Asterisk - Eduguru - Page 14

WARNING: REMOTE HOST IDENTIFICATION HAS CHANGED!

July 31, 2014August 5, 2021 eduguru 0 Comments @@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@ @ WARNING: REMOTE HOST IDENTIFICATION HAS CHANGED! @ @@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@ IT IS POSSIBLE THAT SOMEONE, Add correct host key in /root/.ssh/known_hosts to get rid of this message., Host key verification failed., It is also possible that the RSA host key has just been changed., IT IS POSSIBLE THAT SOMEONE IS DOING SOMETHING NASTY!, Offending key in /root/.ssh/known_hosts:1, Please contact your system administrator., RSA host key for has changed and you have requested strict checking. Host key verification failed., Someone could be eavesdropping on you right now (man-in-the-middle attack)!, The fingerprint for the RSA key sent by the remote host is, WARNING: REMOTE HOST IDENTIFICATION HAS CHANGED

WARNING: REMOTE HOST IDENTIFICATION HAS CHANGED! @@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@ @ WARNING: REMOTE HOST IDENTIFICATION HAS CHANGED! @ @@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@ IT IS POSSIBLE THAT

Understanding asterisk Queue

July 4, 2014 eduguru 0 Comments How asterisk queue works, Understanding asterisk Queue

Best way to manage asterisk queue

July 4, 2014 eduguru 0 Comments Best way to manage asterisk queue

Differences Between HTTP Servlets and SIP Servlets

June 22, 2014March 19, 2016 eduguru 0 Comments and are not obligated to respond to incoming requests. SIP servlets often work in concert with other SIP servlets to respond to particular SIP requests, and SIP servlets can originate SIP requests, Differences Between HTTP Servlets and SIP Servlets, http vs sip, SIP servlets differ from typical HTTP servlets used in web applications in the following ways: HTTP servlets have a particular context (called the context-root) in which they run, unlike HTTP, unlike HTTP servlets which only send responses to the originating client. SIP servlets often act as proxies to other SIP endpoints, unlike HTTP servlets which typically are solely responsible for responding to HTTP requests., while HTTP servlets are typically the final endpoint for incoming HTTP requests. SIP servlets can generate multiple responses for a particular request. SIP servlets can communicate asynchronously, while SIP servlets have no context. HTTP servlets typically return HTML pages to the requesting client, while SIP servlets typically connect SIP-enabled clients to enable telecommunications between the client and server. SIP is a peer-to-peer protocol

SIP servlets differ from typical HTTP servlets used in web applications in the following ways: HTTP servlets have a particular

SIP Vs SS7 Protocal : SIP and SS7

June 22, 2014February 12, 2016 eduguru 0 Comments a caller first locates the appropriate server and then sends a SIP request. The most common SIP operation is the invitation. Instead of directly reaching the intended callee, a SIP request may be redirected or may trigger a chain of new SIP requests by proxies. Users can register their location(s) with SIP servers. SS7 uses a common channel for signalling call setup and te, and information-exchange functions of the PSTN, and its used to create, and terminating an interactive user session that involves multimedia elements such as video, and virtual reality. SIP is a text based, billing, both are the common protocols used for transmitting voice across networks. Session Initiation Protocol (SIP) is a protocol developed by IETF MMUSIC Working Group and proposed standard for initiating, but relies on the IP connectivity from the originator to a Server and thence to the terminating end. It is used for packet based communications and allows for many different call types such as video, gaming interaction etc as well as voice. Signaling System 7 (SS7) is architecture for performing signaling in support of the call-establishment, instant messaging, manage and terminate sessions in an IP based network. A session could be a simple two-way telephone call or it could be a collaborative multi-media conference session. It works as follows: Callers and, modifying, online games, routing, Session Initiation Protocol (SIP) and Signaling System 7 (SS7), signaling protocol similar to HTTP and SMTP, SIP and SS7, SIP Vs SS7 Protocal, SIP Vs SS7 Protocal : SIP and SS7, voice, where as SIP in used in IP network., whereas SIP is a protocol which is used for maintaining sessions in VOIP. SS7 are used to set up the vast majority of the world's PSTN telephone calls

Session Initiation Protocol (SIP) and Signaling System 7 (SS7), both are the common protocols used for transmitting voice across networks.

Session Initiation Protocol :SIP Application layer protocal

June 22, 2014June 22, 2014 eduguru 0 Comments and accepted. Redirection (3xx): Further action needs to be taken (typically by sender) to complete the request. Client Error (4xx): The request contains bad syntax or cannot be fulfilled at the serve, fax over IP and online games. Originally designed by Henning Schulzrinne and Mark Handley in 1996, file transfer, incorporating many elements of the Hypertext Transfer Protocol (HTTP) and the Simple Mail Transfer Protocol (SMTP).[2] SIP works in conjunction with several other application layer protocols that iden, instant messaging, modifying and terminating sessions consisting of one or several media streams. SIP can be used for two-party (unicast) or multiparty (multicast) sessions. Other SIP applications include video conferen, presence information, Session Initiation Protocol :SIP Application layer protocal, SIP has been developed and standardized in RFC 3261 under the auspices of the Internet Engineering Task Force (IETF). It is an application layer protocol designed to be independent of the underlying t, streaming multimedia distribution, termination and other essential elements of a call. SIP can be used for creating, the protocol may be encrypted with Transport Layer Security (TLS). List of SIP request methods REGISTER: Used by a UA to indicate its current IP address and the URLs for which it would like to receive, The Session Initiation Protocol (SIP) is a signaling communications protocol, understood, User Datagram Protocol (UDP) or Stream Control Transmission Protocol (SCTP).[1] It is a text-based protocol, video) SIP typically employs the Real-time Transport Protocol (RTP) or Secure Real-time Transport Protocol (SRTP). For secure transmissions of SIP messages, widely used for controlling multimedia communication sessions such as voice and video calls over Internet Protocol (IP) networks. The protocol defines the messages that are sent between endpoints whic, without setting up a call. A new method has been introduced in SIP in RFC 3262:[14] PRACK (Provisional Response Acknowledgement): PRACK improves network reliability by adding an acknowledgement system

The Session Initiation Protocol (SIP) is a signaling communications protocol, widely used for controlling multimedia communication sessions such as voice

Better SIP Security on Asterisk: How To Secure Asterisk

June 21, 2014April 28, 2018 eduguru 0 Comments again at least 12 characters with a complex mix of symbols, and a mix of upper and lowercase letters at least 12 digits long. 4) Block your AMI manager ports. Use “permit=” and “deny=” lines in manager.conf to reduce inbound connections to known hosts only. Us, and letters. 5) Allow only one or two calls at a time per SIP entity, asterisk-securitySeven Easy Steps to Better SIP Security on Asterisk: 1) Don’t accept SIP authentication requests from all IP addresses. Use the “permit=” and “deny=” lines in sip.conf to only allow a, Better SIP Security on Asterisk: How To Secure Asterisk, denying remote attackers the ability to detect existing extensions with brute-force guessing attacks. 3) Use STRONG passwords for SIP entities. This is probably the most important step you can take. D, for instance, limiting your exposure to toll fraud is a wise thing to do. This also limits your exposure when legitimate password holders on your system lose control of their passphrase – writing it on the bottom o, numbers, or some sort of combination of a common phrase + extension MD5 hash (example: from a shell prompt, this is an easy target for attackers to guess SIP authentication names. Use the MAC address of the device, try “md5 -s ThePassword5000″) 7) Ensure your [default] context is secure. Don’t allow unauthenticated callers to reach any contexts that allow toll calls. Permit only a limited number of active calls , where possible. At the worst, which allows extension information leakage. Setting this to “yes” will reject bad authentication requests on valid usernames with the same rejection information as with invalid usernames, which I’ve seen. 6) Make your SIP usernames different than your extensions. While it is convenient to have extension “1234″ map to SIP entry “1234″ which is also SIP user “1234″, you’d understand that trivial obfuscation like that is a minor hinderance to a modern CPU. Use symbols

Seven Easy Steps to Better SIP Security on Asterisk: 1) Don’t accept SIP authentication requests from all IP addresses. Use

Asterisk Databse mysql

ERROR 2002 (HY000): Can’t connect to local MySQL server through socket ‘/var/lib /mysql/mysql.sock’ (2)

June 10, 2014February 24, 2016 eduguru 0 Comments [SOLVED] CentOS 6.5 - MySQL Daemon failed to start., 000 results (0.23 seconds) Search Results MySQL Daemon Failed to Start - centos 6 - Stack Overflow stackoverflow.com/questions/.../mysql-daemon-failed-to-start-centos-6 Dec 1, 2010 - 15 posts - ‎2 authors I'm really not sure what I'm doing wrong. I'm attempting to start mysql Code: [root@bebop-test /]# /etc/init.d/mysqld restart Stopping mysqld: [ OK ] MySQL Daemon failed t, 2013 - [root@sacroiliac log]# service mysqld start MySQL Daemon failed to start. Starting mysqld: [FAILED] [root@sacroiliac log]# service mysql start ... centos - Cent OS - MySQL Daemon failed to star, 2013 - 12 posts - ‎1 author Starting mysqld: [FAILED] Does anyone know what could be happening here? How to repeat: Run a yum update -y on a centos 6.4, 2013 - EDIT: Look at the checkmarked answer comments to get your issue .... The most likely cause for this error is that your mysql server is not running. You've visited this page 2 times. Last visit:, 2013 - Recently I started working on a very basic Drupal website for a friend, 2013 - Symptoms This article applies if you face any of the following symptoms: MySQL starts/stops properly when started/stopped with the mysqld ... startup - mysql server start failed - Ask Ubuntu as, 2014 - 10 posts - ‎2 authors 140115 17:21:35 mysqld_safe Starting mysqld daemon with databases from /var/lib/mysql ... chcon: failed to change context of `mysql' to ... MySQL Daemon Failed to Start af, 2014 - But mysqld.sock file doesn't exist inside /var/run/mysqld folder. On executing ps aux | grep mysql command, 2014 - Now I have restarted mysqld and it wont start. It has just stopped. I really need to get this up and running now. Its highly urgent for me. Below is ... Cannot start/stop/restart service: MySQL, 2014 - When trying restart mysqld service I'm getting this erroe: Cannot start/stop/restart service: MySQL Daemon failed to start. Anybody have idea ... MySQL Daemon failed to start. | DigitalOcean ht, and change its value to the database server host machine's IP address. For me, by doing that it sadly seems that it has overloaded mysql. Can it really be ... MySQL Daemon failed to start. - LinuxQuestions.org www.linuxquestions.org › Forums › Linux Forums › Linux - Software Nov, Can't connect to local MySQL server through socket, check the ... rhel - Problem Starting MySQL on new Centos 6 build - Unix ... unix.stackexchange.com/.../problem-starting-mysql-on-new-centos-6-bui... Dec 10, ERROR 2002 (HY000): Can't connect to local MySQL server through socket '/var/lib /mysql/mysql.sock' (2), I realized that mysql server was not ... MySQL Bugs: #71086: Starting Mysql service on CentOS6.5 bugs.mysql.com/bug.php?id=71086 Dec 5, I tried the following steps: Log in as super user or use sudo Open /etc/mysql/my.cnf using gedit Check the all the configuration Find bind-address, it was localhost or 127.0.0.1 Save and close the file. Come back to terminal and execute sudo service mysql start And it worked for me., MySQL Daemon Failed to Start - centos 6, Searches related to mysql daemon failed to start. mysql daemon failed to start. starting mysqld failed mysql daemon failed to start. starting mysqld mysql daemon failed to start. starting mysqld cento

I tried the following steps: Log in as super user or use sudo Open /etc/mysql/my.cnf using gedit Check the all

ISDN hangup cause codes

June 10, 2014 eduguru 0 Comments (similar to cause 88). Some part of the format being used is not understood by either the remote side equipment or the switching equipment between the source and destination of the call. Code No. 97 -, a physical layer or data link layer failure at the remote party or user equipment off-line. Code No. 28 - invalid number format (address incomplete). This cause indicates that the called party cannot , all Telcos have their own system for programming the number formats that the switches will recognize. Some systems want to see 7 digits, although the number is in a valid format, and cause 100. Code No. l - Unallocated (unassigned) number. This cause indicates that the destination requested by the calling user cannot be reached because, and others 11. Code No. 52 - outgoing calls barred. Code No. 53 - outgoing calls barred within CUG. This cause indicates that although the calling party is a member of the CUG for the outgoing CUG cal, as defined in Recommendation X.213. cannot be provided (e.g., as it is invalid. 3. On long distance calls, but at this time they are proprietary in nature. The PRI equipment vendors are the most likely to use these codes as they have been using proprietary messages in the facilities data link for some time, but if it doesn't happen in the time allotted by the switch programming, but the reasons for it could be: 1. Bad username or password 2. Router's settings do not match what is expected by the remote end. 3. Telephone line problems. 4. Hung session on remote end. Code No. 1, but this call identify does not. This cause indicates that a call resume has been attempted with a call identity which differs from that in use for any presently suspended call(s). Code No. 84 - call , cause 88, Code No. 1, data rate) which cannot be accommodated. What it means: 1. This usually means that the Number To Dial in the Connection Profile is in the wrong format. You may need to dial a 10 or 11 digit number, dialing at 64K when only 56K is supported. Code No. 66 - channel type not implemented. This cause indicates that the equipment sending this cause does not support the channel type requested. Code No. , for example, giving a SPID failure in the router logs. 2. The ISDN phone number being dialed by the router is invalid and the telco switch cannot locate the number to complete the call, high layer compatibility or sub-address as indicated in the diagnostic. It is noted that the particular type of access information discarded is optionally included in the diagnostic. Code No. 44 - req, if a user has subscribed to those channels on a primary rate interface numbered from l to 12 and the user equipment or the network attempts to use channels 3 through 23, if the equipment sending this cause is an intermediate point, immediately re-attempting the call is not likely to be successful. Code No. 39 - permanent frame mode connection out-of-service. This cause is included in a STATUS message to indicate that a permanent, indicating that the call was cleared due to a supplementary service constraint. The diagnostic field may contain additional information about the supplementary service and reason for rejection. What i, ISDN hangup cause codes, ISDN hangup cause codes provide information as to why a call has been terminated. The list of hangup cause codes below provides detailed information as to the underlying cause behind a call hangup: Co, it is not currently assigned (allocated). What it usually means: 1. The SPIDS may be incorrectly entered in the router or the Telco switch, low layer compatibility, one or more of the fields in the information element are coded in such a way which has not been implemented by the equipment sending this cause. What it means: Like cause 1 and cause 88, or a call will fail with this cause. Code No. 101 - message not compatible with call state. This cause indicates that a message has been received which is incompatible with the call state. Code No. 10, or a misconfiguration on the equipment being called. Code No. 22 - number changed. This cause is returned to a calling party when the called party number indicated by the calling party is no longer as, or a STATUS message was received indicating an incompatible call state. Code No. 99 - Information element / parameter non-existent or not implemented. This cause indicates that the equipment sending t, or dial a 9 in front of the number if it is a Centrex line. 2. This problem may also give a Cause 111. 3. Dialing at the wrong line speed can also give this Cause. Code No. 90 - non-existent CUG. This, some 10, the call cannot be properly routed to its destination. Code No. 2 - No route to specified transit network (national use). This cause indicates that the equipment sending this cause has received a requ, the call should be placed properly. There are no standards for this, the call will not ring through and will be discarded by the switch. Code No. 103 - parameter non-existent or not implemented - passed on (national use). This cause indicates that the equipment sending, the information element is not required to be present in the message in order for the equipment sending the cause to process the message. Code No. 100 - Invalid information element contents. This caus, the number being called is not an ISDN number but an analog destination. 2. The equipment is dialing at a faster rate than the circuitry allows, the router relinquishes the second channel, the Telco switch operates a timer. For example, the user may wish to try another call attempt almost immediately. What it means: This means that there is a temporary failure at the physical layer on the ISDN network. If you remove the ISDN cable fr, then this cause indicates that the parameter(s) were passed unchanged. Code No. 110 - message with unrecognized parameter discarded. This cause indicates that the equipment sending this cause has disc, this cause is generated. Code No. 83 - a suspended call exists, this usually indicates that the ISDN number being dialed is in a format that is not understood by the equipment processing the call. SPIDs will sometimes fail to initialize with a Cause 100, throughput of transit delay cannot be supported). Code No. 50 - requested facility not subscribed. This cause indicates that the user has requested a supplementary service which is implemented by the , unallocated (unassigned) number shall be used. Code No. 26 - non-selected user clearing. This cause indicates that the user has not been awarded the incoming call. Code No. 27 - destination out of ord, unspecified. This cause indicates that an interworking call (usually a call to 5W56 service) has ended. Notes about Cause Codes over 128 Cause code values of 128 and higher aren't sent over the networ, unspecified. This cause is used to report a protocol error event only when no other cause in the protocol error class applies. Code No. 127 - Intel-working, unspecified. This cause is used to report a resource unavailable event only when no other cause in the resource unavailable class applies. Code No. 49 - Quality of Service not available. This cause is, unspecified. This cause is used to report an invalid message event only when no other cause in the invalid message class applies. Code No. 96 - mandatory information element is missing. This cause ind, user-to-user information, when an analog call is placed to a Netopia router that has two B Data Channels in place, which can be caused by a bad switch translation, you would see this. It's usually temporary. Code No. 42 - switching equipment congestion. This cause indicates that the switching equipment generating this cause is experiencing a period of high traff

ISDN hangup cause codes provide information as to why a call has been terminated. The list of hangup cause codes

Asterisk Dialplan varriable

June 10, 2014June 10, 2014 eduguru 0 Comments %d%m%Y-%H:%M:%S)}) ${DIALEDPEERNAME}: Name of the called party. Broken for now, %Y%m%d-%H%M%S)}) ${TRANSFERCAPABILITY}: Type of Channel ${TXTCIDNAME}: Result of application TXTCIDName (see below) ${UNIQUEID}: Current call unique identifier ${TOUCH_MONITOR}: used for "one touch re, 1, 2, 3, and wW dial flags). If is set on either side of the call then that var contains the app_args for app_monitor otherwise the default of WAV||m is used ${TOUCH_MONITOR_PREFIX}: used for "one touch record, and wW dial flags). This set Prefix to ${TOUCH_MONITOR} default: auto "New in 1.8" Application-specific variables Some applications take extra input or provide output using channel variables. AgentCal, are case sensitive. Note: Several of these builtin variables have been converted to functions in 1.2, Asterisk Dialplan varriable, Caller ID that redirected the call. Limitations apply, DIALSTATUS seems to have been removed. Now you should use the DEVSTATE function. Try in astersisk console "core show function DEVSTATE" for more informations) ${DNID}: Dialed Number Identifier. Limita, exten => 100, extra channel variables are available. ${ARG1}: The first argument passed to the macro ${ARG2}: The second argument passed to the macro (and so on) ${MACRO_CONTEXT}: The context of the extension that , for example) should be aware of this). ${CALLINGPRES}: PRI Call ID Presentation variable for incoming calls (See callingpres ) ${CHANNEL}: Current channel name ${CONTEXT}: The name of the current cont, if specified - see Asterisk billing (DEPRECATED in 1.2.0 and removed in 1.4. Use ${CDR(accountcode)} ${ANSWEREDTIME}: This is the amount of time(in seconds) for actual call. ${BLINDTRANSFER}: The acti, in contrast to user-defined variables, instead use :${STRFTIME(${EPOCH}, it was DEPRECATED in 1.2.0 and removed in 1.4. (Note: this is not necessarily numeric as the name would indicate and can legitimately contain the space character. Commands acting on this variable (suc, it was DEPRECATED in 1.2.0 and removed in 1.4. ${CALLERID(num)}: The current Caller ID number - ${CALLERIDNUM} was used in versions of Asterisk prior to 1.2.0, MixMonitor(${filename}) Predefined Channel Variables There are some channel variables set by Asterisk that you can refer to in your dialplan definitions. Asterisk-defined variables, NoOp(${LEN(${Fruit})}) The first NoOp would show a value of 5 (the length of the string "fruit"). The second NoOp would show a value of 4 (the length of the string "pear"). This is an excellent way to, NoOp(${LEN(Fruit)}) exten => 100, see DIALEDPEERNAME ${DIALEDPEERNUMBER}: Number of the called party. Broken for now, see DIALEDPEERNUMBER ${DIALEDTIME}: Time since the number was dialed (only works when dialed party answers the line?!) ${DIALSTATUS}: Status of the call. See DIALSTATUS (note: In the current SVN relea, see DNID ${EPOCH}: The current UNIX-style epoch (number of seconds since 1 Jan 1970) ${EXTEN}: The current extension - cannot be modified with the set command- just use the GoTo to change the EXTEN va, see RDNIS ${SIPDOMAIN}: SIP destination domain of an inbound call (if appropriate) ${SIP_CODEC}: Set the SIP codec for the inbound (=first) call leg (see channelvariables.txt or README.variables in 1., Set(filename=${EVAL(${MY_VAR})}) exten => 104, Set(Fruit=pear) exten => 100, this is the errormessage Dial returns ${DIALSTATUS}: Text code returning status of last dial attempt. Dial takes input from ${TRANSFER_CONTEXT}: If this variable exists, to allow setting their values. ${ACCOUNTCODE}: Account code, Using $ If you want to set a global variable containing the another variable name in the [globals] category of extensions.conf you have to do something like this: [globals] SS=$ MY_VAR=${SS}{EPOCH}-${, when a #transfer is executed it goes to the selected extension on this context. EnumLookup returns ${ENUM}: The result of the lookup Hangup reads the ${PRI_CAUSE} variable for setting PRI return codes

Using $ If you want to set a global variable containing the another variable name in the [globals] category of extensions.conf you