r3gis3r / csipsimple Goto Github PK

CSipSimple Mirror (no pull-requests here)

License: GNU Lesser General Public License v3.0

Makefile 1.21% HTML 1.70% Shell 0.13% C 33.47% C++ 4.21% Diff 2.10% Objective-C 0.21% Python 0.18% Java 56.80%

csipsimple's Introduction

CSipSimple is an open-source native SIP client for Android

See http://www.csipsimple.com for more info

Copyright (C) 2009-2010 Regis Montoya (http://www.r3gis.fr)
Copyright (C) 2010 Robert B. Denny, Mesa, AZ, USA ([email protected])

This file is part of CSipSimple


CSipSimple 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 3 of the License, or
(at your option) any later version.

CSipSimple 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.
If you own a pjsip commercial license you can also redistribute it
and/or modify it under the terms of the GNU Lesser General Public License
as an android library.

You should have received a copy of the GNU General Public License
along with CSipSimple.  If not, see <http://www.gnu.org/licenses/>.


Keep in mind that the entiere application rely on a library 
released under GPL license terms. Please contact them
if you want to use the LGPL dual license of csipsimple.
Copyright (C) 2003-2008 Benny Prijono <[email protected]>
Copyright (C) 2008-2009 Teluu Inc. (http://www.teluu.com)
- Pjsip (See The PJSIP.ORG website)
provides the Open Source, comprehensive, high performance, small 
footprint multimedia communication libraries written in C language 
for building embedded/non-embedded VoIP applications. 

Cryptography notice
This distribution includes cryptographic software. The country 
in which you currently reside may have restrictions on the import,
possession, use, and/or re-export to another country, of 
encryption software. BEFORE using any encryption software, please 
check your country's laws, regulations and policies concerning 
the import, possession, or use, and re-export of encryption 
software, to see if this is permitted.
See <http://www.wassenaar.org/> for more information.

The U.S. Government Department of Commerce, Bureau of Industry and Security 
(BIS), has classified this software as Export Commodity Control Number 
(ECCN) 5D002.C.1, which includes information security software using or 
performing cryptographic functions with asymmetric algorithms. 
The form and manner of this CSipSimple distribution makes it eligible for 
export under the License Exception ENC Technology Software Unrestricted (TSU) 
exception (see the BIS Export Administration Regulations, Section 740.13) 
for both object code and source code.

csipsimple's People

Contributors

Stargazers

Watchers

Forkers

mycelialold voxofon cyphient vishnoi saurabhariel alfabit kuanjava balusreekanth groblerd dastaniqbal morteza-rastgoo duanbo fmedlin goranch minimaldevelop centuryglorion barrychi mobileservice lspecian billyyi thenameisnigel-old billthebest song-buaa berroteran lidocan2012 suresh01 applu zakiravian baneizalfe lewisardern nathanbai misaelcarvalho ekospinach cppalien roshan2188 appmodule michaelye lennychien pikeman-hm wl820215 y7y7y davi-he msoloyu e-wiki fogleng winyatasenjaya desys21 kinghyjiang jackbboy digitalsoft toharish cristeab andy2151 ibonobo ezcall-net-tw shunyauads ruiasantos pasichnichenko loveamy tommyteavee emadhedayati rohitshampur qwivan guffywave gtv-telecom-bvba erhuabushuo gospeljohn kasirye hamzamp ybpranawa ttsigg chencao0524 zxcpoiu watershao sungbaek mrlooper anubhav1510 shuaiyanping jcheung0 rfminelli skadyrov nafil ivan-smagleson prativa ibstelix jlte kuberjsr butcher88 francogo stevezhougs gijung00 weiqiwang 309746069 mehmetzeybel riekr jmsecurity widefox mahendran-sakkarai segamza bagassy

csipsimple's Issues

Local call can't be terminated

Hello,

from time to time I like to do local sip call. Just calling sip: within my local network.

CSipSimple supports these kinds of calls, but there's a bug. Once answered, you cannot hung up on local calls. The red button is missing. You can only terminate such calls by killing the app.

At the time the screenshot was taken, the call was already active and the audio connection was established (and perfectly working). But CSipSimple still shows ”Eingehender Anruf“ which means ”incomming call“.

Regards,
Matthias

Cannot load native library .cpu arch invalid for this build in android studio

I have exported this project to android studio.But I am unable login due Cannot load native library .cpu arch invalid for this build.How can i solve this problem? Can anyone please help me to solve this issue ?

Nothing happens in CSipSimple when I try to place a call

This has been happening for about 2 weeks (via PPPoE, through an optical fibre connection).

It would sometimes work for a few weeks before that (via PPPoE, through an optical fibre connection)).

It would work 100% of the time before that (via modem, through an optical fibre connection).

At first, I thought it might be something to do with the connection method I was using, but I haven't been able to place a call using CSipSimple at all for a couple of weeks.

What steps will reproduce the problem?
Try to place a call using CSipSimple.

What is the expected output?
The call should start immediately, as it always did previously.

What do you see instead?
The app looks as if it's trying to place the call, but there is no sound whatsoever.

What version of the product are you using? On what device / operating system?
Samsung SM-G920I (Galaxy S6)
Android 6.0.1
Official Australian Samsung ROM
CSipSimple 1.02.03 r2470

Please provide any additional information below.
I have tried killing and restarting the app. It doesn't work.
I have tried restarting the phone. It doesn't work.
Come to think of it, maybe the issue started happening around the same time as the ROM was updated one time.

Pjsip sources directory

Hi dear,

Is there any reason that sources directory in jni/pjsip path is not included in this repo?

Thanks

Android integration

Hi，i find set Android integration can work only once,
however i set value is true or false ,� native dialer can't integration call logs

CSipSimple does not honour silent mode

When I have my phone in silent mode and receive a call through CSipSimple it rings in full tone, not honouring the system settings. CSipSimple should query the system volumes and vibration settings and act accordingly.

(Sorry if this is not the correct place for bug reports. Google Code is now closed and I could find no other repo expect this one.)

CSIP Status: 500 Unhandled by dialog usages while expecting 200 OK

Hi all, I'm having problems with CSipSimple (latest version) & an Aastra PBX.
Every call with Csip end after 90 seconds.

CSipSimple is sending 500 Unhandled by dialog usages while expecting on a 200 OK.
After 3 times Aastra end te connection.

Can you help me here?

Regards,

Ron Telleman (from Holland)

CSipsimple logging:

28 2016-04-20 10:21:33.905058 192.168.20.100 192.168.20.220 SIP/SDP 790 Request: INVITE sip:[email protected]:40000 | 0
31 2016-04-20 10:21:33.997832 192.168.20.220 192.168.20.100 SIP 428 Status: 100 Trying | 46
32 2016-04-20 10:21:34.128166 192.168.20.220 192.168.20.100 SIP 609 Status: 180 Ringing | 46
33 2016-04-20 10:21:34.140172 192.168.20.100 192.168.20.117 SIP 565 Status: 180 Ringing | 0
54 2016-04-20 10:21:39.603859 192.168.20.220 192.168.20.100 SIP/SDP 901 Status: 200 OK | 46
243 2016-04-20 10:22:09.621835 192.168.20.100 192.168.20.220 SIP 584 Request: INFO sip:[email protected]:40000;ob | 0
244 2016-04-20 10:22:09.640482 192.168.20.220 192.168.20.100 SIP 481 Status: 500 Unhandled by dialog usages | 46
245 2016-04-20 10:22:09.774806 192.168.20.100 192.168.20.117 SIP 609 Request: INFO sip:[email protected]:37667;ob | 0
246 2016-04-20 10:22:09.790103 192.168.20.117 192.168.20.100 SIP 506 Status: 500 Unhandled by dialog usages | 46
383 2016-04-20 10:22:39.647280 192.168.20.100 192.168.20.220 SIP 584 Request: INFO sip:[email protected]:40000;ob | 0
384 2016-04-20 10:22:39.664854 192.168.20.220 192.168.20.100 SIP 481 Status: 500 Unhandled by dialog usages | 46
387 2016-04-20 10:22:39.796210 192.168.20.100 192.168.20.117 SIP 609 Request: INFO sip:[email protected]:37667;ob | 0
388 2016-04-20 10:22:39.811575 192.168.20.117 192.168.20.100 SIP 506 Status: 500 Unhandled by dialog usages | 46
579 2016-04-20 10:23:09.671762 192.168.20.100 192.168.20.220 SIP 584 Request: INFO sip:[email protected]:40000;ob | 0
580 2016-04-20 10:23:09.688094 192.168.20.220 192.168.20.100 SIP 481 Status: 500 Unhandled by dialog usages | 46
581 2016-04-20 10:23:09.693336 192.168.20.100 192.168.20.220 SIP 511 Request: BYE sip:[email protected]:40000;ob | 0
582 2016-04-20 10:23:09.694104 192.168.20.100 192.168.20.117 SIP 536 Request: BYE sip:[email protected]:37667;ob | 0

per account settings

it would be nice to have a way to override settings for each account.

my use case: i'd like to have ZRTP enabled globally, but freevoipdeal, sipdiscount (and probably many others) don't work when ZRTP is enabled... so i either have to disable it globally, or i cannot make calls to PSTN numbers using CSipSimple.

add support for the ENUM dialing popup from Lumicall

Lumicall provides a convenient ENUM dialing popup.

This code is very generic and could be shared between the Lumicall and CSipSimple projects in a library JAR.

STUN server is necessary for one account, but not for the other

Hello,

First of all, I've read this wiki page.

But I cannot agree on that, because I have a SIP server running on my Fritz Box (an AVM router). There I have to connect without STUN, since only the network internal IP is valid.
And then there is sipgate (my other account), which needs STUN, because it is outside my local network.

How can I solve this? That's an issue.

make VideoLibs Error

Hi,

When I build using the command 'make VideoLibs', it is not building successfully due to some errors. Can you help me fix the issue?

Keeps Ringing Indefinitely

On Android 4.1.2, Neptune Pine
Version 1.02.03 r2457
If a call goes unanswered, CSipSimple just goes on ringing indefinitely, hours long after the caller hung up.

Most important ISSUE: Does not receive calls in sleep mode

Ive tried all the varioius workarounds and no joy(read all the issues with 'sleep').
no matter what I change, it always happens.

After 3 mins, or less csipsimple does not pick up calls but rather sends them straight to voicemail suggesting a problem with registration or program operation in sleep mode.

(3g works in sleep.
I use the 3g connection (shared wirelessly), in sleep mode.

Therefore its not 3g & it should be possible to get the program to work seamlessly like the wifi and and googletalk currently do. )

pasted this from the old google code page:
Issue 2959: App alwys goes inactive/to sleep preventing incoming calls. ‹ Prev 11 of 14 Next ›
1 person starred this issue and may be notified of changes. Back to list
Status: New
Owner: ----
Type-Defect
Priority-Medium

Reported by [email protected], Jul 23, 2015
What steps will reproduce the problem?

The app constantly goes inactive

What is the expected output? What do you see instead?
N/A

What version of the product are you using? On what device / operating
system? 1.02.03 r2457 on Huawei Y550-L02 android version 4.4.4

Please provide any additional information below.

I only use the app on my android phone for incoming calls, however, after minimal time, I notice it's gone inactive which prevents people from calling me. This happens constantly.

Will not work while in adhoc mode

i shifted my samsung I-9100 to ad-hoc mode succesfully but when i launch csipsimple it says selected network route is not possible or something like that?

Blank activity over lock screen after missing call

Phone: Samsung Galaxy S4 on Android 4.2.2, uses screen lock with pattern. Source: night build v1.02_03

How to cause:

Target phone is online, power is off.
Second phone registering to server and immediatelly making a call to the first one.
In 5-10 seconds we are making hang-up.
In some time (up to 1 minute) Asterisk notifying the first phone about missing call. Screen isn't on, blue LED blinking.
We turning-on the first phone and show the blank (black) activity that not handle any touches.
After Back button pressing we can show lock screen.

Screen captured via DDMS. I tried to allow showing over lock screen by adding the WindowManager.LayoutParams.FLAG_SHOW_WHEN_LOCKED flag to SipHome.onCreate or to disable KeyGuard.. Also I tried to clear activities stack when SCREEN_ON was received.

WBR, Nicolas.

Add fax support using OPAL (Open Phone Abstraction Library)

OPAL is available on sourceforge

Fax support:
http://wiki.opalvoip.org/index.php?n=Main.FaxSupport

Android Digit Grouping

I am customising CSipSimple VOIP open source project. I would like to remove the digit grouping. Can any one please help me to solve this issue ? I am not getting digit grouping in Lollipop Version . I am facing this problem only in kitkat version.

How to add a contact from the CsipSimple call log to Android phone book?

Most SoftPhone APPs in the market, such as LinPhone, MagicJack, Whatscall, offer the feature to add a contact directly from their call log to android phone book. Does CsipSimple offer the similar feature? I have tried many things but could not find it.

Please help, if anybody happen to know the method. Thanks.

Has anybody tried CsipSimple on Android 7.0 Nougat? Please share your experiences with the community.

Lollipop and dialer integration

Hello,
I have csipsimple on a samsung Note 4 running lollipop 5.1.1. The dialer integration is turned on. When i make a call using the native dialer both the native dialer and the csipsimple make the call. In 4.4 kitkat the native dialer used to reject the call but it seems 5.1.1 work's differently.

Anybody using csipsimple with Lollipop seeing this issue?

Thanks

CSipSimple SIP client for Android Studio

I want to compile CSipSimple Source code in Android Studio.
**
someone help me that what steps I need to follow???????????**

Audio partially missing on vpn 3g connection android 6.0

Given is following configuration:
asterisk openwrt server with openvpn running on the same node
two clients using csipsimple, one on wifi vpn, second on 3g vpn
the client with 3g vpn is not passing audio to wifi vpn client.
the wifi vpn is running android 4.4 and 3g vpn is running android 6.0.
Configuration with two android 4.4 clients is working.

[FIX] make the phone vibrate and ring in the same time (related to google code issue 2458)

As that google code issue 2458 said, it's currently not possible to have it ring and vibrate at the same time (can be either one but not both).
I've solved this by creating a tasker profile that looks for the application csipsimple putting a notification: "call in progress" and variable SILENT !~ on - run a shell script:

[code]

!/system/xbin/bash

while [ $(logcat -d -s libpjsip |tail -1|grep "180 Ringing"|wc -l) -eq 1 ] ; do
echo 500 > /sys/devices/virtual/timed_output/vibrator/enable
sleep 1
done
[/code]

I'm not sure if the "echo" command will run without root privileges though as mine runs as root.

Looking at logcat that "180 Ringing" line is displayed while the phone is ringing and another line will get put afterwards as soon as the call is either connected or disconnected (rejected).

A bit of a nasty hack but it seems to work.

Audio absent on the first seconds of a call

With 3g connection, user that receive the call don't hear anything for at least 3-4 seconds. I'm wondering if It's possible to add some wait tone (for example) to avoid this no signal.

Group Chat

Dear,
How can we implement a group chat using PJSIP or do you find any workaround in your project, as it is not available in CSIPSimple too ?

CSipSimple does not compile

I downloaded the library source code and tried to compile it in Android Studio 1.4 on ubuntu 15.

When I try to compile i get the following error:

AGPBI: {"kind":"error","text":"Error parsing XML: prefix must not be bound to one of the reserved namespace names","sources":[{"file":"/home/jessicabatista/Desktop/CSipSimple-master3/app/build/intermediates/res/merged/debug/values-cs/values-cs.xml","position":{"startLine":1}}],"original":""} AGPBI: {"kind":"error","text":"Error parsing XML: prefix must not be bound to one of the reserved namespace names","sources":[{"file":"/home/jessicabatista/Desktop/CSipSimple-master3/app/build/intermediates/res/merged/debug/values-de/values-de.xml","position":{"startLine":1}}],"original":""} AGPBI: {"kind":"error","text":"Error parsing XML: prefix must not be bound to one of the reserved namespace names","sources":[{"file":"/home/jessicabatista/Desktop/CSipSimple-master3/app/build/intermediates/res/merged/debug/values-es/values-es.xml","position":{"startLine":1}}],"original":""} AGPBI: {"kind":"error","text":"Error parsing XML: prefix must not be bound to one of the reserved namespace names","sources":[{"file":"/home/jessicabatista/Desktop/CSipSimple-master3/app/build/intermediates/res/merged/debug/values-he/values-he.xml","position":{"startLine":1}}],"original":""} AGPBI: {"kind":"error","text":"Error parsing XML: prefix must not be bound to one of the reserved namespace names","sources":[{"file":"/home/jessicabatista/Desktop/CSipSimple-master3/app/build/intermediates/res/merged/debug/values-nl/values-nl.xml","position":{"startLine":1}}],"original":""} AGPBI: {"kind":"error","text":"Error parsing XML: prefix must not be bound to one of the reserved namespace names","sources":[{"file":"/home/jessicabatista/Desktop/CSipSimple-master3/app/build/intermediates/res/merged/debug/values-pt-rBR/values-pt-rBR.xml","position":{"startLine":1}}],"original":""} AGPBI: {"kind":"error","text":"Error parsing XML: prefix must not be bound to one of the reserved namespace names","sources":[{"file":"/home/jessicabatista/Desktop/CSipSimple-master3/app/build/intermediates/res/merged/debug/values-tr/values-tr.xml","position":{"startLine":1}}],"original":""} AGPBI: {"kind":"error","text":"Error parsing XML: prefix must not be bound to one of the reserved namespace names","sources":[{"file":"/home/jessicabatista/Desktop/CSipSimple-master3/app/build/intermediates/res/merged/debug/values-zh-rTW/values-zh-rTW.xml","position":{"startLine":1}}],"original":""} AGPBI: {"kind":"error","text":"Error retrieving parent for item: No resource found that matches the given name \u0027Theme.Sherlock\u0027.","sources":[{"file":"/home/jessicabatista/Desktop/CSipSimple-master3/app/src/main/res/values/styles.xml","position":{"startLine":1}}],"original":""} AGPBI: {"kind":"error","text":"Error retrieving parent for item: No resource found that matches the given name \u0027Theme.Sherlock\u0027.","sources":[{"file":"/home/jessicabatista/Desktop/CSipSimple-master3/app/src/main/res/values-v11/style.xml","position":{"startLine":1}}],"original":""} AGPBI: {"kind":"error","text":"Error retrieving parent for item: No resource found that matches the given name \u0027Theme.Sherlock.Dialog\u0027.","sources":[{"file":"/home/jessicabatista/Desktop/CSipSimple-master3/app/src/main/res/values/styles.xml","position":{"startLine":1}}],"original":""} AGPBI: {"kind":"error","text":"No resource found that matches the given name: attr \u0027actionBarItemBackground\u0027.","sources":[{"file":"/home/jessicabatista/Desktop/CSipSimple-master3/app/src/main/res/values/styles.xml","position":{"startLine":44,"startColumn":46,"startOffset":1999,"endColumn":69,"endOffset":2022}}],"original":""} AGPBI: {"kind":"error","text":"Error retrieving parent for item: No resource found that matches the given name \u0027Theme.Sherlock.NoActionBar\u0027.","sources":[{"file":"/home/jessicabatista/Desktop/CSipSimple-master3/app/src/main/res/values/styles.xml","position":{"startLine":1}}],"original":""} AGPBI: {"kind":"error","text":"No resource found that matches the given name: attr \u0027actionBarItemBackground\u0027.","sources":[{"file":"/home/jessicabatista/Desktop/CSipSimple-master3/app/src/main/res/values/styles.xml","position":{"startLine":44,"startColumn":46,"startOffset":1999,"endColumn":69,"endOffset":2022}}],"original":""}

Any help would be appreciated.
Thanks in advance.

Mute ringing call

On Android stock Phone app, when a call is ringing one is able to mute the ringtone by pressing the power button.
On CSipSimple I could find no way to mute/silent a ringing call. Please implement the same behaviour of the Android Phone.

Google Code is closing

As Google Code is closing, how about making CsipSimple development active on Github ?

Linphone .. CSipSimple !!

I'm using 3G connection
Linphone works but CSipSimple works only on wifi !

Echo in Android 5.1

If I call somebody using csipsimple on the mobile phone. This person hears himself speaking. I thought this issue should be solved with android 5.1 but I still have this problem on my nexus 4.

apk

Are the versions on F-Droud from 2015 the most recent? I cannot use gapps on my devices; no loss akiac.

Hi sir, there is a bug,make error.error: pathspec 'v1.1.0' did not match any file(s) known to git.

D CSipSimple/src/org/webrtc/videoengine/camera/CameraUtilsWrapper.java
D CSipSimple/svn-revision.build.xml
HEAD is now at 70eb7ed... reformat readme
error: pathspec 'v1.1.0' did not match any file(s) known to git.
make[1]: *** [sources] Error 1
make: *** [jni/libvpx/sources] Error 2
➜ trunk git:(70eb7ed) ✗ git checkout v1.1.0
error: pathspec 'v1.1.0' did not match any file(s) known to git.
➜ trunk git:(70eb7ed) ✗

build on mac

Problem on call filtering rules: sip URI become all Number

@r3gis3r
CSIPSimple 1.02.03.r2470 armeabi-v7a

Account 1: filter - can't call with starts with: 0/+/*
Account 2: filter - can't call starts with: 0/* or contains: @
Mobile: filter - can't call contains @

When I try to dial sip URI "[email protected]" it work nice by using account 1.
The problem arise when "Account 1" sipserver.org not reachable or not active it will rewrite the SIP URI to all number. It should be just simple exit.

Please take a look at this..
Thank you for your really hardwork.

Suppress ringing for calls looping back to the very mobile phone I am on myself

When placing a call via the mobile phone network which then loops back to myself (CSipSimple on my phone) the ringing should be suppressed. I don't see the point of calling myself on the very phone I place the call with. Usually in this case there is some other client registered to the number my CSipSimple client is registered to which is the actual target of the outbound call.

CSipSimple and Android DNS Cache

Hi,

Does CSipSimple support registering to an alternative sip registrar ? ( SRV Records Enabled )

For example is SRV record mentions two servers like

_sip._udp.sbc.example.org. 86400 IN SRV 10 50 5060 sbcprimary
_sip._udp.sbc.example.org. 86400 IN SRV 20 50 5060 sbcsecondary

If first server sbcprimary is down , does csipsimple register to sbcsecondary ?
What configuration is required to achieve this ?

Also, does CSipSimple use itsown DNS resolver ?

regards
Anish

Switch from txt to number pad doesn't work

Samsung Galaxy Tab7, Android 4.4.2, CSipSimple 1.02.03
Switching between number pad and txt (keyboard) results in no keyboard, hence not able to set up calls to SIP accounts as no "@" symbol on keypad.
Also, video would be useful

[feature request] ignore first X number of rings

Thanks for making this great program Free.

As I have multiple devices connected for the same SIP account, I would find the capability of ignoring the first X number of rings or seconds on incoming calls useful, as it would allow to receive them on my mobile phone only if someone doesn't answer from the other devices, without having to use call forwarding (which imposes extra charges). Not sure how easy it is to implement but I think it would be a nice feature.

Current Google Play requires "OpenSSL 1.02f/1.01r". Has anybody upgraded Csipsimple to support it?

DSCP for SIP TLS

Hi,

If i use SIP with TCP transport, i've my DSCP value. But if i use TLS transport, i loose my DSCP value.

Is there an issue to recover this DSCP value when i use TLS for SIP?

Regards

recommend Amazon AMI for TURN?

Is there a recommended AMI to run ones own small [ec2] instance for turn?

Android Studio compatibility (gradle-based build)

If you can transition to a gradle-based build (by exporting these files using Eclipse) it will make the project much more portable, easier to setup and able to be used in the new Android Studio IDE.

Crashes (always) on startup on Zenfone 5

I have reported this via the 'Report' option, but am happy to try to assist if there is more info needed.

Clean install app->Open it->Crash

error compiler

Error:(263, 35) error: cannot find symbol class SWIGTYPE_p_f_int_p_q_const__pj_str_t_p_q_const__pj_str_t_p_void_enum_pjsip_status_code_p_q_const__pj_str_t_p_pjsip_tx_data_p_pjsip_rx_data_int__void
Error:(267, 10) error: cannot find symbol class SWIGTYPE_p_f_int_p_q_const__pj_str_t_p_q_const__pj_str_t_p_void_enum_pjsip_status_code_p_q_const__pj_str_t_p_pjsip_tx_data_p_pjsip_rx_data_int__void
Error:(264, 66) error: cannot find symbol variable SWIGTYPE_p_f_int_p_q_const__pj_str_t_p_q_const__pj_str_t_p_void_enum_pjsip_status_code_p_q_const__pj_str_t_p_pjsip_tx_data_p_pjsip_rx_data_int__void
Error:(269, 37) error: cannot find symbol class SWIGTYPE_p_f_int_p_q_const__pj_str_t_p_q_const__pj_str_t_p_void_enum_pjsip_status_code_p_q_const__pj_str_t_p_pjsip_tx_data_p_pjsip_rx_data_int__void

Wrong notification when SIP/SIMPLE message is received

When the first message in a conversation is received, it works fine.

When subsequent messages arrive in the same thread, in the original notification message that appears on the top bar, the first message always appear (instead of the last).
When you pull down the notification area however, the correct (last) message appears.

When trying to integrate with Tasker, the first message is always picked up, not the latest.

Thank you.

Multiple registration is happening at the server

Initially i was using CSipSimple V 0.04.03 in which i was able to over come "Registration is done on the sip server twice " issue by following your instructions but this code is getting crashed while making call in Android 4.4.2.

So I used CSipSimple V 1.02.03 and it was working on all versions but the registration duplication came up again, even after repeating settings that helped to resolve the issue in V 0.04.03

How to overcome this again?

Outgoing calls have a 5-second delay on Android 6.0, HTC M8, when ICE is enabled.

Outgoing calls have a 5-second delay on Android 6.0, HTC M8, when ICE is enabled.
05-27 17:08:49.380 26486 26486 E : ====== dialButton pressed ======
05-27 17:08:54.620 26508 26551 D libpjsip: 17:08:54.621 icetp00 !ICE session created, comp_cnt=2, role is Controlling agent
05-27 17:08:54.620 26508 26551 D libpjsip: INVITE sip:[email protected] SIP/2.0
Without ICE, or with Android 4.0, there is no delay.
htc826_ice_delay_5.txt

[feature request] broadcast intents for use in tasker

I'm a heavy Tasker user and I really miss the interaction.
Is it possible to broadcast intents when things happen? Like: incoming call, call ended, new message etc.

Thank you.

Issues with dialog asking to call by csipsimple or mobile network

Hi,

I am having an issue with CSipSimple... well in fact with the dialog asking how to do a call when CSipSimple is active.

If I dial a number with CSipSimple active and integrated with the native dialer, when I dial a number the phone is expected to show me a dialog asking if I want to call with CSipSimple or with the Mobile Network.

However, this dialog is troublesome:

Sometimes it works just as it should (OK)
Sometimes it takes ages to come out (can this be because the corresponding code is swapped out by the memory manager of the phone?). This can delay the phone call very much (20-30 sec).
Sometimes the dialog is incorrect. The bottom part of the dialog that should allow to choose calling via the Mobile Networks incorrectly gets a VOIP icon or both a VOIP icon and text. In this case it is impossible to call via the mobile network.

I suggest verifying if this dialer integration can be made more reliable. Otherwise, it should probably be disabled.

[enhancement]add "unactivated" state for rules

at home i use my smartphone to replace my dect phone by using sip over wifi.
then
numbers beginning with +331 , +332 ..., +335
are directed to csip
numbers beginning with +336 are directed to GSM

but when i go outside i want :

sometime to keep the home rules
sometime to direct all numbers to GSM

according to this it is necessary to delete all the rules
and when at home back i must to create again the rules.

to avoid this i see 2 solutions :

add a state "unactivated for each rules" (there is only "modify" and "delete" today)
add the ability to create usage profiles . in my case i would create a "home" profile and an "outside" profile then i would switch to one or the other .

thanks

note : i have just submitted this issue here https://code.google.com/p/csipsimple/issues/detail?id=2962 before to understand this old site is unactivated . sorry .

High permanent battery use after first CSipSimple call (Samsung devices, Android 4.4.2)

I use CSipSimple on several devices, and always with the exact same configuration.

Some configuration details: Version 1.02.03, Integrate with Android, Only for outgoing, UDP only, Only "Wifi for outgoing calls" is checked under Network.

I have the following observation:

On all devices, after a fresh boot, and CSipSimple installed but no call having been placed, power consumption is, say, 0.5% per hour (screen turned off, phone left alone untouched).
On e.g. an htc Desire X (Stock Android 4.1.1), this does not change after I place a call with CSipSimple and hang up again
But on a Samsung S4 Mini (Stock Android 4.4.2) the following happens 100% reproducibly: After placing a call with CSipSimple and hanging up again, the battery consumption of the phone rises ten-fold to about 5% per hour (phone again left alone in sleep mode and not touched). The issue seems to affect several Samsung models in particular.

When checking what causes the excessive battery drain under Android's Settings/Battery, it is not attributed to "CSipSimple", but actually to "Android OS", where you can see that the CPU constantly remains in "Awake" state after the first call with CSIPSimple. ("CPU Awake" time indicator constantly going up if checked periodically).

Interestingly: I have found out that briefly starting Android's built-in Music player, then stopping and closing it again, again allows the CPU to go back to deep sleep when the screen is turned off, and the excessive battery drain initiated by CSIPsimple is stopped again (until another cal is placed using CSIpSimple).

So it looks like we may be looking at an issue with some sort of audio wakelock remaining active after a CSipSimple call, particularly on Samsung devices. I am wondering if there could be a fix or workaround implemented in CSipSimple?