nucleo | 5 Sep 19:20 2011

Unused vars and returned values

Hi,

I have problems with building linphone-3.4.3 and 
mediastreamer-2.7.3 with --enable-strict option.
There are some errors with unused variables
and returned value.

In attachment my patches that fixes build for me.

--

-- 
Alexey Kurov <nucleo <at> fedoraproject.org>
Attachment (linphonecore.c.diff): text/x-patch, 564 bytes
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wael mashal | 6 Sep 09:47 2011
Picon

Cannot Register with linephone SIP (preloaded_ Route-header with that address make the problem)

Hi All

I have an SIP account from a provider in Sweden but i cant success the registration on linephone that i guess because the preload_router header (i note that in another software and when i remove this code line its work fine )  in the registration invite  is there any work around or i can remove this line from code ?  see the details below

see the details below
When using an outbound proxy, the SIP-packets should not contain a _preloaded_ Route-header with that address. Route: <sip:proxy name> the SIP-packets should not contain a _preloaded_ Route-header with that address. ..from the packets, for example initial INVITE from the client. Outbound proxy means that the client should send all initial packets to the outbound proxy address, but there should be no references of the outbound proxy address within the SIP-packets themselves. Don’t confuse this with normal routing headers that may be populated from Record-route headers (loose-routing) during the first stages of SIP-signaling. What I’m talking about is the initial _preloaded_ Route-header.

any idea , please advise ?


Best  Regards

Wael Mashal
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Simon Morlat | 6 Sep 10:13 2011

git repositories changes

Hi all,

This email is just to inform that we migrated our git repos to a new 
machine that runs faster.
git.linphone.org now points to 91.121.22.24 .

Regards,

Simon
Israel Gottlieb | 6 Sep 14:27 2011
Picon

linphone on the blackberry

Hi all
I have installed the linphone on my blackberry 9700 os6 device and have patched asterisk to support amr
when i try to call i get a msg early media and dont here anything
the phone rings and the second i answer the blackberry crashes and has to be restarted

how could i get the logs to send

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Beetge, Henry | 7 Sep 12:19 2011

Registering Linphone 3.4.3 to Cisco callmanager 5

HI,

 

I hope this is the correct place to ask a question, if not please steer me in the correct direction.

 

I would like to know if anybody is able to assist me in registering Linphone 3.4.3 to Cisco Callmanager 5.

 

Thanks in advance.

 

Regards

Henry Beetge
Software Developer


Element Six (Production) (Pty) Ltd
1 Debid Rd, Nuffield, Springs, 1559
Direct: +27 11 812 9384
Mobile: +27 79 770 5040

www.e6.com

 

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tincup87 | 8 Sep 01:46 2011
Picon

phone number

How does one get a phone number to use with linphone?

Is there a regular phone that can be used with linphone, even when the computer is turned off?
Edouard DULIEGE | 8 Sep 07:10 2011
Picon

Re: phone number

Hello,

A few things to read that may answer your questions :

http://www.linphone.org/eng/documentation/
https://www.linphone.org/eng/linphone/register-a-linphone-account.html

https://secure.wikimedia.org/wikipedia/en/wiki/VoIP_phone

 Cheers,

Edouard


2011/9/8 <tincup87 <at> safe-mail.net>
How does one get a phone number to use with linphone?

Is there a regular phone that can be used with linphone, even when the computer is turned off?

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--
Edouard Duliège
edouard.duliege <at> gmail.com

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Nikolai Cassanova | 8 Sep 12:03 2011
Picon

Re: phone number

Hi if you are talking about getting a regular phone number to use along with your sip address that you would receive at linphone website, then you would need a 'direct inward dialing' (DID) number, which is a virtual number.  This virtual number can be used in conjunction with your sip address that you receive from linphone.  By using a DID you allow others to just put in a regular phone number into there landline phone or cell phone and call you on that number, the number then gets transferred automatically to your sip address at linphone and then you are able to answer it.  Not only does a DID allow you to do that but also gives you the ability to let family members and friends in different countries see that you use that countries number based in a particular country, so say you live in England but have family in USA, you can get a DID number for USA, so that families and friends in USA can call that USA number and it will ring in England where your located.

Also to use voip with regular phone without computer check link others provided on mailing list and look up 'ata' which allows you to use voip without a computer.

On Sep 7, 2011 7:46 PM, <tincup87 <at> safe-mail.net> wrote:
> How does one get a phone number to use with linphone?
>
> Is there a regular phone that can be used with linphone, even when the computer is turned off?
>
> _______________________________________________
> Linphone-users mailing list
> Linphone-users <at> nongnu.org
> https://lists.nongnu.org/mailman/listinfo/linphone-users
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wael mashal | 8 Sep 12:04 2011
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Re: Cannot Register with linephone SIP (preloaded_ Route-header with that address make the problem)

Friends,

any updates ?

Thanks


On Tue, Sep 6, 2011 at 10:47 AM, wael mashal <wmashal <at> gmail.com> wrote:
Hi All

I have an SIP account from a provider in Sweden but i cant success the registration on linephone that i guess because the preload_router header (i note that in another software and when i remove this code line its work fine )  in the registration invite  is there any work around or i can remove this line from code ?  see the details below

see the details below
When using an outbound proxy, the SIP-packets should not contain a _preloaded_ Route-header with that address. Route: <sip:proxy name> the SIP-packets should not contain a _preloaded_ Route-header with that address. ..from the packets, for example initial INVITE from the client. Outbound proxy means that the client should send all initial packets to the outbound proxy address, but there should be no references of the outbound proxy address within the SIP-packets themselves. Don’t confuse this with normal routing headers that may be populated from Record-route headers (loose-routing) during the first stages of SIP-signaling. What I’m talking about is the initial _preloaded_ Route-header.

any idea , please advise ?


Best  Regards

Wael Mashal

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Steve Sakoman | 10 Sep 18:10 2011
Picon

Getting linphone working on OMAP3

I've built the latest linphone git head on my Overo OMAP3 machine.

On the surface things seem to be working -- I get a video window with
my ugly mug, the UI is functional, I can change ring tones and hear
the result.

Unfortunately the important part doesn't seem to work!  When a I
attempt a call to the Overo from Linphone on my laptop I don't get a
ring on the Overo.

Looking at the debug window I can see that Linphone actually does see
the call invitation, it just doesn't respond to it!

I've attached the debug window contents below.  Can anyone see an
obvious reason why this isn't working?  Could it be that my laptop is
running an older version of Linphone?

Thanks!

Steve

Debug output:

message: assigning PCMU/8000 payload type number 0
message: assigning GSM/8000 payload type number 3
message: assigning PCMA/8000 payload type number 8
message: assigning speex/8000 payload type number 110
message: assigning speex/16000 payload type number 111
message: assigning speex/32000 payload type number 112
message: assigning telephone-event/8000 payload type number 101
message: assigning H263/90000 payload type number 34
message: assigning theora/90000 payload type number 97
message: assigning H263-1998/90000 payload type number 98
message: assigning MP4V-ES/90000 payload type number 99
message: assigning H264/90000 payload type number 102
message: assigning VP8-DRAFT-0-3-2/90000 payload type number 103
message: assigning x-snow/90000 payload type number 96
message: assigning iLBC/8000 payload type number 100
message: assigning AMR/8000 payload type number 104
message: assigning 1015/8000 payload type number 105
message: assigning G726-16/8000 payload type number 106
message: assigning G726-24/8000 payload type number 107
message: assigning G726-32/8000 payload type number 108
message: assigning G726-40/8000 payload type number 109
message: assigning AAL2-G726-16/8000 payload type number 113
message: assigning AAL2-G726-24/8000 payload type number 114
message: assigning AAL2-G726-32/8000 payload type number 115
message: assigning AAL2-G726-40/8000 payload type number 116
message: Mediastreamer2 2.7.3 (git: 2.7.3-170-ga82dcdb) starting.
message: Registering all soundcard handlers
message: Card 'ALSA: default device' added
warning: Could not attach mixer to card: Invalid argument
message: Card 'ALSA: overo' added
warning: Could not attach mixer to card: Invalid argument
message: Card 'ALSA: VF0380 Live! Cam Optia Pro' added
message: Card 'OSS: /dev/dsp' added
message: Card 'OSS: /dev/dsp1' added
message: Registering all webcam handlers
message: Webcam V4L2: /dev/video0 added
message: Webcam StaticImage: Static picture added
message: Loading plugins
message: Cannot open directory /usr/local/lib/mediastreamer/plugins:
No such file or directory
message: ms_init() done
message: Cannot open directory /usr/local/lib/liblinphone/plugins: No
such file or directory
message: linphone_core_set_playback_gain_db(): no active call.
message: Adding new codec PCMU/8000 with fmtp
message: Adding new codec GSM/8000 with fmtp
message: Adding new codec PCMA/8000 with fmtp
message: Adding new codec G722/8000 with fmtp
message: Adding new codec L16/44100 with fmtp
message: Adding new codec L16/44100 with fmtp
message: Adding new codec speex/8000 with fmtp vbr=on
message: Adding new codec speex/16000 with fmtp vbr=on
message: Adding new codec speex/32000 with fmtp vbr=on
message: Adding new codec H263/90000 with fmtp
message: Adding new codec x-snow/90000 with fmtp
message: Adding new codec theora/90000 with fmtp
message: Adding new codec H263-1998/90000 with fmtp CIF=1;QCIF=1
message: Adding new codec MP4V-ES/90000 with fmtp profile-level-id=3
message: DNS resolution with 0.0.0.0:5060
message: getaddrinfo returned the following addresses:
message: 0.0.0.0 port 5060
message: eXosip: Reseting timer to 10s before waking up!
message: New local ip address is 192.168.0.12
message: Network state is now [UP]
message: DNS resolution with 0.0.0.0:5060
message: getaddrinfo returned the following addresses:
message: 0.0.0.0 port 5060
message: Setting video size 352x288
message: eXosip: Reseting timer to 10s before waking up!
message: Driver is uvcvideo
message: v4lv2: YUV420P choosen
message: Size of webcam delivered pictures is 352x288
message: ms_filter_link: MSV4L2Capture:0x19adc50,0-->MSPixConv:0x19bb480,0
message: ms_filter_link: MSPixConv:0x19bb480,0-->MSX11Video:0x19ad298,0
message: Window has size 352,288
message: msv4l2_thread starting
message: MS ticker priority set to SCHED_RR and max (99)
message: Driver is uvcvideo
message: v4lv2: YUV420P choosen
message: Size of webcam delivered pictures is 352x288
message: V4L2 video capture started.
message: Found output adaptor; name=OMAP XV adaptor num_ports=1, with formats:
message: type=YUV/Packed id=YUY2
message: type=YUV/Packed id=UYVY
message: type=YUV/Planar id=I420
message: type=YUV/Planar id=YV12
message: Grabbed port 62
message: Updating window decorations for preview
warning: Video MSTicker: We are late of 414 miliseconds.

<snip many keep alive and "Captured mean" messages of the same type>

warning: keep alive: 0
message: eXosip: Reseting timer to 10s before waking up!
message: Captured mean fps=24.338799, expected=29.969999
message: Captured mean fps=24.173447, expected=29.969999
message: Captured mean fps=24.480032, expected=29.969999
message: Received message:
INVITE sip:root <at> 192.168.0.12 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.17:5060;rport;branch=z9hG4bK65962445
From: Steve Sakoman <sip:sakoman <at> 192.168.0.17>;tag=2126056146
To: <sip:root <at> 192.168.0.12>
Call-ID: 1681834623
CSeq: 20 INVITE
Contact: <sip:sakoman <at> 192.168.0.17:5060>
Content-Type: application/sdp
Max-Forwards: 70
User-Agent: Linphone/3.2.1 (eXosip2/3.3.0)
Subject: Phone call
Content-Length:   316

v=0
o=sakoman 123456 654321 IN IP4 192.168.0.17
s=A conversation
c=IN IP4 192.168.0.17
t=0 0
m=audio 7078 RTP/AVP 111 110 0 3 8 101
a=rtpmap:111 speex/16000/1
a=rtpmap:110 speex/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:3 GSM/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11

message: Message received from: 192.168.0.17:5060
message: Message received from: 192.168.0.17:5060
message: MESSAGE REC. CALLID:1681834623
message: Message received from: 192.168.0.17:5060
message: This is a request
message: allocating transaction ressource 1 1681834623
message: allocating IST context
message: eXosip: Reseting timer to 10s before waking up!
message: eXosip: Reseting timer to 10s before waking up!
message: Captured mean fps=20.824905, expected=29.969999
message: Received message:
INVITE sip:root <at> 192.168.0.12 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.17:5060;rport;branch=z9hG4bK65962445
From: Steve Sakoman <sip:sakoman <at> 192.168.0.17>;tag=2126056146
To: <sip:root <at> 192.168.0.12>
Call-ID: 1681834623
CSeq: 20 INVITE
Contact: <sip:sakoman <at> 192.168.0.17:5060>
Content-Type: application/sdp
Max-Forwards: 70
User-Agent: Linphone/3.2.1 (eXosip2/3.3.0)
Subject: Phone call
Content-Length:   316

v=0
o=sakoman 123456 654321 IN IP4 192.168.0.17
s=A conversation
c=IN IP4 192.168.0.17
t=0 0
m=audio 7078 RTP/AVP 111 110 0 3 8 101
a=rtpmap:111 speex/16000/1
a=rtpmap:110 speex/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:3 GSM/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11

message: Message received from: 192.168.0.17:5060
message: Message received from: 192.168.0.17:5060
message: MESSAGE REC. CALLID:1681834623
message: Message received from: 192.168.0.17:5060
message: eXosip: Reseting timer to 10s before waking up!
message: Received message:
INVITE sip:root <at> 192.168.0.12 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.17:5060;rport;branch=z9hG4bK65962445
From: Steve Sakoman <sip:sakoman <at> 192.168.0.17>;tag=2126056146
To: <sip:root <at> 192.168.0.12>
Call-ID: 1681834623
CSeq: 20 INVITE
Contact: <sip:sakoman <at> 192.168.0.17:5060>
Content-Type: application/sdp
Max-Forwards: 70
User-Agent: Linphone/3.2.1 (eXosip2/3.3.0)
Subject: Phone call
Content-Length:   316

v=0
o=sakoman 123456 654321 IN IP4 192.168.0.17
s=A conversation
c=IN IP4 192.168.0.17
t=0 0
m=audio 7078 RTP/AVP 111 110 0 3 8 101
a=rtpmap:111 speex/16000/1
a=rtpmap:110 speex/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:3 GSM/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11

message: Message received from: 192.168.0.17:5060
message: Message received from: 192.168.0.17:5060
message: MESSAGE REC. CALLID:1681834623
message: Message received from: 192.168.0.17:5060
message: eXosip: Reseting timer to 10s before waking up!
message: Captured mean fps=19.772598, expected=29.969999
message: Received message:
INVITE sip:root <at> 192.168.0.12 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.17:5060;rport;branch=z9hG4bK65962445
From: Steve Sakoman <sip:sakoman <at> 192.168.0.17>;tag=2126056146
To: <sip:root <at> 192.168.0.12>
Call-ID: 1681834623
CSeq: 20 INVITE
Contact: <sip:sakoman <at> 192.168.0.17:5060>
Content-Type: application/sdp
Max-Forwards: 70
User-Agent: Linphone/3.2.1 (eXosip2/3.3.0)
Subject: Phone call
Content-Length:   316

v=0
o=sakoman 123456 654321 IN IP4 192.168.0.17
s=A conversation
c=IN IP4 192.168.0.17
t=0 0
m=audio 7078 RTP/AVP 111 110 0 3 8 101
a=rtpmap:111 speex/16000/1
a=rtpmap:110 speex/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:3 GSM/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11

message: Message received from: 192.168.0.17:5060
message: Message received from: 192.168.0.17:5060
message: MESSAGE REC. CALLID:1681834623
message: Message received from: 192.168.0.17:5060
message: eXosip: Reseting timer to 10s before waking up!
message: Captured mean fps=24.340832, expected=29.969999
message: Captured mean fps=20.463984, expected=29.969999
message: Captured mean fps=22.483994, expected=29.969999
message: Received message:
INVITE sip:root <at> 192.168.0.12 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.17:5060;rport;branch=z9hG4bK65962445
From: Steve Sakoman <sip:sakoman <at> 192.168.0.17>;tag=2126056146
To: <sip:root <at> 192.168.0.12>
Call-ID: 1681834623
CSeq: 20 INVITE
Contact: <sip:sakoman <at> 192.168.0.17:5060>
Content-Type: application/sdp
Max-Forwards: 70
User-Agent: Linphone/3.2.1 (eXosip2/3.3.0)
Subject: Phone call
Content-Length:   316

v=0
o=sakoman 123456 654321 IN IP4 192.168.0.17
s=A conversation
c=IN IP4 192.168.0.17
t=0 0
m=audio 7078 RTP/AVP 111 110 0 3 8 101
a=rtpmap:111 speex/16000/1
a=rtpmap:110 speex/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:3 GSM/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11

message: Message received from: 192.168.0.17:5060
message: Message received from: 192.168.0.17:5060
message: MESSAGE REC. CALLID:1681834623
message: Message received from: 192.168.0.17:5060
warning: keep alive: 4
message: eXosip: Reseting timer to 10s before waking up!
warning: Video MSTicker: We are late of 114 miliseconds.
warning: Video MSTicker: We are late of 78 miliseconds.
message: Captured mean fps=10.183525, expected=29.969999
message: Captured mean fps=21.783138, expected=29.969999
message: Captured mean fps=11.711856, expected=29.969999
message: Captured mean fps=16.569586, expected=29.969999
warning: Video MSTicker: We are late of 113 miliseconds.
message: Captured mean fps=17.414385, expected=29.969999
message: Received message:
INVITE sip:root <at> 192.168.0.12 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.17:5060;rport;branch=z9hG4bK65962445
From: Steve Sakoman <sip:sakoman <at> 192.168.0.17>;tag=2126056146
To: <sip:root <at> 192.168.0.12>
Call-ID: 1681834623
CSeq: 20 INVITE
Contact: <sip:sakoman <at> 192.168.0.17:5060>
Content-Type: application/sdp
Max-Forwards: 70
User-Agent: Linphone/3.2.1 (eXosip2/3.3.0)
Subject: Phone call
Content-Length:   316

v=0
o=sakoman 123456 654321 IN IP4 192.168.0.17
s=A conversation
c=IN IP4 192.168.0.17
t=0 0
m=audio 7078 RTP/AVP 111 110 0 3 8 101
a=rtpmap:111 speex/16000/1
a=rtpmap:110 speex/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:3 GSM/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11

message: Message received from: 192.168.0.17:5060
message: Message received from: 192.168.0.17:5060
message: MESSAGE REC. CALLID:1681834623
message: Message received from: 192.168.0.17:5060
message: eXosip: Reseting timer to 10s before waking up!
warning: Video MSTicker: We are late of 707 miliseconds.
message: Captured mean fps=12.492111, expected=29.969999
message: Captured mean fps=11.900142, expected=29.969999
message: Captured mean fps=18.924681, expected=29.969999
message: Captured mean fps=21.933025, expected=29.969999
message: Captured mean fps=21.000414, expected=29.969999
message: Captured mean fps=23.725334, expected=29.969999
warning: keep alive: 8
message: eXosip: Reseting timer to 10s before waking up!
message: Captured mean fps=22.688780, expected=29.969999
message: Captured mean fps=24.450920, expected=29.969999
message: Captured mean fps=23.588652, expected=29.969999
message: Received message:
INVITE sip:root <at> 192.168.0.12 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.17:5060;rport;branch=z9hG4bK65962445
From: Steve Sakoman <sip:sakoman <at> 192.168.0.17>;tag=2126056146
To: <sip:root <at> 192.168.0.12>
Call-ID: 1681834623
CSeq: 20 INVITE
Contact: <sip:sakoman <at> 192.168.0.17:5060>
Content-Type: application/sdp
Max-Forwards: 70
User-Agent: Linphone/3.2.1 (eXosip2/3.3.0)
Subject: Phone call
Content-Length:   316

v=0
o=sakoman 123456 654321 IN IP4 192.168.0.17
s=A conversation
c=IN IP4 192.168.0.17
t=0 0
m=audio 7078 RTP/AVP 111 110 0 3 8 101
a=rtpmap:111 speex/16000/1
a=rtpmap:110 speex/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:3 GSM/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11

message: Message received from: 192.168.0.17:5060
message: Message received from: 192.168.0.17:5060
message: MESSAGE REC. CALLID:1681834623
message: Message received from: 192.168.0.17:5060
message: eXosip: Reseting timer to 10s before waking up!

Gmane