configuring digium TE235 and TE435 card in asterisk vicidial goautodial freepbx elastix
Digium’s (TE235) dual span digital interface cards support 48 (T1 / J1) or 60 (E1) connections to PSTN trunks over
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SIP.conf : Asterisk
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
Read moreHow to I disable echo cancellation module without removing the hardware?
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
Read moreWARNING: REMOTE HOST IDENTIFICATION HAS CHANGED!
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
Read moreDifferences Between HTTP Servlets and SIP Servlets
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
Read moreSIP Vs SS7 Protocal : SIP and SS7
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.
Read moreSession Initiation Protocol :SIP Application layer protocal
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
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