Asterisk - Eduguru - Page 13

Asterisk: The Hacker’s PBX

December 6, 2014February 12, 2016 eduguru 0 Comments and they’ll be able to build a system that’s far more secure than anything that exists today. Rather than being constricted by the dubious and easily cracked security of closed systems, Asterisk will be able to evolve into a far more secure platform than any proprietary system, Asterisk: The Hacker’s PBX, because the term actually existed long before the media corrupted its meaning. Hackers built the networking engine that is the Internet. Hackers built the Apple Macintosh and the Unix operating system, been twisted by the mass media into meaning “malicious cracker.” This is unfortunate, but rather because of them., Hacker made asterisk Telecommunications companies that choose to ignore Asterisk do so at their peril. The flexibility it delivers creates possibilities that the best proprietary systems can scarcely , not in spite of its hacker roots, of course, the hackers will be able to quickly respond to changing trends in security and fine-tune the telephone system in response to both corporate policy and industry best practices. Like other open source s

Telecommunications companies that choose to ignore Asterisk do so at their peril. The flexibility it delivers creates possibilities that the

iNet Expert Technology

Asterisk Future of Asterisk

A Telephony Revolution with Asterisk

December 6, 2014 eduguru 0 Comments A Telephony Revolution with Asterisk, and a good book on the subject., and is an accepted (albeit perhaps not always loved) technology in the telecom industry. Unfortunately, and we believe it is still one of the best hopes for any sort of sensible integration between telecom and all the other technologies businesses might want to interconnect with. With Asterisk, but a person contemplating such a task must either a) enlist competent help or b) develop the required skills through instruction, confusing, especially by up-and-coming generations. Asterisk remains pretty awesome technology, etc.). The phone call is seen as a bit of a dead thing, First they ignore you, IM, it is very sensible and practical. People’s eyes light up when they first see an Asterisk dialplan and begin to contemplate the possibilities. But when there are literally thou‐ sands of ways to achie, no one is telling you how your phone system should work, or cryptic; on the contrary, or what technologies you are limited to. If you want it, over the past nine years the telecom industry has continued to lose its way. The methods by which we communicate have changed. Whereas 20 years ago phone calls were the preferred way to converse acros, practice, the current trend is to message via text (email, the process naturally requires extra effort. Perhaps it can be compared to building a house: the components are relatively easy to understand, then they fight you, then they laugh at you, then you win. —Mahatma Gandh we confidently predicted that Asterisk would fundamentally change the telecommunications industry. Asterisk is now the most successful private branch exchange (PBX) in the, this incredible flexibility comes with a price: Asterisk is not a simple system to configure. This is not because it’s illogical, while also enjoying the freedom to develop its own innovations. What you choose to implement is up to you—Asterisk imposes no limits. Naturally, you can have it. Asterisk lovingly em‐ braces the concept of standards compliance

First they ignore you, then they laugh at you, then they fight you, then you win. —Mahatma Gandh we confidently

digium-te435f-digital-card

Changing PRI card line mode to T1 or E1 without moving the jumper

December 3, 2014December 3, 2014 eduguru 7 Comments 2, 4, all single span cards (via device driver wcte12xp) are set to E1 mode; all 2-, and 4-port (span) digital telephony cards is to set the jumper(s) on the card for either T1 or E1 mode for each span on the card. With the jumper off, and 8-span cards, and 8-span cards (via device driver wct4xxp) are set to T1 mode. Then you would reload the DAHDI device drivers by first stopping asterisk, and it may be inconvenient to remove the card from the server to access its jumper(s). In this case, Changing PRI card line mode to T1 or E1 without moving the jumper, dual, for the wct4xxp device driver, How To: My Recomdation: The recommended way to set line mode on your Digium 1-, is the only method to set the TE820 line mode. Depending on how your Asterisk server loads the DAHDI device drivers, or quad span digital cards. However, quad-, see the user manual for the single, sometimes a card will have been installed in a server without first setting the jumper(s) correctly, the "default_linemode" option can be passed when the card's device driver is loaded. For single-span cards, the "default_linemode" option must be set/passed in one of two places: If your Asterisk server uses the DAHDI init script to load the DAHDI device drivers on boot (or if you execute the "service dahdi, the device driver is wct4xxp. Note: The eight-span card (TE820) does not have jumpers. The "default_linemode" module parameter, the device driver is wcte12xp; for dual-, the span is ready for E1 mode. For more details about the jumpers, the span is ready for T1 mode; with the jumper on, then restarting (stopping and starting) DAHDI using the init script: # service asterisk stop # service dahdi restart # service asterisk start * Remember that you will still need to configure /etc/dahd, to set the wcte12xp driver to T1 mode. # modprobe dahdi # modprobe wcte12xp default_linemode=t1 # dahdi_cfg -vv, which runs the init script), you must set the "default_linemode" option in the/etc/modprobe.d/dahdi.conf file. /etc/modprobe.d/dahdi.conf : ------------------------------------------------------------------------ # You should pla

How To: My Recomdation: The recommended way to set line mode on your Digium 1-, 2-, and 4-port (span) digital

chane run level centos

Asterisk Error linux

Changing centos/redhat run level init 3

November 13, 2014March 4, 2019 eduguru 0 Comments change run level in centos 6, change run level in centos 7, Changing centos/redhat run level init 3, init level in linux, it is possible to change the default runlevel at boot time. To change the runlevel of a single boot session, Linux, linux init level, press any key to enter the GRUB menu (within the first three seconds). Press the a key to append to the kernel command. Add at the end of the boot options line to boot to the desired runlevel. For exa, run level in linux, the following entry would initiate a boot process into runlevel 3: grub append> ro root=/dev/VolGroup00/LogVol00 rhgb quiet 3, Under Red Hat Enterprise Linux, use the following instructions: When the GRUB menu bypass screen appears at boot time, whart is run level

What is a runlevel ? Runlevel is the preset state in which a Linux system operates. Below mentioned are the

Asterisk Databse mysql

Failed to connect to MySQL-asterisk

November 6, 2014 eduguru 1 Comment [unixODBC][Driver Manager]Data source name not found, and no default driver specified", asterisk database dns configuration, asterisk mysql dns, asterisk odbc configuration, asterisk odbc connection, Error SQLConnect=-1 errno=0 [unixODBC][Driver Manager]Data source name not found, Failed to connect to MySQL-asterisk, Files need to check are: /etc/asterisk/res_odbc.conf /etc/odbcinst.ini To resolve this, MySQL ODBC functions not returning any data, res_odbc.c:565 odbc_obj_connect: res_odbc: Error SQLConnect=-1 errno=0, You have to check the correct configuration of above file. For Example : in -- /etc/odbcinst.ini [MySQL] Description = ODBC for MySQL Driver = /usr/lib/libmyodbc3.so Setup = /usr/lib/libodbcmyS.so Fil

Files need to check are: /etc/asterisk/res_odbc.conf /etc/odbcinst.ini To resolve this , You have to check the correct configuration of above

Convert seconds to hh:mm:ss format IN EXCEL

October 1, 2014April 27, 2018 eduguru 0 Comments "hh:mm:ss"), Convert seconds to hh:mm:ss format, Convert seconds to hh:mm:ss format IN EXCEL, excel, excel time convert, hh:mm:ss format in excel, time convert in excel, Use =A1/86400 Where A1 holds the time. Format as hh:mm:ss or whatever you need. You can also try A1=9084 B1==TEXT(A1/(24*60*60)

Use =A1/86400 Where A1 holds the time. Format as hh:mm:ss or whatever you need. You can also try A1=9084

configuring digium TE235 and TE435 card in asterisk vicidial goautodial freepbx elastix

September 26, 2014 eduguru 1 Comment 1, configuring digium TE235 and TE435 card in asterisk vicidial goautodial freepbx elastix, Configuring TE235 in E1 mode. Step 1 : enabling the Line mode as E1 vi /etc/modprobe.d/dahdi.conf go to last line and paste the below line options wcte43x default_linemode=e1 save and exit the file. N, Dial(DAHDI/g0/${EXTEN:1}) were 9 is the prefix, if you want to change it, then edit in dahdi-channels.conf exten => _9X.

Digium’s (TE235) dual span digital interface cards support 48 (T1 / J1) or 60 (E1) connections to PSTN trunks over

SIP.conf : Asterisk

September 20, 2014 eduguru 0 Comments actually the new jb of IAX2). Defaults to fixed. (Added in Version 1.4) • jblog = no|yes: Enables jitterbuffer frame logging. Defaults to "no". (Added in Version 1.4) • language = : Default language u, allowed, allowed_failed_screen, allowed_passed_screen, also called the inter-digit timer. With overlap dial set to off, and making a call through any extension specifying SetMusicOnHold will override this value for the call. • subscribemwi: Instructs Asterisk to not send NOTIFY messages for message waiting indication (, and unavailable. See SetCallerPres for more information. Default allowed_not_screened. • canreinvite = update|yes|no|nonat : If the client is able to support SIP re-invites. Default yes. • cid_number , anyone will be able to log in as a peer (with no check of credentials; useful for operation with SER). Default no. • Asterisk sip autodomain = yes|no : Enable/disable Asterisk's ability to add local h, Asterisk makes the assumption that the endpoint supports all known SIP methods. If you know that your SIP endpoint does not provide support for a specific method, asterisk sip.conf, before allowing other codecs) • disallow = all : Disallow all codecs (global configuration) • Asterisk sip allowexternaldomains = yes|no : Enable/Disable INVITE and REFER to non-local domains. Default, before allowing other codecs) • disallow = all : Disallow all codecs for this peer or user definition. • allowguest = yes|no : Allow or reject guest calls (default is yes, billing, BLF; added in 1.4) • localnet = NetAddress/Netmask : local network and mask. • fromdomain= : Sets default From: domain in SIP messages when acting as a SIP ua (client) • insecure = very|yes|no|invite|, bug 5427. Default no. • vmexten = : Dialplan extension to reach mailbox. Default asterisk. (New in v1.2.x) • callevents = yes|no: Set to yes to receive events on AMI when a call is put on/off hold. • , but right away passes media to the other party like a SIP proxy • dtmfmode = inband|info|rfc2833 : How the client handles DTMF signalling. Default rfc2833. Warning: inband very high CPU load. • fromus, deny, documentation. See Asterisk billing • astdb : Appears to insert a value in the Asterisk database. See example below. • auth = : Value assigned to the Digest username= SIP header. • callerid = : Caller, for peers that register with Asterisk, hostname, like tos=184 ) • trustrpid = yes|no : If Remote-Party-ID SIP header should be trusted. Default no. • useclientcode = yes|no : If yes, mask : IP address and network restriction • pickupgroup : Group that can pickup fellow workers' calls using *8 and the Pickup() application on the *8 extension • port : SIP port of the client • progre, multiline SIP headers and URI-encoded headers. Default no (in Asterisk 1.8 defaultyes). • port = : Default SIP port of peer. (this is not the port for Asterisk to listen. See bindport). • progressinba, not on a per phone basis, num2-num3 : Defines call groups for calls to this device. • callingpres = number|descriptive_text : Set Caller-ID presentation on a call. Valid descriptive values are: allowed_not_screened, omit, or IP address of the outbound SIP Proxy. (New in v1.2.x). • outboundproxyport = Number : UDP port number for the Outbound SIP Proxy. (New in v1.2.x). • pedantic = yes|no : Enable slow, or IP address of the outbound SIP Proxy. Valid only in [general] section and type=peer. (New in v1.2.x). • permit, pedantic checking of Call-ID:s, possible to use multiple options separated by commas from v1.4.x • ipaddr : Dotted Quad IP address of the peer. Valid only for realtime peers. • language : A language code defined in indications.conf , possible to use multiple options separated by commas from v1.4.x • maxexpiry = Number : Max duration (in seconds) of incoming registration we allow. Default 3600 seconds. • musicclass = one of the cla, prohib, prohib_failed_screen, prohib_not_screened, prohib_passed_screen, see bug 12707 • rtcachefriends = yes|no : Cache realtime friends by adding them to the internal list just like friends added from the config file. Default no. (New in v1.2.x). Buggy up to 1.4.19, see bug 12707 • rtsavesysname = yes|no : If set will write the value of asterisk.conf (options) systemname to the sip peer table in the field "regserver". Useful for multi-server systems. (New in v1.?, see bug 5374 for details., SIP Configuration - general The [general] section of sip.conf includes the following variables: • allowsubscribe = yes|no : Allow or Ignore Subscribe requests • allow = : Allow codecs in order of pref, so beware... • Asterisk sip checkmwi = Number : Global interval (in seconds) between mailbox checks. Default 10 seconds. (New in v1.2.x) • Asterisk sip compactheaders = yes|no : Indicates Asterisk sho, some endpoints either do not include an Allow header or lie about what methods they implement. In the former case, subscriptions, the $CALLERID(name) will start off blank and requires the dialplan to set the$CALLERID(name). (New in v1.6.x) • trustrpid = yes|no : If Remote-Party-ID SIP header should be trusted. Default no. • type, the checks occur every 60 seconds. Default no. • realm = my realm (Change authentication realm from asterisk (default) to your own. Requires Asterisk v1.x) • recordhistory = yes|no. Enable logging of , the checks occur every 60 seconds. Valid only in [general] section and type=peer. • regexten = • regseconds = seconds : Number of seconds between SIP REGISTER. Valid only for realtime peer entries. • , the configuration variable can be used for both type=peer and type=user.) • accountcode = : Users may be associated with an accountcode. See Asterisk billing • allow = : Allow codecs in order of prefe, the Context for the inbound call from this SIP user definition. If type=peer, the Context in the dialplan for outbound calls from this SIP peer definition. If type=friend the context used for both inbound and outbound calls through the SIP entities definition. If no type=user e, the gateway expects to receive the digits one right after the other coming in to this line with very little delay between digits. With overlap dial set to on, the only actual use for this currently is for determining if Asterisk may send connected line UPDATE requests. Its use may be expanded in the future. • preferred_codec_only= (1.8.x) Respond to a SIP i, the other endpoint should include an Allow header telling us what SIP methods the endpoint implements. However, then a type=peer or type=friend will match if the hostname or IP address defined in host= matches. • defaultip = Dotted.Quad.IP.Addr : Default IP address of client if host=dynamic is specified. Used i, then the Call Originator as stated in the CDR will be changed to whatever is specified in a X-ClientCode SIP Header. Default no. (New in v1.2.x) • usereqphone = yes|no : Indicates whether to add a ";u, then the device waits up to about 2 seconds between digits). • autocreatepeer = yes|no : If set, then the IP address associated with the hostname is looked up only once during the reading of the sip.conf. If you want support for a hostname associated with a dynamic IP address, then the peer setting completely overrides the general setting (i.e. the result is *not* the union of the two options). Note also that while Asterisk currently will parse an Allow header to learn what, then there is no need to set this option. This option may be set in the general section or may be set per endpoint. If this option is set both in the general section and in a peer section, then this field is used to authenticate SIP INVITEs that Asterisk sends to the remote SIP server. Asterisk 1.6.2.x: Changed the secret parameter to remotesecret. • sendrpid = yes|no : If a Remote-Part, then this SIP client must login with this Password (A shared secret). If Asterisk is acting as a SIP client to a remote SIP server that requires SIP INVITE authentication, then you may provide a comma-separated list of methods that your endpoint does not implement in the disallowed_methods option. Note that if your endpoint is truthful with its Allow header, this can also be set to 'osp' if asterisk was compiled with OSP support). (New in v1.2.x). • amaflags : Categorization for CDR records. Choices are default, this field specifies the user name for authentication. (Contrast with fromuser.) Also, this username is used in INVITEs until we have a registration. • vmexten = : Dialplan extension to reach mailbox. Default asterisk. Valid only in [general] or type=peer. Notes • Asterisk 1.6 and later, to wait for RTP traffic before classify the connection as discontinued. Default 0 (no RTP timeout). (New in v1.2.x). • rtupdate = yes|no : Send registry updates to the database when using Realtime sup, use Asterisk sip externhost. • externhost = hostname.tld : (New in Asterisk 1.2.x) • externrefresh = Number : Specify how often (in seconds) a hostname DNS lookup should be performed for the value ent, use the keyword dynamicinstead of Host IP. • incominglimit and outgoinglimit = Number : Limits for number of simultaneous active calls for a SIP client. Valid only for type=peer. • insecure : very|yes, used on the receiving side of a SIP channel. Two implementations are currently available - "fixed" (with size always equals to jbmaxsize) and "adaptive" (with variable size, usually sent from exotic devices and programs. Defaults to 1000. (Added in Version 1.4) • jbimpl = fixed|adaptive: Jitterbuffer implementation, when a RTP Keepalive packet will be sent if no other RTP traffic on that connection. Default 0 (no RTP Keepalive). (New in v1.2.x). • rtptimeout = Number : Number of seconds, when a RTP Keepalive packet will be sent if no other RTP traffic on that connection. Default 0 (no RTP Keepalive). Valid only in [general] section and type=peer. • rtptimeout = seconds : Terminate cal, with big jumps in/broken timestamps, yes and very removed in v1.6.x, you cannot use silence suppression. Make sure ALL SIP phones have disabled silence suppression. There is a solution for the silence suppression problem

SIP Configuration – general The [general] section of sip.conf includes the following variables: allowsubscribe = yes|no : Allow or Ignore Subscribe

How to I disable echo cancellation module without removing the hardware?

August 17, 2014 eduguru 0 Comments disable echo cancellation without removing the hardware?, How do I disable echo cancellation module without removing the hardware?, How I disable echo cancellation module without removing the hardware?, How to I disable echo cancellation module without removing the hardware?

You’ll need to: Stop Asterisk Stop DAHDI Unload the driver for your card from the Linux kernel Load the driver

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Get Best and affordable Dialer in just 1 hour

August 10, 2014 eduguru 0 Comments best affordable dialer, Dialer in one hour, dialer installation in 1 hour, Get Best and affordable Dialer in just 1 hour, get best dialer, idial best dialer, world class dialer features