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From: Mihai B. <mi...@ha...> - 2009-03-03 19:32:40
|
On Feb 21, 2009, at 3:31 PM, Steven Henke wrote: > Hello Everyone, > > I added the IAX2 specification's radio control messages to iaxclient > last year. Now for an amateur radio project I want to use the data > payload capability in those control message packets. There is a data > payload in the IAX2 spec (I am already using it on the Asterisk side) > and now I want to add it to the iaxclient side. > > It would be great to not create and maintain a branch for my own > purposes but I know changing the iaxclient.h API structure shakes the > world. > > Does anyone have any ideas before I jump in and make a mess of things. > > Thanks, > > Steven Henke, W9SH Hey Steve, If your modifications break the binary/source compatibility of the API, then yeah, it would be a big deal and you would probably need to maintain your own branch, at least for a while. But it will eventually make it into the next mainline version. Mihai |
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From: Jean-Denis G. <jd....@sy...> - 2009-02-25 06:01:08
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-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi Erik, Thanks for answering. Erik Bunce a écrit : > My client does attended transfers using iaxc_setup_call_transfer(source, > target) all the time. > > A few things: > 1) What version of asterisk are you connecting to? 1.4.22 > 2) What is the state of call #0? Did you quelch the call? Yes, my client automatically quelch the first call when a new call is made. > 3) What is the state of call #1? Has the far side answered the call? Yes, I talk to the other side for a few seconds, then try to transfer. > 3) Also, which call is selected at the time? The second call. > 4) What is the network configuration (clients and server)? My IAX2 client is MozPhone (moziax.mozdev.org). Call #0: SIP phone (snom) => Asterisk => MozPhone Call #1: MozPhone => Asterisk => SIP phone (gxp) What client are you using? I'd like to test if it works in my setup and find what I'm doing wrong in MozPhone. > > Thanks and Enjoy, > Erik > > Jean-Denis Girard wrote: > Hi list, > > I'm trying to add attended transfer to my iaxclient based application, > but I can't make it work. > > 1. Answer incoming call (#0) > 2. Make a second call (#1) > 3. Try to transfer using iaxc_setup_call_transfer(0,1) (I also tried > iaxc_setup_call_transfer(1,0), same result) > ... nothing happens:( > > The asterisk console shows TXREQ, TXCNT (repeated 3 times), then TXREJ. > > Has anybody used attended transfer? How is supposed to work? > > > Thanks, >> - ------------------------------------------------------------------------------ Open Source Business Conference (OSBC), March 24-25, 2009, San Francisco, CA - -OSBC tackles the biggest issue in open source: Open Sourcing the Enterprise - -Strategies to boost innovation and cut costs with open source participation - -Receive a $600 discount off the registration fee with the source code: SFAD http://p.sf.net/sfu/XcvMzF8H _______________________________________________ Iaxclient-devel mailing list Iax...@li... https://lists.sourceforge.net/lists/listinfo/iaxclient-devel > ------------------------------------------------------------------------------ > Open Source Business Conference (OSBC), March 24-25, 2009, San Francisco, CA > -OSBC tackles the biggest issue in open source: Open Sourcing the Enterprise > -Strategies to boost innovation and cut costs with open source participation > -Receive a $600 discount off the registration fee with the source code: SFAD > http://p.sf.net/sfu/XcvMzF8H > _______________________________________________ > Iaxclient-devel mailing list > Iax...@li... > https://lists.sourceforge.net/lists/listinfo/iaxclient-devel Thanks, - -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 50 10 40 / GSM: +689 79 75 27 -----BEGIN PGP SIGNATURE----- iEYEARECAAYFAkmk3ooACgkQuu7Rv+oOo/g4sgCbBkPp9opQomxC3gAzBvam9Eot f50AnRT2eAvXyPsX3rH6rFS7RHv2rC6q =S9zr -----END PGP SIGNATURE----- |
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From: Erik B. <kd...@bu...> - 2009-02-25 04:02:26
|
My client does attended transfers using iaxc_setup_call_transfer(source, target) all the time. A few things: 1) What version of asterisk are you connecting to? 2) What is the state of call #0? Did you quelch the call? 3) What is the state of call #1? Has the far side answered the call? 3) Also, which call is selected at the time? 4) What is the network configuration (clients and server)? Thanks and Enjoy, Erik Jean-Denis Girard wrote: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > Hi list, > > I'm trying to add attended transfer to my iaxclient based application, > but I can't make it work. > > 1. Answer incoming call (#0) > 2. Make a second call (#1) > 3. Try to transfer using iaxc_setup_call_transfer(0,1) (I also tried > iaxc_setup_call_transfer(1,0), same result) > ... nothing happens:( > > The asterisk console shows TXREQ, TXCNT (repeated 3 times), then TXREJ. > > Has anybody used attended transfer? How is supposed to work? > > > Thanks, > - -- > Jean-Denis Girard > > SysNux Systèmes Linux en Polynésie française > http://www.sysnux.pf/ Tél: +689 50 10 40 / GSM: +689 79 75 27 > -----BEGIN PGP SIGNATURE----- > > iEYEARECAAYFAkmiOqMACgkQuu7Rv+oOo/iYrgCZAcdc6sBmzx+REJ6vJoChJqlV > SlQAn1RdfUSrqQH00J9ytEQuD50/wsvf > =y8I5 > -----END PGP SIGNATURE----- > > ------------------------------------------------------------------------------ > Open Source Business Conference (OSBC), March 24-25, 2009, San Francisco, CA > -OSBC tackles the biggest issue in open source: Open Sourcing the Enterprise > -Strategies to boost innovation and cut costs with open source participation > -Receive a $600 discount off the registration fee with the source code: SFAD > http://p.sf.net/sfu/XcvMzF8H > _______________________________________________ > Iaxclient-devel mailing list > Iax...@li... > https://lists.sourceforge.net/lists/listinfo/iaxclient-devel > |
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From: Szalai G. <po...@gm...> - 2009-02-24 17:20:10
|
Dear Developers! I'm a Hungarian student, and I want to build up my own SoftPhone in C# based on an Asterisk in my home. To build up the whole application was quite simple, but one step is currently missing. To transfer calls. Actually I can transfer the call, but on other SoftPhone I can't answer it. But only in my applications based on IaxClient (I use SecondSignal). With HardPhones and other SoftPhones like Zoiper it works very well. I use one line clients. I transfer call number 0 and I try to pick up call number 0 on the other phone. I have tried both blind transfer call, and send DTMF (#[extension]) The application doesn't send any command to the Asterisk such as the CallNo is not good. Do you have any idea how can I blind pick up a transfered call? Thanks in advance -- Gergely |
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From: Jean-Denis G. <jd....@sy...> - 2009-02-23 06:14:06
|
-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi list, I'm trying to add attended transfer to my iaxclient based application, but I can't make it work. 1. Answer incoming call (#0) 2. Make a second call (#1) 3. Try to transfer using iaxc_setup_call_transfer(0,1) (I also tried iaxc_setup_call_transfer(1,0), same result) ... nothing happens:( The asterisk console shows TXREQ, TXCNT (repeated 3 times), then TXREJ. Has anybody used attended transfer? How is supposed to work? Thanks, - -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 50 10 40 / GSM: +689 79 75 27 -----BEGIN PGP SIGNATURE----- iEYEARECAAYFAkmiOqMACgkQuu7Rv+oOo/iYrgCZAcdc6sBmzx+REJ6vJoChJqlV SlQAn1RdfUSrqQH00J9ytEQuD50/wsvf =y8I5 -----END PGP SIGNATURE----- |
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From: Steven H. <sp...@xe...> - 2009-02-21 20:31:50
|
Hello Everyone, I added the IAX2 specification's radio control messages to iaxclient last year. Now for an amateur radio project I want to use the data payload capability in those control message packets. There is a data payload in the IAX2 spec (I am already using it on the Asterisk side) and now I want to add it to the iaxclient side. It would be great to not create and maintain a branch for my own purposes but I know changing the iaxclient.h API structure shakes the world. Does anyone have any ideas before I jump in and make a mess of things. Thanks, Steven Henke, W9SH |
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From: db p. <pe...@gm...> - 2009-02-09 01:07:25
|
We build a Voip client with Iaxclient(Jiaxclient+Iaxclient2.1beta1+Theora1.0+libvidcap0.21),when we established video call with resolution 640X480(bitrate:500000,frameRate:11,fragsize:1472) in both sides,the CPU is consumed about 100%,the workload of Asterisk server is good at that moment.This issue does not exist with the resolution 320X240(bitrate:204800,frameRate:11,fragsize:1500), the environment is as following, - CPU Interl Celeron 2.66GHz,RAM1GB, - Windows XP SP2 - Camera Logitech 4000 pro We convert the video as resolution 320X240 used for displaying as local user's profile video while capturing and sending video to remote user,the process of capturing video from camera send it to remote user and display it in remote user side is like the below, grab video(YUV format) -> Theora encode -> sent video ..... -> receive video -> Theora decode -> (video data with YUV format) -> Jiaxclient( show video picture). According to profile about Iaxclient and our analysis it is found what consumed most of the CPU resources are Theora encode/decode functions,we recompiled Theora lib with the argument "USE_ASM" and make it MMX instruction set,but the effect is not obvious. Is there anyone else met this issue too? Is there any suggestion or solution about the issue? |
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From: <jp...@vi...> - 2009-02-06 02:53:02
|
We build a Voip client with Iaxclient(Jiaxclient+Iaxclient2.1beta1+Theora1.0+libvidcap0.21),when we established video call with resolution 640X480(bitrate:500000,frameRate:11,fragsize:1472) in both sides,the CPU is consumed about 100%,the workload of Asterisk server is good at that moment.This issue does not exist with the resolution 320X240(bitrate:204800,frameRate:11,fragsize:1500), the environment is as following,
- CPU Interl Celeron 2.66GHz,RAM1GB,
- Windows XP SP2
- Camera Logitech 4000 pro
We convert the video as resolution 320X240 used for displaying as local user's profile video while capturing and sending video to remote user,the process of capturing video from camera send it to remote user and display it in remote user side is like the below,
grab video(YUV format) -> Theora encode -> sent video ..... -> receive video -> Theora decode -> (video data with YUV format) -> Jiaxclient( show video picture).
According to profile about Iaxclient and our analysis it is found what consumed most of the CPU resources are Theora encode/decode functions,we recompiled Theora lib with the argument "USE_ASM" and make it MMX instruction set,but the effect is not obvious.
Is there anyone else met this issue too? Is there any suggestion or solution about the issue?
|
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From: Shoutrain G. <sho...@ya...> - 2009-01-18 03:05:02
|
hi, does anybody have ideas about iaxclient with g729?
___________________________________________________________
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|
From: atuc <at...@gm...> - 2009-01-17 14:07:36
|
-------- Original-Nachricht --------
Betreff: Re: [Iaxclient-devel] Re; iaxclient on Linux ...
Datum: Sat, 17 Jan 2009 14:49:47 +0100
Von: Alexander Tuchacek <tuc...@kr...>
An: Gareth Bult <ga...@en...>
Referenzen: <11933176.342711232103683798.JavaMail.root@zimbra>
Gareth Bult schrieb:
> Hi,
>
> I've been developing with iaxclient on Linux and had a fair amount of success in general, but I'm now running up against the buffers. Linux (Ubuntu in particular) seem to be standardising on "PulseAudio", which generally works fine - when you start an Alsa or OSS application, it activates a plugin which connects to PluseAudio allowing different applications to output sound at the same time. Great!
>
> First problem I notice is that the ALSA option in portaudio doesn't work, apparently because it's trying to use a sample rate of 8000 when the minimum for alsa appears to be 44100. Anyway, this is bypassable by using OSS and /dev/dsp ...
>
> aoss <application> "looks" like it's going to work .. when I run the app and the app opens /dev/dsp, off it goes and the sound comes out nicely. However, when my iaxclient based app is outputting sound and I try to get another app to output sound - nothing happens! Conversely, if I'm playing a CD and I get an incoming call .. silence from my app! (the underlying code reports it was unable to open /dev/dsp)
>
> Checking other docs, apparently there is a specific issue with PortAudio which means it's about the only library out there which doesn't play ball with PulseAudio .. :-(
>
> It doesn't look like there's much going on with PortAudio any more, I've seen no end of recommendations that developers use *anything* else, and at the moment they don't even have a mailing list (!) , so are there any plans to move to OSS/ALSA/Pulse directly and remove PortAudio .. ? Alternatively are there any moves afoot to make PortAudio play nice with Pluse ??
>
> .. or (!) can anyone suggest how I might get IAX2 working in a Linux app such that I can get sound from both the app and the rest of the Linux application base to run at the same time ??
>
i use a the alsa dmix plugin for default, it forces samplerate convert
to 44100, pulse audio shoes it as "My Alsa DMIX Soundcard"
my .asoundrc:
best alex
> pcm.dmixer {
> type dmix
> ipc_key 1024
> ipc_perm 0666 # Andere Benutzer können ebenfalls dmix gleichzeitig nutzen
> slave.pcm "snd_card"
> slave {
> pcm "hw:0,0"
> # buffer_size kann bei Problemen der jeweiligen Karte angepasst werden.
> period_time 0
> period_size 128
> buffer_size 8192
> # bei Störungen kann die Konvertierung auf die Rate 44100 eingeschaltet werden.
> rate 44100
> # einige Soundkarten benötigen das exakte Datenformat (zB ice1712)
> # format S32_LE
> }
> bindings {
> 0 0
> 1 1
> }
> }
>
> # Das dsnoop-Plugin, welches es erlaubt, mehrere Programme gleichzeitig aufnehmen zu lassen.
> pcm.dsnooper {
> type dsnoop
> ipc_key 2048
> ipc_perm 0666
> slave.pcm "snd_card"
> slave
> {
> pcm "hw:0,0"
> period_time 0
> period_size 128
> buffer_size 8192
> # bei Störungen kann die Konvertierung auf die Rate 44100 eingeschaltet werden.
> rate 44100
> # einige Soundkarten benötigen das exakte Datenformat (zB ice1712)
> # format S32_LE
> format S16_LE
> }
> bindings {
> 0 0
> 1 1
> }
> }
>
> # Dies definiert unser Fullduplex-Plugin als Standard für alle ALSA-Programme.
> pcm.duplex {
> type asym
> playback.pcm "dmixer"
> capture.pcm "dsnooper"
> hint {
> show on
> description "My Alsa DMIX Soundcard "
> }
> }
>
> pcm.!default {
> type plug
> slave.pcm "duplex"
> }
>
>
|
|
From: Gareth B. <ga...@en...> - 2009-01-16 10:59:57
|
Hi, I've been developing with iaxclient on Linux and had a fair amount of success in general, but I'm now running up against the buffers. Linux (Ubuntu in particular) seem to be standardising on "PulseAudio", which generally works fine - when you start an Alsa or OSS application, it activates a plugin which connects to PluseAudio allowing different applications to output sound at the same time. Great! First problem I notice is that the ALSA option in portaudio doesn't work, apparently because it's trying to use a sample rate of 8000 when the minimum for alsa appears to be 44100. Anyway, this is bypassable by using OSS and /dev/dsp ... aoss <application> "looks" like it's going to work .. when I run the app and the app opens /dev/dsp, off it goes and the sound comes out nicely. However, when my iaxclient based app is outputting sound and I try to get another app to output sound - nothing happens! Conversely, if I'm playing a CD and I get an incoming call .. silence from my app! (the underlying code reports it was unable to open /dev/dsp) Checking other docs, apparently there is a specific issue with PortAudio which means it's about the only library out there which doesn't play ball with PulseAudio .. :-( It doesn't look like there's much going on with PortAudio any more, I've seen no end of recommendations that developers use *anything* else, and at the moment they don't even have a mailing list (!) , so are there any plans to move to OSS/ALSA/Pulse directly and remove PortAudio .. ? Alternatively are there any moves afoot to make PortAudio play nice with Pluse ?? .. or (!) can anyone suggest how I might get IAX2 working in a Linux app such that I can get sound from both the app and the rest of the Linux application base to run at the same time ?? tia Gareth. -- Managing Director, Encryptec Limited Tel: 0845 5082719, Mob: 0785 3305393 Email: ga...@en... Statements made are at all times subject to Encryptec's Terms and Conditions of Business, which are available upon request. |
|
From: Holger W. <wi...@df...> - 2009-01-14 14:33:24
|
Hi all, I solved this... I thagugt oggenc gives me a speex file - but I ahve to use speexenc for convertig the audio file instead of oggenc... Regards, HOlger Holger Wirtz wrote: > Hi, > > I am trying to use stresstest in a stripped down way. I have removed all > video calls because I only need audio capabilities. Now I am trying to > use a simple audio file to send it to app_conference for aome testing. > > The problem is that the audio in the conference is very much stuttering > and there are many long silent pasuses. I can guess that it is my > original aufio file but I don't know why this happens. > > Can anyone tell me what kind of aufio is needed for testing? > I used to create my testfile in this way: > > $ sox lighthouse.wav -c 1 -r 8000 lighthouse2.wav > $ oggenc lighthouse2.wav lighthouse.ogg > $ ogg123 lighthouse.ogg (just for testing) > $ file lighthouse.ogg > lighthouse.ogg: Ogg data, Speex audio > > What am I doing wrong? > > I also tried to compile the original stresstest program but I have > trouble to link it against theora (Debian testing and theora-trunk)... > > Thanks, Holger > -- +++NEUE ANSCHRIFT+++NEUE ANSCHRIFT+++NEUE ANSCHRIFT+++NEUE ANSCHRIFT+++ ##### #### ## ## Holger Wirtz Phone : (+49 30) 884299-40 ## ## ## ### ## DFN-Verein Fax : (+49 30) 884299-70 ## ## #### ###### Alexanderplatz 1 E-Mail: wi...@df... ## ## ## ## ### 10178 Berlin ##### ## ## ## GERMANY WWW : http://www.dfn.de GPG-Fingerprint: ABFA 1F51 DD8D 503C 85DC 0C51 E961 79E2 6685 9BCF |
|
From: Holger W. <wi...@df...> - 2009-01-13 15:26:21
|
Hi, I am trying to use stresstest in a stripped down way. I have removed all video calls because I only need audio capabilities. Now I am trying to use a simple audio file to send it to app_conference for aome testing. The problem is that the audio in the conference is very much stuttering and there are many long silent pasuses. I can guess that it is my original aufio file but I don't know why this happens. Can anyone tell me what kind of aufio is needed for testing? I used to create my testfile in this way: $ sox lighthouse.wav -c 1 -r 8000 lighthouse2.wav $ oggenc lighthouse2.wav lighthouse.ogg $ ogg123 lighthouse.ogg (just for testing) $ file lighthouse.ogg lighthouse.ogg: Ogg data, Speex audio What am I doing wrong? I also tried to compile the original stresstest program but I have trouble to link it against theora (Debian testing and theora-trunk)... Thanks, Holger -- +++NEUE ANSCHRIFT+++NEUE ANSCHRIFT+++NEUE ANSCHRIFT+++NEUE ANSCHRIFT+++ ##### #### ## ## Holger Wirtz Phone : (+49 30) 884299-40 ## ## ## ### ## DFN-Verein Fax : (+49 30) 884299-70 ## ## #### ###### Alexanderplatz 1 E-Mail: wi...@df... ## ## ## ## ### 10178 Berlin ##### ## ## ## GERMANY WWW : http://www.dfn.de GPG-Fingerprint: ABFA 1F51 DD8D 503C 85DC 0C51 E961 79E2 6685 9BCF |
|
From: Andrea S. <si...@op...> - 2008-12-23 13:20:04
|
Andrea Suisani wrote: > Hi all, > > as far as I can see from comment below (quoting iaxclient.h): > > /*! > Causes the audio channel for \a callNo to QUELCH (be squelched). > \param callNo The number of the active, accepted call to quelch. > \param MOH If non-zero Music On Hold should be played on the QUELCH'd call. > */ > EXPORT int iaxc_quelch(int callNo, int MOH); [cut] Maybe code is worth thousand words ;) attached a patch that actually fit the description quoted happy Christmas |
|
From: bugra H. <bug...@gm...> - 2008-12-22 14:18:36
|
2008/12/19 Mihai Balea <mi...@ha...> > Isn't pa_callback the main event callback that you were referring? > > > Nope, you shouldn't touch pa_* functions, unless you know what you're > doing. > I'm talking about the callback you set using iaxc_set_event_callback() > I get the data from event.ev.audio.data when an IAXC_EVENT_AUDIO event triggers. The problem is, audio event triggers all the time,even when nobody is calling me :S I observed that such event's timestamp is 0 so i created a filter based on timestamp's value. when timestamp is 0, i ignore that event. is this the right approach? btw why do i get such events when nobody is calling me? is this a bug? > IAXC_AUDIO_PREF_*_ENCODED will get you encoded audio data (obviously). > Make sure your player knows how to decode it and also, make sure you save > it in a container format that your player understands. You would probably > be better of just saving raw instead of encoded data. > I think i figured out what the problem is. I wasn't writing any header information to output audio file so media players were thinking file was corrupted. I will try to insert header information manually > Btw i couldn't find the test mode that you were referring. Can you please > give me the full name of it? > > Use your editor's Find function and look for "test mode" in iaxclient.h. > > Mihai > I already searched 'test mode' with no results. Uh oh, wait a minute, you are talking about 2.1 version. sorry, my bad. i will search newer versions before asking such a dump question :D |
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From: Mihai B. <mi...@ha...> - 2008-12-19 14:59:15
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Well, if you dig through iaxclient.h, you'll notice a set of flags starting with IAXC_AUDIO_PREF_ They can be used to receive incoming and outgoing audio data in both raw and encoded format. The data is received through the main event callback. You will have to write your own code that saves the data to a file. If you want your client to just save audio to file and not actually use your audio hardware, you might want to look at test mode (again in iaxclient.h) The old audio_file stuff is deprecated and as far as I'm concerned, unsupported. You're on your own there. Mihai On Dec 19, 2008, at 9:33 AM, bugra HASBEK wrote: > hi guys, > > I want to build an application which auto-answers and records calls > to a file. I googled it but failed to find a proper solution. it > seems like there was an audio_file.c in the older version which can > be used by initializing iax with 'audio_internal_file' argument. In > the newer releases audio_file.c is missing so i downloaded > audio_file.c from an older release and add it to makefile. I also > edited iaxclient_lib.c to use audio_file.c instead of portaudio: > > -------------------- > if(file_initialize(&audio_driver, 8000) ) > // if(pa_initialize(&audio_driver, 8000) ) > --------------------- > > The problem is, i can't play the output file. I tried amarok, vlc, > wmp but they all refused to play the output file :(( > > Here is a self compilable sample of my program: http://pastebin.com/f2fcf5b60 > Here is audio_file.c: http://pastebin.com/f7e5ed3e7 > > My question is what is best way of recording a call and what is > wrong with my code? Any suggestion is appreciated.... > > btw i installed 2.0.2 version, there were build errors for tkphone > but i managed to bypass them. > > > > > > > > > > > > > > > > > > > > > > > > > > ------------------------------------------------------------------------------ > _______________________________________________ > Iaxclient-devel mailing list > Iax...@li... > https://lists.sourceforge.net/lists/listinfo/iaxclient-devel |
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From: bugra H. <bug...@gm...> - 2008-12-19 14:33:59
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hi guys,
I want to build an application which auto-answers and records calls to a
file. I googled it but failed to find a proper solution. it seems like there
was an audio_file.c in the older version which can be used by initializing
iax with 'audio_internal_file' argument. In the newer releases audio_file.c
is missing so i downloaded audio_file.c from an older release and add it to
makefile. I also edited iaxclient_lib.c to use audio_file.c instead of
portaudio:
--------------------
if(file_initialize(&audio_driver, 8000) )
// if(pa_initialize(&audio_driver, 8000) )
---------------------
The problem is, i can't play the output file. I tried amarok, vlc, wmp but
they all refused to play the output file :((
Here is a self compilable sample of my program:
http://pastebin.com/f2fcf5b60
Here is audio_file.c: http://pastebin.com/f7e5ed3e7
My question is what is best way of recording a call and what is wrong with
my code? Any suggestion is appreciated....
btw i installed 2.0.2 version, there were build errors for tkphone but i
managed to bypass them.
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From: Mihai B. <mi...@ha...> - 2008-12-16 14:18:49
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Nope, sorry, jiaxclient is not maintained by us. I suggest you contact the authors Mihai On Dec 16, 2008, at 8:41 AM, talat demir wrote: > Hi, i want to remove number buttons from applet. But i can not > remove that there is a compiler problem. I use eclipse ide on ubuntu > 8.04. Can you help me, please? > > |
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From: talat d. <com...@ho...> - 2008-12-16 13:41:50
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Hi, i want to remove number buttons from applet. But i can not remove that there is a compiler problem. I use eclipse ide on ubuntu 8.04. Can you help me, please? _________________________________________________________________ Windows Live Messenger'ın için Ücretsiz 30 İfadeyi yükle http://www.livemessenger-emoticons.com/funfamily/tr-tr/ |
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From: Andrea S. <si...@op...> - 2008-12-15 13:55:21
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Andrea Suisani wrote: > Hi all, > [cut] > send_command(session, AST_FRAME_IAX, IAX_COMMAND_QUELCH, 0, ied.buf, ied.pos); > > is called, whereas it seems to me that in case of MOH == 0 we should use > > send_command(session, AST_FRAME_IAX, IAX_COMMAND_QUELCH, 0, NULL, 0); > > is it correct? am I missing something obvious? (maybe yes:) > > or should be better to check for MOH value in iaxc_quelch (iaclient_lib.c) > and call iax_quelch_moh if (!MOH) or iax_quelch otherwise? maybe It was clear since the beginning but I feel worthy to say that while re-reading my own message I realized I missed to define my main goal: being able to put a call on hold without playing music on the other end of the channel (i.e. callee) Andrea |
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From: Andrea S. <si...@op...> - 2008-12-12 14:30:24
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Hi all,
as far as I can see from comment below (quoting iaxclient.h):
/*!
Causes the audio channel for \a callNo to QUELCH (be squelched).
\param callNo The number of the active, accepted call to quelch.
\param MOH If non-zero Music On Hold should be played on the QUELCH'd call.
*/
EXPORT int iaxc_quelch(int callNo, int MOH);
iaxc_quelch should do different things depending on MOH value, but going
one level down I discovered that iaxc_quelch is a wrapper around
int iax_quelch_moh(session, MOH) that is a libiax2 function.
but looking at the code it seems to me that despite the check for
non-zero value of MOH (and setting of session->transfer_moh to 1 if true)
in every case
send_command(session, AST_FRAME_IAX, IAX_COMMAND_QUELCH, 0, ied.buf, ied.pos);
is called, whereas it seems to me that in case of MOH == 0 we should use
send_command(session, AST_FRAME_IAX, IAX_COMMAND_QUELCH, 0, NULL, 0);
is it correct? am I missing something obvious? (maybe yes:)
or should be better to check for MOH value in iaxc_quelch (iaclient_lib.c)
and call iax_quelch_moh if (!MOH) or iax_quelch otherwise?
Andrea
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From: Mihai B. <mi...@ha...> - 2008-12-12 13:57:25
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On Dec 11, 2008, at 8:47 PM, Da Beave wrote: > > I was going to build iax2slin and got the following error: > > iax2slin.c:(.text+0x173): undefined reference to `iaxc_set_files' > iax2slin.c:(.text+0x188): undefined reference to `iaxc_set_files' > > Has the name of this function changed? I'm working with > iaxclient-2.1beta3. I'm trying to get a handle on recording IAX2 > calls. Thanks! If memory serves, that function was deprecated a long time ago. If you want to record audio, take a look at the IAXC_AUDIO_PREF_* set of flags. These flags control what kind of audio data is returned to the main application via callbacks. Mihai |
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From: Da B. <be...@bu...> - 2008-12-12 01:50:18
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I was going to build iax2slin and got the following error: iax2slin.c:(.text+0x173): undefined reference to `iaxc_set_files' iax2slin.c:(.text+0x188): undefined reference to `iaxc_set_files' Has the name of this function changed? I'm working with iaxclient-2.1beta3. I'm trying to get a handle on recording IAX2 calls. Thanks! -- Da Beave (be...@vi...) Key Id: 357B86AD Key fingerprint = 1470 658D 16BB A129 462E 84BF 6C53 8FDD 357B 86AD If it wasn't for C, we'd be using BASI, PASAL and OBOL. |
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From: Da B. <be...@bu...> - 2008-12-10 14:47:46
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I asked this question on the iaxclient mailing list a _long long_ time ago and figured I'd throw it out there again. I'm interested in writing a utility that will analyze the in-band audio of a VoIP call with IAXClient. A simple example might be analyzing the audio for DTMF using the Goertzel DTMF algorithm. It wouldn't be limited to DTMF, but other audio tones/signals as well. Does anyone know of a way to get to the back end audio/data for analyzing? All calls would be via U-LAW. Thanks... - Beave |
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From: Da B. <be...@vi...> - 2008-12-10 14:37:21
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I asked this question on the iaxclient mailing list a _long long_ time ago and figured I'd throw it out there again. I'm interested in writing a utility that will analyze the in-band audio of a VoIP call with IAXClient. A simple example might be analyzing the audio for DTMF using the Goertzel DTMF algorithm. It wouldn't be limited to DTMF, but other audio tones/signals as well. Does anyone know of a way to get to the back end audio/data for analyzing? All calls would be via U-LAW. Thanks... - Beave Key Id: 357B86AD Key fingerprint = 1470 658D 16BB A129 462E 84BF 6C53 8FDD 357B 86AD If it wasn't for C, we'd be using BASI, PASAL and OBOL. |
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