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

[George Joseph]

On Wed, Mar 23, 2016 at 5:28 AM, George Joseph <george.joseph at fairview5.com>
wrote:

>
>
> On Tue, Mar 22, 2016 at 10:44 PM, Jean-Denis Girard <jd.girard at sysnux.pf>
> wrote:
>
>> 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?
>>
>
> ​Yep.  Let me check.​
>

​This made it in...​

875d5e9 pjproject_bundled: Remove --with-external-pa from configure options.

​I forgot about packet length.   Creating review now.​


>
>
>>
>>
>> Thanks,
>> --
>> Jean-Denis Girard
>>
>> SysNux                Systèmes   Linux   en   Polynésie   française
>> http://www.sysnux.pf/ Tél: +689 40.50.10.40 / GSM: +689 87.79.75.27
>>
>> Le 13/03/2016 17:32, George Joseph a écrit :
>> >
>> >
>> > 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 écrit :
>> >     >     Le 07/03/2016 09:28, George Joseph a écrit :
>> >     >     > 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                Systèmes   Linux   en   Polynésie   française
>> >     http://www.sysnux.pf/ Tél: +689 40.50.10.40 / GSM: +689 87.79.75.27
>> >
>> >
>>
>>
>>
>
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