pjproject | Asterisk fork of PJSIP NO PULL REQUESTS OR ISSUES!!! | TCP library
kandi X-RAY | pjproject Summary
kandi X-RAY | pjproject Summary
Copyright (C) 2003-2008 Benny Prijono benny@prijono.org Copyright (C) 2008-2011 Teluu Inc. (This program is free software: you can redistribute it and/or modify it under the terms of the GNU General Public License as published by the Free Software Foundation, either version 2 of the License, or (at your option) any later version. This program is distributed in the hope that it will be useful, but WITHOUT ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for more details. You should have received a copy of the GNU General Public License along with this program. If not, see Getting Started: Building and Using PJSIP and PJMEDIA. [Last Update: $Date: 2007-02-02 20:42:44 +0000 (Fri, 02 Feb 2007) $]. This article describes how to download, customize, build, and use the open source PJSIP and PJMEDIA SIP and media stack. The online (and HTML) version of this file can be downloaded from
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QUESTION
I'm working within MS Project and trying to use a macro to import project information from a spreadsheet to MS Project. What I have so far works fine for importing the Task information from the spreadsheet. I'd also like to import the Enterprise Custom Fields Project information, like Project Departments -- the information you would see in the Project Information dialog box. I am able to pass hard-coded information to those fields using ProjectSummaryTask.SetField
, like this:
ANSWER
Answered 2021-Mar-23 at 17:16Here is code that you can run from within Project to automate Excel and get the values from a worksheet:
QUESTION
I have built the FFmpeg with libx264 into static libs, here is my directory tree.
...ANSWER
Answered 2021-Feb-22 at 07:15I made a mistake in the build script:
QUESTION
I am trying to make a simple SIP user agent using https://github.com/pjsip/pjproject. I can succesfully connect to a sip server (Twilio) and place calls to PSTN numbers using the pjsua_* interface. This works fine.
What I would like now is to get a notification (through a callback or such) from pjsip when the user that I am calling answers the call.
I am using on_call_state() to get updates on the invite, but this goes through the same states
CALLING -> CONNECTING -> CONFIRMED -> DISCONNCTD
even if the user rejects the call. So I guess I am not looking at the right callback for this.
How can I definitely tell if the user has answered or rejected the call?
...ANSWER
Answered 2021-Jan-21 at 10:36for me it is working this way. in on_call_state callback:
QUESTION
Im trying to cross compile pjproject in yocto with ssl support but it errors out trying to build pjlib-test. Ive tried adding the openssl package to DEPENDS in my .bb file but nothing changes. Any help would be appreciated.
...ANSWER
Answered 2020-Oct-27 at 11:05I would first check why the pjproject configure fails to find openssl (it should not be rocket science on linux). But what you asked for can normally be done with something like this:
QUESTION
How do I create a HTML link to an emergency number like
911
or 112
?
The RFC says
The phone number can be represented in either global or local notation. All phone numbers MUST use the global form unless they cannot be represented as such. [Emergency numbers ("911", "112")] cannot be represented in global form and need to be represented as a local number with a context.
From the local-context section I don't find it easy to understand what a "local-context" is, let alone what the correct one for this case is. It lists domain prefixes like houston.example.com
or a numeric prefix like +1
, and in one paragraph it says
A context consisting of the initial digits of a global number does not imply that adding these to the local number will generate a valid E.164 number. It might do so by coincidence, but this cannot be relied upon. (For example, "911" should be labeled with the context "+1", but "+1-911" is not a valid E.164 number.)
But the phrasing of this paragraph is again very confusing.
Is
...ANSWER
Answered 2020-Apr-04 at 19:03The cited Section 5.1.5 the RFC states
A context consisting of the initial digits of a global number does not imply that adding these to the local number will generate a valid E.164 number. It might do so by coincidence, but this cannot be relied upon. (For example, "911" should be labeled with the context "+1", but "+1-911" is not a valid E.164 number.)
I interpret this that emergency numbers should be prefixed by their country-secific prefixes, i.e.
The rest of the paragraph is about this syntax not being compliant to the E.164 recommendation. As far as I understand, E.164 is irrelevant in this context though.
QUESTION
I am trying to compile pjsip library for iOS, but getting an error and I am following below steps
Step 1: Create a new file config_site.h in the project folder /pjlib/include/pj/ with the following code:
...ANSWER
Answered 2020-Feb-11 at 03:30Refer to https://trac.pjsip.org/repos/wiki/FAQ#missing-sep-err
This happens because of corrupt dependency file, probably because make dep stopped or was stopped abruptly previously. The solution is either to delete the offending file manually or to run make distclean to clean everything. Either case, you'd have to run make dep again to rebuild the dependency.
QUESTION
I want to create a Docker with asterisk 16, I can register my sip extension but I don't listen anything.
I can go to 8088 port too, modifying http.conf
Also I edited rtp.conf
with
ANSWER
Answered 2019-Jun-13 at 16:00Unfortanly asterisk need AT LEAST 2 ports for ONE call.
For two legs calls(call to provider and back) it already need FOUR ports.
So you need change your rtpsettings and change docker accordinly.
Actually it is REALY bad idea run sip inside docker if you have zero knowledge about SIP. Use some manual from people who have experience.
QUESTION
I'm trying to make a call to a telephone number. I'd like to be able to make a call from the raspberry pi, and also make a call to my voip.ms phone number and be able to answer or auto-answer and play some generic .wav file.
My current understanding of thingsThis maybe should be titled "My current misunderstanding of things". I'm new to sip and pjsip, and I think I must be missing some part of the process I don't understand. I was under the impression that, if I register with voip.ms, when I make a call it would route to voip.ms and they would do a lookup on the number/address, and then respond with an address that I would then begin to communicate with.
What I've done so far CompiledI've compiled Pjsip on a Raspberry Pi 3B+ properly, for what I can tell. I can include pjsua2.hpp in my c++ applications. I've roughly followed this tutorial
Tested compilation with pjsua binary && demo.cppI'm running into identical problems running a modified pjsua2_demo.cpp and the binary included in the pjsip build. For the sake of simplicity, I'll ask about the binary located (for me) at /pjproject-2.8/pjsip-apps/bin/pjsua-armv7l-unknown-linux-gnueabihf
.
I have an account and phone number with Voip.ms and can become registered with voip.ms by executing the following script:
call_and_auto_answer.sh
...ANSWER
Answered 2019-Mar-22 at 22:44So I figured out the answer to my question. Here's the skinny:
Voip.ms registrationMy registration with voip.ms wasn't configured properly. I was given credentials by a coworker, but upon further inspection of the sip endpoint, I found that the DiD number purchased for the account wasn't associated with the subaccount my coworker created for me. So, depending on the recipient's phone carrier, I was given different errors. Additionally, when I was testing inbound calls and receiving the error, User Busy
, this was because the account I registered wasn't associated with the phone number.
To fix this, on voip.ms I associated the DiD number to my subaccount, and then went to my subaccount information and set the callerId number to be my DiD number, though I think you can override this value via pjsip.
Outbound sip callsSecondly, to call a phone number, outbound calls should follow the following format:
sip:@
So for me, this looked very much like:
sip:5551234567@newyork.voip.ms
QUESTION
This is not a duplicated question, other user had the same problem but this question add more info. I'm working on Ubuntu 18.
When I receive a call using VoIP with and asterisk server, my python script crash. This python script works in other computer but not in mine. So I tried to diagnose the pjsip dependency because the problem come from there.
I use pjsip as dependency of Python Most Voip Library
The pjsip installation process is the following:
...ANSWER
Answered 2019-Jan-24 at 15:06After debugging the python library, I found that when a call arrive, a wav file try to be used. I installed python library using --user
option, and the point where the library look for the file was wrong.
Using --user
option make installation in a non-standard system path. If a call arrives the wav file won't be found and python library crash. To solve this, just add a symbolic link to the correct path:
QUESTION
I am getting following error while building pjsip
as mentioned in this link.
When I run the make
command:
ANSWER
Answered 2018-Mar-29 at 14:04Ubuntu 16.04 - amd64
git clone https://github.com/svn2github/pjsip.git
... or pjproject-2.7.2.tar.bz2
: Both will build OK here, with gcc version 4.8 → sudo apt install g++-4.8
Build example, ref. https://trac.pjsip.org/repos/wiki/Getting-Started/Autoconf
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