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Main • Yjinglechan

Jingle module

This module supports the Jingle protocol by implementing XMPP as an external component of another server.

Message handlers

Routing

Syntax: jingle/user@server

A 'callerprompt' parameter may be added with a text to be sent to the called party before making the call

Examples

Route a call 125 to the user 'remoteuser' at server 'remotedomain.com'

  ^125$=jingle/remoteuser@remotedomain.com;callerprompt=Hello, this is a call from ME 

Configuration

yjinglechan.conf

[general]
; Global settings of Jabber and Jingle engines

; localip: string: Set this machine's IP address. If missing, we will try to guess one
; The local machine's IP address is used to negotiate the audio stream
;localip=

; anonymous_caller: string: Set the caller name if missing for outgoing calls
; Defaults to unk_caller
;anonymous_caller=unk_caller

; request_subscribe: boolean: Make a subscribe request before probing a remote user
; Defaults to enable
; pending_timeout: integer: How much an outgoing call will wait for a response to a probe
; Defaults to 10000
; These options are used when we want to make an outgoing call and we can't do that because we don't have
; a resource with audio capability for the called party, so we have to probe its presence
; Since we might not receive a response if the caller is not subscribed to called's party presence,
; we should make a subscribe request first (enabling request_subscribe will do that)
; The jingle channel will wait for the time given by pending_timeout for an appropriate response
; If no response is received within pending_timeout interval the call will be dropped
; When receiving a valid response, the jingle channel will start to negotiate the call
; pending_timeout actually extends the time an outgoing call may be in a not answered state
;request_subscribe=enable
;pending_timeout=10000

; xmlparser_maxbuffer: integer: Maximum length of text we ever parse as XML
; Defaults to 8192
;xmlparser_maxbuffer=8192

; stream_restartcount: integer: Maximum value for stream restart counter
; Defaults to 2. Take values between 1 and 10
; stream_restartupdateinterval: integer: Interval (in miliseconds) to update (increase)
; the stream restart counter. Defaults to 15000. Take values between 5000 and 300000
; The TCP connection with a server may be closed due to XML parser or socket errors or the open
; operation may fail (the server may be down)
; These options defines the TCP connection restart behaviour. The mechanism is designed to avoid
; repetitive connect attempts
; The restart count defines the number of times the engine will try to establish a TCP connection
; with the server. Each time a connect attempt fails the counter is decremented. No more attempts
; to (re)connect are made when the counter is 0
; The restart count update interval defines the interval at which the restart counter is incremented,
; up to the value given by stream_restartcount
;stream_restartcount=2
;stream_restartupdateinterval=15000

; auto_subscribe: string: Presence. Set the behaviour of processing the subscribe/unsubscribe requests
; Allowed values:
; none: All subscription are processed as a result of messages. The subscription changes are made
; on receiving 'unsubscribed' responses from remote users or 'resource.subscribe' from another module
; from: Automatically accept subscribe/unsubscribe requests
; to: Automatically subscribe/unsubscribe to/from the remote user when adding to the roster
; both: The combination of 'from' and 'to'
; Defaults to none
; This parameter is forced to 'from' if the roster is kept by the server
;auto_subscribe=none

; delete_unavailable: bool: Presence. Automatically delete an user or a resource when unavailable
; This setting is ignored if the roster is kept by the server
; Defaults to enable
;delete_unavailable=enable

; auto_probe: boolean: Presence. Automatically respond to probe requests for users in our domain when
; we know nothing about the user presence
; Defaults to enable
; This parameter is forced to enable if the roster is kept by the server
;auto_probe=enable

; probe_interval: integer: Presence. Interval to probe a remote user if no data was received
; Defaults to 1800000
; expire_interval: integer: Presence: Interval to wait a response to a probe request sent as a result
; of interval (set in probe_interval) ellapsing
; Defaults to 300000
; These options are used to refresh the roster. The probe interval is the maximum time allowed between
; succesive presence notifications from remote users. After this interval ellapses a probe request will
; be sent to the user. If a response is not received within expire_interval, the remote user will be
; supposed to be unavailable
; These parameters are ignored when the roster is kept by the server
;probe_interval=1800000
;expire_interval=300000

; default_resource: string: Default resource name when missing from local users' JIDs
; Defaults to yate
;default_resource=yate

; extra_domain: string: Extra domain served by us
; Here may be specified an extra domain accepted by the engine (other then the domain name of
; the default component server) as destination for incoming data
;extra_domain=

; printxml: boolean: Print sent/received XML data to output if debug level is at least 9
; Defaults to no
;printxml=no

[codecs]
; This section allows to individually enable or disable the codecs

; default: bool: Enable all unlisted codecs by default
default=false

; mulaw: bool: Companded-only G711 mu-law (PCMU/8000)
mulaw=default

; alaw: bool: Companded-only G711 a-law (PCMU/8000)
alaw=true

; gsm: bool: European GSM 06.10 (GSM/8000)
gsm=default

; lpc10: bool: Linear Prediction Codec (LPC/8000)
lpc10=default

; slin: bool: Signed Linear 16-bit uncompressed (L16/8000)
slin=default

; g723: bool: ITU G.723 all variations (G723/8000)
g723=default

; g726: bool: ITU G.726 32-bit (G726-32/8000)
g726=default

; g728: bool: ITU G.728 all variations (G728/8000)
g728=default

; g729: bool: ITU G.729 all variations (G729/8000)
g729=default

[jabber.domain]
; Each other section is used to initialize the server list as a Jabber Component
; The name of the section is used as base Jabber domain
; Changes don't affect the already opened streams

; address: string: The IP address of the machine running the Jabber server (required)
;address=

; port: integer: The port on server to connect to (required)
;port=

; password: string: The password used by Yate to authenticate with the server
;password=

; identity: string: The (sub)domain used by the component
; This setting controls the roster storing and will be used to generate the domain name of local users
; If identity is equal to jabber.domain (section name) the roster will be kept by the component server
; The domain part of local user's JID will be given by the identity: jabber.domain
; If identity is not equal to section name:
; If terminated in '.jabber.domain' it will be left untouched, otherwise the suffix '.jabber.domain' will be added
; In this case, the roster will be kept locally
; Examples:
; Section name: null.ro, subdomain=somename
; identity=null.ro
; The roster is kept by the component server. Domain part of local JIDs: null.ro
; Yate identify itself to the server as somename.null.ro
; identity=someidentity or identity=someidentity.null.ro
; The roster is kept locally. Domain part of local JIDs: someidentity.null.ro
; Yate identify itself to the server as someidentity.null.ro
; Either way, the domain part of the local users will be also the domain accepted by the engine
; as destination for received data (and, of course, the domain given by extra_domain)
; Defaults to section name
;identity=

; subdomain: string: The subdomain to use if the identity equals the section name
; Defaults to the value of default_resource
;subdomain=yate

; startup: boolean: Open the stream to the server when initializing the driver
; Defaults to enable
;startup=enable

; default: boolean: This is the default component to use. If no component is
; marked as default the first will be implicitely default
; Defaults to false ;default=false

Examples

This is an example of how to configure the stream with the Jabber server who will expose Yate to the Jabber network

[mydomain.com]
address=1.2.3.4
port=10000
password=whateverstring
startup=enable

Yate will start a TCP connection to the IP address 'mydomain.com' resolves to. The port to use to connect to 'mydomain.com' (10000) will be obtained internally by searching after the value entered in the address parameter (1.2.3.4). The password (whateverstring) will be used to authenticate with the server

July 2014:
Yate 5.4 and YateBTS 4 launched. Added JSON and DNS support in Javascript, Handover support in YateBTS.

March 2014:
YateBTS 2.0 launched. Added authentication and WebGUI. Added USSD support in commercial version.

March 2014:
Yate 5.2 launched. Better JavaScript support and a fixed memory leak.

Jan 2014:
YateBTS 1.0 launched. The first GSM Basestation which works with an IMS/VoLTE core network.

Jan 2014:
Yate 5.1 launched. Better JavaScript support and added libygsm. Elisa chatbot added in RManager

Oct 2013:
OpenHSS is the Yate based HLR/HSS solution for MVNO and LTE carriers.

Oct 2013:
Yate 5 released. Added IPv6 support in SIP for LTE. Improved JavaScript support. Download NOW

Jan 2013:
Yate 4.3 released: Added XML support in Javascript. SCCP - GTT routing between different networks. Stability improvements.
Download NOW

Aug 2012:
Yate 4.2 released: SIP flood protection. Better Jabber/Google Voice support. Usable Javascript. Fixed SIGTRAN links fluctuations.
Download NOW

Apr 2012:
YateClient was accepted in the Mac Store.

Yate 4.1 released: better Gvoice support, iSAC codec, support for new Wanpipe drivers. Fixes T.38 and Mac client issues.

Mar 2012:
SS7Cloud is launched today, 1st March, 2012, by NullTeam, Yate creators. Having all you need to be a US CLEC, it brings SS7 services in a cloud.

Feb 2012:
Yate 4.0 released.
SCCP, TCAP, MAP and CAMEL, TCP and TLS in SIP, Javascript fast prototyping of telephony applications and brand new face for YateClient.

Nov 2011:
Here is a video that, quote "demonstrates the truly awesome power of the YATE engine, as it easily handles 3 simultaneous calls to an audio player application including dtmf (button press) handling "(from PaintedRockComm).

Nov 2011:
Yate will attend ORR - OPENRHEINRUHR (November 12 - 13).

04 May 2011:
sipgate chooses open source project Yate for core infrastructure.

12 Apr 2011:
Yate 3.3.2 released.
Fix for Jingle calls to Google Voice dropping after 5 minutes.
4 Apr 2011:
Yate 3.3 released.
Support for GMail chat conference, fixes for internal microphone in MacOS. Minor fixes in SS7 M2PA and ANSI. Fixes in H.323, SIP and RTP.

9 Mar 2011:
Yate 3.2 released.
Bug fixes in SIGTRAN/MGCP/SS7 and added support for CNAM/LNP lookup by SIP INVITE/3xx.

Feb 2011:
Yate will attend FOSDEM and XMPP summit.

31 Jan 2011:
Yate 3.1 released.
Yate client support for Google Voice. Support for any country tones in tonegen.

20 Dec 2010:
Yate 3.0 released.
SS7 ITU certified. SS7 STP added. Client supports Jabber IM (Google Talk + Facebook).

3 May 2010:
Yate 3.0.0 alpha 3 released. Featuring the new Jabber server and wideband audio.

8 March 2010:
Yate 2.2 released. Mostly bug fixes. Dahdi compatible. Latest 2 release before 3.0.

6-7 February 2010:
Yate booth at FOSDEM 2010. Free CD with Freesentral available.

2 Nov 2009:
Yate 2.1 launched. Can replace a Cisco PGW2200 to control a Cisco AS54xx.

6 Aug 2008:
Yate and OpenSIPS (former OpenSER) join to build IP based clusters.

4 Aug 2008:
Yate 2 launched.


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