Pjsip call hold. I did not find anything in pjsip/p Group PJSIP_MSG_HDR group PJSIP_MSG_HDR Declarations for various SIP header fields. 在基于PJSIP开发的VoIP应用中，准确检测通话保持状态是一个常见需求。本文将详细介绍如何在PJSIP项目中实现通话保持状态的检测机制。 ## 通话保持状态的基本原理 PJSIP作为开源SIP协议栈，提供了完整的通话状态管理机制。当通话进入保持状态时，系统会通过特定的回调函数通知应用程序。理解 MicroSIP is a portable SIP softphone based on the PJSIP stack available for Microsoft Windows operating systems. newCallId, and return the Call object via prm. Describe the bug when call on hold error how to get from pjsua2 Steps to reproduce CallOpParam op = new CallOpParam(true); op. enumerator PJSUA_CALL_SET_MEDIA_DIR Set media direction as specified in pjsua_call_setting. ( also ci = py_pjsua. The channel is either on hold or a call waiting call. g: hold, transfer, change media parameters, application must instantiate a new Call object for the new call using call ID specified in prm. However, the memory allocated for the pj_thread_t itself will only be released when the pool used to create the thread is destroyed. g: PJSUA_CALL_UNHOLD, PJSUA_CALL_UPDATE_CONTACT, PJSUA_CALL_NO_SDP_OFFER. For this NAT example, the important config options to note are local_net, external_media_address and external_signaling_address in the transport type section and direct_media in the endpoint section. Call Management Relevant source files This document describes how call handling works in the PJSIP framework, focusing on the call management architecture, data structures, APIs, and operations. call_get_info (call_id) ). PJSUA . Notes about call setting: if call setting is changed in the subsequent call to this function, only the first call setting supplied will applied. This function is different than answering the call with 3xx-6xx response (with pjsua_call_answer ()), in that this function will hangup the call regardless of the state and role of the call, while pjsua_call_answer () only works with incoming calls on EARLY state. 文章浏览阅读2. 2版本中添加）； PRACK（100rel，RFC 3262）； UPDATE (RFC 3311)； OPTIONS Describe the feature I would like to have an easy way how to detect a SIP call is inactive (no audio data in RTP) and terminate it after reaching certain timeout. media_dir. Describe the bug Hang up immediately after making a call, which will trigger a deadlock. getMedia () function will return valid audio media. I'm working on PJSIP Android app and facing a problem with call hold. conf Configuration We are assuming you have already read the Configuring res_pjsip page and have a basic understanding of Asterisk. Application must link with pjsip-ua static library to use this API. PJSUA is a console based application, designed to be simple enough to be readble, but powerful enough to demonstrate all features available in PJSIP and PJMEDIA. More on the callback will be explained a bit later. . conf [endpoint]: Endpoint The Endpoint is the primary configuration object. Audio media status of the call When the audio media of the call is ready (or activated), only the audio media of the call (for example, a call connected to a sound device in the conference phone) can be used. Depending on the status code specified as parameter, this function may send provisional response, establish the call, or terminate the call. MicroSIP is a portable SIP softphone based on the PJSIP stack available for Microsoft Windows operating systems. fritzbox have a default lock for outbounding sip calls) and that you call a different number from your account extention Adding projects necessary to build pjsip and use from C# - Windows, Windows Phone, Xamarin/iOS, Xamarin/Android - StrikeForceZero/PJSip-CSharp A ready-to-use C# implementation of the PJSIP PJSUA2 API - AaronReynoldsUK/PJSIP-PJSUA2-CSharp PJSUA Command Line Interface (CLI) Manual Table of Contents PJSUA Command Line Interface (CLI) Manual Introduction Commands Root commands Call and related commands [call] IM and Presence commands [im] Account commands [acc] Conference and Media commands [audio] Status and config commands [stat] Video commands [video] Introduction CLI is a feature of pjsua that enables user to execute commands The application is not able to retain its current call state like mute, stop video feed, front/back camera when unHolding the call. Note that to release hold or update contact or omit SDP offer, this parameter cannot be NULL and it must specify appropriate flags, e. Subsequent operations to the call can use the method in the call instance, and events to the call will be reported to the callback. The changes to the call’s media state is Hangup call by using method that is appropriate according to the call state. Communication with another SIP device is accomplished via Addresses I am struggling with this : when entering a queue, I'd like to play a welcome message to the caller. Call media status changes are reported in the onCallMediaState () callback, and if the calling audio media is ready (or active), the Call. Center Call. Jul 9, 2015 · Hold and Unhold sip call using the PJSIP Asked 10 years, 7 months ago Modified 8 years, 4 months ago Viewed 2k times Sep 17, 2024 · Describe the feature Hi, Do you have a way to detect when the current caller press (on-hold) during the call so I could do some logic or handle something? which callback do I need to use? onCallMed May 22, 2025 · Call Management Relevant source files This page covers SIP call lifecycle management using the PJSUA2 high-level API, including call creation, state handling, media negotiation, and in-call messaging features. Contribute to jchavanton/pjsua development by creating an account on GitHub. Getting this error while unholding a call using pjsip. 0 The Endpoint is the primary configuration object. PJSUA是一个开源的命令行SIP用户代理（软电话），用PJSIP协议，PJNATH，和PJMEDIA实现。 它虽然只有很简单的命令行界面，但是功能齐全。 SIP功能： 多个id（帐户注册）； 多个呼叫； 支持IPv6（在1. This setting is useful for ensuring that ; holes in NATs and firewalls are kept open throughout a call. Learn how to configure VitalPBX to work with Telnyx using either credentials (user/pass) or IP authentication. Mar 13, 2024 · Describe the bug Hello, In the rare case where two endpoints attempt to place the call on hold at the same time, PJSIP generates a 491 error on both sides, and the Working with Call’s Audio Media Application can only operate the call’s audio media when the call’s audio media state is ready (or active). I'm running FreePBX 13. Communication with another SIP device is accomplished via Addresses -1 - None 1 - Reverse Charging Requested no_media_path - R/O PRI Nonzero if the channel has no B channel. Contribute to mojolingo/asterisk development by creating an account on GitHub. I want to make an app that makes a call, plays wave file and hangs up after that. e unMute, call with video-enabled, font camera Comprehensive documentation hub for Sangoma products and services, providing resources, guides, and support for users. Does this on both Android and iOS. on_incoming_call Notify application on incoming call Call Management Relevant source files This document describes how call handling works in the PJSIP framework, focusing on the call management architecture, data structures, APIs, and operations. setOptions (pjsua_call_flag. This issue happens when call resumes using hold/unhold function. The caller enters first a short queue (initQ) without announce, then enters the mynewQ where I'd SIP Service for Android based on PJSIP. Configuration File: pjsip. It facilitates high quality VoIP calls (p2p or on regular telephones) based on the open SIP protocol. I am have on_call_state callback (where a call to ci = py_pjsua. ;rtp_timeout= ; Hang up channel if RTP is not received for the specified ; number of seconds when the channel is off hold (default: ; "0" or not enabled) ;rtp_timeout_hold= ; Hang up channel if RTP is not received for the specified ; number 9- Call. Detach pjsip library from the current thread without terminating the OS thread, frees the resources allocated by pjlib for the calling thread, including freeing the tls slot in which pj_thread_t is stored. PJSUA 呼叫使用Call来实现，一般根据需要我们需要自定义Call的实现 在具体实现类中，通过重写呼叫回调，用于处理与呼叫有关的事件，如呼叫状态更改或来电转接请求。 呼叫 呼叫是一个 Describe the feature Hi, Do you have a way to detect when the current caller press (on-hold) during the call so I could do some logic or handle something? which callback do I need to use? onCallMed Send response to incoming INVITE request with call setting param. On unHold, all the call settings reset to default i. 49 (Asterisk 13. 0. call_get_info (call_id) is) and also i am polling pjsip from my app in while loop. I set rtp_timeout=15 in Clone of Asterisk. 持有 MainActivity. Usually this happens once the call has been established, although media can active before that (it is called early media), and established call can have no media (such as when it is being put on-hold). PJSIP is a free and open source multimedia communication library written in C language implementing standard based protocols such as SIP, SDP, RTP, STUN, TURN, and ICE. More detailed information is explained in PJSIP Developer’s Guide PDF document, and readers are encouraged to read the document to get the concept behind dialog, dialog usages, and INVITE sessions. Communication with another SIP device is accomplished via Addresses of Record Two threads entered pjsua_call_get_info simultaneously? Looking at the time i suspected it happens when i am calling and call is not yet confirmed. To be able to control the call, e. on_dtmf_digit Notify application upon receiving incoming DTMF digit. When I use the following code, Call is made but I cannot hear any music that is supposed to play ( Cannot place an incoming call on hold as it says: Call/Transaction Does Not Exist. options = (uint)pjsua_call_flag. I was able to set RTP timeouts on the endpoint so that it recognizes loss of connectivity and hangs up, but the call on the Asterisk server side of things continues indefinitely until my other endpoint hangs up. 5. Center is a free multiplatform SIP and VoIP client for enterprise users. The app presents a unique, simple-to-use, and intuitive drag-to-call user interface. pjsip. Parameters: -- Started music on hold, class 'default', on channel 'PJSIP/sip_trunk_in-0000000b' -- Channel Local/67666@transfer_remote-00000005; 1 joined 'simple_bridge' basic-bridge <b6f9ca65-6101-4cc4-ab7d-52020e634d 49> This flag is only valid for pjsua_call_set_hold (), pjsua_call_reinvite (), and pjsua_call_update (). Default is PJSUA_CALL_HOLD_TYPE_RFC3264, and there's no reason to change this except if you're communicating with an old/non-standard peer. While the caller is call to the receiver when caller is put call on hold, receiver how can identify is remote server call on hold? Send response to incoming INVITE request with call setting param. While the caller is call to the receiver when caller is put call on hold, receiver how can identify is remote server call on hold? Specify the default call hold type to be used in pjsua_acc_config. pjsip / pjproject Public Notifications You must be signed in to change notification settings Fork 863 Star 2. 2k次。本文通过解析simple_pjsua. Are you able to make hold on both sides? Cause when I am trying to hold the call from the calling side is ok, but from the receiving side it stops the call, sending disconnect. The snippet above creates a Call object and initiates outgoing call to dest_uri using the default call settings. Works fine for outbound calls. It covers basic and advanced call operations including making calls, answering calls, call media management, and call events. The INVITE session does NOT manage media. Download MicroSIP, full or lite version, installer or zip archive with portable version. Unable to create re-INVITE: Invalid operation (PJ_EINVALIDOP) [status=70013] Error hanging up call: Invalid res_pjsip: SIP Resource using PJProject This configuration documentation is for functionality provided by res_pjsip. prm. So normally application will not supply call 我开发了VOIP应用程序来进行和接收sip调用，我构建了pjsip，如"“中所描述的。1. ASTERISK-30350: res_pjsip_sdp_rtp: rtp_timeout_hold is not used when moh_passthrough has call on hold [Home] Contribute to jchavanton/pjsua development by creating an account on GitHub. I am new to PJSIP. newCall. buffers - W/O Change the channel's buffer policy (for the current call only) This option takes two arguments: Number of buffers, Buffer policy being one of: 'full' 'immediate' 'half' PJSIP is a free and open source multimedia communication library written in C language implementing standard based protocols such as SIP, SDP, RTP, STUN, TURN, and ICE. c的main函数流程，详细介绍了PJSIP的使用方法，包括创建、初始化、设置回调函数、创建传输端口、启动、设置SIP用户账号及发起呼叫等步骤，并附带了来电、呼叫状态和媒体状态改变的回调函数示例。 I'm working on PJSIP Android app and facing a problem with call hold. I read the code and will call acquire_ Call, wait for the lock for 2S, and then return to the deadlock. So normally application will not supply call make sure you have enough credit to place a call, you don't have a firewall that lock traffic on 5060 port (eg. res_pjsip: SIP Resource using PJProject This configuration documentation is for functionality provided by res_pjsip. This flag can be useful in IP address change scenario where IP version has been changed and application needs to update target IP address. Call management is built on top of the PJSIP stack and provides object-oriented abstractions for handling voice and video calls. Application can monitor the status of the call transfer request, for example to decide whether to terminate existing call. It works with a SaaS system for enterprise to manage, control their incoming and outgoing calls for their employees. conf [endpoint]: Endpoint Since 12. Contribute to VoiSmart/pjsip-android development by creating an account on GitHub. It contains the core SIP related options only, endpoints are NOT dialable entries of their own. 3k on_call_transfer_status Notify application of the status of previously sent call transfer request. 0) with PJSIP and running into a problem when my endpoint disconnects form the network while the call is in progress. l69u, x3o48t, jlvp, 3kdym, bk5xv, m4ts, pxvi0, fx871c, pgbcs, 2zpd,