to get started?

Neil Youngman


Neil Youngman
Developer
Wirefast Limited

Wirefast provides secure corporate messaging services.
See our messaging solutions at  http://www.wirefast.com/ Please consider the
environment.
Does this email or attachment need to be printed?
This message contains confidential information and is intended only for the
individual named. If you are not the named addressee you should not
disseminate, distribute or copy this email. Please notify the sender
immediately by email if you have received this email by mistake and delete
this email from your system.

Any views or opinions are solely those of the author and do not necessarily
represent those of Wirefast Limited

Email transmission cannot be guaranteed to be secure or error-free as
information could be intercepted, corrupted, lost, destroyed, arrive late or
incomplete, or contain viruses. The sender therefore does not accept
liability for any errors or omissions in the contents of this message which
arise as a result of email transmission.
Wirefast Limited is registered in England & Wales Company number: 03865860
Registered Office: 7/10 Chandos Street, Cavendish Square, London, W1G 9DQ

Wirefast definitions of classification can be found here:
www.wirefast.com/classifications

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[asterisk-users] Forwarding a call "off pbx"

[Tech Support]

All;

I had someone ask me if they received an incoming phone call and it was
forwarded "off pbx" to their cell phone, would the call be strictly between
the caller and the cell phone, or would it between the caller, the pbx, and
the cell phone where the VoIP minutes continue to be charged. Rather than
pull an answer out of my butt, I tested it and found that it's a
three-legged call which really surprised me. So my question is, is that the
correct behavior?

Thanks Again;

John V.

 

 

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Re: [asterisk-users] Can't compile Asterisk on Fedora server

[Tech Support]

Fantastic. I can't thank you enough.
Regards;
John

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tzafrir Cohen
Sent: Wednesday, January 10, 2018 10:52 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Can't compile Asterisk on Fedora server

On Wed, Jan 10, 2018 at 10:28:42AM -0500, Tech Support wrote:
>  
> 
> All;
> 
> I have a Fedora 26 server that I am trying to compile
> asterisk-certified-13.13-cert6 on. However, I'm getting the following 
> errors. I'm also having a tough time trying to compile Dahdi. I'm not 
> sure what I'm missing, but if anyone else is running Fedora, I'd 
> really appreciate any help at all.
> 
> Thanks Much;
> 
> John V.
> 
>  
> 
> make[1]: Leaving directory
> '/usr/src/asterisk-certified-13.13-cert6/menuselect'
> 
>[CC] tcptls.c -> tcptls.o
> 
> tcptls.c: In function 'tcptls_stream_close':
> 
> tcptls.c:401:20: error: dereferencing pointer to incomplete type 'SSL 
> {aka struct ssl_st}'
> 
> if (!stream->ssl->server) {
> 
> ^~

When I want to bisect something, I know that if I need to go back far
enough, I need ./configure --without-ssl # :-(

Asterisk 13.14.0 includes basic OpenSSL 1.1.0 support. I have no idea if
anybody wants to backport it. Look at the log of main/tcptls.c in branch
13 in git to see the relevant patches. I suppose hopefully they'll apply
cleanly.

-- 
   Tzafrir Cohen
+972-50-7952406   mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com

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[asterisk-users] Can't compile Asterisk on Fedora server

[Tech Support]

 

All;

I have a Fedora 26 server that I am trying to compile
asterisk-certified-13.13-cert6 on. However, I'm getting the following
errors. I'm also having a tough time trying to compile Dahdi. I'm not sure
what I'm missing, but if anyone else is running Fedora, I'd really
appreciate any help at all.

Thanks Much;

John V.

 

make[1]: Leaving directory
'/usr/src/asterisk-certified-13.13-cert6/menuselect'

   [CC] tcptls.c -> tcptls.o

tcptls.c: In function 'tcptls_stream_close':

tcptls.c:401:20: error: dereferencing pointer to incomplete type 'SSL {aka
struct ssl_st}'

if (!stream->ssl->server) {

^~

tcptls.c:404:5: warning: 'ERR_remove_thread_state' is deprecated
[-Wdeprecated-declarations]

 ERR_remove_thread_state(NULL);

 ^~~

In file included from /usr/include/openssl/opensslconf.h:42:0,

 from /usr/include/openssl/ct.h:13,

 from /usr/include/openssl/ssl.h:61,

 from
/usr/src/asterisk-certified-13.13-cert6/include/asterisk/tcptls.h:66,

 from tcptls.c:44:

/usr/include/openssl/err.h:247:1: note: declared here

DEPRECATEDIN_1_1_0(void ERR_remove_thread_state(void *))

^

tcptls.c: In function '__ssl_setup':

tcptls.c:819:31: warning: implicit declaration of function
'SSLv2_client_method'; did you mean 'SSLv3_client_method'?
[-Wimplicit-function-declaration]

cfg->ssl_ctx = SSL_CTX_new(SSLv2_client_method());

   ^~~

   SSLv3_client_method

tcptls.c:819:31: warning: passing argument 1 of 'SSL_CTX_new' makes pointer
from integer without a cast [-Wint-conversion]

In file included from
/usr/src/asterisk-certified-13.13-cert6/include/asterisk/tcptls.h:66:0,

 from tcptls.c:44:

/usr/include/openssl/ssl.h:1338:17: note: expected 'const SSL_METHOD * {aka
const struct ssl_method_st *}' but argument is of type 'int'

__owur SSL_CTX *SSL_CTX_new(const SSL_METHOD *meth);

 ^~~

tcptls.c:825:4: warning: 'SSLv3_client_method' is deprecated
[-Wdeprecated-declarations]

cfg->ssl_ctx = SSL_CTX_new(SSLv3_client_method());

^~~

In file included from /usr/include/openssl/opensslconf.h:42:0,

 from /usr/include/openssl/ct.h:13,

 from /usr/include/openssl/ssl.h:61,

 from
/usr/src/asterisk-certified-13.13-cert6/include/asterisk/tcptls.h:66,

 from tcptls.c:44:

/usr/include/openssl/ssl.h:1616:1: note: declared here

DEPRECATEDIN_1_1_0(__owur const SSL_METHOD *SSLv3_client_method(void)) /*
SSLv3 */

^

tcptls.c:829:4: warning: 'TLSv1_client_method' is deprecated
[-Wdeprecated-declarations]

cfg->ssl_ctx = SSL_CTX_new(TLSv1_client_method());

^~~

In file included from /usr/include/openssl/opensslconf.h:42:0,

 from /usr/include/openssl/ct.h:13,

 from /usr/include/openssl/ssl.h:61,

 from
/usr/src/asterisk-certified-13.13-cert6/include/asterisk/tcptls.h:66,

 from tcptls.c:44:

/usr/include/openssl/ssl.h:1631:1: note: declared here

DEPRECATEDIN_1_1_0(__owur const SSL_METHOD *TLSv1_client_method(void)) /*
TLSv1.0 */

^

make[1]: *** [/usr/src/asterisk-certified-13.13-cert6/Makefile.rules:150:
tcptls.o] Error 1

make: *** [Makefile:402: main] Error 2

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Re: [asterisk-users] Showing CallerID on multiple phones

[Tech Support]

Hello;

I’m using Polycom VVX 600’s.

Thanks;

John V.

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tomáš Holý
Sent: Monday, December 11, 2017 09:22 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Showing CallerID on multiple phones

 

What kind of phones are you using? To some phones you can push text dialog over 
HTTP/XML. Investigate this features, maybe it is what are you looking for.

 

TH

 

 

Dne pondělí 11. prosince 2017 15:16:28 CET, Tech Support napsal(a):

Hello;

I certainly appreciate your response. In fact, I used that exact solution 
for three of the incoming lines. I setup ring groups and a silent ringtone for 
each phone. Unfortunately, the last incoming line is more complicated and uses 
an IVR with multiple input choices, so the solution is not as clear cut as for 
the other ones. That’s why I was trying to look at other options. 

Best Regards;

John V.

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ron Wheeler
Sent: Friday, December 08, 2017 03:06 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Showing CallerID on multiple phones

 

I could be way off track but have you looked at ring groups.

Have all of the phones ring (maybe a mute or special ring tone, if this is 
possible) so that everyone on the list of extensions sees the incoming call.
If no one picks it up by ring x, have it go to another phone or to voice mail.

 <https://www.freepbx.org/ring-group-and-follow-me-ring-strategies-1-of-2/> 
https://www.freepbx.org/ring-group-and-follow-me-ring-strategies-1-of-2/   
might be useful.

Ron


On 08/12/2017 2:17 PM, Tech Support wrote:

All;

I have an interesting scenario where I have a small office with maybe half 
a dozen phones and several incoming lines. The calls are routed based on the 
DID that people call. What they would like is when a call comes in to a single 
phone to have all the phones show the CallerID. That way they can decide if 
they should pick up the call or not using call pickup. I’ve been looking at 
products such as the one from Camrivox that interfaces with different CRM 
packages or Outlook, but I was wondering if a way was available to show the 
calls on their phones.

Thanks in Advance;

John V.

 

 

-- 
Ron Wheeler
President
Artifact Software Inc
email:  <mailto:rwhee...@artifact-software.com> rwhee...@artifact-software.com
skype: ronaldmwheeler
phone: 866-970-2435, ext 102

 

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Re: [asterisk-users] Showing CallerID on multiple phones

[Tech Support]

Hello;

I certainly appreciate your response. In fact, I used that exact
solution for three of the incoming lines. I setup ring groups and a silent
ringtone for each phone. Unfortunately, the last incoming line is more
complicated and uses an IVR with multiple input choices, so the solution is
not as clear cut as for the other ones. That's why I was trying to look at
other options. 

Best Regards;

John V.

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ron Wheeler
Sent: Friday, December 08, 2017 03:06 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Showing CallerID on multiple phones

 

I could be way off track but have you looked at ring groups.

Have all of the phones ring (maybe a mute or special ring tone, if this is
possible) so that everyone on the list of extensions sees the incoming call.
If no one picks it up by ring x, have it go to another phone or to voice
mail.

https://www.freepbx.org/ring-group-and-follow-me-ring-strategies-1-of-2/
might be useful.

Ron


On 08/12/2017 2:17 PM, Tech Support wrote:

All;

I have an interesting scenario where I have a small office with maybe
half a dozen phones and several incoming lines. The calls are routed based
on the DID that people call. What they would like is when a call comes in to
a single phone to have all the phones show the CallerID. That way they can
decide if they should pick up the call or not using call pickup. I've been
looking at products such as the one from Camrivox that interfaces with
different CRM packages or Outlook, but I was wondering if a way was
available to show the calls on their phones.

Thanks in Advance;

John V.





 

-- 
Ron Wheeler
President
Artifact Software Inc
email: rwhee...@artifact-software.com
skype: ronaldmwheeler
phone: 866-970-2435, ext 102
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[asterisk-users] Showing CallerID on multiple phones

[Tech Support]

All;

I have an interesting scenario where I have a small office with maybe
half a dozen phones and several incoming lines. The calls are routed based
on the DID that people call. What they would like is when a call comes in to
a single phone to have all the phones show the CallerID. That way they can
decide if they should pick up the call or not using call pickup. I've been
looking at products such as the one from Camrivox that interfaces with
different CRM packages or Outlook, but I was wondering if a way was
available to show the calls on their phones.

Thanks in Advance;

John V.

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Re: [asterisk-users] Can't park/unpark/re-park call

[Tech Support]

That was it. I can't thank you enough.
Regards;
John V.

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle
Sent: Tuesday, November 14, 2017 09:27 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Can't park/unpark/re-park call

>>> The problem lies when I go to park the call again, and nothing happens. I’m 
>>> running Asterisk 11. Any insight at all would be greatly appreciated. 

Check your 

/etc/asterisk/features.conf 

Look for the option: 

parkedcallreparking 

Doug

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[asterisk-users] Can't park/unpark/re-park call

[Tech Support]

All;

I'm having a problem with parking a call and I'm hoping that someone has
seen this problem before. A call comes in and I park it. A few seconds
later, I retrieve the call. So far, so good. The problem lies when I go to
park the call again, and nothing happens. I'm running Asterisk 11. Any
insight at all would be greatly appreciated.

Thanks;

John V.   

 

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Re: [asterisk-users] Looking for the carrier that owns a particular DID

[Tech Support]

That's a great tool. Thanks.
John V.


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Frank Vanoni
Sent: Thursday, November 02, 2017 07:43 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Looking for the carrier that owns a particular
DID

On Thu, 2017-11-02 at 11:33 -0400, Tech Support wrote:

> How do I find out which carrier owns the DID in question?

Try here:

https://www.twilio.com/lookup



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[asterisk-users] Looking for the carrier that owns a particular DID

[Tech Support]

All;

I have a customer who is looking for a particular DID. (I dialed it and
it is not in service). I searched through my preferred upstream provider's
list but I came up empty. I wrote them, and this is their reply.

 

"We currently do not have that specific number in stock as this number is
owned by another carrier that we do not have a business relationship with."

 

So my question is this. How do I find out which carrier owns the DID in
question?

Thanks;

John V. 

 

Tech Support

Tech Support

VoIP Business Solutions

240-215-3479 x325

 <mailto:f...@voipbusiness.us> supp...@voipbusiness.us

 

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Re: [asterisk-users] What version of Linux?

[Tech Support]

Hello;
Have you run the script that's included in the Asterisk distribution
that lists and installs the needed dependencies? It's called
"install_prereq" and it's in the contrib/scripts directory. Hope this helps.
Regards;
John V.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ira
Sent: Monday, August 28, 2017 03:28 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] What version of Linux?

Hello Asterisk,

I've been running CentOS since 2006 or so and support for the 32 bit version
recently ended. CentOS no longer offers a 32 bit version so I thought I'd
try Fedora 26 as they have 32 bit and support. Got it installed, then
downloaded Asterisk 14.6.0 but can't seem to get it built. The configure
script fails with some error about CPP not working correctly? I did discover
that kernel-devel was not installed so I fixed that but I'm still stuck.

Is the latest Fedora a good choice for an Asterisk box or should I try
something else. The machine is an Intel Atom board with a Digium PCI analog
board for my one last analog line.

I believe the board is limited to a 32 bit OS.

So two questions, is Fedora a good choice and if not, what should I use for
a machine running only Asterisk and Samba?

Is there a list of dependencies I need to install before Asterisk will
compile?

Thanks, Ira 


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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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[asterisk-users] How are people billing for minutes?

[Tech Support]

All;

We have always tried to avoid charging customers for minutes simply
because we didn't want the hassle of doing the accounting. I was wondering
what software packages or services people are using for this. 

Best Regards;

John V.

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Re: [asterisk-users] Writing CDR's to two database servers

[Tech Support]

I appreciate all the feedback, and replication seems to be a logical 
solution, but I was initially thinking about how to implement a solution within 
Asterisk to write the CDR's to two databases. Is that possible? Now I'm just 
curious.
Thanks Much;
John V.

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Antony Stone
Sent: Monday, June 19, 2017 01:08 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Writing CDR's to two database servers

On Monday 19 June 2017 at 18:12:35, Sebastian Gutierrez wrote:

> use replication

1. Agreed - use replication.

2. If you want an HA (High Availability, not dependent on a single Master DB 
server replicating to a slave) solution, consider setting up Master-Master 
replication, with an LVS (Linux Virtual Server) HA machine in front of the two, 
so that writes can go to either server using only a single IP address 
configured in Asterisk.

Then, if one fails, you can still write to (and read from) the other, repair 
the failed one, and restore replication.


Antony

> > On Jun 19, 2017, at 17:47, Tech Support <aster...@voipbusiness.us> wrote:
> > 
> > All;
> > 
> > I know that there are probably several solutions to this problem, but
> > what I am trying to do is provide some redundancy for my customers
> > CDR data. I know that doing simple backups of MySQL is probably the
> > easiest way to go, but I’m thinking that there may be some benefit
> > to simultaneously writing the CDR data to multiple servers at once.
> > However, I’m drawing a blank on this one. Has anyone else done this
> > before? Any insight at all would be greatly appreciated.
> > 
> > Thanks Much;
> > John V.

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[asterisk-users] Writing CDR's to two database servers

[Tech Support]

All;

I know that there are probably several solutions to this problem, but
what I am trying to do is provide some redundancy for my customers CDR data.
I know that doing simple backups of MySQL is probably the easiest way to go,
but I'm thinking that there may be some benefit to simultaneously writing
the CDR data to multiple servers at once. However, I'm drawing a blank on
this one. Has anyone else done this before? Any insight at all would be
greatly appreciated.

Thanks Much;

John V.

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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] Automatically dial a number, then an extension

[Tech Support]

I definitely appreciate your insight on this and I split the context into 
two contexts, but I still have a problem. That is, after I send the extension 
digits using SendDTMF( ), I need to be able to tell whether or not the far end 
extension picked up and I don’t know how to do that. Because of that, the 
message sometimes gets played while the extension is still ringing.
Thanks Much;
John V.  

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Antony Stone
Sent: Tuesday, May 23, 2017 06:11 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Automatically dial a number, then an extension

On Tuesday 23 May 2017 at 20:01:14, Tech Support wrote:

> Ok, the purpose of the answering machine detection (AMD) is to 
> determine when the audio file should start playing *after* the call 
> has been picked up. Typically, if a call has been picked up by a 
> person, they say a short greeting, for example "Hello, this is John, 
> how can I help you?" or simply "Hello?" or something similar. If a 
> call has been picked up by an answering machine, usually the message 
> is somewhat longer, maybe
> 10 seconds or so, maybe longer. Ideally, the AMD tries to make sure 
> that the audio file starts right after the greeting is over. It's not 
> exact, but my experience is that it works fairly well. The problem 
> that I am having is that when I also have to dial an extension, the 
> call has already picked up and the AMD will start working immediately 
> after the SendDTMF() even if dialing the extension means that it may 
> ring anywhere from 5 - 20 seconds plus the greeting on the far end. 
> There doesn’t appear to be a way for the AMD to wait until extension 
> gets picked up, either by a human or a machine. So what happens is 
> that the AMD gets confused and the audio file starts playing while the 
> extension is still ringing. I hope this helps.

Okay, so my suggestion still stands:

Create two contexts:

 - one which does AMD and gets called when there is no follow-on extension to 
dial

 - another which dials a follow-on extension and doesn't do AMD (or at least, 
not at the start)

Then you choose which context to place the call through depending on whether a 
follow-on extension has been supplied for that customer's number or not - if 
there's no follow-on extenstion, use the first context; if there is, use the 
second one.


Antony.

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Re: [asterisk-users] Automatically dial a number, then an extension

[Tech Support]

Ok, the purpose of the answering machine detection (AMD) is to determine 
when the audio file should start playing *after* the call has been picked up. 
Typically, if a call has been picked up by a person, they say a short greeting, 
for example "Hello, this is John, how can I help you?" or simply "Hello?" or 
something similar. If a call has been picked up by an answering machine, 
usually the message is somewhat longer, maybe 10 seconds or so, maybe longer. 
Ideally, the AMD tries to make sure that the audio file starts right after the 
greeting is over. It's not exact, but my experience is that it works fairly 
well.   
The problem that I am having is that when I also have to dial an extension, 
the call has already picked up and the AMD will start working immediately after 
the SendDTMF() even if dialing the extension means that it may ring anywhere 
from 5 - 20 seconds plus the greeting on the far end. There doesn’t appear to 
be a way for the AMD to wait until extension gets picked up, either by a human 
or a machine. So what happens is that the AMD gets confused and the audio file 
starts playing while the extension is still ringing. I hope this helps.
Thanks;
John V. 

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Antony Stone
Sent: Tuesday, May 23, 2017 01:39 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Automatically dial a number, then an extension

On Tuesday 23 May 2017 at 19:20:25, Tech Support wrote:


Isn't it safe to assume that if you've been given an extension number to dial 
after the initial call is answered, then it wasn't answered by an answering 
machine?

The extension might be answered by an answering machine, I suppose, but that's 
not the problem you're talking about (I think).

I would create two contexts:

1. Does AMD and gets called when there is no follow-on extension to dial

2. Dials a follow-on extension and doesn't do AMD (or at least, not at the
start)

Then you choose which context to place the call through depending on whether a 
follow-on extension has been supplied for that customer's number or not.

> Simply placing the AMD command after the SendDTMF() wasn’t the answer

Why wasn't it the answer?  What happens or doesn't happen when you try this?




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Re: [asterisk-users] Automatically dial a number, then an extension

[Tech Support]

All;

What I did was add a line in the dialplan that used the SendDTMF() 
application and that worked great. The problem that I’ve run into now is that 
dialing the extension screwed up the answering machine detection. The sample 
context looks something like this.

 

[play-audiomsg]

exten => s,1,AMD

exten => s,n,ExecIf($["${EXT}" != ""]?SendDTMF(${EXTEN}))

exten => s,n,Background(${AUDIOMSG})

exten => s,n,Hangup

 

As you can see, it's very simple. Modifying the amd.conf configuration wasn’t 
the answer since I don’t know how long it will take for the extension to pick 
up. Simply placing the AMD command after the SendDTMF() wasn’t the answer  I 
don’t know how to approach this problem.

 

Thanks;

John V.  

 

 

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tech Support
Sent: Tuesday, May 16, 2017 03:56 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Automatically dial a number, then an extension

 

Hey;

   What happens is that a script logs into the AMI and originates a call. When 
the call is answered, it jumps to a context in the dial plan.

Thanks Much;

John V. 

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Victor Villarreal
Sent: Monday, May 15, 2017 12:08 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Automatically dial a number, then an extension

 

Hi John, 

 

I think we need to known how you play the audio to the customers, before we can 
help you. 

 

Are you using AMI? Or AGI maybe? Or Call files? 

 

What Asterisk version do you have? 

 

El 15 may. 2017 12:35, "Tech Support" <aster...@voipbusiness.us> escribió:

All;

I have an application that dials a list of numbers and then plays a 
recorded message. My customer uses it to dial a list of customers to confirm 
their appointment for the next day. No biggie, maybe 25 – 30 calls per day for 
customers who want the confirmation call. What they need now is a way to dial 
an extension after the number is dialed and answered. I’ve seen that before, 
but I just can't remember where. I was wondering if anyone else has implemented 
something along these lines. Any insight at all would be greatly appreciated.

Thanks Much;

John V.   


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Re: [asterisk-users] Automatically dial a number, then an extension

[Tech Support]

Hey;

   What happens is that a script logs into the AMI and originates a call. When 
the call is answered, it jumps to a context in the dial plan.

Thanks Much;

John V. 

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Victor Villarreal
Sent: Monday, May 15, 2017 12:08 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Automatically dial a number, then an extension

 

Hi John, 

 

I think we need to known how you play the audio to the customers, before we can 
help you. 

 

Are you using AMI? Or AGI maybe? Or Call files? 

 

What Asterisk version do you have? 

 

El 15 may. 2017 12:35, "Tech Support" <aster...@voipbusiness.us> escribió:

All;

I have an application that dials a list of numbers and then plays a 
recorded message. My customer uses it to dial a list of customers to confirm 
their appointment for the next day. No biggie, maybe 25 – 30 calls per day for 
customers who want the confirmation call. What they need now is a way to dial 
an extension after the number is dialed and answered. I’ve seen that before, 
but I just can't remember where. I was wondering if anyone else has implemented 
something along these lines. Any insight at all would be greatly appreciated.

Thanks Much;

John V.   


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[asterisk-users] Automatically dial a number, then an extension

[Tech Support]

All;

I have an application that dials a list of numbers and then plays a
recorded message. My customer uses it to dial a list of customers to confirm
their appointment for the next day. No biggie, maybe 25 - 30 calls per day
for customers who want the confirmation call. What they need now is a way to
dial an extension after the number is dialed and answered. I've seen that
before, but I just can't remember where. I was wondering if anyone else has
implemented something along these lines. Any insight at all would be greatly
appreciated.

Thanks Much;

John V.   

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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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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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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] Can't compile asterisk-certified-11.6-cert16 on Ubuntu 16

[Tech Support]

Sorry. I wasn’t clear in my post but I was talking about installing both 
certified 11 and 13. The reason I am trying to install 11 is for support for a 
couple of legacy systems. Any insight at all into the original question would 
be greatly appreciated.
Thanks;
John 

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere
Sent: Saturday, April 29, 2017 01:10 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Can't compile asterisk-certified-11.6-cert16 on 
Ubuntu 16


Don't get me wrong - he should absolutely do that.  Just offering an 
explanation for his post.  I read it as "I am trying to install certified and 
it won't compile.  Normal 13 does though.".  He didn't mention if the 13 he was 
trying to compile was certified. Why he would want 11 over 13 is anyone's guess.

j

On 04/29/2017 11:15 AM, Jonathan H wrote:
> Sure, so why not install the current supported certified 13.13-cert 3 
> which he confirms builds OK, rather than the about-to-become-EOL 11 
> version?
>
> http://www.asterisk.org/downloads/asterisk/all-asterisk-versions
>
> On 29 April 2017 at 17:11, Jeff LaCoursiere <j...@jeff.net> wrote:
>> On 04/29/2017 10:57 AM, Jonathan H wrote:
>>> On 29 April 2017 at 16:47, Tech Support <aster...@voipbusiness.us> wrote:
>>>
>>>> I’m trying to install certified asterisk 11.6 cert16 on a Ubuntu 16 
>>>> server. However, when I try to compile it, I’m getting hundreds and 
>>>> hundreds of errors. Here is a sample of the output.
>>>> When I try to build Asterisk 13, I have no problem. Any insight at 
>>>> all would be greatly appreciated.
>>> I suppose the first and most obvious question would be:
>>>
>>> If the current version installs fine, why would you want to install 
>>> an old obsolete version which is currently subject to an end of life 
>>> warning?
>>>
>>> https://community.asterisk.org/t/asterisk-11-eol-6-month-notice/7049
>>> 0
>>>
>>> "As many of you know, for the past 6 months Asterisk 11 has been in 
>>> security fix only mode. This means it currently does not receive bug 
>>> fixes, but it does receive applicable security fixes and will 
>>> continue to do so for the next 6 months."
>>>
>> Probably because he wants to run certified...
>>
>>
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>> Check out the new Asterisk community forum at:
>> https://community.asterisk.org/
>>
>> New to Asterisk? Start here:
>>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>
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Re: [asterisk-users] softphone instead of desktop phones

[Tech Support]

I thought this was a non-commercial list.


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alex Epshteyn
Sent: Saturday, April 29, 2017 08:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] softphone instead of desktop phones

Thirdlane Connect can be used as a softphone. It works in modern browsers
(no installation is required), on Mac, Windows and Linux desktops, and on
mobile phones.

Besides basic softphone functionality, it provides instant messaging, group
chat (channels), voice and video conferencing, and screen sharing. It
integrates with a variety of applications and CRMs such as Salesforce, Zoho,
Zendesk, Redmine, etc.

Try it out!


-- 

Alex Epshteyn
web: www.thirdlane.com


- Original Message -
> From: "Amit Patkar" 
> To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
> 
> Sent: Saturday, April 29, 2017 9:16:05 AM
> Subject: Re: [asterisk-users] softphone instead of desktop phones
> 
> 
> Linphone is available for all major OS platforms.
> Then there is PortGo as well
> Regards,
> Amit Patkar
> 
> 
> On April 29, 2017 9:05:22 PM GMT+05:30, Thomas  
> wrote:
> 
> Hello,
> Iam lookong for an Softphone for iPhor oder Android smartphone using 
> togehter with an headset.
> I tried Zoiper and CSipSimple but quality was bad compared to an 
> desktop SIP phone.
> 
> Is there an better softphone?
> 
> Or are there softphone solutions for PC desktop MAC or Android with an 
> headset?
> I want to save cost for desktop phones.
> 
> thanks Thomas
> 
> 
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> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
> 
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
> 
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[asterisk-users] Can't compile asterisk-certified-11.6-cert16 on Ubuntu 16

[Tech Support]

All;

I'm trying to install certified asterisk 11.6 cert16 on a Ubuntu 16
server. However, when I try to compile it, I'm getting hundreds and hundreds
of errors. Here is a sample of the output.

 

make[1]: Leaving directory
'/usr/src/asterisk-certified-11.6-cert16/menuselect'

   [LD] aelparse.o aelbison.o pbx_ael.o hashtab.o lock.o ael_main.o
ast_expr2f.o ast_expr2.o strcompat.o pval.o extconf.o -> aelparse

aelbison.o: In function `ast_atomic_fetchadd_int':

/usr/src/asterisk-certified-11.6-cert16/include/asterisk/lock.h:567:
multiple definition of `ast_atomic_fetchadd_int'

aelparse.o:/usr/src/asterisk-certified-11.6-cert16/include/asterisk/lock.h:5
67: first defined here

aelbison.o: In function `ast_atomic_dec_and_test':

/usr/src/asterisk-certified-11.6-cert16/include/asterisk/lock.h:613:
multiple definition of `ast_atomic_dec_and_test'

aelparse.o:/usr/src/asterisk-certified-11.6-cert16/include/asterisk/lock.h:6
13: first defined here

aelbison.o: In function `ast_tvdiff_sec':

/usr/src/asterisk-certified-11.6-cert16/include/asterisk/time.h:45: multiple
definition of `ast_tvdiff_sec'

aelparse.o:/usr/src/asterisk-certified-11.6-cert16/include/asterisk/time.h:4
5: first defined here

aelbison.o: In function `ast_tvdiff_us':

 

pbx_ael.o: In function `ast_str_set_va':

/usr/src/asterisk-certified-11.6-cert16/include/asterisk/strings.h:803:
undefined reference to `__ast_str_helper'

pbx_ael.o: In function `ast_str_append_va':

/usr/src/asterisk-certified-11.6-cert16/include/asterisk/strings.h:820:
undefined reference to `__ast_str_helper'

pbx_ael.o: In function `ast_str_set_va':

/usr/src/asterisk-certified-11.6-cert16/include/asterisk/strings.h:803:
undefined reference to `__ast_str_helper'

pbx_ael.o: In function `ast_str_append_va':

/usr/src/asterisk-certified-11.6-cert16/include/asterisk/strings.h:820:
undefined reference to `__ast_str_helper'

pbx_ael.o: In function `ast_str_set_substr':

 

 

I'm not exactly sure where I should begin looking for the source of the
problem. When I try to build Asterisk 13, I have no problem. Any insight at
all would be greatly appreciated.

Thanks;

John 

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Re: [asterisk-users] ** in extensions.conf

[Tech Support]

Is ** also defined in features.conf?
Thanks;
John

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Sent: Wednesday, April 26, 2017 05:41 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] ** in extensions.conf

On Wed, 26 Apr 2017, Jerry Geis wrote:

> dialplan show testing-sip
>   '**' =>   1. Noop(Testing)  
> [pbx_config]
> 2. Playback(demo-congrats)
> [pbx_config]
> 
> Looks like its there.
> 
> if I do ** "Dial" it works, but if I do "New Call" ** then "Dial" it 
> does not work. Weird. How do I get it to work for both cases. (glad I 
> tried the other)

I never use 'New Call' -- just 'Dial' and 'Redial,'

I suspect you'll need to fiddle with the Polycom dialplan. As soon as I press 
the first '*' my Poly sends the INVITE.

--
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
 https://www.linkedin.com/in/steve-edwards-4244281


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Re: [asterisk-users] Asterisk download stats

[Tech Support]

On a similar note, does anyone have any idea as to the total number of Asterisk 
installations out there?

Thanks;

John 

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Klein
Sent: Tuesday, April 25, 2017 10:00 AM
To: asterisk-users
Subject: Re: [asterisk-users] Asterisk download stats

 

I have tried to find these in the past, best I came up with was using Shodan.io 
search

 

Looking for Asterisk I get:


TOTAL RESULTS


42,036

 


TOP COUNTRIES


United States 12,914

Russian Federation   3,173

Brazil   2,356

United Kingdom 2,305

Germany2,218

Canada   2,119

Japan 1,648

France1,404

Ukraine   971

Australia 782

China   528

Belgium   407

United Kingdom 367

Columbia361

Argentina   308

Sweden   207

Mexico 201

South Africa   190

Turkey 160

Philippines  157

Kazakhstan 147

Thailand  112

New Zealand  103

Iran  101

Peru91

Greece72

Venezuela  45

Morocco 26

Nigeria21

Saudi Arabia  20

Kuwait 17

Mongolia 17

Other  8,488

 

 

Not a complete list, and only for PBXs identifying as Asterisk, not FreePBX etc

 

Hope this helps.

 

On Mon, Apr 24, 2017 at 8:00 PM,  
wrote:

Send asterisk-users mailing list submissions to
asterisk-users@lists.digium.comMessage: 2
Date: Sun, 23 Apr 2017 17:51:28 -0400
From: Dovid Bender <  do...@telecurve.com>
To: Asterisk Users Mailing List - Non-Commercial Discussion
<  
asterisk-users@lists.digium.com>
Subject: [asterisk-users] Asterisk download stats
Message-ID:
< 
 
cam3tth1d7zn71o1by+25tgj7r9saqu_k1xovitl+k7f1d5q...@mail.gmail.com>
Content-Type: text/plain; charset="utf-8"

Hi,

Are there any stats on where Asterisk is downloaded from based on the

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Re: [asterisk-users] Can't compile Asterisk on Ubuntu 16

[Tech Support]

Yes. I did the usual configure ; make menuselect ; make ; make install ; make 
samples ; make progdocs.

Thanks;

John

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonathan H
Sent: Wednesday, April 19, 2017 09:27 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Can't compile Asterisk on Ubuntu 16

 

I don't - it just seems to.. work!

 

Try a reboot - it always comes up OK for me. Are you doing "make install"?

 

On 19 April 2017 at 14:19, Tech Support <aster...@voipbusiness.us> wrote:

Hey;
   Thank you very much. I was able to install asterisk from your link. One
other question. How are you starting asterisk? Do you use an init script or
systemd? Do you think that you could share the script you use?
Thanks Again;
John V.


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonathan H
Sent: Tuesday, April 18, 2017 09:10 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Can't compile Asterisk on Ubuntu 16

Feel free to take a look at
https://github.com/lardconcepts/asterisk-digitalocean-voipfone-config/blob/m 
<https://github.com/lardconcepts/asterisk-digitalocean-voipfone-config/blob/master/Asterisk-14-on-Ubuntu.md>
 
aster/Asterisk-14-on-Ubuntu.md

Ignore the bit about Voipfone and just skip to the "Install Asterisk" bit.

I've used this same script with Asterisk 12,13 and 14 on Ubuntu 15,16 and 17
so this should work!

Let me know how you get on.

On 18 April 2017 at 13:41, Tech Support <aster...@voipbusiness.us> wrote:
> All;
>
> I am trying to build and install certified Asterisk 13.13 cert3 on
> a Ubuntu 16.04.2 LTS host without much success. I am getting the
> following errors when I try to compile.
>
>
>
>[CC] res_pjsip/config_transport.c -> res_pjsip/config_transport.o
>
> res_pjsip/config_transport.c: In function 'transport_apply':
>
> res_pjsip/config_transport.c:572:6: error: 'pjsip_tcp_transport_cfg
> {aka struct pjsip_tcp_transport_cfg}' has no member named 'sockopt_params'
>
>cfg.sockopt_params.options[0].level = pj_SOL_TCP();
>
>   ^
>
> res_pjsip/config_transport.c:573:6: error: 'pjsip_tcp_transport_cfg
> {aka struct pjsip_tcp_transport_cfg}' has no member named 'sockopt_params'
>
>cfg.sockopt_params.options[0].optname = pj_TCP_NODELAY();
>
>   ^
>
> res_pjsip/config_transport.c:574:6: error: 'pjsip_tcp_transport_cfg
> {aka struct pjsip_tcp_transport_cfg}' has no member named 'sockopt_params'
>
>cfg.sockopt_params.options[0].optval = 
>
>   ^
>
> res_pjsip/config_transport.c:575:6: error: 'pjsip_tcp_transport_cfg
> {aka struct pjsip_tcp_transport_cfg}' has no member named 'sockopt_params'
>
>cfg.sockopt_params.options[0].optlen = sizeof(option);
>
>   ^
>
> res_pjsip/config_transport.c:576:6: error: 'pjsip_tcp_transport_cfg
> {aka struct pjsip_tcp_transport_cfg}' has no member named 'sockopt_params'
>
>cfg.sockopt_params.cnt = 1;
>
>   ^
>
> /usr/src/asterisk-certified-13.13-cert3/Makefile.rules:149: recipe for
> target 'res_pjsip/config_transport.o' failed
>
> make[1]: *** [res_pjsip/config_transport.o] Error 1
>
> Makefile:402: recipe for target 'res' failed
>
> make: *** [res] Error 2
>
>
>
> Has anyone seen this error before? Any insight at all would be greatly
> appreciated.
>
> Thanks;
>
> John V.
>
>
>
> Tech Support
>
> Tech Support
>
> VoIP Business Solutions
>
> 240-215-3479 x325
>
> supp...@voipbusiness.us
>
>
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users

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Re: [asterisk-users] Can't compile Asterisk on Ubuntu 16

[Tech Support]

Hey;
   Thank you very much. I was able to install asterisk from your link. One
other question. How are you starting asterisk? Do you use an init script or
systemd? Do you think that you could share the script you use?
Thanks Again;
John V.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonathan H
Sent: Tuesday, April 18, 2017 09:10 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Can't compile Asterisk on Ubuntu 16

Feel free to take a look at
https://github.com/lardconcepts/asterisk-digitalocean-voipfone-config/blob/m
aster/Asterisk-14-on-Ubuntu.md

Ignore the bit about Voipfone and just skip to the "Install Asterisk" bit.

I've used this same script with Asterisk 12,13 and 14 on Ubuntu 15,16 and 17
so this should work!

Let me know how you get on.

On 18 April 2017 at 13:41, Tech Support <aster...@voipbusiness.us> wrote:
> All;
>
> I am trying to build and install certified Asterisk 13.13 cert3 on 
> a Ubuntu 16.04.2 LTS host without much success. I am getting the 
> following errors when I try to compile.
>
>
>
>[CC] res_pjsip/config_transport.c -> res_pjsip/config_transport.o
>
> res_pjsip/config_transport.c: In function 'transport_apply':
>
> res_pjsip/config_transport.c:572:6: error: 'pjsip_tcp_transport_cfg 
> {aka struct pjsip_tcp_transport_cfg}' has no member named 'sockopt_params'
>
>cfg.sockopt_params.options[0].level = pj_SOL_TCP();
>
>   ^
>
> res_pjsip/config_transport.c:573:6: error: 'pjsip_tcp_transport_cfg 
> {aka struct pjsip_tcp_transport_cfg}' has no member named 'sockopt_params'
>
>cfg.sockopt_params.options[0].optname = pj_TCP_NODELAY();
>
>   ^
>
> res_pjsip/config_transport.c:574:6: error: 'pjsip_tcp_transport_cfg 
> {aka struct pjsip_tcp_transport_cfg}' has no member named 'sockopt_params'
>
>cfg.sockopt_params.options[0].optval = 
>
>   ^
>
> res_pjsip/config_transport.c:575:6: error: 'pjsip_tcp_transport_cfg 
> {aka struct pjsip_tcp_transport_cfg}' has no member named 'sockopt_params'
>
>cfg.sockopt_params.options[0].optlen = sizeof(option);
>
>   ^
>
> res_pjsip/config_transport.c:576:6: error: 'pjsip_tcp_transport_cfg 
> {aka struct pjsip_tcp_transport_cfg}' has no member named 'sockopt_params'
>
>cfg.sockopt_params.cnt = 1;
>
>   ^
>
> /usr/src/asterisk-certified-13.13-cert3/Makefile.rules:149: recipe for 
> target 'res_pjsip/config_transport.o' failed
>
> make[1]: *** [res_pjsip/config_transport.o] Error 1
>
> Makefile:402: recipe for target 'res' failed
>
> make: *** [res] Error 2
>
>
>
> Has anyone seen this error before? Any insight at all would be greatly 
> appreciated.
>
> Thanks;
>
> John V.
>
>
>
> Tech Support
>
> Tech Support
>
> VoIP Business Solutions
>
> 240-215-3479 x325
>
> supp...@voipbusiness.us
>
>
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users

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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 
<asterisk-users@lists.digium.com>
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 <a...@thirdlane.com 
<mailto:a...@thirdlane.com> > 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 <mailto:a...@thirdlane.com> 
web: www.thirdlane.com <http://www.thirdlane.com> 
phone +1 415.261.6601 <tel:%2B1%20415.261.6601> 



- Original Message -
> From: "J Montoya or A J Stiles" <asterisk_l...@earthshod.co.uk 
> <mailto:asterisk_l...@earthshod.co.uk> >
> To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
> <asterisk-users@lists.digium.com <mailto:asterisk-users@lists.digium.com> >
> 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

[asterisk-users] Can't compile Asterisk on Ubuntu 16

[Tech Support]

All;

I am trying to build and install certified Asterisk 13.13 cert3 on a
Ubuntu 16.04.2 LTS host without much success. I am getting the following
errors when I try to compile.

 

   [CC] res_pjsip/config_transport.c -> res_pjsip/config_transport.o

res_pjsip/config_transport.c: In function 'transport_apply':

res_pjsip/config_transport.c:572:6: error: 'pjsip_tcp_transport_cfg {aka
struct pjsip_tcp_transport_cfg}' has no member named 'sockopt_params'

   cfg.sockopt_params.options[0].level = pj_SOL_TCP();

  ^

res_pjsip/config_transport.c:573:6: error: 'pjsip_tcp_transport_cfg {aka
struct pjsip_tcp_transport_cfg}' has no member named 'sockopt_params'

   cfg.sockopt_params.options[0].optname = pj_TCP_NODELAY();

  ^

res_pjsip/config_transport.c:574:6: error: 'pjsip_tcp_transport_cfg {aka
struct pjsip_tcp_transport_cfg}' has no member named 'sockopt_params'

   cfg.sockopt_params.options[0].optval = 

  ^

res_pjsip/config_transport.c:575:6: error: 'pjsip_tcp_transport_cfg {aka
struct pjsip_tcp_transport_cfg}' has no member named 'sockopt_params'

   cfg.sockopt_params.options[0].optlen = sizeof(option);

  ^

res_pjsip/config_transport.c:576:6: error: 'pjsip_tcp_transport_cfg {aka
struct pjsip_tcp_transport_cfg}' has no member named 'sockopt_params'

   cfg.sockopt_params.cnt = 1;

  ^

/usr/src/asterisk-certified-13.13-cert3/Makefile.rules:149: recipe for
target 'res_pjsip/config_transport.o' failed

make[1]: *** [res_pjsip/config_transport.o] Error 1

Makefile:402: recipe for target 'res' failed

make: *** [res] Error 2

 

Has anyone seen this error before? Any insight at all would be greatly
appreciated.

Thanks;

John V.

 

Tech Support

Tech Support

VoIP Business Solutions

240-215-3479 x325

 <mailto:f...@voipbusiness.us> supp...@voipbusiness.us

 

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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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[asterisk-users] Tool to restart Asterisk

[Tech Support]

All;

There was a thread a while back talking about a tool to restart
Asterisk. We have a program that runs out of CRON that does just that. What
it does is check the Asterisk process every X minutes and verifies not only
that Asterisk is running, but that it is running efficiently by checking the
Asterisk process parameters such as memory usage, cpu usage, number of
threads spawned, the number of open files, etc. The program reads the
threshold values in a config file and if the parameters exceed the threshold
values, the program restarts Asterisk and shoots an email off to the admin
letting them know that there was a restart. The program is written in Perl
and requires no non-standard CPAN modules. Feel free to download the program
at https://mte6.mobilepbx.net/downloads/AsteriskProcess.tar.gz. All comments
are welcome and can be sent to supp...@voipbusiness.us.

Regards;

John V.

 

Tech Support

Tech Support

VoIP Business Solutions

240-215-3479 x325

 <mailto:f...@voipbusiness.us> supp...@voipbusiness.us

 

 

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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 <http://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'
<asterisk-users@lists.digium.com>
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>
[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"<sip:1005@asterisk-ip:5060>' failed for 'offending-IP:53417' - Wrong
password

NOTICE[29784] chan_sip.c: Registration from
'"user3"<sip:1005@asterisk-ip:5060>' 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
<mailto:dest=mo...@email.com> , sender=fail2...@asterisk-ip.com
<mailto: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

  

Re: [asterisk-users] fail2ban Asterisk 13.13.1

[Tech Support]

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 '.*' failed for '' - ACL
error (permit/deny)

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

NOTICE.*  failed to authenticate as '.*'$

NOTICE.* .*: No registration for peer '.*' \(from \)

NOTICE.* .*: Host  failed MD5 authentication for '.*' (.*)

NOTICE.* .*: Failed to authenticate user .*@
 .*

NOTICE.* .*: Sending fake auth rejection for device .*\ \>;tag=.*

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

NOTICE.* .*: Registration from '\".*\".*' failed for '' -
Wrong password

 

ignoreregex =

 

Thanks

Motty

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Re: [asterisk-users] Which tool to automatically restart Asterisk ?

[Tech Support]

Hello;

Over time, we’ve built a huge enterprise level monitoring system for our 
internal and customer PBX’s. Using Nagios as the core, along with Grafana, 
Graphite, Carbon, Whisper, etc. so we can also create custom dynamic 
dashboards, we typically monitor over 1,000 different metrics for each PBX. For 
something like monitoring a system process like Asterisk, besides just checking 
to see if the process is running or not, we also check about a dozen or so 
related metrics like memory and cpu usage. If anything gets out of whack, the 
system runs the event handler to restart Asterisk. All the plugins are written 
in Perl, so they’re very easy to modify. What I can do if there is an interest 
is take the Asterisk plugin, strip out everything that wouldn’t apply to 
someone not using our system, and make it available to the general public. It's 
up to you guys. What do you think? Would people find that useful?

Regards;

John V.

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
Sent: Friday, February 17, 2017 10:39 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Which tool to automatically restart Asterisk ?

 

Hello,

Years ago, I used Monit to monitor Asterisk and restart it whenever it failed.

Now, I wonder which tool I should pick for an Debian 8 (current) or CentOS 7 
(future) environment.

The main reason I'm looking for this tool is to avoid as much as possible, 
current 5 minutes delay between Asterisk's stop and first cutomers complains.

 

1. I always install Asterisk from source but I've read in Debian Stretch 
/etc/defaul/asterisk file, the following:
# RUNASTSAFE: run safe_asterisk rather than asterisk (will auto-restart upon
# crash). This is generally less tested and has some known issues
# with properly starting and stopping Asterisk.

Where I can read about those known issues ?

(not found in [1]).

2. For systemd envs where /etc/init.d files are still used, what do you 
recommend ?

3. For systemd envs where /etc/init.d files are not used anymore, what do you 
recommend ?

4. Suggestions ?

Regards



[1] https://bugs.debian.org/cgi-bin/pkgreport.cgi?pkg=asterisk;dist=unstable

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Re: [asterisk-users] Voicemail notification by email is missing CallerID info

[Tech Support]

I’ll go through it and see what I missed. I can't thank you enough!

John

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonathan H
Sent: Saturday, February 18, 2017 11:50 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Voicemail notification by email is missing 
CallerID info

 

This is what comes with voicemail.conf.sample - works for me!

; Change the from, body and/or subject, variables:

; VM_NAME, VM_DUR, VM_MSGNUM, VM_MAILBOX, VM_CALLERID, VM_CIDNUM,

; VM_CIDNAME, VM_DATE

; Additionally, on forwarded messages, you have the variables:

; ORIG_VM_CALLERID, ORIG_VM_CIDNUM, ORIG_VM_CIDNAME, ORIG_VM_DATE

; You can select between two variables by using dialplan functions, e.g.

; ${IF(${ISNULL(${ORIG_VM_DATE})}?${VM_DATE}:${ORIG_VM_DATE})}

;

; Note: The emailbody config row can only be up to 512 characters due to a

;   limitation in the Asterisk configuration subsystem.

;emailsubject=[PBX]: New message ${VM_MSGNUM} in mailbox ${VM_MAILBOX}

; The following definition is very close to the default, but the default shows

; just the CIDNAME, if it is not null, otherwise just the CIDNUM, or "an unknown

; caller", if they are both null.

;emailbody=Dear ${VM_NAME}:\n\n\tjust wanted to let you know you were just left 
a ${VM_DUR} long message (number ${VM_MSGNUM})\nin mailbox ${VM_MAILBOX} from 
${VM_CALLERID}, on ${VM_DATE}, so you might\nwant to check it when you get a 
chance.  Thanks!\n\n\t\t\t\t--Asterisk\n

;

; Note: ${IF()} strips spacing at the beginning and end of its true and false

; values, so a newline cannot be placed at either location.  The word 'so' is

; therefore duplicated, in order for the newline to be interpreted correctly.

;emailbody=Dear ${VM_NAME}:\n\n\tjust wanted to let you know you were just 
${IF($["${VM_CIDNUM}" = "${ORIG_VM_CIDNUM}"]?left:forwarded)} a ${VM_DUR} long 
message (number ${VM_MSGNUM})\nin mailbox ${VM_MAILBOX} from ${VM_CALLERID}, on 
${VM_DATE},\n${IF($["${VM_CIDNUM}" = "${ORIG_VM_CIDNUM}"]?so:(originally sent 
by ${ORIG_VM_CALLERID} on ${ORIG_VM_DATE})\nso)} you might want to check it 
when you get a chance.  Thanks!\n\n\t\t\t\t--Asterisk\n

 

On 18 February 2017 at 16:35, Tech Support <aster...@voipbusiness.us> wrote:
> All;
>
> I am running Asterisk 11.6-cert16 and I have voicemail setup so
> voicemail messages are sent as email attachments. That works fine. However,
> the body of the email contains the CallerID(name), but is missing the
> CallerID(num). For example, the email body looks like this:
>
>
>
>   Just wanted to let you know you were just left a 0:21 long message
> (number 13) in mailbox 101 from WIRELESS CALLER, on Friday, February 17,
> 2017 at 04:48:38 PM so you might want to check it when you get a chance.
> Thanks!
>
>
>
> Checking the CDR’s shows that both the name and number were recorded by
> Asterisk. Am I missing something obvious? Is it a simple config option in
> voicemail.conf? Any insight at all would be greatly appreciated.
>
> Thanks;
>
> John V.
>
>
>
>
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> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
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>http://lists.digium.com/mailman/listinfo/asterisk-users

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[asterisk-users] Voicemail notification by email is missing CallerID info

[Tech Support]

All;

I am running Asterisk 11.6-cert16 and I have voicemail setup so
voicemail messages are sent as email attachments. That works fine. However,
the body of the email contains the CallerID(name), but is missing the
CallerID(num). For example, the email body looks like this:

 

  Just wanted to let you know you were just left a 0:21 long message
(number 13) in mailbox 101 from WIRELESS CALLER, on Friday, February 17,
2017 at 04:48:38 PM so you might want to check it when you get a chance.
Thanks!

 

Checking the CDR's shows that both the name and number were recorded by
Asterisk. Am I missing something obvious? Is it a simple config option in
voicemail.conf? Any insight at all would be greatly appreciated.

Thanks;

John V.

 

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Re: [asterisk-users] Call List Campaign to an IVR

[Tech Support]

I remember doing the testing and two calls going out at the same time don’t 
actually have to go out at the *exact* same time. The remote end will pick up 
one of the two calls, but there is no guarantee which one it will be. Also, if 
you let the first call ring too long, yes, the second call will go to 
voicemail,  but the first call will start ringing, which is something we wanted 
to avoid.

John

 

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matt Riddell 
(lists)
Sent: Monday, February 06, 2017 12:32 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Call List Campaign to an IVR

 

Not really, doing the way below you don't even have to worry about it. They 
both go out at the same instant and as soon as it hits voicemail it disconnects 
the other leg. 

 

If you wanted you could leave it ringing for twenty minutes and it would still 
have the same effect. 

Kind regards,





Matt


On Feb 6, 2017, at 12:29 PM, Tech Support <aster...@voipbusiness.us> wrote:

That's the basics, but you have to nail the timing just right. The timing is
really important to do it the right way.


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Sent: Monday, February 06, 2017 12:25 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Call List Campaign to an IVR





On Mon, 6 Feb 2017, Tech Support wrote:

 

 We were able to develop a feature to send the call to voicemail

about 90% of the time. That way, an end user could (1) not be bothered by
having to answer the call, (2)



 delete the message without listening to it, or (3) listen to the

message when it was most convenient for them. That way, they were in control
and things were done on



 their terms.





On 6/02/2017, at 11:34 AM, Steve Edwards <asterisk@sedwards.com>

wrote:

 

Love the idea. How?


On Mon, 6 Feb 2017, Matt Riddell wrote:




exten => 

_X.,1,Dial(SIP/0111${EXTEN}@myprovider/1${EXTEN}@myprovider,3)


Amazing. Who knew?

So how/why does this work?

I see 2 calls going out to my cell. Does the first 'busy out' my number at
my cell provider so the second goes straight to VM? What part does the
'0111' play?

--
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
https://www.linkedin.com/in/steve-edwards-4244281

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Re: [asterisk-users] Call List Campaign to an IVR

[Tech Support]

That's the basics, but you have to nail the timing just right. The timing is
really important to do it the right way.
 

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Sent: Monday, February 06, 2017 12:25 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Call List Campaign to an IVR


> On Mon, 6 Feb 2017, Tech Support wrote:
>
>   We were able to develop a feature to send the call to voicemail
about 90% of the time. That way, an end user could (1) not be bothered by
having to answer the call, (2)
>   delete the message without listening to it, or (3) listen to the
message when it was most convenient for them. That way, they were in control
and things were done on
>   their terms.

> On 6/02/2017, at 11:34 AM, Steve Edwards <asterisk@sedwards.com>
> wrote:
> 
> Love the idea. How?

On Mon, 6 Feb 2017, Matt Riddell wrote:

> exten => 
> _X.,1,Dial(SIP/0111${EXTEN}@myprovider/1${EXTEN}@myprovider,3)

Amazing. Who knew?

So how/why does this work?

I see 2 calls going out to my cell. Does the first 'busy out' my number at
my cell provider so the second goes straight to VM? What part does the
'0111' play?

--
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
 https://www.linkedin.com/in/steve-edwards-4244281

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Re: [asterisk-users] Call List Campaign to an IVR

[Tech Support]

It was a very long time ago, so I'd have to dig through some old
notebooks to get the exact details, but it wasn't too difficult. Basically,
two calls are made. The first call is made with the wait time set to about a
quarter of a second. We modified the Asterisk source code to allow floating
point values for the wait time, basically modifying only 1 or 2 lines of
code. Even a non-programmer could do it. When the first call is made for
such a short period, the remote end still goes off hook, but the call will
end before it starts to ring. Then, halfway through the first call, a second
call is made. Since the remote end is off hook from the first call, the
second call will get sent to voicemail and the message is played there. I
remember having to do some testing to get optimal wait times and delays in
milliseconds, but overall, the ability to go straight to voicemail was a
valuable tool for us.
Regards;
John V. 

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Sent: Monday, February 06, 2017 11:35 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Call List Campaign to an IVR

On Mon, 6 Feb 2017, Tech Support wrote:

> We were able to develop a feature to send the call to voicemail about 
> 90% of the time. That way, an end user could (1) not be bothered by 
> having to answer the call, (2) delete the message without listening to 
> it, or (3) listen to the message when it was most convenient for them.
> That way, they were in control and things were done on their terms.

Love the idea. How?

--
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-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
 https://www.linkedin.com/in/steve-edwards-4244281

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