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  • SIP routing in Yate can be done from a single routing module or from a combinat The calls can be routed directly to a SIP channel from a routing module (e.g regexroute.conf).
    3 KB (422 words) - 08:08, 23 November 2017
  • * SIP Methods like SUBSCRIBE and NOTIFY are required. == Implemented SIP Methods==
    3 KB (456 words) - 09:16, 9 April 2015
  • Yate has support for SIP Protocol. The module is called ysipchan and the associated configuration fi This module is a VoIP SIP driver based on [[YASS]] library.
    25 KB (3,811 words) - 10:09, 6 July 2018
  • ...ng for incoming calls using SIPS URI. They must be explicitly forwarded as SIPS URI * When a SIPS URI is used to send a request transport defaults to TLS unless explicitly
    1 KB (245 words) - 10:35, 6 July 2018

Page text matches

  • <font class="basetext">Adds SIP flood protection and improves Jingle / Google Voice support.</font></div><b ...ss="content"><font class="title">[[Connecting To Google Voice]] using your SIP account</font><br/>
    8 KB (1,086 words) - 09:08, 31 July 2018
  • debug sip level 10 sip=level 9
    6 KB (819 words) - 14:08, 16 August 2023
  • ...(e.g. for callto.1=sip/sip:1@host1, callto.2=sip/sip:2@host1, callto.4=sip/sip:3@host1 the module will handle the first 2 targets and will ignore callto.4 fork sip/sip:1@host1 | h323/2@host2 | sip/sip:3@host3;stoperror=busy
    12 KB (1,844 words) - 11:24, 5 June 2014
  • protocol=sip '''// 5060 is the default SIP port. When using the default port you don't need to specify it'''
    3 KB (425 words) - 10:03, 22 March 2015
  • * [[How to convert SIP headers into SIP parameters names in messages]] Examples on how to convert SIP headers into SIP parameters.
    9 KB (1,376 words) - 10:01, 27 October 2017
  • ... The accounts that are defined in accfile can use different protocols like SIP, H323, Jabber or IAX. * protocol: The protocol to register on (sip, iax, h323, jabber ...)
    8 KB (1,225 words) - 10:07, 6 July 2018
  • ==Session Initiation Protocol(SIP)== The Session Initiation Protocol(SIP) is an signaling protocol widely used for controlling communication session
    8 KB (1,083 words) - 11:49, 27 October 2017
  • SIP routing in Yate can be done from a single routing module or from a combinat The calls can be routed directly to a SIP channel from a routing module (e.g regexroute.conf).
    3 KB (422 words) - 08:08, 23 November 2017
  • * SIP Methods like SUBSCRIBE and NOTIFY are required. == Implemented SIP Methods==
    3 KB (456 words) - 09:16, 9 April 2015
  • * [[SIP Configuration File|ysipchan]] VoIP SIP driver based on our own [[YASS]] (Yet Another Sip Stack) library
    12 KB (1,662 words) - 13:25, 1 October 2019
  • ; (other then presence) subscriptions e.g. SIP 'dialog' or 'message-summary'
    6 KB (835 words) - 15:05, 7 April 2015
  • param['id'] = 'sip/1' param['module'] = 'sip'
    17 KB (1,082 words) - 13:07, 18 August 2022
  • Yate has support for SIP Protocol. The module is called ysipchan and the associated configuration fi This module is a VoIP SIP driver based on [[YASS]] library.
    25 KB (3,811 words) - 10:09, 6 July 2018
  • ... Internet Protocol (VoIP) and PSTN, it can easily be extended. It supports SIP, H.323, IAX, MGCP, Jingle, Jabber, E1, T1, analogic, ISDN PRI, BRI, and SS7 ...Client is an Instant Messenger and Voice application for Jabber/Jingle IM, SIP, H.323, and IAX VoIP protocols.-->
    5 KB (888 words) - 13:05, 10 March 2015
  • == SIP messages== * [[SIP Generic Message|sip.<methodname>]]
    3 KB (329 words) - 08:19, 25 May 2018
  • ..._how_to_enable_chat_messages_in_SIP#Custom_applications|Custom handling of SIP MESSAGE requests from javascript]] How to hangle call.route for SIP MESSAGE request to build SMSC or allow SIP messages only for some users
    2 KB (323 words) - 09:50, 21 August 2017
  • debug sip level 10 * for sip module
    7 KB (905 words) - 15:03, 14 October 2016
  • sip/74: ReRoute nip.js is a global script and sip/74 is a sip channel that is assisted by the configured routing script
    3 KB (387 words) - 09:46, 21 August 2017
  • |Nokia N80|| cvs 08.02.2007|| SIP || OK || diana@voip.null.ro |WM6 smartphone<sup>1</sup>|| 1.3 || SIP || OK || vmarco@hotmail.com
    9 KB (1,256 words) - 07:01, 25 September 2015
  • ...etion help indicates in configuration mode. If it says "available for h323/sip only" you need to apply the workaround: receive-rtcp configure RTCP receive timer- '''available for h323/sip only''' at this time
    6 KB (921 words) - 11:10, 26 October 2017

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