Google Chrome Extension Softphone Update via Dev Dashboard Issue
google-chrome,google-chrome-extension,voip,phone
An update may be downloaded at any time, but an extension is only reloaded when the background page goes inactive. I guess that you're using Web sockets for signaling. Since the connection need to be available at all times, you are probably using a persistent background page. This is enough...
ambiguity in G729 Annexes
From RFC4856 Section 2.1.9 annexb: indicates that Annex B, voice activity detection, is used or preferred. Permissible values are "yes" and "no" (without the quotes); "yes" is implied if this parameter is omitted. More details can be found From RFC3551 Section 4.5.6 (G.729, G.729a and G.729b) and 4.5.7 (G.729d and...
How to pass data between 2 background apps
I decided to use 2 voip apps and communicate via sockets. Function: SBSetApplicationBadgeNumber(mach_port_t* port, char* appID, int number); is not available on ios7 and I didn't find new api similar to this. Darwin notification center & UIPasteboard works perfect when UiApp is in foreground, but when app is suspened Darwin...
Asterisk / Freepbx / Call doesn't disconnects after hangup
BYE message authentification is not very hardly used extension(actually it not needed, becuase BYE always have session id and ip/port same as before). There was no such feature in asterisk, becuase on that time almost no softswitches or devices have that implemented. After some moment BYE auth was added to...
TOS to 802.11 QOS Control on iPhone
You are looking at two different values - one is the layer 1 value and the other is a layer 2 value. There is no direct correspondence between the two - some mapping needs to be provided. In your case this mapping must be performed by the Access Point (or...
Asterisk Instant Messaging Errors
asterisk,sip,voip,rtp,instant-messaging
Looks like your softphone use INVITE instead of MESSAGE for messaging. You can get more info by enable sip debug in asterisk console asterisk -r sip set debug on ...
Create a UDP ping packet to fetch mumble(murmur) status
This should work to generate your ping packet: def ping(identifier) v = identifier a = [] while v > 256 # extract bytes from the identifier a << v % 256 v = v / 256 end a << v % 256 prefix = [0] * (8-a.length) # pad the...
When packet drop occurs in a link with VoIP and TCP working concurrently?
networking,tcp,udp,voip,packet-loss
A couple of comments first: VoIP will not usually 'saturate' the receiver (or the network) - it will simply send as many packets as it needs for the particular codec you are using. In other words it won't just keep growing until it fills the network. VoIP systems are sensitive...
Incoming cellular call interrupting a VoIP call in my iOS App
As this wiki: http://trac.pjsip.org/repos/wiki/Getting-Started/iPhone?format=pdf of pjsip explains, with iOS7 onwards pjsua is using high level APIs of AVAudioSession to manage opening and closing of sound streams which doesn't allow the older methods of (automatically) reconnecting your media streams after GSM call (or any other sound) interruptions. So to make it...
Voice quality issue in Android VoIP app with PJSIP
This issue is not probably due to PJSIP or multi threads in Android. Actually the TLS tunnel we use to connect the mobile and the server is on TCP which is a bad choice for sending RTP data. However, I don't have any idea how iOS and Mac manages to...
Server side requirements for a SIP VoIP app
You need to figure out your use-cases to decide. A SIP server is like an HTTP server, it will analyse the request URI, the request headers and whatever hints it can see to execute some resource at the backend. Think if you plan to have a user database and authentication....
PRI Trunks on IAD
sip,voip,pbx,telecommunication
I provide a brief overview of an IAD below that will help clarify the answers to the questions posted. Should the IAD be always a 48-port IAD or could it be a 24-port IAD? Hardware-wise, the provider will provide a Voice handoff(Analog or Digital, PBX or Analog phone) and this...
Call Hung up listener for asterisk
Yes it is posible by listen via AMI manager interface and waiting for Hangup You can use for example perl Asterisk::AMI class for that(example have on cpan for that class)...
Asterisk Integration with Symfony2 application
symfony2,asterisk,voip,telephony,asteriskami
I have used Asterisk in a Symfony Application. How We've done it: custom class for connecting to AMI and executing needed commands, like originate a call, whisper, hangup or listen add as second db your asterisk db if it is required, generate entities. The biggest problem for you will be...
Ozeki Voip SDK raises memory corrupt exception
The problem is from SDK and nothing can be done. I emailed their support and it seems that is a serious problem they can't solve. I have purchased 128 concurrent channel license and using Ver 11.2.0. No proper documentation is available and the SDK is not reliable. After working for...
Detect wheter a sound buffer is filled with data or not
c++,audio,voip,voice-recording
Firstly, I predict that the buffer hasn't been filled by the time you analyze it. Rather than a simple sleep, you should poll for WaveInHdr.dwFlags for the WHDR_DONE bit to be set. result = waveInStart(hWaveIn); if(result) { MessageBoxA(NULL, "Failed to start recording", NULL, MB_OK | MB_ICONEXCLAMATION); return; } // Wait...
Plivo call transfers
Transfer works in potentially two different ways: 1) End-point based In this mode: A transfer B to C by requesting (REFER) B to establish a new call to C. If Plivo is B, it needs to support REFER for this to work. A B C <=conversation=> ----REFER------> <-202 Accepted-- ----INVITE----->...
cli showing unknown sip registration in asterisk
Very likly that is automated programs(bots) with goal find your secrets/on success call out to Cuba or other costly destinations. Every asterisk installation need have at least rate limiter - usually fail2ban.org used. That will prevent bots from spamming too fast....
asterisk dialplan, forward call to gateway/pstn
Without having seen more of your dial plan, typically what you can do is, in an accessible context, a way to dial the whole thing -- and then use a substring to strip out parts of it. exten => _9XXXXXXXXXX,1,Dial(SIP/8003/${EXTEN:1}) The first part of the extension matches a 10-digit number...
Not able to connect incoming caller to th dialed calee in meetme function in asterisk
I think for this you should use option G from DIAL command: http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial G(context^exten^pri): If the call is answered, transfer both parties to the specified context and extension. The calling party is transferred to priority x, and the called party to priority x+1. This allows the dialplan to distinguish between...
When I return back to 1st Activity the onCreate() function of 2nd Activity is not called in 2nd time
Add the following method to your Dial activity: @Override protected void onPause() { finish(); } ...
Convert voip audio to text for debugging
yes, absolutely. The easiest would be to have a separate FreeSWITCH server that is used for placing the test calls and sending/receiving your test signals. tone_stream will generate the tones at frequencies that you need: https://freeswitch.org/confluence/display/FREESWITCH/Tone_stream tone_detect can detect the frequencies and execute actions, or even better, generate events that...
Freepbx can't connect to asterisk wrong password
You have do apply changes(red button) after each change You also have ensure your nat settings setuped correctly You can check that device added in sip_additional.conf file and in asterisk via asterisk -rx "sip show users" Again, if you use nat=no and your device after nat, it will not work....
Unable to find extension '' in context 'mycontext' while transferring call with option t in dial() in asterisk
call,asterisk,voip,transfer,ivr
By default asterisk search for the extension in the same context and it receives an empty extention. You can specify extension by pressing the extention after pressing # and in your dialplan you can route it to different context.For ex [from-pstn] exten=>_X.,1,Playback(hello-world) exten=>_X.,n,set(__GOTO_ON_BLINDXFR=mycontext^123456^1) exten=>_X.,n,Goto(mycontext,111,1) [mycontext] exten=>123,1,Goto(from-pstn) exten=>111,1,dial(DAHDI/g0/0${9xxxxxxxxx},,mtG(recordvoice,111,1)) [recordvoice]...
Getting CallSid from Twilio in ougoing calls using js
javascript,ruby-on-rails,ruby,twilio,voip
As seen in https://www.twilio.com/docs/client/connection, connection should be Twilio.Connection object which have a property named parameters with a CallSid attribute both for incoming and outgoing connections. It is probably only available after the connection is made, so you should use an event handler for connect event: Twilio.Device.connect(function(connection) { var callSid =...
FreeSWITCH - how can i make the call, talk to the person and then add him to the conference?
voip,freeswitch,telecommunication
See bgdial and dial in mod_conference documentation. What you are doing, looks quite OK, and needs debugging. But with conference dial you would have it easier....
Unable to hear ringing signal when calling out on a SIP trunk
Try adding the "R" parameter to your dialstring. Just as a side note, the person who configured your FreePBX should be hung. When dialing out to a trunk, putting the "Tt" parameters as part of your dial string is a nice hole for fraud. Also, if you have that in...
How to get all dialer events from Asterisk REST API (ARI)?
ARI uses a subscription based model for events. Quoting from the documentation on the wiki: Resources in Asterisk do not, by default, send events about themselves to a connected ARI application. In order to get events about resources, one of three things must occur: The resource must be a channel...
Configuring iOS VoIP application to run in sleep/background mode
ios,objective-c,sockets,voip,sleep-mode
There are tips for developing a VoIP App, reference from Apple A Voice over Internet Protocol (VoIP) app allows the user to make phone calls using an Internet connection instead of the device’s cellular service. Such an app needs to maintain a persistent network connection to its associated service so...
SIP UAC crash during a call
For the core SIP standard (RFC 3261) there is no requirement to check the other end of the call is still there and it's left up to implementors. In practise a user agent would typically detect there have been no RTP packets for a period (such as 60s) and hang...
Asterisk pre-emption and callers in a channel
You can read variables of any channel using SHARED(varname[,channel]) -= Info about function 'SHARED' =- [Synopsis] Gets or sets the shared variable specified. [Description] Implements a shared variable area, in which you may share variables between channels. The variables used in this space are separate from the general namespace of...
Implementing VOIP over udp, what is the approach to take when the player cannot cope up with received packet's speed
Since AudioTrack.write is blocking, you should make sure you have one thread for every channel you are writing to, so that write calls don't get stuck waiting for other channels to complete.
NAT Traversal - Probability of success using STUN
Depending on who your customers are, where they are, and the types of devices they use (PCs vs Mobile), the results can vary. In practice (based on my experience), ICE connectivity with STUN alone is about 85% successful for desktop and laptop PCs. But if it works once for a...
Twilio iOS unrecognized selector sent to Class
After reading all of Twilios documentation I discovered I was missing one thing. I had to add -Objc to Other Linker Flags and that solved my problem.
Asterisk: Connecting an Asterisk System To SIP Provider
Try changing the @gw1.sip.us to @myprovider and see if there's any changes. This will force asterisk to use the configured trunk [myprovider] instead of directly forward the call to gw.sip.us directly. Hope this helps....
What Request URL for Voice in TwiML App setup should I use when I develop on localhost?
Developing with a web service like Twilio can be difficult when it needs to make requests from a server to your development environment. One of the best tools I've found to make this easy is called ngrok. ngrok allows you to create tunnels to your machine and give them a...
Is it kosher to send RTP audio and DTMF events at the same time?
4733 says that in case of inband dtmf, you should send dtmf 'instead' of audio data. you increment seqNo and TS at the same rate but the payload would be dtmf data. If the remote UE supports playout of dtmf tones, it will playout the tone. else it will just...
View other conference call participants on SIP?
Yes, if your UAC and your conference provider supports RFC 4575: A Session Initiation Protocol (SIP) Event Package for Conference State In that case you would be able to establish a subscription (SUBSCRIBE) with the conference event package and (subject to appropriate permissions, authorizations, etc. at the discretion of the...
The voip service which don't care about ACD/ASR?
Many carriers have two routes: Conversational and Dialer. The dialer routes are intended for higher CPS/CPM and are tolerant of much lower ACD/ASR. Check the individual carrier to see exactly what they allow. Some that I'm familiar with: Voxbeam -- they call it "Call Center" Alcazar - they call it...
VOIP on my Android Application
android,sockets,networking,voip
I. "VoIP" is a very broad term, but, if your app transfers voice over IP network, it's definitely VoIP one, despite it may use totally proprietary protocols (as e.g. Skype does). II. VoIP stack is basically split to two meta-layers - 1) signaling and 2) media transport. Each of them...
Calling from 3rd party device (with “Videophone” capabilities) to Linphone(Desktop) displays video while calling Linphone(Browser) does not
This morning I posted the same question with the linphone developers mailing list. I received a reply, which when implemented solved my problem. Link to the mailing list with the reply. http://lists.gnu.org/archive/html/linphone-developers/2015-05/msg00086.html Basically, now that the plugin had support for H263, I had to enable the codec. Enabling the H263...
Playing text to speech inside read function in asterisk
asterisk,text-to-speech,voip,ivr,dtmf
You have record TTS to wav file(8khz 16 bit mono pcm uncompressed format REQUIRED) and use something like this [mycon] exten => _X.,1,Read(KEY,soundfile&/tmp/tts/tts-file&other-sound,,) You also can generate that file, save in variable and use that variable instead of file name. Also you can use app_swift application which do that for...
iOS Voip Application | AudioQueue | AVSession Category
ios,objective-c,voip,audioqueue
Thanks for looking at it , i realize the problem, this is my callback, void AudioStream::AQBufferCallback(void * inUserData, AudioQueueRef inAQ, AudioQueueBufferRef inCompleteAQBuffer) { AudioStream *THIS = (AudioStream *)inUserData; if (THIS->mIsDone) { return; } if ( !THIS->IsRunning()){ NSLog(@" AudioQueue is not running"); **return;** // Error part } int bytes = THIS->bufferByteSize;...
Securing RTP packets without encrypting each packets
security,encryption,sip,voip,rtp
I believe some of your concerns are addressed in the following IETF Spec - https://tools.ietf.org/html/rfc7201 - Options for Securing RTP Sessions But IMO, there is a cost of security w.r.t processing and thats a given for the enhanced layer of protection. I haven't come across any other fancier ways other...
How to make an iOS VoIP app obey Do Not Disturb when ringing?
As you say in your 3 options, only a UILocalNotification actually obeys silent/DND mode. The problems with it can be solved. Spamming the notification center: I think that works quite well. You can cancel your previous notification immediately before you fire off a new one, so there will always be...
Two pjsua application at the same time on Android
Set unique local ports (might be also described as bind ports in configuration) for these applications. For plain pjsua this is --local-port parameter.
How to call group using PJSIP
I have no experience to run PJSIP on iOS yet (may be there is some restrictions on call count in iOS version of PJSIP?). Based on my experience of using PJSIP on desktop, you should call all the parties with different calls to pjsua_call_make_call (execute pjsua_call_make_call 4 times for 4...
Forward a call from landline to did number
There are 2 posible variants 1) Ask your landline provider support about number portability(depend of country and provider) or forward number to sip. 2) Put on your line FXO gateway device(pstn2sip gate, $100-$200 cost) and setup it to work as did for asterisk....
How can I stream audio once SIP connection is established
Turns out, the easiest way to solve this was to install VoiceMeeter (http://vb-audio.pagesperso-orange.fr/Voicemeeter/index.htm) and then just play audio through the speaker as normal. Once I register my SIP and make an outgoing call, once someone picks up I can play my WAV straight through the speakers and the callee hears...
Attempt to create usable VoIP phone by Twilio in ASP.NET-MVC fails as Only XML is visible on the website instead of phone
asp.net-mvc,twilio,voip,asp.net-mvc-5.2
The Twilio TwiML generation doesn't need to go in your Razor View and instead you just use the controller. So you might have something like the following instead: public ActionResult PhoneMenu() { var response = new TwimlResponse().Say("Hello"): return new TwimlResult(response); } Or, as part of Twilio.Mvc there's an alternative controller...
How to Launch Windows Phone Foreground App from VoipHttpIncomingCallTask?
c#,windows-phone,scheduled-tasks,sip,voip
The best way to do this is to use the Native Chatterbox sample and implement the background handlers it has. These will automatically launch the App to the page you set in CallController.cpp line 390 CallController::CallController() : callInProgressPageUri(L"/YourAppProject;component/MainMenu/MainMenu.xaml?FromSipPush=true"), This will let the App launch after you answer the Sip Notification....
How to measure QoS parameters in the ns-3 simulation
simulation,voip,ns-3,qos,wimax
From my short experience, Generally you need to use the flowMonitor helper class to get the statistics. Refer below. http://www.nsnam.org/doxygen/classns3_1_1_flow_monitor_helper.html If it is just the basic statistics like bytes received, sent etc, you have already functions in the packet class that can be used to get the values. Else you...
How do wireless companies block mVoIP traffics? [closed]
There are various different ways that a service provider can do this - for example: monitor and limit RTP traffic (the actual speech, rather than the SIP signalling) monitor SIP protocol traffic and limit new connections being set up block traffic to and from known servers that support VoIP If...
Sip call drop after 30seconds [closed]
It's because 30 seconds is the timeout value for SIP transactions and it's probable that the ACK request, which completes a call INVITE transaction, is not getting through. As to why the ACK request is not getting through there are a number of possibilities but it's unlikely to be NAT....
How does a SIP based app scheme work?
Since your device not know where each other located(ip/port), they call sip server or proxy. Sip server match dialplan and send request changed(server) or unchanged(proxy) to other side. In INVITE request each peer send address/port and info about media stream RTP If that info unchanged(proxy) they can see each other...
Dialing a SIP number with TWINKLE client, from Ubuntu Linux Terminal. Is it possible?
If you check out the man page for twinkle ( man twinkle ), online version http://manpages.ubuntu.com/manpages/karmic/man1/twinkle.1.html you will see that there is both an option to specify the called party from the command line, and also a CLI interface When you run twinkle, you can add -call to dial a...
tc (netem) - reported loss is different to set loss
It turns out that wireshark was causing my problem, when a VoIP stream changed to, say the hold music, the stream was switched to the hold music from the server, and when the user was taken off hold the sequence numbers had jumped a few hundred making wireshark think there...
How can I prevent my iOS VoIP app from playing audio when the user switches their phone to silent?
I think I found a solution that works for a VoIP application. If the phone is set to silent, I wanted the phone to show an alert and vibrate instead of playing the audible ring. I can solve this by using local notifications and moving the sound out of the...