Antipyretic remedies for children are prescribed by a pediatrician. But there are situations of indispensable help for fevers, if the child needs to give faces negligently. Then the fathers take on themselves the resuscitation and stop the antipyretic preparations. What can be given to infants? How can you lower the temperature in older children? What are the most safe faces?
Perelіk dіy to the cob vikoristannya possessing VoIP
To install VoIP phones or VoIP gateways, you need to enter a number of front doors:
- Take the password to skin oblіkovogo record SIP. Passwords are required for setting up a password. To recover passwords, enter the following:
- Change the password before the SIP public record and override it when changing the password.
- Send yourself line-up SIP email.
- Link a SIP physical record to a product;
- Bind a SIP public record to an IP address.
Removal of the password to the SIP public record
Before the beginning of the cob, the VoIP phone or VoIP gateway administrator should note the following in the electronic mailbox looking like a friend password to the SIP public record. The password is needed for a remote control of the possession.
Removing the password can be done in one of the following ways:
- Password change and the creation of a new password for another.
- Strengthening the current password on the specified email address.
Changing the password before the SIP public record
To change the password before the SIP public record and to see the password for another, you need to enter the following:
Enforcement of the password by electronic mail
To send the current password to the SIP public record by e-mail, please enter the following:
Linking a SIP physical record to a product
To link a SIP face record to a product, type in the following:
Linking a public SIP record to an IP address
In addition, as you have chosen the object, in order to bind the SIP oblique record, it is also necessary to bind the oblique record to the IP address, or else. Binding to an IP address is configured in the "Binding" parameter group: 
Note
The options for linking a SIP observable record to an IP address are not available, as the "Do not change" flag has been set.
- Select binding type:
- Ні - SIP public record will not be associated with the same IP address.
- Auto - SIP public record will be assigned automatically. Sound off at the first registration of the oblique record
- So, if this option is selected, it is necessary to add the required number of IP address or the address in the IP address entry field. The field is no longer available if you select the "Yes" option. Enter the IP address and press "Add". Then enter the next IP-address or the mask of the 1.2.3.4/24 type via / Specify the mask of the tier.
Add the necessary IP-addresses in this order.
Respect!
Selecting the "Hi" option may cause harm, security, shards, the possibility of an evil SIP overwrite is allowed. Binding to an IP address prevents access to a public record from other IP addresses.
Group binding of SIP public records to a product and an IP address
For your convenience, the system has been transferred the ability to bind at once a number of cloud records to a product or an IP address. For group linking of public records, type the following:
SIP is the current standard for IP telephony. Widely victorious as commercial VoIP operators, as well as cost-free services such as Google Voice, Voxalot, Ekiga.net, Sipnet.ru, etc., support faceless soft/hard phones and adapters, support video telephony. On the vіdmіnu vіd skype є є tverdzhenim, zagaloviznim and vіdkritim signaling protocol of VoIP systems.
SIP does not have a single global hub for managing that registration. іsnuє impersonal rіznih servіs registrar providers. You can draw an analogy with email, or more precisely, jabber servers.
In addition, be a kind of koristuvach or an organization can run an authoritative server.
Whether SIP addresses (sip uri) are accumulated from the login of the server and may look, for example "". In principle, letters are allowed in the login, but I would rather be surrounded by numbers, so that later we would not have problems dialing such a number from telephones without an alphabetic keyboard. You can have a large number of SIP accounts on one individual server, for various purposes.
Skіlki SIP - vіdkritiy standard, іsnuє impersonal software or hardware SIP clients, the stink of the sum one with one and with any servers. In order to establish a connection, clients do not have to change the language on the same server. At the next hour, the stench will be automatically caught between itself without any intermediary.
To connect to SIP telephony, you need to select a registrar and a client. We can recommend sipnet.ru or comtube.ru to the fellows. CoryStuvacam Windows Sipnet propane your softphone from pre-settings. Vіn can be done more with this registrar, but it may not be necessary to do it. Another popular softphone for this OS is X-lite.
For *Nix bad systems, the option is Ekiga or SFLphone with the same registrars. (Ekiga is also available for Windows)
Don't tag the ekiga.net registrar that promotes in Ekiga. Winning protests and revealing problems with input calls. We'll check it out during the first launch, then select "add SIP public record" from the cloud record manager.
Different registrars can give a different set of services and responsibilities. Acts are more focused on calls to the telephone line (PSTN termination), acts are on a difference online services. Choose those that suit you the most, or greet yourself with dekilkom, in SIP it is generally a normal practice. The geographic position of the registrar is not of great importance, but the media traffic between clients in most cases is all the same without a hitch.
The registration sounds free of charge, no need to go anywhere and is carried out directly from the site, after which you take the number, password, data for the connection (or zawant the program with reinstallations) and you can immediately log in with your SIP account. Calls in the middle of the fence, in the other lane and from them, from the main telephone fence through the gateways - without any money. It is not necessary for you to indicate when registering your daily payment fees, as you cannot call on your own private city / style numbers, to get a personal direct access number or to be assigned to other exotic services.
At hand lashing the main parameters are your number (login), server address for connection and password. Often the STUN server is also indicated. Don't be fooled by him, in some ways a robot is impossible without him.
After a successful connection, you will want to protest the work of your client. May all registrars have service numbers for verification, a list of them is on the website. In this case, the most significant is "ehotest" or auto-reporting from the record and further confirmations. Often such numbers are used only in the middle of the line, ale є y vodkrit, available zvіdusіl, for example:
At the end, a piece of information about cryptography.
SIP-telephony includes 2 protocols - signaling sip (control, dialing a number and information about the status of the call) and transport rtp (without intermediary audio / video streams). If offending clients can support the encrypted stream (SRTP / zRTP), the roaming can be carried out by an encrypted channel. If both the server and the client support TLS, signaling traffic will be stolen.
The most relevant SIP softphones:
Ekiga (GPL, *nix/windows) http://ekiga.org/
Qutecom (GPL, *nix/windows/osx) http://www.qutecom.org/
SIP Communicator (LGPL, java) http://www.sip-communicator.org/
SFLphone (GPL, *nix) http://www.sflphone.org/
Linphone (GPL, *nix/windows/osx/android/iphone) http://www.linphone.org/
SipDroid (GPL, android) http://sipdroid.org/
X-Lite (proprietary, windows) http://www.counterpath.com/x-lite.html
fring (freeware, mobile devices) http://www.fring.com/
An example of a hardware SIP solution: a gateway that allows connection 2 special phone up to 2 independent SIP accounts and can be connected to them without connecting to a PC. Duzhe zruchna and rich on the possibility of river. Support CallerID. Allows you to win SipBroker independently from the registrar (dial plan support).
http://voips.ru/Linksys-by-Cisco-PAP2T.html
You can find out about the full spectrum of possession, which is issued by SIP-sum, here: http://www.sipnet.ru/orderandpay/hardware.php
(for a typical coristuvacha, obviously, the most significant categories will be "SIP telephony" and "VoIP gateways"). When choosing, give priority to the brands that have proven themselves (Linksys, D-link, Cisco, Grandstream, etc.)
Dekіlka v_domih sip voip providers:
http://www.sipnet.ru
http://www.comtube.ru
http://zadarma.com/ru
http://www.voxalot.com
http://www.ideasip.com
http://www.voipbuster.com
Comparison of prices for calls to the telephone line for different directions:
http://www.voipratetracker.com/compare_rates
http://www.voip-catalog.com/voip_routes.html
Possibly problems.
If your client registers on the server normally, but calls to one or the other side do not go through, or one-sided weirdness is guarded, the problem is worse, it’s better for everything, it’s nasty to fix the firewall or it’s hard to close NAT, which doesn’t “out of the box” for a robot with such services . Back off, you have everything with a mixer and a microphone, as described earlier. Try wiping Firewall. Flip what STUN says. Below, a spear of silan is indicated, for which everything is described more briefly, lower it is possible to expand in this article.
Okremo you can see the situation, if the input calls work right after the connection, but they stop going after the decal hvilin is idle. At this point, the decision is made to keepalive in the client until the pick-up time, and it is also well described in the first message.
Do not forget to reconnect to the server after the skin shortcut (just restart yoga as a softphone)
http://wiki.sipnet.ru/index.php/Connection_through_router_with_NAT
http://wiki.sipnet.ru/index.php/Translation_merezhevy_address_%28NAT%29_i_SIP
SIP/2.0 100 Trying- The request is processed, for example, the server is sent to the data bases, but there is a place for roztashuvannya viklikanago koristuvach present moment not appointed.
SIP/2.0 180 ringing- Roztashuvannya viklikanago koristuvach appointed. You are given a signal about the entrance call.
SIP/2.0 181 Call Is Being Forwarded- Proxy-server redirects the tweet to another caller.
SIP/2.0 182 Queued- Koristuvach, who calls out, Timchasovo is not available, but entry weekly orders at the devil. If the koristuvach becomes available, I will pass on the final report.
SIP/2.0 200 OK- The team is successfully vikonan. On request INVITE means that it is possessed, that it is necessary to call before participating in the session; at the tіlі vіdpovіdі are assigned the functional possibilities of that possession; On request BYE means the end of the call session; On request, CANCEL means asking for a request; on the request REGISTER means that the registration was successful; On request OPTION wins for the transmission of information about functional possibilities obladnannya, qі vіdomostі mіstya іtіlі vіdpovіdі.
SIP/2.0 300 Multiple Choices- Koristuvach, which is called, available for dekіlkom addresses. Viklik can choose any of them. You must specify a SIP-address for the prompt, for which you can find the called koristuvach, and the koristuvache that called, you can choose one of them.
SIP/2.0 301 Moved Permanently- Koristuvach changed his/her local address, his/her new address is shown in the Contact field.
SIP/2.0 302 Moved Temporarily- Koristuvach timchasovo changed his/her financial position (there may be an appointment at the Expires field), and the new address is entered at the Contact field.
SIP/2.0 305 Use Proxy- The party that calls is able to accept an input call only in that case, if the win goes through a proxy server. The recommended party is recommended to go back to the proxy server, the address of which is specified in the Contact field. The message is only passed on to terminal owners (UAS).
SIP/2.0 380 Alternative Service- Wiklik did not reach the addressee, but alternative service, what kind of indications have you got? For example, a weekly click can be redirected to a current postal screen.
SIP/2.0 400 Bad Request- The request is not for understanding through the presence of new syntactic pardons.
SIP/2.0 401 Unauthorized- request for the verification of the procedure for authenticating the koristuvach. Іsnuyut different options for authentication, and in different cases it can be assigned, which one can win in this particular type.
SIP/2.0 402 Payment Required- Advance payment of services is required.
SIP/2.0 403 Forbidden- The request is not served by the server and is not guilty of being retransmitted.
SIP/2.0 404 Not Found- The server has not detected a callback that is called in the domain specified in the Request-URI field.
SIP/2.0 405 Method Not Allowed- It is not allowed to request this type of request to the address specified in the Request-URI field. In the field Allow drinks, the allowed types of drinks are indicated
SIP/2.0 406 Not Acceptable- Vіdpovіdі, scho generated vyklikannoy side, will not razumіlі side, scho viklіkaє.
SIP/2.0 407 Proxy Authentication Required- The client is guilty to confirm its right to access the proxy server.
SIP/2.0 408 Request Timeout- The server is unable to send a message, for example, to indicate the miscalculation of the call for an hour, specified in the Expires field for the request. Weekly can be retransmitted after one hour.
SIP/2.0 409 Conflict- The processing of the REGISTER request could not be completed due to a conflict between the task assigned to the action parameter of the request, but the current resource camp.
SIP/2.0 410 Gone- The server can no longer access the requested resource and does not know where to redirect the request.
SIP/2.0 411 Length Required- It is necessary to indicate the length of the title of the note near the Content-Length field.
SIP/2.0 413 Request Entity Too Large- Rozmir zaputu is even great for processing.
SIP/2.0 414 Request-URI Too Large- The address specified in the Request-URI field turned out to be too big, so it's impossible to interpret.
SIP/2.0 415 Unsupported Media Type- Request to change the format of the body of the message, which is not supported.
SIP/2.0 420 Bad Extension- The server does not understand the protocol extension specified in the Require field.
SIP/2.0 480 Temporarily not available- Koristuvach, who calls out, Timchasovo inaccessible.
SIP/2.0 481 Call Beg/Transaction Does Not Exist- Forced to apply for the withdrawal of the BYE request, which should not be applied to the in-line charges, or the CANCEL request, which should not be applied to the in-line charges.
SIP/2.0 482 Loop Detected- The server has shown that the acceptance request is transmitted over a closed route (the Via field already has the address of the server).
SIP/2.0 483 Too Many Hops- The server has shown in the Via field, which accepts requests through a larger number of proxy servers, lower is allowed in the Max-Forwards field.
SIP/2.0 484 Address Incomplete- The server accepted the request from the wrong address in the Either Request-URI field. Need additional address information.
SIP/2.0 485 Ambiguous- The addresses of the koristuvach that are called are ambiguous. The Contact header may have a list of addresses for which requests can be passed.
SIP/2.0 486 Busy Here- Calling a koristuvach at once cannot accept an incoming call for a given address. It does not include the possibility of contacting a correspondent for another address, or, for example, deprive the information from the current mailing screen.
SIP/2.0 500 Internal Server Error- The server cannot be served through internal pardons. The client can try again and ask for a ticket in one hour.
SIP/2.0 501 Not Implemented- The server does not implement functions that are necessary for servicing this request. For example, the server cannot recognize the type of request.
SIP/2.0 502 Bad Gateway- The server, which functions as a gateway or a proxy server, accepts an incorrect input from the server, before which the vin is requested.
SIP/2.0 503 Service Unavailable- The server can't currently service the weekly via a rewrap or maintenance.
SIP/2.0 504 Gateway Timeout - The server, which functions as a gateway or a proxy server, after a set interval of time without deleting the server type (for example, the type of the local server), until which vin was returned to complete the processing of the request.
SIP/2.0 505 SIP Version not supported - The server does not support this version of the SIP protocol.
SIP/2.0 600 Busy Everywhere- Koristuvach of employment and do not hesitate to take a call at the moment. Vidpovіd can indicate the vіdpovіdny hour for the call
SIP/2.0 603 Decline- Coristuvach can’t or doesn’t want to accept input calls. At the end of the day, it is possible to have a fixed hour for the call.
SIP/2.0 604- Koristuvacha does not know.
SIP/2.0 606 Not Acceptable- Calling the caller cannot accept an input call through those that have the type of information, the values in the session description, the call in the SDP format, the bandwidth, etc. unacceptable.
While you are working with a virtual PBX from your operator - MSN Telecom, Mango, etc. - Tsya article is not necessary for you. Ale, when transferring numbers to another PBX, chi service will help you.
What is the language about?
All OnlinePBX clients win the correct numbers at our virtual PBX. Bo mi їх is not for sale that is not rahuєmo hvilini.
Let me tell you, it’s not handy - to pay for two rahunki: we will tell the operator. And our clients take advantage of this benefit: they know a cheap operator or a direction of calls (in Moscow, intermittently, beyond the cordon), connect to an automatic telephone exchange and save.
You can connect it to any operator, but it’s better to take a look at the CIP-installation. These are the parameters that are required for the connection. Oskіlki CIP is also similar to e-mail, let's draw an analogy.
What is a SIP login?
For example, take Vigadan mail on Yandex: [email protected] with password ******. There are actually three options here:
- Koristuvach login: siplogin.
- Yogo password: ****** .
- Server addresses, so as not to get confused with Mail.ru and Google: yandex.ru.
Tі, hto hto once nalashtovuvav mail client Let me know what the language is about: I ask you to remember all the parameters. Sometimes it is necessary to specify ports and separate servers for incoming and outgoing listings.
Z SIP all the same, only nasty with names. SIP-login is called differently: SIP ID, SIP Login, login, SIP-account, SIP account, public record, koristuvach, spіvrobіtnik, manager and so on. Vin is composed of digits and letters of the Latin alphabet, three characters minimum.
Need more wine?
Just as the mail is necessary for the oversight and removal of leaves, the CIP-login is for receiving input and output calls.
Є important moment: without a calling number (trunk) you can’t make calls to Moscow and mobiles, only to the same SIP logins.
What is a trunk in IP telephony?
Detect the cable, I have a lot of darts.
Leather postings - the main input or output channel of the link. That is why the numbers are called rich-channel.
The whole cable is full-cycle - ce trunk. One "kinets" cable connects to the PBX, the other operator connects to its possession. Electricity is also used: є cable - є voltage.
Let's make an analogy with electronic mail.
You write an electronic sheet and strengthen yoga mom. Alemati do not have electronic mail. An okrem company is required to deliver your sheet. She takes away your electronic sheet, spreads it on paper and packs it in an envelope. Let the company know your mother's address, write on the envelope and in the entry to the Post of Russia with your name. If your mother says, this company will take it out and write you an e-mail.
Such a company is victorious about the function of the trunk - it makes the electronic light out of the real world.
Plusy: mum does not need a computer, internet, e-mail and read chimney cym. There is a victorious speech - a pen, a piece of paper and an envelope. You can see that you are greeted with what they called - by electronic mail. I don't need to know mami's address.
Minusi: you have to pay for the service
That trunk is needed for you to dial calls to calling numbers - local and mobile. Telephoning on CIP-login is possible without a trunk, mi about tse.
How to remove SIP-login?
We found out that you need a calling number (trunk) for calling. And for connecting to other services - special addresses - CIP-login.
Otrimati CIP-login and other parameters can be connected with the operator. Deyakі operators give CIP-login without intermediary for the trunk, ce MSN Telecom, Telfin, Arentel. Інші - ні, tse Mango, UISCOM and more other operators.
In another way, you will have to select the operator's PBX, or CIP-login - tse koristuvach / spіvrobіtnik / dodatkovy. Create a new koristuvach, not tied to a person, enter a password. Remember the given data: login, password and server.
Telephony for business with integration with CRM
