sip

Should a SIM Card and Cell Phone Produce an Authorization Response after Recieving “401 Unauthorized”?

由 醉酒当歌 提交于 2019-12-25 11:13:54
问题 I am trying to use FreePBX 13 (a GUI version of Asterisk) to create a mock VoLTE network with our cellphones (an iPhone 6S+ and an LG G3 VS985). FreePBX has been set up as such: FreePBXForums: Using FreePBX 13 with a Mock Cellular Network At the moment, we are able to register and make calls with computer clients, but the UEs (aka the cellphones) fail to register because of a "realm" mismatch among other things. Because of this I have a couple questions: If a client (even a UE) recieves a 401

how to catch the value of dialpads pressed button?

由 本秂侑毒 提交于 2019-12-25 08:13:57
问题 I am developing a SIP application for making and receiving a call. For that purpose I did analysis on open source project SipDroid. in that project how they catch the value of dialpads pressed button which is sent to the particular method for making a SIP call. I tried to find the code for that task but I didn't get anything.in which file the code is resides to catch that value in SipDroid project? 回答1: The calls in SipDroid are handled by the SipdroidEngine: org.sipdroid.sipua.SipdroidEngine

how to catch the value of dialpads pressed button?

由 一笑奈何 提交于 2019-12-25 08:10:15
问题 I am developing a SIP application for making and receiving a call. For that purpose I did analysis on open source project SipDroid. in that project how they catch the value of dialpads pressed button which is sent to the particular method for making a SIP call. I tried to find the code for that task but I didn't get anything.in which file the code is resides to catch that value in SipDroid project? 回答1: The calls in SipDroid are handled by the SipdroidEngine: org.sipdroid.sipua.SipdroidEngine

Linphone NDK compilation error

由 让人想犯罪 __ 提交于 2019-12-25 05:24:22
问题 Getting error when trying to compile Linphone Native Library using NDK G:\WorkSpaces\Demo\LinphoneLauncherActivity>G:\android-ndk-r8d-windows\android-n dk-r8d\ndk-build Build X264 plugin for mediastreamer2 "Compile arm : vpx <= vpx_mem.c In file included from jni/..//submodules/externals/build/libvpx/../../libvpx/vpx _mem/vpx_mem.c:18:0: jni/..//submodules/externals/build/libvpx/../../libvpx/vpx_mem/include/vpx_mem_i ntrnl.h:14:24: fatal error: vpx_config.h: No such file or directory

Unable to register to FreeSwitch server & unable to call SIP client (XLite) respectively using SIPml5 client

由 丶灬走出姿态 提交于 2019-12-25 04:53:11
问题 I am unable to register to FreeSwitch server & unable to call to SIP client (XLite) by using SIPml5 SIP client. Following is my HTML5 code: <!DOCTYPE html> <html> <head> <meta content="charset=utf-8"/> <script type="text/javascript" src="SIPml-api.js"></script> <title>SIP Client 1</title> <script type="text/javascript"> window.onload = function() { var readyCallback = function(e){ createSipStack(); // see next section }; var errorCallback = function(e){ console.error('Failed to initialize the

Syntax of H.264 SPS/PPS in SIP/SDP offer

由 穿精又带淫゛_ 提交于 2019-12-25 04:04:08
问题 According to RFC 6184: Annex B of H.264 defines an encapsulation process to transmit such NALUs over bytestream-oriented networks. In the scope of this memo, Annex B is not relevant. I see a lot of examplex, including in RFC6236, of SPS/PPS like this a=fmtp:99 packetization-mode=0;profile-level-id=42e011; \ sprop-parameter-sets=Z0LgC5ZUCg/I,aM4BrFSAa However, according to H.264, Annex B, there should be no comma, and a preamble must be added to the beginning of both SPS and PPS (and after

Start activity on BroadcastReceiver while receiving SIP Calls

由 对着背影说爱祢 提交于 2019-12-25 02:35:09
问题 I have my IncomingCallReceiver class from which I want to send my incoming calls to another activity to give user option to Receive or Decline incoming call this is my IncomingReceiver class and clearly out of ideas so If someone might suggest how do I do that. public class IncomingCallReceiver extends BroadcastReceiver { @Override public void onReceive(Context context, Intent intent) { SipAudioCall incomingCall = null; try { SipAudioCall.Listener listener = new SipAudioCall.Listener() {

Parse::ABNF perl usage [closed]

由 柔情痞子 提交于 2019-12-25 02:08:47
问题 Closed . This question needs to be more focused. It is not currently accepting answers. Want to improve this question? Update the question so it focuses on one problem only by editing this post. Closed 5 years ago . I need to parse the SIP headers (grammar in ABNF format) and verify if my Header strings are ok or not. (Example: check strings like "Accept: application/sdp,application/3gpp-imp+xml" to provide testcase pass/fail). Currently I am trying to use perl Parse::ABNF. Now I am not able

libeXosip2(1-2) -- How-To initiate, modify or terminate calls.

由 ▼魔方 西西 提交于 2019-12-25 00:49:13
How-To initiate, modify or terminate calls. The eXtented eXosip stack eXosip2 offers a flexible API to help you controling calls. Initiate a call To start an outgoing call, you typically need a few headers which will be used by eXosip2 to build a default SIP INVITE request. The code below is used to start a call: osip_message_t *invite; int cid; int i; i = eXosip_call_build_initial_invite (ctx, &invite, "<sip:to@antisip.com>", "<sip:from@antisip.com>", NULL, // optional route header "This is a call for a conversation"); if (i != 0) { return -1; } osip_message_set_supported (invite, "100rel");

sip-4.16.2 makefile error

由 99封情书 提交于 2019-12-24 21:54:55
问题 I'm trying to install PyQt4 on a Windows 8 machine. I know almost nothing about makefiles, only enough to run make, make install, and make clean. I am stuck on the installation process for SIP. When running nmake from a VisualStudio command prompt, I get the error as follows: python27.lib(python27.dll) : fatal error LNK1112: module machine type 'x64 conflicts with target machine type 'x86' NMAKE : fatal error U1077: '"C:\Program Files (x86)\Microsoft Visual Studio 12.0\VC\BIN\link.EXE?' : ?
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