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Esben Stien | 1 Apr 2006 16:22
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A Burst of White Noise (linphone-CVS-20060328)

I got some problems with the latest CVS. After some minutes, a very
loud sustaining white noise signal is fed out of linphone and only
thing I can do, is hang up and redial. The meter in ardour reads over
700dB above FS.

The other end hears nothing. 

Any pointers on what I can try?. Anyone seen this problem?

--

-- 
Esben Stien is b0ef <at> e     s      a             
         http://www. s     t    n m
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Esben Stien | 1 Apr 2006 19:07
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Re: A Burst of White Noise (linphone-CVS-20060328)

Esben Stien <b0ef <at> esben-stien.name> writes:

> Anyone seen this problem?

On some calls, this happens after some minutes, while other calls can
last maybe 15 minutes before it starts.

I'm registering linphone with asterisk and I use other softphones,
like iaxcomm, kphone with no similar problems.

--

-- 
Esben Stien is b0ef <at> e     s      a             
         http://www. s     t    n m
          irc://irc.  b  -  i  .   e/%23contact
          [sip|iax]:   e     e 
           jid:b0ef <at>     n     n
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Simon Morlat | 4 Apr 2006 10:23

Re: Linephone(c) doesn't recognise end of call

Can you execute linphonec -d 6 and send the debug messages ?

SImon

Le Lundi 27 Mars 2006 18:42, Conrad a écrit :
> Hi,
>
> could anyone please help me with that:
>
> I run linphone 1.3.1 (the command line version to be precise) on a Kanotix
> box connected to an Asterisk 1.2.1.
>
> The purpose is for an outbound call center to start the phone call from the
> application (hence the command line tool)
>
> The problem: everytime I call to an outside sip line and the remote
> endpoint hangs up, linphonec doesn't realize it. It doesn't end the call
> and keeps sending rtp packets.  Internal calls get signalized correctly
> though. On the other hand tests with other softphones (Ekiga and twinkle)
> showed that Asterisk is indeed signalizing the end of a call by remote
> party hangup.
>
> What could be wrong?
>
> Thank you for help
>
> Conrad
>
>
>
(Continue reading)

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Simon Morlat | 4 Apr 2006 10:25

Re: alsa_read_desc Undeclared

It is fixed from in cvs. Run make clean before.

Le Jeudi 23 Mars 2006 23:17, Esben Stien a écrit :
> Trying to compile linphone-CVS-20060323, I get:
>
> In file included from mscommon.c:27:
> alldescs.h:17: error: syntax error before ')' token
> alldescs.h:35: error: `alsa_read_desc' undeclared here (not in a function)
> alldescs.h:35: error: initializer element is not constant
> alldescs.h:35: error: (near initialization for `ms_filter_descs[14]')
> alldescs.h:35: error: syntax error before ')' token
> make[4]: *** [mscommon.lo] Error 1
> make[4]: Leaving directory
> `/home/b0ef/.inbound/linphone-cvs-20060323/linphone/mediastreamer2'
> make[3]: *** [all-recursive] Error 1
> make[3]: Leaving directory
> `/home/b0ef/.inbound/linphone-cvs-20060323/linphone/mediastreamer2'
> make[2]: *** [all] Error 2
> make[2]: Leaving directory
> `/home/b0ef/.inbound/linphone-cvs-20060323/linphone/mediastreamer2'
> make[1]: *** [all-recursive] Error 1
> make[1]: Leaving directory
> `/home/b0ef/.inbound/linphone-cvs-20060323/linphone' make: *** [all] Error
> 2
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Simon Morlat | 4 Apr 2006 11:01

Re: ortp - sound not clear


>
> I dont know if the problem is related to my gettimeofday function or my
> payload type. (What does BITRATE mean in payload structure?)
It is only indicative; and has no influence on the processing by oRTP.

Simon

>
> Please Advice.
>
>
>
> Regards,
> sriram
>
>
> (My gettimeofday and payload structure looks like this)
> int gettimeofday (struct timeval *tv, void* tz)
> {
>  union
>  {
>   __int64 ns100; /*time since 1 Jan 1601 in 100ns units */
>   FILETIME fileTime;
>  } now;
>     SYSTEMTIME SystemTime;
>     GetSystemTime(&SystemTime);
>  SystemTimeToFileTime(&SystemTime, &now.fileTime ) ;
>  tv->tv_usec = (long) ((now.ns100 / 10L) % 1000000L);
>  tv->tv_sec = (long) ((now.ns100 - 116444736000000000L) / 10000000L);
(Continue reading)

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Simon Morlat | 4 Apr 2006 11:19

Re: A Burst of White Noise (linphone-CVS-20060328)

Le Samedi 1 Avril 2006 16:22, Esben Stien a écrit :
> I got some problems with the latest CVS. After some minutes, a very
> loud sustaining white noise signal is fed out of linphone and only
> thing I can do, is hang up and redial. The meter in ardour reads over
> 700dB above FS.
>
> The other end hears nothing.
>
> Any pointers on what I can try?. Anyone seen this problem?

Do you see something when executing linphone --verbose ?
Please send a complete log so that we see what codec is being used.
Try with today's cvs because I commited some very big changes this morning and 
the bug may have disapear.

Simon
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Esben Stien | 4 Apr 2006 19:25
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Re: alsa_read_desc Undeclared

Simon Morlat <simon.morlat <at> linphone.org> writes:

> It is fixed from in cvs

I'm still getting this error with alldescs.h and the file looks the
same. I did a clean checkout.

--

-- 
Esben Stien is b0ef <at> e     s      a             
         http://www. s     t    n m
          irc://irc.  b  -  i  .   e/%23contact
          [sip|iax]:   e     e 
           jid:b0ef <at>     n     n
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Marco Wang | 6 Apr 2006 10:38
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Failed to build debug version of linphone 1.3.5 in KDevelop

Hello,

I was building linphone 1.3.5. The build process goes smoothly in command line mode. However, when I try to use KDevelop in debug mode, failure happens.

Step 1: First I import the existing linphone 1.3.5 project into KDevelop;
Step 2: The default project is "debug", with "--enable-debug=full"

The build process fails during configuring, because oRTP configure script only recognises "--enable-debug=yes" rather than "full";

Step 3: After fixing the oRTP script problem, it fails when compiling mediastreamer2/mscommon.cpp, included file "alldescs.h" doesn't exist.

But KDevelop can configure, make and install if there is no "--enable-debug=full" in project configuration.
My linux is Fedora 4.

Anyone can help me out?

Thanks,
Marco Wang
--
MarcoDotWang <at> gmail, skype, hotmail 

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Alexis Deltour | 7 Apr 2006 17:28
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linphonec with SIP proxy wengo

Dear,

I'm using linphonec (from linphone 1.3.4 version) and configuration file below :

[net]
con_type=4
use_nat=0

[sip]
sip_port=5060
guess_hostname=0
contact=root <at> 127.0.0.1
inc_timeout=15
use_info=0
use_ipv6=0
default_proxy=0

[rtp]
audio_rtp_port=7078
video_rtp_port=9078
audio_jitt_comp=60
video_jitt_comp=60

[sound]
playback_dev_id=0
ringer_dev_id=0
capture_dev_id=0
rec_lev=80
play_lev=80
ring_lev=80
source=m
local_ring=/usr/share/sounds/linphone/rings/oldphone.wav
remote_ring=/usr/share/sounds/linphone/ringback.wav

[video]
enabled=0
show_local=0

[audio_codec_0]
mime=PCMU
rate=8000
enabled=1

[audio_codec_1]
mime=GSM
rate=8000
enabled=1

[audio_codec_2]
mime=PCMA
rate=8000
enabled=1

[audio_codec_3]
mime=speex
rate=8000
enabled=1

[audio_codec_4]
mime=speex
rate=16000
enabled=1

[audio_codec_5]
mime=1015
rate=8000
enabled=1

[audio_codec_6]
mime=iLBC
rate=8000
enabled=1

[proxy_0]
reg_proxy=sip:voip.wengo.fr
reg_identity=sip:alexisdevel <at> voip.wengo.fr
reg_expires=600
reg_sendregister=1
publish=1
reg_route=<sip:213.91.9.210;lr>

[auth_info_0]
username=alexisdevel
passwd=****************
realm="voip.wengo.fr"



I start and register successfully (linphonec displays : registration on sip:voip.wengo.fr sucessful), but on a wengophone in another platform, alexisdevel is not connected ...

I manage to receive a call from the wengophone. Linphonec ring with an anonymous caller ...

But I don't manage to call the wengophone (with another user id) :

I've tried : call user, call sip:user, call sip:user <at> 213.91.9.210:5060, call sip:user <at> voip.wengo.fr:5060

Is my config file Ok ?? could you help me to resolve my problem

Faithfully

Alexis Deltour.

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Esben Stien | 7 Apr 2006 23:19
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Re: alsa_read_desc Undeclared

Esben Stien <b0ef <at> esben-stien.name> writes:

> I'm still getting this error with alldescs.h and the file looks the
> same. I did a clean checkout.

Did a clean checkout again today and still the same;). You might have
updated the wrong branch. 

--

-- 
Esben Stien is b0ef <at> e     s      a             
         http://www. s     t    n m
          irc://irc.  b  -  i  .   e/%23contact
          [sip|iax]:   e     e 
           jid:b0ef <at>     n     n
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