Hello,
how and when is main/version.c generated ? I checked out branch-11 svn,
compiled, packaged for debian, and now I realized that core show
version is diffent than version that I got from SVN.
main/version.c
static const char asterisk_version[] = SVN-branch-11-r410490M;
ipfon-test*CLI
On 02/26/2014 03:38 PM, Kaloyan Kovachev wrote:
On 2014-02-26 16:17, Frederic Van Espen wrote:
And i guess you want to use asterisk to act as an SS7 - SIP translator
towards the PSTN so there shoudlnt be a BSSAP involved only plain
ISUP which
could be handled from libss7.
Assuming the above
Hi Jaco,
I reviewed spandsp sources at the places where your segfaults
happened, and this is very different situation than mine. But I see
that span_log function (spandsp logging) is called frequently from
this code, you should find some more information in spandsp
Hello Pavel,
On 01/31/2014 07:59 AM, Pavel Troller wrote:
This code will translate non-slinear frames to slinear, just before they
are sent to libspandsp for v21detection. With this patch applied, v21
detection is done also for RTP (SIP) alaw/ulaw frames, so maybe SIP/G711
- SIP/T38 gateway
Dear Pavel,
On 01/30/2014 06:55 AM, Pavel Troller wrote:
I probably found what causes random segfaults during fax detection with
spandsp... Many thanks for your contributions and brainstorming about this,
it helped a lot. If you have such issue, please test this possible fix and
let me know...
Hi folks,
I probably found what causes random segfaults during fax detection with
spandsp... Many thanks for your contributions and brainstorming about
this, it helped a lot. If you have such issue, please test this possible
fix and let me know...
At res_fax.c, somewhere around line 3070,
Hi Matthew,
On 01/28/2014 01:00 AM, Matthew Jordan wrote:
In res_rtp_asterisk, the packet is read from the socket in
ast_rtp_read. This is also the place where the read data is converted
into a frame of the appropriate type. The payload for the voice frame
is obtained in the actual RTP packet
Hello,
I have problem with random Asterisk segfaults on the machine, which I
use as T.38 gateway between DAHDI and SIP. I would like to kindly ask
somebody to take a look at it, and help me to find what's wrong...
Asterisk is version 11 from SVN, r382022 (I'm using this because of
other
Hi Pavel,
thank you for an answer - it inspired me a lot, and we're now much
closer to the resolution (I hope). It seems that there is something
wrong with memory allocation for RTP frames (probably
res_rtp_asterisk.c). I explain details below, and I hope that one of
Asterisk gurus will help
