Fiat Panda windscreen wiper timer

My Panda, after almost 16 years of service, begins to show its years.
This time, the front  wiper does not work properly.
You may read almost evreywhere, Google is your friend....
About this matter, it seems nobody ever went deeper, and to my issue, which seems to be quite widespread, Google wouldn't helped that much; the only answer I found by it is "change the motor and its controlling circuit". This solution can be quite expansive, looking around for prices,  it seems you can not find it for less than 100€.
On the other hand, looking at the controlling circuit, it does not seem to be very complicated, so I decided to use some time to resolve my issue, and therefore save 100€.

As from the image I included in this post, circuit itself is not too complicated.
I also made some schemas to help myself in understanding its behaviour.

The switch appearing in the bottom of the schema, is embedded in the gear inside the transmission of the motor. Its rotation make the controller understand about the wiper position. It is a plastic gear with the core made of conductor material. It appears as follow:

Where the RED part is the metallic core, and the black lines is where contact from the control circuit goes.

Well, I guess this is the time where I tell you which is my problem, after all.
Problem is as follow:  the "SLOW" speed of the wipers does not work, and when it is set to the "FAST" speed, if you turn off the wiper it stops where it is, without return to its home postion.
After several hours of analysis, I found where the problem lies.
It is due to the oxidation of the normally open contact of the relay.

This way I saved myself more than 100€ for the spare part, I hope anyone else could found my info useful and save his money.

Block aggressive advertising calls

In my country, but I bet anywhere else do the same, the telephone calls for proposing new products are constantly increasing, these annoying calls, sometimes malicious, aren't always welcome.

My country has adopted a kind of institutional way to handle this phenomena, it is called  "registro pubblico delle opposizioni", using this, any citizen may publish his telephone number, merely this action should him guarantee he would never ever receive again any advertisement call.

Unfortunately, reality can't be more distant from what really is going to happen.

Telephone spammers do not use to follow rules, and the weak statement "this user does not want to be called for commercial purposes "  is not enough to prevent they to call you. 

If you do not want to be called, you have to find yourself a way to do that.
In other words, you have to find a way to recognize them as spammer, and forward them, for example, to an announce stating you do not want to be called.

As an Asterisk user I have a framework at my disposal, which potentially lets me route calls using any sort of policy; but the main issue is how to identify a spammer?

Internet community, seems to have faced this problem and has built a kind of database in which people complaints about telephone numbers.

It exists, for example, a service called TELLOWS on which anyone can leave a comment on a source telephone number.

So why don't use those information to identify spammers and avoid unwanted telephone calls?

My proposal is integrate the service provided by TELLOWS in the Asterisk dialplan.

Here an example of my thougths.

in the Asterisk dialplan

exten =>  incoming,       1,              Noop(Incoming call is just arrived)
exten =>  incoming,       n,              Set(tellowsc=${SHELL(tellows_c.sh ${CALLERID(num)})})
exten =>  incoming,       n,              GotoIf($[${tellowsc}>1]?reject)
exten =>  incoming,       n,              Dial(${destination},30,${FLAGS})
exten =>  incoming,       n,              HangUP()
exten =>  incoming,       n(reject),      Answer()
exten =>  incoming,       n,              Wait(2)
exten =>  incoming,       n,              Set(CDR(userfield)=RejectedByPolicy)
exten =>  incoming,       n,              Playback(spam)
exten =>  incoming,       n,              HangUP()

the bash script for the italian version of TELLOWS

#!/bin/sh
score=$(wget -O - -q "http://www.tellows.it/num/$1" | grep "/images/score/score[0-9].png" | grep scoreimage | sed -r 's/.*score([0-9])\.png.*/\1/');
comments=$(wget -O - -q "http://www.tellows.it/num/$1" | grep "itemprop" | grep "Numero di commenti:"| sed -r 's/.*>([0-9]+)<.*/\1/');
echo -n "$comments";

I used, to reject calls, the number of comments the number has, but there's also a kind of number score, you can use to take your decision

JabberSend crashes your system if it has no connection with the XMPP server.

JabberSend is quite a useful function.
It lets you send xmpp instant messages, and it can be used for various things such as alert you when a call hits your voice mail or also that someone is calling you.

Here an example dial plan.
[outside_line]
exten =>  s, 1,  Noop(Incoming call is just arrived)
exten =>  s, n,  JabberSend(XMPP_Account,Jid@domain,Incoming call from "${CALLERID(num)}")
exten =>  s, n,  Dial(${GLOBAL(incoming_call_group)},30,${FLAGS_incoming})
exten =>  s, n,  JabberSend(XMPP_Account,Jid@domain,${CALLERID(num)} is leaving VM message)
exten =>  s, n,  VoiceMail(${GLOBAL(incoming_voicemail)}@asterisk)
exten =>  s, n,  Hangup()

Nice feature, but it comes with a big problem: if for any reason the XMPP connection is down, JabberSend crashes your Asterisk system.
This kind of behaviour is obviously not admittable in any kind of production environment.

You can however use it in a production environment making a slight modification to the code source code of res_xmpp.c

In the Asterisk version I'm using, the 11.6, modifying the function xmpp_client_send_message as follow, the result is:
if JabberSend is invoked and there's no connection to the xmpp server it logs a warning message instead of crashing the system.

here the code, the red indicates where I modified the original code:
static int xmpp_client_send_message(struct ast_xmpp_client *client, int group, const char *nick, const char *address, const char *message)
{
        RAII_VAR(struct xmpp_config *, cfg, ao2_global_obj_ref(globals), ao2_cleanup);
        RAII_VAR(struct ast_xmpp_client_config *, clientcfg, NULL, ao2_cleanup);
        int res = 0;
        char from[XMPP_MAX_JIDLEN];
        iks *message_packet;

        if (!cfg || !cfg->clients || !(clientcfg = xmpp_config_find(cfg->clients, client->name)) ||
            !(message_packet = iks_make_msg(group ? IKS_TYPE_GROUPCHAT : IKS_TYPE_CHAT, address, message))) {
                return -1;
        }

        if (!ast_strlen_zero(nick) && ast_test_flag(&clientcfg->flags, XMPP_COMPONENT)) {
                snprintf(from, sizeof(from), "%s@%s/%s", nick, client->jid->full, nick);
        } else {
                snprintf(from, sizeof(from), "%s", client->jid->full);
        }

        iks_insert_attrib(message_packet, "from", from);

        //begin//
        if (client->state == XMPP_STATE_CONNECTED) {
                                                    res = ast_xmpp_client_send(client, message_packet);
                                                   }
                                              else {
                                                    res = -1;
                                                    ast_log(LOG_WARNING, "XMPP: Not connected, unable to send messages to %s\n", client->name);
                                                   }
        //end//

        iks_delete(message_packet);

        return res;
}I hope this can help.

Access your PBX using voice channels

As VoIP PBX administrators, we often have to deal with uncooperative network administrators. IP PBXs rely on the underlying network infrastructure for both normal operation and remote access. However, they frequently encounter restrictive network policies, which can complicate remote management. It can be particularly frustrating when network administrators implement policies without consultation, often resulting in denied remote access for everyone except themselves. In such scenarios, even a simple task like renaming an extension (e.g., changing 400 to something else) can become time-consuming. You may need to visit the client’s location to make the change on-site. In the best-case scenario, you might ask the client to use a PC with software like TeamViewer, but this still adds unnecessary complexity. If you’ve faced similar challenges, you know how frustrating it can be.

In this article, I won’t claim to provide the ultimate solution, but I’ll share a method to mitigate the situation. It requires no additional hardware and no human assistance.

Do you remember the days of using analog modems to connect to the internet? At the time, affordable digital lines were unavailable, so modems were the only option. The solution I propose is to use voice lines—VoIP or otherwise—in a way similar to how modems were used back then.

1. Linux Ham Radio AX25 software suite
2. Alsa loopback device
3. Asterisk Alsa channel

You might wonder: why use amateur radio software for this? There are several reasons. First, this software is mature, well-tested, and readily available. Additionally, old modems required a continuous carrier signal to maintain a connection. This could lead to latency issues over VoIP connections, similar to the problems you encounter with failed FAX transmissions. In contrast, amateur radio packet protocols only generate a carrier during actual data transmission, making them less susceptible to latency problems.
The final idea should look such as the following images:

This solution won’t provide the highest connection speeds for your Asterisk server. The maximum achievable speed with this method is 38,400 bps. However, it allows you to remotely manage your installation without relying on others for remote access!

The implementation

The Amateur Radio AX.25

To begin, ensure that the Ham Radio kernel modules are installed. You can verify their presence by checking for mkiss.ko in the /lib/modules/ directory, which indicates that the necessary modules are available.
A Note for Red Hat-Based Distributions
For Red Hat-derived distributions, such as Red Hat Enterprise Linux 5 and 6, as well as the corresponding CentOS versions and possibly others, the Ham Radio kernel modules are not built by default. In this case, you’ll need to build them manually.
For Red Hat users, the best approach is to download the kernel source using the yum utility. Then, integrate the missing Ham Radio kernel modules by transplanting them from the corresponding kernel sources available at www.kernel.org. (You can refer to guides on building kernel modules on CentOS systems after performing the transplant.)
Once your kernel setup is complete, you can turn your attention to the user-space utilities required for this task. Required User-Space Packages It’s recommended to build these utilities from source to ensure compatibility with your environment. Regardless of your approach, ensure you have the following packages installed:

• ax25-apps
• ax25-tools
• libax25
• soundmodem

The AX.25 protocol stack is relatively simple to configure. In my case, the only necessary change was adding a single line to the axports file located in the /etc/ax25 directory. The axports file defines port names and their associated mappings. Here’s an example entry:

sm0     HQH1            4800    255     2       Test connection

The fields in this file represent the following:

• Port Name: The name of the new interface used to send packets through the voice link.
• Callsign: The L2 address used by the AX.25 protocol. In amateur radio, this is assigned by an authority, but in our case, any identifier will work.
• Speed: The communication speed between the TNC and the computer. In this setup, it refers to the communication speed between the kernel and the soundmodem (a software implementation of the TNC).
• MTU: The Maximum Transmission Unit for the AX.25 connection.
• AX.25 Window Parameter: This is equivalent to the "maxframe" setting.
• Description: A description of the connection.

An example of its configuration is the following:

<?xml version="1.0"?>
<modem>
<configuration name="AX25">
<chaccess txdelay="150" slottime="100" ppersist="40" fulldup="1" txtail="100"/>
<audio type="alsa" device="plughw:Loopback,0,0" halfdup="0" capturechannelmode="Mono"/>
<ptt file="none" hamlib_model="" hamlib_params="" gpio="0"/>
<channel name="sm0">
<mod mode="fsk" bps="4800" f0="1200" f1="2200" diffenc="1" inlv="8" fec="1" tunelen="32" synclen="32" filter="df9ic/g3ruh"/>
<demod mode="fsk" bps="4800" f0="1200" f1="2200" diffdec="1" inlv="8" fec="3" mintune="16" minsync="16" filter="df9ic/g3ruh"/>
<pkt mode="MKISS" ifname="sm0" hwaddr="HQH1" ip="23.0.0.1" netmask="255.255.255.0" broadcast="23.0.0.255"
file="/dev/soundmodem0" unlink="1"/>
</channel>
</configuration>
</modem>

This setup has been proven to work, but it is slow—limited to 4,800 bps! If your voice path is clear, there is room to achieve higher speeds with adjustments. To simplify the explanation, I’ve divided the configuration into color-coded sections:

• Orange: This section specifies the media access. I used some default settings, but keep in mind that the media we are working with is significantly different from what this protocol was originally designed for. Our media is full-duplex, so we don’t need to adhere to time constraints like time slots or transmission delays.
• Magenta: This section defines the ALSA hardware we’ll use. I’ll explain this in more detail in the next section, "ALSA Loopback Device." For now, note that in this section, you must specify an ALSA aloop device channel, prefixed by "plug" to handle resampling issues.
• Yellow: This section sets the channel parameters and binds the configuration to the Linux network interface—in this case, sm0.
• Green: Here, you configure the modulation and demodulation method, throughput, and frequencies. Feel free to experiment with different combinations to find what works best in your specific environment.
• Cyan: This section outlines the packet mode parameters. It assigns the Layer 2 and Layer 3 addresses for your connection.

Alsa loopback device aloop

The primary reason for using this piece of software is to interface Asterisk with the soundmodem device. Both Asterisk and the soundmodem rely on ALSA devices for input and output, making the ALSA loopback device the most natural way to connect them.
The aloop module is included in every Linux distribution I’m aware of, so finding one ready for use should not be an issue. In most cases, you simply need to load the aloop module into the kernel.

modprobe snd-aloop

This should result in 8 new channels ready to be used

 card 2: Loopback [Loopback], device 0: Loopback PCM [Loopback PCM]
  Subdevices: 8/8
  Subdevice #0: subdevice #0
  Subdevice #1: subdevice #1
  Subdevice #2: subdevice #2
  Subdevice #3: subdevice #3
  Subdevice #4: subdevice #4
  Subdevice #5: subdevice #5
  Subdevice #6: subdevice #6
  Subdevice #7: subdevice #7
card 2: Loopback [Loopback], device 1: Loopback PCM [Loopback PCM]
  Subdevices: 7/8
  Subdevice #0: subdevice #0
  Subdevice #1: subdevice #1
  Subdevice #2: subdevice #2
  Subdevice #3: subdevice #3
  Subdevice #4: subdevice #4
  Subdevice #5: subdevice #5
  Subdevice #6: subdevice #6
  Subdevice #7: subdevice #7

We only need one of the ALSA loopback channels, so you can adjust the module loading options by specifying parameters in your distribution's configuration file for kernel modules, such as modprobe.conf or its equivalent.

Asterisk

The Asterisk portion of the configuration utilizes the less commonly used chan_alsa module. The Asterisk ALSA channel driver was originally developed to allow the server console to function as an endpoint in the telephony system. As the name suggests, chan_alsa uses ALSA-compatible devices to manage the voice stream. To adapt this for our setup, the only modification needed in the default alsa.conf configuration is specifying the ALSA device to be used—specifically, the aloop device.

[general]
autoanswer=yes
context=local
extension=s
input_device=plughw:Loopback,1,0
output_device=plughw:Loopback,1,0

At this point, by directing a call to this endpoint, we have effectively created one leg of the tunnel. For example, in extensions.conf:

exten => 929,           1,              Dial(Console/snd/controlC0,30)

Note:
During actual tests of the tunnel, although AX.25 is treated as a type of Ethernet and implements an ARP-style Layer 2 address resolution, I had to set it manually.
If you encounter the same issue, the ARP syntax for AX.25 bindings is as follows:

arp -H ax25 -s 23.0.0.1 HQH1

Google Talk ends its services, now what?

Due to the decision Google made to turn off its xmpp service talk, I've been forced to migrate another xmpp service.
I thought it would be an easy task, just put new data into xmpp.conf and it's done. But it didn't go that way.
As soon as I run the new configuration, I got:

<--- XMPP sent to 'tigase.im' --->
<?xml version='1.0'?><stream:stream xmlns:stream='http://etherx.jabber.org/streams' xmlns='jabber:client' to='tigase.im' version='1.0'>
<------------->

<--- XMPP received from 'tigase.im' --->
<?xml version='1.0'?><stream:stream xmlns='jabber:client' xmlns:stream='http://etherx.jabber.org/streams' from='tigase.im' id='6be9205d-5793-4ae6-84e0-afd0983ed490' version='1.0' xml:lang='en'>
<------------->

<--- XMPP sent to 'tigase.im' --->
<starttls xmlns='urn:ietf:params:xml:ns:xmpp-tls'/>
<------------->

<--- XMPP received from 'tigase.im' --->
<stream:features><sm xmlns="urn:xmpp:sm:3"/><register xmlns="http://jabber.org/features/iq-register"/><mechanisms xmlns="urn:ietf:params:xml:ns:xmpp-sasl"/><ver xmlns="urn:xmpp:features:rosterver"/><starttls xmlns="urn:ietf:params:xml:ns:xmpp-tls"><required/></starttls><compression xmlns="http://jabber.org/features/compress"><method>zlib</method></compression></stream:features>
<------------->

<--- XMPP received from 'tigase.im' --->
<proceed xmlns="urn:ietf:params:xml:ns:xmpp-tls"/>
<------------->
[Feb 16 19:20:56] ERROR[20733]: res_xmpp.c:2558 xmpp_client_requested_tls: TLS connection for client 'tigase.im' cannot be established. OpenSSL initialization failed.
[Feb 16 19:20:56] WARNING[20733]: res_xmpp.c:3571 xmpp_client_receive: Parsing failure: Hook returned an error.
[Feb 16 19:20:56] WARNING[20733]: res_xmpp.c:3635 xmpp_client_thread: JABBER: socket read error
What the hell was there?
I reviewed all the res_xmpp parameters, but at last I had to accept to the fact it doesn't work.
A xmpp provider related issue?
Might be, but changing to another xmpp service didn't help, and connection kept fail.

Debug log states it is a TLS connection related problem...
I started my investigation on iksemel, the xmpp parser library.
It has its own tls mechanism based on gnutls, which is independent with the one implemented in res_xmpp, but at that time I didn't know yet.

There is a tool in the "tool" directory of the iksemel library which performs a simple xmpp connection.
It is the iksroster.
Using that to test the xmpp connection I got a TSL connection error.

Looking at the code, the I found in the src/stream.c file that function handshake performs the negotiation for the TLS connection:

static int handshake (struct stream_data *data)
{
        const int protocol_priority[] = { GNUTLS_TLS1, GNUTLS_SSL3, 0 };
        const int kx_priority[] = { GNUTLS_KX_RSA, 0 };
        const int cipher_priority[] = { GNUTLS_CIPHER_3DES_CBC, GNUTLS_CIPHER_ARCFOUR, 0};
        const int comp_priority[] = { GNUTLS_COMP_ZLIB, GNUTLS_COMP_NULL, 0 };
        const int mac_priority[] = { GNUTLS_MAC_SHA, GNUTLS_MAC_MD5, 0 };
        int ret;

        if (gnutls_global_init () != 0) return IKS_NOMEM;              
        if (gnutls_certificate_allocate_credentials (&data->cred) < 0) return IKS_NOMEM;       
        if (gnutls_init (&data->sess, GNUTLS_CLIENT) != 0) {
                                                            gnutls_certificate_free_credentials (data->cred);
                                                            return IKS_NOMEM;
                                                           }
        gnutls_protocol_set_priority (data->sess, protocol_priority);
        gnutls_cipher_set_priority(data->sess, cipher_priority);
        gnutls_compression_set_priority(data->sess, comp_priority);
        gnutls_kx_set_priority(data->sess, kx_priority);
        gnutls_mac_set_priority(data->sess, mac_priority);
        gnutls_credentials_set (data->sess, GNUTLS_CRD_CERTIFICATE, data->cred);

        gnutls_transport_set_push_function (data->sess, (gnutls_push_func) tls_push);
        gnutls_transport_set_pull_function (data->sess, (gnutls_pull_func) tls_pull);
        gnutls_transport_set_ptr (data->sess, data->prs);

        ret = gnutls_handshake (data->sess);
        if (ret != 0) {
                       gnutls_deinit (data->sess);
                       gnutls_certificate_free_credentials (data->cred);
                       return IKS_NET_TLSFAIL;
                      }
        data->flags &= (~SF_TRY_SECURE);
        data->flags |= SF_SECURE;
        iks_send_header (data->prs, data->server);
        return IKS_OK;
}
The code shows the function is using a deprecated chiper, it could be the reason why TLS connection fails.
A test that worth to try is add another more recent cipher.
const int cipher_priority[] = { GNUTLS_CIPHER_AES_256_CBC,GNUTLS_CIPHER_AES_128_CBC,GNUTLS_CIPHER_3DES_CBC, GNUTLS_CIPHER_ARCFOUR, 0};
Indeed The new version made the trick, and that time on, iksroster successfully connected to the server.
In the asterisk console however, no change.
Looking with attention to the res_xmpp.c code, I finally got the problem: asterisk does its own TLS connection using OpenSSL, and it is bound to use SSLv3.
You can be more lucky than me, but it seems not many xmpp servers are using SSLv3!
You can test yourself  you xmpp provider using the tools provided by https://xmpp.net/, in my case, tigase.im used TLSv1.2.
The only way I found to make my connection work has been patch my res_xmpp.c to use TLSv1.1

static int xmpp_client_requested_tls(struct ast_xmpp_client *client, struct ast_xmpp_client_config *cfg, int type, iks *node)
{
#ifdef HAVE_OPENSSL
        int sock;
#endif
        int dbg_info=0;

        if (!strcmp(iks_name(node), "success")) {
                /* TLS is up and working, we can move on to authenticating now */
                xmpp_client_change_state(client, XMPP_STATE_AUTHENTICATE);
                return 0;
        } else if (!strcmp(iks_name(node), "failure")) {
                /* TLS negotiation was a failure, close it on down! */
                return -1;
        } else if (strcmp(iks_name(node), "proceed")) {
                /* Ignore any other responses */
                return 0;
        }

#ifndef HAVE_OPENSSL
        ast_log(LOG_ERROR, "Somehow we managed to try to start TLS negotiation on client '%s' without OpenSSL support, disconnectin$
        return -1;
#else
//        client->ssl_method = SSLv3_method();  //this was the original
        client->ssl_method = TLSv1_1_method();
        if (!(client->ssl_context = SSL_CTX_new((SSL_METHOD *) client->ssl_method))) {
                dbg_info=1;
                goto failure;
        }

        if (!(client->ssl_session = SSL_new(client->ssl_context))) {
                dbg_info=2;
                goto failure;
        }

        sock = iks_fd(client->parser);
        if (!SSL_set_fd(client->ssl_session, sock)) {
                dbg_info=3;
                goto failure;
        }

        if (!SSL_connect(client->ssl_session)) {
                dbg_info=4;
                goto failure;
        }
Now my connection works.

I hope you find this useful.