Zloty | 1 Dec 2005 09:12
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RE: Incoming sound problem


Hello,

I check on version 1.2.0-pre5 and 1.2.0-pre6 and problem still exists,
Actual I have win xp with sjphone and redhat with asterisk+linphone 
On redhat I don't have any firewall and ipforward is enabled.
I don't have only incoming sound, but I hear ringing on linphone.
Tcpdump show that packets are send from xp to redhat by eth0, and on
ifconfig I can see how counter is incrementing( for both interfaces eth and
lo). Then I think that xp communicate with asterisk by eth0 and it sent
packets to linphone by lo. But linphone doesn't got it.

I look into   Global statistics :
 number of rtp packet received=0
 number of rtp bytes received=0 bytes
Why, any sugestions? Maybe linphone don't listen on lo interface.

I can include log from asterisk.

Why in logs from linphone I see something like that
Warning:alsa_set_params: The rate 8000 Hz is not supported by your hardware.
 ==> Using 8000 Hz instead.
Is it the same, I think.

How I can reduce latency on linphone, when I check echo(asterisk) on the
same computer I got echo about 0,5s, ( I've PIII450, isn't enough).

Best regards
Robert

(Continue reading)

Zloty | 1 Dec 2005 09:24
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RE: Incoming sound problem


Hello,

I check on version 1.2.0-pre5 and 1.2.0-pre6 and problem still exists,
Actual I have win xp with sjphone and redhat with asterisk+linphone 
On redhat I don't have any firewall and ipforward is enabled.
I don't have only incoming sound, but I hear ringing on linphone.
Tcpdump show that packets are send from xp to redhat by eth0, and on
ifconfig I can see how counter is incrementing( for both interfaces eth and
lo). Then I think that xp communicate with asterisk by eth0 and it sent
packets to linphone by lo. But linphone doesn't got it.

I look into   Global statistics :
 number of rtp packet received=0
 number of rtp bytes received=0 bytes
Why, any sugestions? Maybe linphone don't listen on lo interface.

I also include log from asterisk.

Why in logs from linphone I see something like that
Warning:alsa_set_params: The rate 8000 Hz is not supported by your hardware.
 ==> Using 8000 Hz instead.
Is it the same, I think.

How I can reduce latency on linphone, when I check echo(asterisk) on the
same computer I got echo about 0,5s, ( I've PIII450, isn't enough).

Best regards
Robert

(Continue reading)

li jiuying | 1 Dec 2005 15:29
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RE: [Linphoe-users]can't here anything in the speaker

hello,
well, the sound driver is alsa, the switch is up and
playback levels is ok (80) and I installed linphone
1.2 pre6.
instead of using a headphone and microphone
seperately, now linphone uses ALSA device: Plantronics
Headset for  playback, capture and ring sound device.

IPBX: asterisk
SJphone sous windows
Xlite sous windows
Linphone sous linux
Kphone sous un autre linux
all the codecs are enabled in linphone, and all the
softphone are registrated in Asterisk

here is my test resulat of linphone with other
softphone:

linphone calls SJphone: 
rings: both side works
in SJphone: I can here what I said in Linphone
in Linphone: I can here what I said in SJphone but I
also hear of myself

SJphone calls linphone or linphone calls Xlite or
Xlite calls linphone:
rings: both side works
in SJphone/X-lite: I can here what I said in linphone
in linphone: can't here anything
(Continue reading)

Simon Morlat | 1 Dec 2005 17:24

linphone-1.2.0pre6

Hi,

I've just release of fresh tarball called 1.2.0pre6.
Download at http://simon.morlat.free.fr/downloads/unstable
It contains bugfixes, compatibily improvements, and video code cleaned. In 
previous version there were a confusion between H263 and H263-1998: now 
linphone supports and only supports H263-1998.
Support for video4linux webcams has been improved also.
Video support is still considered as experimental as compatibility with other 
H263-1998 devices is untested.

Thanks for your testing.

Simon
Mikhail Ramendik | 1 Dec 2005 18:38
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Re: linphone-1.2.0pre6

В сообщении от 1 декабря 2005 19:24 Simon Morlat написал(a):

> I've just release of fresh tarball called 1.2.0pre6.
> Download at http://simon.morlat.free.fr/downloads/unstable

404 not found :(

--

-- 
Yours, Mikhail Ramendik
David Schumacher | 1 Dec 2005 19:53
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AW: linphone-1.2.0pre6

http://simon.morlat.free.fr/download/unstable/source/

greetings from germany,
dave

-----Ursprungliche Nachricht-----
Von: linphone-users-bounces+dave=dgx.de <at> nongnu.org
[mailto:linphone-users-bounces+dave=dgx.de <at> nongnu.org]Im Auftrag von Mikhail
Ramendik
Gesendet: Donnerstag, 1. Dezember 2005 18:38
An: linphone-users <at> nongnu.org
Betreff: Re: [Linphone-users] linphone-1.2.0pre6

В сообщении от 1 декабря 2005 19:24 Simon Morlat написал(a):

> I've just release of fresh tarball called 1.2.0pre6.
> Download at http://simon.morlat.free.fr/downloads/unstable

404 not found :(

--
Yours, Mikhail Ramendik

_______________________________________________
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Linphone-users <at> nongnu.org
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Khurram Qureshi | 1 Dec 2005 21:01
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frame too large errors

Hi,
 
I'm trying to use linphone w/ speex and Asterisk and I keep on receiving the "frame too large" error.  Is there a setting that needs to be modified to enable linphone and speex to work well together?
 
Khurram.
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Linphone-users <at> nongnu.org
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Khurram Qureshi | 2 Dec 2005 02:23
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"must catchup" warning

Hi,
 
I'm new to linphone and I'm getting this "must catchup error" when I try to place a SIP call.  It's not a processor performance issue.  Any feedback/comments would be appreciated.
 
Khurram.
---------------
 
 
 
Warning:This remote sip phone did not answer properly to my sdp offer!

Warning:This remote sip phone did not answer properly to my sdp offer!

Warning:This remote sip phone did not answer properly to my sdp offer!

Warning:payload G729 is not usable

Warning:This remote sip phone did not answer properly to my sdp offer!

Warning:payload has no rtpmap.

Connected.

Message:ms_filter_add_link: OssRead,0 -> MULAWEncoder,0

Message:ms_filter_add_link: MULAWEncoder,0 -> RTPSend,0

Message:ms_filter_add_link: RTPRecv,0 -> MULAWDecoder,0

Message:ms_filter_add_link: MULAWDecoder,0 -> OssWrite,0 Message:Opening sound card [/dev/dsp (Open Sound System)] in capture mode with stereo=0,rate=8000,bits=16

Message:oss_open: bits = 16, stereo = 0, rate = 8000

Message:dsp blocksize is 512.

Message:sndcard.c: Opening sound card [/dev/dsp (Open Sound System)] in playback mode with stereo=0,rate=8000,bits=16 Message:CALL_STARTAUDIO

Warning:mstimer.c> Must catchup 58 miliseconds.

Warning:mstimer.c> Must catchup 70 miliseconds.

Warning:mstimer.c> Must catchup 82 miliseconds.

Warning:mstimer.c> Must catchup 94 miliseconds.

Warning:mstimer.c> Must catchup 107 miliseconds.

Warning:mstimer.c> Must catchup 119 miliseconds.

Warning:mstimer.c> Must catchup 131 miliseconds.

Warning:mstimer.c> Must catchup 143 miliseconds.

Warning:mstimer.c> Must catchup 155 miliseconds.

Warning:mstimer.c> Must catchup 167 miliseconds.

Warning:mstimer.c> Must catchup 179 miliseconds.

Warning:mstimer.c> Must catchup 191 miliseconds.

Warning:mstimer.c> Must catchup 203 miliseconds.

Warning:mstimer.c> Must catchup 215 miliseconds.

Warning:mstimer.c> Must catchup 227 miliseconds.

Warning:mstimer.c> Must catchup 239 miliseconds.

Warning:mstimer.c> Must catchup 251 miliseconds.

Warning:mstimer.c> Must catchup 263 miliseconds.

Warning:mstimer.c> Must catchup 275 miliseconds.

Warning:mstimer.c> Must catchup 287 miliseconds.

Warning:mstimer.c> Must catchup 299 miliseconds.

Warning:mstimer.c> Must catchup 311 miliseconds.

Warning:mstimer.c> Must catchup 323 miliseconds.

Warning:mstimer.c> Must catchup 335 miliseconds.

Warning:mstimer.c> Must catchup 347 miliseconds.

Warning:mstimer.c> Must catchup 359 miliseconds.

Warning:mstimer.c> Must catchup 371 miliseconds.

Warning:mstimer.c> Must catchup 382 miliseconds.

Warning:mstimer.c> Must catchup 394 miliseconds.

Warning:mstimer.c> Must catchup 406 miliseconds.

Warning:mstimer.c> Must catchup 418 miliseconds.

Warning:mstimer.c> Must catchup 430 miliseconds.

Warning:mstimer.c> Must catchup 442 miliseconds.

Warning:mstimer.c> Must catchup 454 miliseconds.

Warning:mstimer.c> Must catchup 466 miliseconds.

Warning:mstimer.c> Must catchup 479 miliseconds.

Warning:mstimer.c> Must catchup 491 miliseconds.

Warning:mstimer.c> Must catchup 503 miliseconds.

Warning:mstimer.c> Must catchup 515 miliseconds.

Warning:mstimer.c> Must catchup 527 miliseconds.

Warning:mstimer.c> Must catchup 539 miliseconds.

Warning:mstimer.c> Must catchup 551 miliseconds.

_______________________________________________
Linphone-users mailing list
Linphone-users <at> nongnu.org
http://lists.nongnu.org/mailman/listinfo/linphone-users
Doug Smith | 6 Dec 2005 05:25

trouble entering a password


To whom it may concern: 

This is Doug Smith.  I have recently joined the linphone-users list,
and I have to ask a question here, as I did before.  

I have registered on fwd.pulver.com for an account so that I can use
linphone as a tool in some upcoming work.  However, I have a problem.  

They say that the passwords on the voice mailboxes have to be at least
4 numbers long and cannot be anything but numbers.  Ok, I entered this
as I was instructed.  

Now, when I call sip:8502 <at> fwd.pulver.com, I am not able to get into my
voice mail.  It is really strange.  If you need me to run a debug log
on this I can, but the thing seems to involve understanding of
combinations of numbers.  

When I put in my password, I simply expect that the tones will be sent
out and the asterisk server on their end will be able to understand
what I am talking about.  I want to enter the password into my voice
mailbox.  

However, nothing seems to go out from linphonec.  I am blind and
cannot use the graphical version.  What happens is this: 

linphonec Cannot understand this. 

It seems that no numbers go out after I hit the return key.  Am I
possibly doing something wrong in entering the information that
linphonec cannot understand a simple line of numbers and generate the
appropriate touch tone sounds? 

Thank you for your assistance.  

--

-- 
Doug Smith: C.S.F.C.
Computer Scientist For CHRIST!

Oralux: http://oralux.org
Sybren Stuvel | 6 Dec 2005 11:02

Re: trouble entering a password

Doug Smith enlightened us with:
> Now, when I call sip:8502 <at> fwd.pulver.com, I am not able to get into
> my voice mail.

This might have something to do with how the DTMF signals are passed
to the server. There are roughly two options: RFC2833 and 'inband'.
'inband' is what your ordinary telephone does - it just sends out the
frequencies through the voice channel. This has one drawback, which is
that those frequencies will be mangled by the compression that's used.
At least, this happens with the GSM codec. To solve this, RFC2833
could be used.

> It seems that no numbers go out after I hit the return key.  Am I
> possibly doing something wrong in entering the information that
> linphonec cannot understand a simple line of numbers and generate
> the appropriate touch tone sounds? 

My version of linphonec (1.1.0) understands it just fine. Here is a
copy of my session:

$ linphonec
linphonec> call sip:s <at> unrealtower.org
Contacting  sip:s <at> unrealtower.org
linphonec> Connected.
linphonec> 1
linphonec> terminate
Communication ended.

the '1' got picked up, and transferred my call to the appropriate person.

Sybren
--

-- 
The problem with the world is stupidity. Not saying there should be a
capital punishment for stupidity, but why don't we just take the
safety labels off of everything and let the problem solve itself? 
                                             Frank Zappa

Gmane