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George Joseph

2016-Mar-07 22:53 UTC

[asterisk-users] Asterisk now available with bundled pjproject!

On Mon, Mar 7, 2016 at 2:53 PM, Jean-Denis Girard <jd.girard at sysnux.pf>
wrote:
> Hi,
>
> Le 07/03/2016 09:28, George Joseph a ?crit :
> > PLEASE TRY THIS!!  I'd love some feedback BEFORE 13.8.0 is
released.
>
> I have tried GIT-master-ee5a944M on my Fedora 23 test server, and got:
>
> [pjproject] Unpacking /tmp/pjproject-2.4.5.tar.bz2
> [pjproject] Applying patches and custom files
> [pjproject] Configuring with --prefix=/opt/pjproject
> --with-external-speex --with-external-gsm --with-external-srtp
> --with-external-pa --disable-video --disable-v4l2 --disable-sound
> --disable-resample --disable-opencore-amr --disable-ilbc-codec
> --without-libyuv --disable-g7221-codec --enable-epoll
> aconfigure: error: Unable to use PortAudio. If PortAudio development
> files are not available in the default locations, use CFLAGS and LDFLAGS
> env var to set the include/lib paths
> Makefile:57: recipe for target 'build.mak' failed
> make: *** [build.mak] Error 1
> failed
>
> So I installed portaudio-devel (it was not needed before), and then
> compilation / installation were ok. When restarting Asterisk, SELinux
> blocked an access to /usr/bin/portaudio.
>
> Can't we simply disable portaudio? I have changed --with-external-pa to
> --disable-pa in third-party/pjproject/Makefile.rules, and it seems to
> compile / work fine.
>
?Good catch on PortAudio.  I'll add the --disable-pa,?

>
> I have a question for servers without Internet access : is it enough to
> copy pjproject-2.4.5.tar.bz2 to /tmp or will there be other dependencies?
>
?No other dependencies.  Putting it in /tmp should be fine.?

>
> I made a couple of test calls without problem (with or without portaudio).
>
>
> Thanks for your work,
>

?Thanks for testing!?

> --
> Jean-Denis Girard
>
> SysNux                Syst?mes   Linux   en   Polyn?sie   fran?aise
> http://www.sysnux.pf/ T?l: +689 40.50.10.40 / GSM: +689 87.79.75.27
>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>                http://www.asterisk.org/hello
>
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Jean-Denis Girard

2016-Mar-13 05:48 UTC

head link

[asterisk-users] Asterisk now available with bundled pjproject!

Hi George,

Le 07/03/2016 12:53, George Joseph a ?crit :>     Le 07/03/2016 09:28, George Joseph a ?crit :
>     > PLEASE TRY THIS!!  I'd love some feedback BEFORE 13.8.0 is
released.
I don't think this is related to the bundled version, but I got
PJSIP_ERXOVERFLOW when initiating a WebRTC video call from Chrome:

[Mar 12 19:08:37] ERROR[9071]: pjproject:0 <?>:         sip_endpoint.c
Error processing packet from 192.168.10.88:50072: Rx buffer overflow
(PJSIP_ERXOVERFLOW)  [code 171062]:
INVITE sip:*91 at sysnux.pf SIP/2.0
Via: SIP/2.0/WSS ca4cqpd5cv2h.invalid;branch=z9hG4bK2286368
Max-Forwards: 70
To: <sip:*91 at sysnux.pf>
From: <sip:websip2 at sysnux.pf>;tag=q1ejnhm074
Call-ID: l7rivm3clnebl6om63eb
CSeq: 1487 INVITE
Authorization: Digest algorithm=MD5, username="websip2",
realm="asterisk",
nonce="1457845717/bfbd52f55e31f89cda00a1305c272bd6",
uri="sip:*91 at sysnux.pf",
response="d30a2f2b4d5d25e81dded44b7d98e336",
opaque="639fdd14224f0290", qop=auth, cnonce="r0d44vjitbof",
nc=00000001
Contact: <sip:cldsr32v at ca4cqpd5cv2h.invalid;transport=ws;ob>
Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY
Content-Type: application/sdp
Supported: outbound
User-Agent: SIP.js/0.7.3
Content-Length: 3335
...

This can be solved by adding the following line to config_site.h:
#define PJSIP_MAX_PKT_LEN	6000

Would you consider adding it?


Thanks,
-- 
Jean-Denis Girard

SysNux                Syst?mes   Linux   en   Polyn?sie   fran?aise
http://www.sysnux.pf/ T?l: +689 40.50.10.40 / GSM: +689 87.79.75.27

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Jean-Denis Girard

2016-Mar-23 04:44 UTC

head link

[asterisk-users] Asterisk now available with bundled pjproject!

Hi George,

It seems configure with --disable-pa, and configuration "#define
PJSIP_MAX_PKT_LEN 6000" did not make it to 13.8.0-rc1, do you still
intend to add include these modifications?


Thanks,
-- 
Jean-Denis Girard

SysNux                Syst?mes   Linux   en   Polyn?sie   fran?aise
http://www.sysnux.pf/ T?l: +689 40.50.10.40 / GSM: +689 87.79.75.27

Le 13/03/2016 17:32, George Joseph a ?crit :> 
> 
> On Sat, Mar 12, 2016 at 10:48 PM, Jean-Denis Girard <jd.girard at
sysnux.pf
> <mailto:jd.girard at sysnux.pf>> wrote:
> 
>     Hi George,
> 
>     Le 07/03/2016 12:53, George Joseph a ?crit :
>     >     Le 07/03/2016 09:28, George Joseph a ?crit :
>     >     > PLEASE TRY THIS!!  I'd love some feedback BEFORE
13.8.0 is released.
> 
>     I don't think this is related to the bundled version, but I got
>     PJSIP_ERXOVERFLOW when initiating a WebRTC video call from Chrome:
> 
>     [Mar 12 19:08:37] ERROR[9071]: pjproject:0 <?>:        
sip_endpoint.c
>     Error processing packet from 192.168.10.88:50072
>     <http://192.168.10.88:50072>: Rx buffer overflow
>     (PJSIP_ERXOVERFLOW)  [code 171062]:
>     INVITE sip:*91 at sysnux.pf <mailto:91 at sysnux.pf> SIP/2.0
>     Via: SIP/2.0/WSS ca4cqpd5cv2h.invalid;branch=z9hG4bK2286368
>     Max-Forwards: 70
>     To: <sip:*91 at sysnux.pf <mailto:91 at sysnux.pf>>
>     From: <sip:websip2 at sysnux.pf
>     <mailto:sip%3Awebsip2 at sysnux.pf>>;tag=q1ejnhm074
>     Call-ID: l7rivm3clnebl6om63eb
>     CSeq: 1487 INVITE
>     Authorization: Digest algorithm=MD5, username="websip2",
>     realm="asterisk",
nonce="1457845717/bfbd52f55e31f89cda00a1305c272bd6",
>     uri="sip:*91 at sysnux.pf <mailto:91 at sysnux.pf>",
>     response="d30a2f2b4d5d25e81dded44b7d98e336",
>     opaque="639fdd14224f0290", qop=auth,
cnonce="r0d44vjitbof", nc=00000001
>     Contact: <sip:cldsr32v at ca4cqpd5cv2h.invalid;transport=ws;ob>
>     Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY
>     Content-Type: application/sdp
>     Supported: outbound
>     User-Agent: SIP.js/0.7.3
>     Content-Length: 3335
>     ...
> 
>     This can be solved by adding the following line to config_site.h:
>     #define PJSIP_MAX_PKT_LEN       6000
> 
>     Would you consider adding it?
> 
> 
> 
> Yes.  I'll add it this week.?
>  
> 
> 
> 
>     Thanks,
>     --
>     Jean-Denis Girard
> 
>     SysNux                Syst?mes   Linux   en   Polyn?sie   fran?aise
>     http://www.sysnux.pf/ T?l: +689 40.50.10.40 / GSM: +689 87.79.75.27
> 
> 

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asterisk users - Mar 2016 - Asterisk now available with bundled pjproject!

[asterisk-users] Asterisk now available with bundled pjproject!

[asterisk-users] Asterisk now available with bundled pjproject!

[asterisk-users] Asterisk now available with bundled pjproject!