Re: [asterisk-users] *****SPAM***** Re: IAX port 4569

[Victor Villarreal]

Ok..

Maybe you can try with this command, from the Asterisk cli:

cli> core set verbose 5
cli> core set debug 5
cli> iax2 set debug on

And then, try to register with your softphone.

At the end of the test, execute:

cli> iax2 set debug off

And finally, review the logfiles.

2017-06-05 17:40 GMT-03:00 <the...@sys-concept.com>:

> Doesn't matter how much I increase the verbose output
> asterisk -vvr
> asterisk will not even print a single line.
>
> How to find out if my firewall has this port open?
> https://www.grc.com
> is reporting that my port is 4569 is in Stealth mode (so it is closed) :-/
>
>
> Thelma
> On 06/05/2017 02:19 PM, Victor Villarreal wrote:
> > I think you need to increase verbose output and search in
> > /var/log/asterisk/full for any error message related to IAX2 registration
> > or simil.
> >
> > 2017-06-05 17:12 GMT-03:00 <the...@sys-concept.com>:
> >
> >> No, I don't think it is IP table issue, I've not upgraded dd-wrt for a
> >> while and it was zoiper was working OK with my previous version of
> >> asterisk.
> >>
> >> After upgrade to 11.25.1 it stop working.
> >> I'm sure port forwarding on dd-wrt is working OK as I have port 80 and
> >> 443 open.
> >>
> >>
> >> Thelma
> >> On 06/05/2017 07:12 AM, Christopher van de Sande wrote:
> >>> Another might be to make sure iptables isn't blocking the connection.
> >>>
> >>> You can run
> >>> iptables -L -n -v
> >>> To see if its set to block any ports.
> >>>
> >>>
> >>> On June 5, 2017 9:06:55 AM EDT, the...@sys-concept.com wrote:
> >>>> I'm getting:
> >>>> netstat -a |grep 4569
> >>>> udp0  0 0.0.0.0:45690.0.0.0:*
> >>>>
> >>>> Should I be getting localhost IP?
> >>>>
> >>>> Thelma
> >>>>
> >>>> On 06/05/2017 06:48 AM, the...@sys-concept.com wrote:
> >>>>> Does asterisk listen on port 4569 by default?
> >>>>>
> >>>>> I'm running version Asterisk 11.25.1 and have a problem registering
> >>>>> Zoiper (IAX) to Asterisk.
> >>>>> I'm getting an error:
> >>>>> Registration refused
> >>>>>
> >>>>
> >>>> --
> >>>> _
> >>>> -- 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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> >>
> >> 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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>
> Check out the new Asterisk community forum at: https://community.asterisk.
> org/
>
> New to Asterisk? Start here:
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>
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Re: [asterisk-users] IAX port 4569

[Victor Villarreal]

I think you need to increase verbose output and search in
/var/log/asterisk/full for any error message related to IAX2 registration
or simil.

2017-06-05 17:12 GMT-03:00 :

> No, I don't think it is IP table issue, I've not upgraded dd-wrt for a
> while and it was zoiper was working OK with my previous version of
> asterisk.
>
> After upgrade to 11.25.1 it stop working.
> I'm sure port forwarding on dd-wrt is working OK as I have port 80 and
> 443 open.
>
>
> Thelma
> On 06/05/2017 07:12 AM, Christopher van de Sande wrote:
> > Another might be to make sure iptables isn't blocking the connection.
> >
> > You can run
> > iptables -L -n -v
> > To see if its set to block any ports.
> >
> >
> > On June 5, 2017 9:06:55 AM EDT, the...@sys-concept.com wrote:
> >> I'm getting:
> >> netstat -a |grep 4569
> >> udp0  0 0.0.0.0:45690.0.0.0:*
> >>
> >> Should I be getting localhost IP?
> >>
> >> Thelma
> >>
> >> On 06/05/2017 06:48 AM, the...@sys-concept.com wrote:
> >>> Does asterisk listen on port 4569 by default?
> >>>
> >>> I'm running version Asterisk 11.25.1 and have a problem registering
> >>> Zoiper (IAX) to Asterisk.
> >>> I'm getting an error:
> >>> Registration refused
> >>>
> >>
> >> --
> >> _
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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
> >>
> >> asterisk-users mailing list
> >> To UNSUBSCRIBE or update options visit:
> >>   http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
> >
>
> --
> _
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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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Re: [asterisk-users] IAX port 4569

[Victor Villarreal]

No. The 0.0.0.0 listen address is fine.

El 5 jun. 2017 10:06,  escribió:

> I'm getting:
> netstat -a |grep 4569
> udp0  0 0.0.0.0:45690.0.0.0:*
>
> Should I be getting localhost IP?
>
> Thelma
>
> On 06/05/2017 06:48 AM, the...@sys-concept.com wrote:
> > Does asterisk listen on port 4569 by default?
> >
> > I'm running version Asterisk 11.25.1 and have a problem registering
> > Zoiper (IAX) to Asterisk.
> > I'm getting an error:
> > Registration refused
> >
>
> --
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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] IAX port 4569

[Victor Villarreal]

Another idea:

* Run netstat -tulpn command on Linux box AND look if there are an Asterisk
process listening on 4569 UDP port on 0.0.0.0

El 5 jun. 2017 10:00, "Victor Villarreal" <mefhigos...@gmail.com> escribió:

> Dear Thelma,
>
> Yes. Asterisk listen on port 4569 UDP on default config.
>
> Please, look at the Asterisk logfile, for clues about your issue. Or
> enable IAX2 debug vía Asterisk CLI.
>
> Other ideas:
>
> * Check that your server firewall permit UDP port 4569 incoming traffic.
>
> * Run tcpdump over the network interface of your server where the
> registration packets suppose come in. Look ir at least the softphone
> registration request are reaching the server.
>
> * Check if the credentials configured un the softphone mach the
> credentials configured on the server.
>
> Cheers
>
> El 5 jun. 2017 9:48, <the...@sys-concept.com> escribió:
>
>> Does asterisk listen on port 4569 by default?
>>
>> I'm running version Asterisk 11.25.1 and have a problem registering
>> Zoiper (IAX) to Asterisk.
>> I'm getting an error:
>> Registration refused
>>
>> --
>> Thelma
>>
>> --
>> _
>> -- 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] IAX port 4569

[Victor Villarreal]

Dear Thelma,

Yes. Asterisk listen on port 4569 UDP on default config.

Please, look at the Asterisk logfile, for clues about your issue. Or enable
IAX2 debug vía Asterisk CLI.

Other ideas:

* Check that your server firewall permit UDP port 4569 incoming traffic.

* Run tcpdump over the network interface of your server where the
registration packets suppose come in. Look ir at least the softphone
registration request are reaching the server.

* Check if the credentials configured un the softphone mach the credentials
configured on the server.

Cheers

El 5 jun. 2017 9:48,  escribió:

> Does asterisk listen on port 4569 by default?
>
> I'm running version Asterisk 11.25.1 and have a problem registering
> Zoiper (IAX) to Asterisk.
> I'm getting an error:
> Registration refused
>
> --
> Thelma
>
> --
> _
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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] Automatically dial a number, then an extension

[Victor Villarreal]

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"  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] Hack attempt sequential config file read looking for valid files.

[Victor Villarreal]

Hi David, Tim,

Try to use Bail2Ban at last resort. Fail2Ban is a ractive approach, that
permit the traffinc AND ONLY BLOCK them after certain level triggered.


Use iptables to block the unused services faced to public networks like
Internet. And configure these services properly, so they listen only
selected interfaces and IPs, and not from 0.0.0.0

2017-04-21 13:47 GMT-03:00 Tim S :

> Is that IP in your network or outside (I can ping it so I'm guessing it's
> outside your network)?  Do you have a firewall between your asterisk box
> and the internet?  Is there a WHITELIST of IP addresses that only allow
> your provider's limited IP pool to connect to your asterisk box from
> outside?
>
> If you are getting TFTP requests hitting your Asterisk box, they are not
> properly being filtered at your firewall - ftp and tftp are considered
> insecure communication methods, that port (69 I think) should be closed on
> your firewall unless you have a really good reason to have it opened (and
> unless you run a public FTP site, THERE IS NO GOOD REASON).
>
> Fail2Ban is a BLACKLIST method, blacklists are most effective after good
> network hygiene is implemented, as you drastically limit the pool of
> potential bad actors with a whitelist.
>
> Best,
>
> -Tim
>
> On Fri, Apr 21, 2017 at 9:38 AM, Dovid Bender  wrote:
>
>> This is old news. They use Shodan and then try to connect. Set up
>> Fail2Ban that say after 10 404's to ban the IP.
>>
>>
>> On Fri, Apr 21, 2017 at 12:27 PM, Jerry Geis 
>> wrote:
>>
>>> I "justed" happened to look at /var/log/messages...
>>>
>>> I saw:
>>> Apr 21 12:18:40 in.tftpd[22719]: RRQ from 69.64.57.18 filename
>>> 0004f2034f6b.cfg
>>> Apr 21 12:18:40 in.tftpd[22719]: Client 69.64.57.18 File not found
>>> 0004f2034f6b.cfg
>>> Apr 21 12:18:40 in.tftpd[22720]: RRQ from 69.64.57.18 filename
>>> 0004f2034f6c.cfg
>>> Apr 21 12:18:40 in.tftpd[22720]: Client 69.64.57.18 File not found
>>> 0004f2034f6c.cfg
>>> Apr 21 12:18:40 in.tftpd[22721]: RRQ from 69.64.57.18 filename
>>> 0004f2034f6d.cfg
>>> Apr 21 12:18:40 in.tftpd[22721]: Client 69.64.57.18 File not found
>>> 0004f2034f6d.cfg
>>> Apr 21 12:18:40 in.tftpd[22722]: RRQ from 69.64.57.18 filename
>>> 0004f2034f6e.cfg
>>>
>>> so basically an sequential read of polycom MAC address config files.
>>> Some is trying to read to determine if I have any polycom files just
>>> sequential read after read.
>>> And if so - it would get any extension and password at that time.
>>> Luckily I have none.
>>>
>>> However - how does one block attempts like this ?
>>>
>>> Thanks!
>>>
>>> Jerry
>>>
>>> --
>>> _
>>> -- 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
>>>
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>>> To UNSUBSCRIBE or update options visit:
>>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
>>
>>
>> --
>> _
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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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>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
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> Check out the new Asterisk community forum at: https://community.asterisk.
> org/
>
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>
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Re: [asterisk-users] Hack attempt sequential config file read looking for valid files.

[Victor Villarreal]

Hi, Jerry,

I don't know what S.O. you have in the Server, but you can check the man
page (https://linux.die.net/man/8/in.tftpd) for tftpd and use the options
--address, so you can tell tftp from what interface/port this service
listen request.

>From the IP in your logs (69.64.57.18) the request came from a web hosting
provider (http://www.heg.com/). So, the request came from Internet, so your
server listen TFTP request from outside, what is bad.

You can use iptables in any Linux distro to block incoming TFTP traffic.
TFTP is a UDP protocol at port 69.

Example:

/sbin/iptables -A INPUT -i eth0 -p udp --destination-port 69 -j DROP

Change eth0 to the correct name of your public internet server interface.



2017-04-21 13:27 GMT-03:00 Jerry Geis :

> I "justed" happened to look at /var/log/messages...
>
> I saw:
> Apr 21 12:18:40 in.tftpd[22719]: RRQ from 69.64.57.18 filename
> 0004f2034f6b.cfg
> Apr 21 12:18:40 in.tftpd[22719]: Client 69.64.57.18 File not found
> 0004f2034f6b.cfg
> Apr 21 12:18:40 in.tftpd[22720]: RRQ from 69.64.57.18 filename
> 0004f2034f6c.cfg
> Apr 21 12:18:40 in.tftpd[22720]: Client 69.64.57.18 File not found
> 0004f2034f6c.cfg
> Apr 21 12:18:40 in.tftpd[22721]: RRQ from 69.64.57.18 filename
> 0004f2034f6d.cfg
> Apr 21 12:18:40 in.tftpd[22721]: Client 69.64.57.18 File not found
> 0004f2034f6d.cfg
> Apr 21 12:18:40 in.tftpd[22722]: RRQ from 69.64.57.18 filename
> 0004f2034f6e.cfg
>
> so basically an sequential read of polycom MAC address config files.
> Some is trying to read to determine if I have any polycom files just
> sequential read after read.
> And if so - it would get any extension and password at that time.
> Luckily I have none.
>
> However - how does one block attempts like this ?
>
> Thanks!
>
> Jerry
>
> --
> _
> -- 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
>
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Re: [asterisk-users] SIP connections over OpenVPN connection get one-way voice.

[Victor Villarreal]

Hi Ernie,

When one-way audio appear (no matters if there is a VPN or NAT server on
the diagram) I simply :

* Enable SIP debug on Asterisk server. Excecute 'sip set debug ip x.x.x.x'
on Astrisk CLI, where x.x.x.x is the IP of the phone or SIP peer you want
to debug.

* Make a test call and replicate the issue.

* Stop debug with 'sip set debug off'.

* Follow the SIP conversation. Verify that the INVITE message has the
correct IP on the contact field and any other related fields.

* On SDP handshake, verify that the ports where the sound is send, is
correct.

Normally, one-way audio is faced when one audio stream (example the called
audio) is send to the correct IP and Port destination, on the other audio
stream (example the caller audio) don't.

Last, if Asterisk is 'behind' another server, you need tell Asterisk what
is the external IP so it can inform this IP to your clients.

If you dont want to follow the SIP conversation on plain text, you can make
a packet capture on the Asterisk server, instead of SIP debug.

El 19 abr. 2017 16:38, "Mark Wiater"  escribió:

> On 4/18/2017 7:40 PM, Ernie Dunbar wrote:
>
>> Server network: 192.168.0.0/24
>> OpenVPN network: 10.8.0.0/24
>> Asus network: 192.168.1.0/24
>>
>> The Asterisk SIP registration appears to be responding properly to this -
>> this is what I see when I do a 'sip show peer' for an Aastra phone that's
>> connecting through the VPN (Asterisk output is truncated):
>>
>>   ToHost   :
>>   Addr->IP : 10.8.0.6:5060
>>
>
> If the Asus network is 192.168.1.0/24, and the phone is registering as
> 10.0.8.6, it looks like NAT is taking place. Would your asterisk server
> know how to route traffic to 192.168.1.0/24?
>
> I've always used site-to-site OpenVPN tunnels where the vpn's terminate on
> the gateway for both the phones and the asterisk server. I've always had
> rock solid connections between phones and Asterisk.
>
> --
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Re: [asterisk-users] Voicemail asking for login

[Victor Villarreal]

Hi Darcy,

What Pete think is correct.

Maybe excecuting the following command at Asterisk console, will help you:

asterisk> voicemail show users

And you will get a list of all mailbox configured in your system. Search
for the user with problems.

Finally, in the Asterisk wiki you can find more info:

https://wiki.asterisk.org/wiki/display/AST/Configuring+Voice+Mail+Boxes

Cheers

El 18 abr. 2017 21:18, "Pete Mundy"  escribió:

On 19/04/2017, at 7:58 am, D'Arcy Cain  wrote:



Everything looks the same as another one that works except for two things.
The one that works doesn't have the "Probation passed" lines. I am not sure
if that is even part of this call.  The other is the line with "Playing
'vm-login.gsm'" in it.  at that point the working one has this:




Presumably also the line containing 'vm_authenticate: Couldn't read
username' also doesn't appear in the output on a working mailbox either?

I think that's the place to concentrate your efforts.
It shows shortly after the attempt by VoiceMailMain to enter mailbox
'stocktrans2' in context 'VoiceMail'. Does this mailbox exist?

Can you show the equivalent line from a working mailbox (so we can see if
it also uses the context 'VoiceMail', or maybe something else instead, like
'default'?).

Pete


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Re: [asterisk-users] PBX selection

[Victor Villarreal]

Hi Speed Boy.

I agree with Emiliano Vazquez too.

Additionally, you and your team must think others points before choose
Asterisk:

* Asterisk is build to work on Linux. So your team needs some skills like
setting up a basic Linux server (Debian, Centos, etc), donwload software
from Internet, compile and install software manually.

* Your team must know how to configure Linux networking. And solve NAT
issue if apply. Basic network protocols like UDP, SIP and SDP/RDP are
welcome.

* If Asterisk needs interact with external world via VOIP provider, then
you must know how to configure SIP or IAX2 trunks. If you have analog (like
FXO) or digitals lines (like ISDN or similar), then you need ti know how to
install and configure hardware on the Linux server like telephony cards
(PCI-e or PCI) or configure VOIP gateways.

* Security: How to install and configure a basic firewall (using iptables),
o Fail2Ban. And best practices in Asterisk about this topics.

Cheers

El 17 abr. 2017 13:03, "Emiliano Vazquez" 
escribió:

> I prefer Asterisk for my projects.
>
> On Mon, Apr 17, 2017 at 11:57 AM, 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.
>>
>
> Hire a team with knowledge about VOIP, without your prefer if you use
> Asterisk or whatever you want
> You will win a brand new full responsibility with VOIP. The learning
> process is long and hard. You will find a lot of problems like NAT,
> intrusions. Consider learn before you pain this.
>
>
>
>>
>> We have did some searching of Asterisk, here are my questions:
>>
>> 1. Does the last Asterisk using PJSIP stack ?
>>
>
> Yes.
>
>
>> 2. Does there has the comparison of PJSIP and reSIProcate, sofia(using by
>> FreeSwicth) ?
>>
> did you google about this?
>
>
>
>
>> 3. Is it easy to compile and setup Asterisk?
>>
> You need some skills but today is really simple.
>
>
>
>> 4. Which Asterisk version is recommended? And does Asterisk support
>> Windows ?
>>
>> The latest stable release.
>
>
>
>
>> Thanks in advance .
>>
>> Best regards.
>
>
>>
>
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Re: [asterisk-users] Commit dialplan & other config. in memory to disk?

[Victor Villarreal]

Hi Nathan,

Personally, I create a git repo on /etc/asterisk/ folder.

With this approach, you not only can backup current dilplan on another
location (another private server, or private repo on Bitbucket account).
You can follow all the change history you made.

Simply install git, then go to /etc/asterisk/ an issue the following
commands:

#> git init
#> git add.
#> git commit -a 'First commit'

Cheers...

El 7 abr. 2017 10:48, "Steve Edwards"  escribió:

> On Thu, 6 Apr 2017, Steve Edwards wrote:
>
> You're welcome to the script at:
>>
>> http://www.sedwards.com/recover-show-dialplan.php
>>
>
> Sorry about that...
>
> Try:
>
> http://www.sedwards.com/recover-show-dialplan.txt
>
> --
> 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] Manager events showing in CLI

[Victor Villarreal]

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

[Victor Villarreal]

Hi Ron,

I don't remember right now, but you can try this command:

cli> manager set debug off

Cheers

El 26 mar. 2017 3:58, "Telium Technical Support" 
escribió:

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] Which tool to automatically restart Asterisk ?

[Victor Villarreal]

Hi, Oliver.

Maybe something like this (add this script to your crontab):

8<--

#!/bin/bash
#
# File: asterisk-watchdog.sh
# Date: 2015.05.26
# Build:v1.0
# Brief:Secuencia para monitorizar procesos.
#
# ${PATH}: Variable de entorno con las rutas a los ejecutables.
PATH=/bin:/sbin:/usr/bin:/usr/sbin

# ${DAEMON}: Demonio a monitorizar.
DAEMON="asterisk"

# ${MSG}: Cuerpo del mensaje a enviar por mail.
MSG="$(date '+%F %T'): ${DAEMON} se ha caido!"

pidof ${DAEMON} > /dev/null 2>&1

[ $? -ne 0 ] && { echo ${MSG}; service ${DAEMON} start; }

exit 0

--->8---

2017-02-20 11:29 GMT-03:00 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] Turn on SIP debugging from DialPlan

[Victor Villarreal]

Hi Derek,

SIP debug can be enabled via Asterisk CLI (console) with the command:

asterisk> sip set debug on

If you know via what trunk your call goes, you can use the following
command instead:

asterisk> sip set debug ip xxx.xxx.xxx.xxx

Where the xxx is the IP of your trunk (voip to pstn provider).

Affter you make all your test, simply issue:

asterisk> sip set debug off

And all the SIP conversation are saved in your full log file.

More info here:

https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information

If what you want is test your dialplan, simply use the command:

asterisk> dialplan show xxx@your_context

Where xxx is the number you want to dial, from the context asigned to your
extension.

Cheers


El 17/2/2017 19:44, "Derek Andrew"  escribió:

> I have some troublesome numbers that I would like to capture the SIP
> dialogue when I am calling them. When I am about to dial the number, is
> there any way to turn on SIP debugging in the dial plan before I make the
> call? (and turn it off after the call is completed?)
>
>
>
>
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Re: [asterisk-users] Disallow CALLS without registry

[Victor Villarreal]

Hi Antony,

Sory but I don't understand why your Asterisk accept anon calls with the
conf you provide us.

Maybe a full excerpt of an incoming call will help.

Last, there exist dialplan like GROUP and GROUP_COUNT that permits you
count the number of calls in a custom group fashion.

El 10/2/2017 11:51, "Антон Сацкий"  escribió:

> Thanks Frank -- but this not   a solution
> below my  current  config
>
> [general]
>
> ;sms
> accept_outofcall_message= yes
> outofcall_message_context   = messages
> auth_message_requests   = no
>
> ;general
> allowguest  = no
> jbenable= no
> jbimpl  = adaptive
> allow   = !all,g722,ulaw,gsm
> udpbindaddr = 0.0.0.0
> transport   = udp
>
> language= ru
> context = public
> alwaysauthreject= yes
> nat = force_rport,comedia
> directmedia = no
> allowoverlap= no
> match_auth_username = yes
>
> progressinband  = yes
> textsupport = yes
> videosupport= yes
> maxcallbitrate  = 1384
> ;
> sendrpid = pai
> rpid_update = yes
> pedantic=no
>  ;tos
> tos_sip=cs3
> tos_audio=ef
> tos_video=cs4
>
> 2017-02-10 16:40 GMT+02:00 Frank Vanoni :
>
>> On Thu, 2017-02-09 at 14:58 +0200, Антон Сацкий wrote:
>>
>>
>> > so the main question is -- how to Disallow CALLS without registering
>> > on PBX
>>
>> sip.conf configuration
>> In the [general] section, define:
>>
>>
>> [general]
>> ...
>> allowguest=no
>> alwaysauthreject=yes
>> ...
>>
>>
>> The "allowguest" line disables anonymous SIP calls to your PBX. Some SIP
>> providers connect as a guest user, however, so this may be inappropriate
>> for your situation. Also, if you want to accept anonymous SIP calls,
>> this line would block them, so you wouldn't want that. But it is listed
>> here because it is the safest configuration.
>>
>> The "alwaysauthreject" line is important. This causes a hacker to get
>> the same response from your PBX when they try to guess passwords whether
>> or not they guessed a valid username. This also has the side-effect of
>> making poorly written scanning scripts (the vast majority of hacker
>> scripts seem to be poorly written) take less resources on your Asterisk
>> box, as even if they scan a valid username, they'll think it doesn't
>> exist.
>>
>> (Source: https://www.voip-info.org/wiki/view/Asterisk+security )
>>
>>
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>>
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>>
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>
>
>
>
> --
> Best regards
> Antony
> tel.   +380669197533
> tel2. +380636564340
> Paypal http://paypal.me/Satskiy
> 
> satski...@gmail.com 
>
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Re: [asterisk-users] Using g729 now that patents have expired

[Victor Villarreal]

Hi Steve,

I understand your question and your point, but I use the g729 codec from
the link that Carlos share, for almost 6 years from Asterisk 1.4 to v13
without a single problem.

So, sory but I don't share your phrase "from a lesser know web site".

About your question, I did not known that the patent has expired, so I
expect and answer just like you.

Cheers.

El 7/2/2017 19:18, "Steve Edwards"  escribió:

> On Tue, Feb 7, 2017 at 4:47 PM, Steve Edwards 
>> wrote:
>>
>
>   Now that the g729 patents have expired, how do we use g729 in
>>  Asterisk?
>>
>>   Will Digium be releasing a g729 codec for 'free' use or do we
>>   download the 'free' codec off the Internet now that we can use it
>>   without moral or legal restrictions?
>>
>
> On Tue, 7 Feb 2017, Carlos Rojas wrote:
>
> You can uses:
>>
>> http://asterisk.hosting.lv/
>>
>
> I'm hoping Digium will do something so we can have an 'out of the box'
> experience rather than downloading code from a lesser known web site.
>
> --
> 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 - Vtiger integration

[Victor Villarreal]

Hi Alejandro,

The documentation about your question is here:
https://wiki.vtiger.com/vtiger6/index.php/PBX_Manager

After a few seconds of read, I think that VTigerAsteriskConnector can run
on a separate server than Asterisk PBX.

VTigerAsteriskConnector connects to Asterisk via Asterisk Manager Interface
(AMI), so you need to edit your /etc/asterisk/manager_custom.conf (because
you use Elastix distro) and create a user for the VTigerConnector. Then go
to CRM Settings -- > Integration --> PBXManager and complete all the info.

Note that seems that VTigerCoonector needs Java 1.7 onwards.

Please, follows the steps on the links. Cheers.

2017-01-13 16:04 GMT-03:00 Alejandro Cabrera Obed :

> Dear, I have Asterisk 1.8 (installed with Elastix 2.4) and I want to
> integrate a Vtiger 6.5 server.
>
> In my PBX I have Asterisk 1.8, Java 1.4 and I have not Java Jetty.
>
> What are the requirements in the Asterisk server in order to install the
> VtigerAsteriskConnector package and then integrate the services.
>
> Thanks a lot.
>
>
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Re: [asterisk-users] Polycom SoundStation IP 6000 does not register

[Victor Villarreal]

Hi Yves,

Maybe your switch put your Polycom inside a Voice VLAN, based on the MAC of
the phone. Maybe with the snom this not happen because your switch don't
see the MAC of the Snom as a "supperted IP Phone".

2016-12-21 13:59 GMT-03:00 Yves :

> sorry... typo
> the problematic phone has the 192.168.0.13
> the asterisk has 192.168.1.211
>
> when i connect a snom phone on the cable that was in the soundstation 6000
> before and configure the
> phone to use 192.168.0.13 it does register on the asterisk via 5060/UDP...
>
> it would be helpful if someone, that has a running soundstation ip 6000
> could send the configuration... :-/
>
> regards,
> yves
>
>
>
> Am 21.12.2016 um 15:13 schrieb Mauricio Tavares:
>
>> On Wed, Dec 21, 2016 at 7:50 AM, Yves  wrote:
>>
>>> Hi Mark,
>>>
>>> yes, you are right... these are different VLANs
>>> I configured the other phone to use the same IP (192.168.1.13)... and it
>>> worked flawlessly... on the SAME Networkcable in the same plug...
>>> so it must have something to do with the polycom phone config...
>>> remember...
>>> when I use tcp the phone tries to register, but does not even try with
>>> udp...
>>>
>>> thank you,
>>> yves
>>>
>>>I am a bit confused: is your problematic phone's IP 192.168.0.13
>> (what the error log is reporting below) or 192.168.1.13?
>>
>> Am 21.12.2016 um 13:34 schrieb Mark Wiater:
>>>
>>> Yves,
>>>
>>> Didn't you say that
>>>
>>> AsteriskServer: 192.168.1.211
>>> SIP-user: 165
>>>
>>> ?
>>>
>>> On 12/21/2016 4:24 AM, Yves wrote:
>>>
>>> . It is sure for 100% that there is no firewall or something else
>>> mangeling
>>> in between... another Hardphone works as expected using the same
>>> Netzworkcable on the same Networkplug with UDP on Port 5060...
>>>
>>>
>>> This other hardphone, what IP does it have?
>>>
>>>
>>> 50.848|cfg  |*|03|RT|Primary IP changed to 192.168.0.13 subnet mask
>>> 255.255.255.0
>>>
>>> The line above suggests to me that your phone and your asterisk server
>>> are
>>> on a different network, there has to be something that routes between
>>> those
>>> two networks. Often what routes, can firewall.
>>>
>>> 000122.941|sip  |4|03|Registration failed User: 165, Error Code:480
>>> Temporarily not available
>>>
>>>
>>>
>>> Mark
>>>
>>>
>>>
>>>
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>>>
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>>>
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>>> To UNSUBSCRIBE or update options visit:
>>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
>>
>
>
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Re: [asterisk-users] Cisco IP 8841 asterisk integration

[Victor Villarreal]

With all the money you plan to invest in firmware, licenses, etc., you have
bought a Grandstream IP phone or Yealink...
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Re: [asterisk-users] iowait issues on CentOS 7

[Victor Villarreal]

Hi Luca,

IO delay maybe come from Hard Disk lattency. You can exec an "lsof "
command to view what file asterisk proccess hold down when load spike.

If there are some call recording, you can configure Asterisk to make it in
a temp location, a RAM Disk in Linux.

If you make hard usage of the AstDB file, you can copy it to RAM too, to
avoid read/write to the disk.

Please, read this post about lsof:
http://0xfe.blogspot.com.ar/2006/03/troubleshooting-unix-systems-with-lsof.html

You can view something weird in Asterisk logs when high load ? Maybe enable
debug ?

You can install and setup "atop". Then you can review the system status
after the load peak and drill down, just to the problem.

Finally, for troubleshooting IO Wait on a Linux system, you can view this
post: http://bencane.com/2012/08/06/troubleshooting-high-io-wait-in-linux/

Cheers

2016-11-23 12:32 GMT-03:00 Luca Pradovera :

> Hello!
> One of our customers has  an issue where our load average on two of the
> boxes spikes on peak loads. What I got from testing is consistent with what
> they were reporting: on CentOS 7, the load spikes in hockey stick fashion,
> from 40-50% up to 200%, with very high iowait values.
> On CentOS 6, load increases and decreases linearly and the machine never
> slows down.
>
> Asterisk version is 1.8.22.0, which is if course quite old (but it is what
> is installed).
> The CentOS 6 box actually has less RAM (8 Gb vs. 16 Gb), but other than
> that they are exactly identical in hardware configuration.
>
> I checked the usual culprits, but to no avail. Is this a known issue?
>
> Best regards,
>
> Luca
>
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Re: [asterisk-users] iaxmodem errors.

[Victor Villarreal]

Hi John!

I'm not sure why are you using iaxmodem... I use it  a few years ago with
Asterisk 1.4

In Asterisk v11 fax is managed  using res_fax. Please see
https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_ReceiveFAX_res_fax

You only need download, compile and install the spandsp lib for your distro
(fax depend  on it) and then recompile Asterisk (if you don't have this
resource module already).

We currently receive fax via g.711 ir SIP (via T.38), convert it into PDF
and send via email. All  with Asterisk v11 and OpenSource software. I can
send you our scripts if you want.

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Re: [asterisk-users] Force hangup not working on stuck channel

[Victor Villarreal]

Hi Carlos,

Did you try with the following CLI command:

CLI> channel request hangup  CHANNEL_NAME

???

El nov. 3, 2016 1:16 PM, "Carlos Chavez"  escribió:

> I am unable to force a hangup on a channel that has been stuck for over
> two days:
>
> IAX2/from-CD-11006   oficina  27701 Up
> Dial IAX2/to-CD/2883   3467130007  46:24:59 Sotelo
> Sotelo  IAX2/to-CD-20713
>
> I have tried "hangup request IAX2/from-CD-11006" several times but no
> joy.  I also see the following in the CLI:
>
> [Nov  3 10:05:54] WARNING[2879]: chan_iax2.c:4936 handle_call_token: Too
> much delay in IAX2 calltoken timestamp from address X.X.X.X
>
> This is an IAX2 trunk between two Asterisk 1.8 servers (I know it is old
> but new client so haven't had time yet to upgrade to 13).  Because this
> channels is stuck
>  all other calls between servers are not working.  The only way I have
> found to resolve the problem is to stop and restart Asterisk.  This is
> obviously a great inconvinience so is there a way for force iax to unload
> even if there are channels in use?  Or any other way to kill these stubborn
> channels?
>
> --
> Telecomunicaciones Abiertas de México S.A. de C.V.
> Carlos Chávez
> dCAP #1349
> +52 (55)9116-91161
>
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Re: [asterisk-users] Asterisk 13.11.2 unable to register on Centos 7 64bit

[Victor Villarreal]

Ok.

Please, note that 192.168.1.37 (I suspect) is the internal  LAN address Of
the Polycom hardphone. If this is true, then you have  NAT issues.

The REGISTER message are received by your PBX, but when respond, Asterisk
send the next SIP message to the IP informed by the phone, that is the
internal LAN address. The messages do not reach back to the hardphone.

You need to setup a STUN server in the Polycom hardphone settings. Please,
check the manual. Search in Google some public  STUN server to put in the
settings.

Last, the idea behind the "sip set debug" command was view the complete SIP
messages conversation, not search for an error.

On NAT escenarios, remember:

* The NATed phones need to know the public  IP of the NATing router. Either
by manual setting  or  by STUN protocol.

* Reduce the time between REGISTERs attempt, if the client  have a dynamic
IP connection.

* Use the "localnet" SIP settings in Asterisk, so the PBX can distingish
what Network need contacted via NAT and what not.

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Re: [asterisk-users] Asterisk 13.11.2 unable to register on Centos 7 64bit

[Victor Villarreal]

Hi Motty,

Please, set  Verbose  to 3 and Debug to 3 At Asterisk CLI. Then "sip set
debug on".

Now try to register again. At last, " sip  de debug off".

Examine tour console  or  full log file to find some clue ir send me back
some trace.

Cheers.

El oct. 13, 2016 1:45 PM, "Motty Cruz"  escribió:

> Hello, fresh install of Asterisk 13.11.2, client unable to register.  For
> now I have IPtables disabled, also selinux is disabled
>
>
>
> [1006]
>
> type=friend
>
> username=1006
>
> secret=mysecret
>
> context=sip-phone
>
> call-limit=1
>
> callerid="iuser" <1006>
>
> disallow=all
>
> host=dynamic
>
> allow=all
>
>
>
> any ideas?
>
>
>
> Thanks,
>
> Motty
>
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Re: [asterisk-users] Asterisk 13.11.2, 13.11.1, 13.10.0 and certified-13.8-cert3 : freeze on 'sip reload'

[Victor Villarreal]

Hi Jonas!

Do you currently use any TLS technology in your Asterisk? Like SIP-TLS o
pjSIP-TLS support ? If don't, please go to modules.conf and start disabling
some modules that you don't use.

For example, I can see some other modules related to calendars. If you
don't use this, please disable it. You gain a lower memory footprint, and
maybe fix your issue.

I hope this help you. Cheers.

2016-10-11 9:41 GMT-03:00 Jonas Kellens :

> Hello
>
> I am experiencing a freeze of the Asterisk proces when issuing a 'sip
> reload'.
>
> I have this issue every time on asterisk versions : 13.11.2, 13.11.1,
> 13.10.0 and certified-13.8-cert3.
>
> I do not have this on versions certified-13.8-cert2, certified-13.8-cert1
> and asterisk 1.8.32.3.
>
> The only solution is a cold restart of Asterisk.
>
> I can execute any command on CLI except 'sip reload'.
>
> This is what I have on CLI :
>
> sip5*CLI> sip reload
> [Oct  7 23:58:40]  Reloading SIP
> [Oct  7 23:58:40]   == Parsing '/etc/asterisk/sip.conf': Found
> [Oct  7 23:58:40]   == Parsing '/etc/asterisk/sipTemplates.conf': Found
> [Oct  7 23:58:40]   == Parsing '/etc/asterisk/users.conf': Found
> [Oct  7 23:58:40]   == Using SIP TOS bits 96
> [Oct  7 23:58:40]   == Using SIP CoS mark 3
> [Oct  7 23:58:40]   == TLS/SSL ECDH initialized (secp256r1), faster PFS
> cipher-suites enabled
> [Oct  7 23:58:40]   == TLS/SSL certificate ok
>
> --> no more output on CLI. Asterisk has gone completely !
>
> Another 'sip reload' gives :
>
> sip5*CLI> sip reload
> [Oct  8 00:01:10] Previous SIP reload not yet done
>
> sip5*CLI> sip reload
> sip5*CLI>
>
>
> Other commands are no problem on the CLI (while the freeze occurs ! ) :
>
> sip5*CLI> core show  version
> Asterisk certified/13.8-cert3 built by root @ sip5.mydomain.tld on a
> x86_64 running Linux on 2016-10-07 21:27:15 UTC
>
>
> sip5*CLI> sip show channelstats
> Peer Call ID  Duration Recv: Pack  Lost   ( %)
> Jitter Send: Pack  Lost   ( %) Jitter
> 0 active SIP channels
>
>
> sip5*CLI> core show threads
> 0x7f97ff0fb700 2849 netconsole   started at [ 1639] asterisk.c
> listener()
> 0x7f97fe843700 2760 worker_start started at [ 1077] threadpool.c
> worker_thread_start()
> 0x7f97ff367700 2759 worker_start started at [ 1077] threadpool.c
> worker_thread_start()
> 0x7f97fe8bf700 2758 worker_start started at [ 1077] threadpool.c
> worker_thread_start()
> 0x7f97fe93b700 2173 monitor_sig_flagsstarted at [ 4768] asterisk.c
> asterisk_daemon()
> 0x7f97fe9b7700 2172 default_tps_processing_function started at [  200]
> taskprocessor.c default_listener_start()
> 0x7f97fea33700 2171 default_tps_processing_function started at [  200]
> taskprocessor.c default_listener_start()
> 0x7f97feaaf700 2170 default_tps_processing_function started at [  200]
> taskprocessor.c default_listener_start()
> 0x7f97feb2b700 2169 scan_thread  started at [  920] pbx_spool.c
> load_module()
> 0x7f97feba7700 2167 cleanup  started at [  400] pbx_realtime.c
> load_module()
> 0x7f97fec23700 2165 lock_broker  started at [  524] func_lock.c
> load_module()
> 0x7f97fee13700 2161 cal->tech->load_calendar started at [  489]
> res_calendar.c build_calendar()
> 0x7f97fec9f700 2164 default_tps_processing_function started at [  200]
> taskprocessor.c default_listener_start()
> 0x7f97fed1b700 2163 cal->tech->load_calendar started at [  489]
> res_calendar.c build_calendar()
> 0x7f97fed97700 2162 cal->tech->load_calendar started at [  489]
> res_calendar.c build_calendar()
> 0x7f97fee8f700 2160 default_tps_processing_function started at [  200]
> taskprocessor.c default_listener_start()
> 0x7f97fef0b700 2159 default_tps_processing_function started at [  200]
> taskprocessor.c default_listener_start()
> 0x7f97fef87700 2158 default_tps_processing_function started at [  200]
> taskprocessor.c default_listener_start()
> 0x7f97ff003700 2157 do_monitor   started at [11645] chan_dahdi.c
> restart_monitor()
> 0x7f97ff07f700 2156 do_monitor   started at [29518] chan_sip.c
> restart_monitor()
> 0x7f97ff1f3700 2153 worker_start started at [ 1077] threadpool.c
> worker_thread_start()
> 0x7f97ff2eb700 2151 worker_start started at [ 1077] threadpool.c
> worker_thread_start()
> 0x7f97ff26f700 2152 worker_start started at [ 1077] threadpool.c
> worker_thread_start()
> 0x7f97ff3e3700 2149 worker_start started at [ 1077] threadpool.c
> worker_thread_start()
> 0x7f97ff45f700 2148 worker_start started at [ 1077] threadpool.c
> worker_thread_start()
> 0x7f97ff4db700 2147 worker_start started at [ 1077] threadpool.c
> worker_thread_start()
> 0x7f97ff5d3700 2145 worker_start started at [ 1077] threadpool.c
> worker_thread_start()
> 0x7f97ff557700 2146 worker_start started at [ 1077] threadpool.c
> worker_thread_start()
> 0x7f97ff64f700 2144 worker_start started at [ 1077] threadpool.c
> 

Re: [asterisk-users] send a call to moh until user is available

[Victor Villarreal]

Hi Tux John,

The behavior you need is cover in Asterisk within a Queue.

1. Create a new queue in queues.conf and assign as static member, the 4450
extension.

2. In your dialplan, you need to route the incomming calls to the new queue
and pass as argument the timeout in seconds you want when hangup the
waiting calls.

When a new call arrives, it enter the queue. The Callee ear moh music,
while the 4450 ring if its available. Ir not, the queue system wait until
the 4450 become available, an then send the call.

Please, refer to http://www.asteriskguru.com/tutorials/queues.html

The #3 title (simple queue) is all you need.

Cheers
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Re: [asterisk-users] asterisk-users Digest, Vol 147, Issue 5

[Victor Villarreal]

Hi all ! Thanks for your feedback and sory for the delay. Respond:


> Date: Mon, 3 Oct 2016 21:05:55 -0300
> From: Marcelo Terres 
>
> I think that you need the dev files too. In Debian 8, the package is
> libmysqlclient-dev.
>
> But Debian 8 uses libmysqlclient-18. Where did you get the 20 ?
>
> Regards,
>
> Marcelo H. Terres 

Hi Marcelo,

My idea was to install a new PBX to one of my clients, but with the latest
mySQL version.

I follow these instructions: http://dev.mysql.com/downloads/repo/apt/

The libmysqlclient-dev package is installed, but from the mysql repo. It's
version 20.



@Tzafir, thanks for your reply. I respond you inline...

> Date: Wed, 5 Oct 2016 17:13:41 +0300
> From: Tzafrir Cohen 
>
> For the record, we ubild both asterisk 11 (last version: 11.21.2) and 13
> (13.11.2) for Debian Stable using the distro-provided MySQL packages.

Ok, I take note of this. My idea is install Asterisk-11 from source instead
of use a "distribution package".
It's not the first time I compile Asterisk (in fact I work with Asterisk
from v1.4 in production
on many mission critical projects with greats results). But this time I
decided to use the latest mysql version
and face a problem that exceed my knowledge scope :-(

> Is Are there any mysql-related module loaded?
>
> Start with e.g.
>
>  ldd /usr/lib/asterisk/module/cdr_mysql.so

No, there is no mysql-related module compiled or loaded. Only ODBC:

sistemas@nodo1:~$ ls -lh /usr/lib64/asterisk/modules/ | grep mysql

sistemas@nodo1:~$ ls -lh /usr/lib64/asterisk/modules/ | grep odbc
-rwxr-xr-x 1 root root 331K oct  3 17:36 cdr_adaptive_odbc.so
-rwxr-xr-x 1 root root 265K oct  3 17:36 cdr_odbc.so
-rwxr-xr-x 1 root root 339K oct  3 17:36 cel_odbc.so
-rwxr-xr-x 1 root root 382K oct  3 17:36 func_odbc.so
-rwxr-xr-x 1 root root 325K oct  3 17:36 res_config_odbc.so
-rwxr-xr-x 1 root root 350K oct  3 17:36 res_odbc.so

It's like the compile script don't found my v20 libmysqlclient package
installed.
What is the routine responsible for this job?

Asterisk here in this server, was compiled with a libmysqlclient v18, but I
unistalled this package now
because Asterisk-11 compiled with v18 and connected to a v5.7 mySQL
instance, return an Asterisk crash on libmysqlclient.so module.

If it's don't possible to compile Asterisk-11 against libmysqlclient-20, I
will have to downgrade my mySQL-5.7 instance back to 5.5 Debian version :(

Any idea?

Thanks in advance and best regards.

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[asterisk-users] Asterisk 11.23 with libmysqlclient20 on Debian 8

[Victor Villarreal]

Hi List!

I'm facing a problem while compiling Asterisk-11 on a Debian 8 server.

The mysql-server version installed is 5.7 and come from the official mySQL
community repo for Debian.

After compile, install and execute Asterisk, the comman "lsof -p `pidof
asterisk` | grep mysql" don't produce any output. Like if confgure script
don't found the mysql lib.

With libmysqlclient18 every is Ok. How can I use libmysqlclient20 with
Asterisk ?

Thanks in advance, and best regards.

root@nodo1:/usr/src/asterisk-11.23.0# ls -lh /usr/lib/x86_64-linux-gnu/ |
grep mysql
-rw-r--r-- 1 root root 5,7M ago 25 09:37 libmysqlclient.a
lrwxrwxrwx 1 root root   20 ago 25 09:37 libmysqlclient.so ->
libmysqlclient.so.20
lrwxrwxrwx 1 root root   24 ago 25 09:37 libmysqlclient.so.20 ->
libmysqlclient.so.20.3.2
-rw-r--r-- 1 root root 4,2M ago 25 09:37 libmysqlclient.so.20.3.2
-rw-r--r-- 1 root root  18K ago 25 09:37 libmysqlservices.a

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Re: [asterisk-users] how to read sip debug

[Victor Villarreal]

Hi Thufir,

The analysis of a SIP Debug depends on what the problem to be solved.

If you experience problems with inbound calls from a SIP trunk or
provider, you can type in Asterisk cli 'core set debug 3' and then
'sip set debug ip xxx.xxx.xxx.xxx' where xxx is the IP of your SIP
provider or from where it is supposed to come call.

Then you make a test call, and look in full log an INVITE message
(note that you analize an OPTION message in your mail, but I think
that this not help in this case).

After the incoming INVITE message from your SIP provider, you can
follow the rest of the Asterisk logic and look for the reason why
Asterisk is denying that call.

Hope this help you.

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[asterisk-users] Compiler errors when 'make asterisk' for D100 transcoding board

[Victor Villarreal]

Hi List,

I solve this issue and I want share it with this community.

The sng-tc-linux-1.3.8 package don't compile across Certified Asterisk.
Only normal Asterisk like 11.22.0 version.

We have this version in production with the D100 board. Working.

Cheers

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[asterisk-users] Identify more demanding routine inside Asterisk

[Victor Villarreal]

Hi List !

I'm facing a problem with the CPU consumption in Asterisk 11.22.0.

I could decrease a lot of load, migrating both the astdb.sqlite3 and call
recordings (with Monitor app) to a tmpfs mount in RAM (with noatime and
nodiratime flags), manually spread each of the hardware interrupts (network
interfaces, wanpipe and megasas) to an individual dedicated CPU core and
stick the Asterisk process to other dedicated CPU core (free of hardware
interrupts).

Now the usage is 90% of his core with 300 active channels (66% SIP + 17%
IAX + 17% DAHDI). 17% of channels are g.729 transcoded via software. Some
considerations are that there is +30 AMI concurrent clients, plus SIP
Realtime peers. I don't have any AGI script or ODBC custom function/query.
The DialPlan is minimal for the work it does too.

My question is: Is there any way to identify the more demanding
routine/task in Asterisk so I can know where to tweak ?

I read the excelente post of Moy [1], but the most usage of CPU is in the
mail process of Asterisk, not in any of the threads. Moreover, the pstack
command don't work on my 64-bits system.

Hope anyone can help. Cheers.

[1]
https://moythreads.com/wordpress/2009/05/06/why-does-asterisk-consume-100-cpu/

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Re: [asterisk-users] nagios asterisk check SIP

[Victor Villarreal]

On Fri, Jun 17, 2016 at 11:22:48AM +0200, Thomas wrote:
> Iam loocking for an programm to check the SIP port of an Asterisk
asterisk.
>
> Ome time ago I have used
> #/usr/bin/sipsak
> but it seemed that it is not working anymore?

Hi Thomas,

Maybe this links help you:
http://fabian-affolter.ch/blog/nmap-scripts-for-voip-analyses/

Not for sipsak, but for great nmap.

Cheers

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Re: [asterisk-users] queue_log - odbc vs AMI

[Victor Villarreal]

Hi Marek,

Here, we have an Asterisk v11-cert11 and found that there is NOT equal the
CDR via AMI and CDR in Database.

Please, check my gist:
https://gist.github.com/MefhigosetH/89462e599a996dedf048f8d2b4e94d47

We have in use some custom dialplan variables in CDR (ie.: groupcount and
rptqos), and these variables are visibles in CDR table BUT ARE NOT SHOW in
AMI Event.

Hope this help. Cheers from Argentina.


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Re: [asterisk-users] SPA112 flapping

[Victor Villarreal]

Hi Mike,

I would try the following:

* If you can login through HTTP, check the uptime of the Cisco device. Make
sure the device is not rebooting.
* If you can, make a 'ping' from the PBX to the device and annotate
milli-seconds of response. Then compare then to the default 'qualify' sip
setting for the Cisco peer (width sip show peer _SPA112_PEER_NAME_). Maybe
you can set 'qualify=X' where 'X' is the measured round-trip time to a peer.
* If the Cisco is behind a NAT device/router, maybe the default 60 seconds
for the 'qualifyfreq=60' sip setting is not enough to keep active the
session. Try changing this value to something lower like 15 or 30 seconds.

Cheers

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[asterisk-users] Compiler errors when 'make asterisk' for D100 transcoding board

[Victor Villarreal]

Hi there !

Someone in this wonderful list tried to install Sangoma transcoding board
D100  on Asterisk v11 ?

I followed each of the steps in the wiki [1], but when running 'make
asterisk' receipt compilation errors about the absence of some header files
[2].

I exchanged some mail with the official support, but still I have not
received any solution.

Using the following software version:

* Debian 7.11 with up-to-date packages.
* Linux nodo3 3.2.0-4-amd64 #1 SMP Debian 3.2.78-1 x86_64 GNU/Linux
* certified-asterisk-11.6-cert11 compiled from official Asterisk-org source
code.
* sng-tc-linux-1.3.8.x86_64

TIP: I've tried compile again through v11.22.0 source code with similar
resoults.

Thanks in advance, and best regards.

[1] http://wiki.sangoma.com/Asterisk-D100-Single-Server-Installation
[2] https://gist.github.com/MefhigosetH/883589726c52b8dc72f3cfd6825fe3f1

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