Displaying 12 results from an estimated 12 matches for "ast_translator_build_path".
2006 Dec 15
2
call from h323 to SIP
...from g729 to slin
Dec 15 14:45:13 WARNING[19795]: channel.c:2380 set_format: Unable to
find a codec translation path from g729 to ulaw
Dec 15 14:45:13 WARNING[19795]: channel.c:2380 set_format: Unable to
find a codec translation path from g729 to slin
Dec 15 14:45:13 WARNING[19794]: translate.c:116
ast_translator_build_path: No translator path from alaw to unknown
Dec 15 14:45:13 WARNING[19794]: channel.c:1202 queue_frame_to_spies:
Cannot build a path from g729 to slin
Dec 15 14:45:13 WARNING[19794]: chan_h323.c:614 oh323_write: Asked to
transmit frame type 64, while native formats is 256 (read/write =
4/64)
Dec 15 14...
2019 Oct 03
2
Asterisk not using common codec between (SIP) endpoints
...f the way by establishing a direct media path if
somehow possible.
Best Regards,
Andreas
NOTE: The following is the console output from Asterisk 13 when the
mediagateway answers and sends RTP PT 102 at the beginning of the call.
[Oct 2 07:24:55] WARNING[23961][C-00000000]: translate.c:490
ast_translator_build_path: No translator path: (ending codec is not valid)
[Oct 2 07:24:55] WARNING[23961][C-00000000]: translate.c:490
ast_translator_build_path: No translator path: (ending codec is not valid)
[Oct 2 07:24:55] WARNING[23961][C-00000000]: translate.c:490
ast_translator_build_path: No translator path: (s...
2003 Oct 28
2
Another Segmentation Fault (Recording sound)
...from '890003' to '185' on channel 27, span 1
-- Executing Answer("Zap/27-1", "") in new stack
-- Executing Record("Zap/27-1", "soundexampless:mp3") in new stack
-- Playing 'beep'
WARNING[360468]: File translate.c, Line 128 (ast_translator_build_path): No translator path from UNKN to ULAW
WARNING[360468]: File file.c, Line 218 (ast_writestream): Unable to translate to format mp3, source format ALAW
WARNING[360468]: File app_record.c, Line 166 (record_exec): Problem writing frame
Segmentation fault
I guess this is pretty explanatory.
Regar...
2011 Dec 20
1
File Convert
...gsm file
to G729 using file convert, but I am facing error as follows,
file convert /tmp/welcome.gsm /tmp/welcome.g729
Failed to convert /tmp/welcome.gsm to /tmp/welcome.g729!
Command 'file convert /tmp/welcome.gsm /tmp/welcome.g729' failed.
[Dec 20 17:24:18] WARNING[2221]: translate.c:256 ast_translator_build_path:
No translator path from g723 to alaw
[Dec 20 17:24:18] WARNING[2221]: file.c:184 ast_writestream: Unable to
translate to format g729, source format gsm
Even though I have the module format_g729.so. Do I need to have licensed
G729 codec for this? or codec_g729.so?
Kindly let me know how to conver...
2003 Jul 23
4
Problems with g729
...ugh a restart of *. When the error ocurrs there are no other calls in place. Any ideas?
Error Opening channel:2 not available, see va_g729_init_global(..) WARNING[71694]:File codec_g729b.c line 102 (g729lin_new): No available g729b resource for channel 2
WARNING:[71694] File translate.c Line 111 (ast_translator_build_path):Failed to build translator path from 8 to 6 Zap1-1 answered SIP/105-ce3c
WARNING[71694]: File chan_zap.c Line 3367 (zt_write):Cannot handle frames in 256 format
Hangup Zap/1-1
2)
have discovered a problem when using g729 under the following setup:
SIP call between a Budgetone 102 and ATA 186...
2010 Feb 19
1
transcoding with TC400P
...90/0 encoders/decoders of 92 channels are in use.
But it all does not work for the second case, in asterisk2 cli I get
messages like
[Feb 19 15:18:32] ERROR[3121]: codec_dahdi.c:244 zap_translate: Unable
to attach to transcoder: Input/output error
[Feb 19 15:18:32] WARNING[3121]: translate.c:294
ast_translator_build_path: Failed to build translator step from 8 to 2
[Feb 19 15:18:32] WARNING[3121]: chan_sip.c:3751 sip_write: Asked to
transmit frame type 256, while native formats is 0x4 (ulaw)(4)
read/write = 0x4 (ulaw)(4)/0x100 (g729)(256)
Btw call does go through but transcoding is done by processor not by
TC400P...
2004 Jun 30
3
Bugfix for CVS-HEAD-06/26/04-21:56:45
...e need to
> translate, IF
> > // the channel doesn't support SLINEAR. Otherwise, we need to just
> > // write the SLINEAR frame.
> > if (!(chan->nativeformats & AST_FORMAT_SLINEAR)) {
> > struct ast_trans_pvt* transPath =
> ast_translator_build_path(chan->writeformat, AST_FORMAT_SLINEAR);
> > struct ast_frame* transFrame = ast_translate(transPath,
> &ps->f, 0);
> > if (transFrame) {
> > ast_write(chan, transFrame);
> > ast_frfree(trans...
2020 Nov 19
0
Asterisk 17.9.0 Now Available
...d by Jean Aunis - Prescom)
* ASTERISK-29097 - res_pjsip_config_wizard: Crash when freeing
string when failing to add extension
(Reported by Vieri)
* ASTERISK-26424 - app_voicemail: Undocumented behavior from
VMSayName
(Reported by Eric Smith)
* ASTERISK-29091 - Crash when ast_translator_build_path fails
(Reported by Jasper van der Neut)
* ASTERISK-29099 - res_musiconhold: Realtime MOH only loads a
single entry
(Reported by laszlovl)
* ASTERISK-29051 - res_pjsip_sdp_rtp: Does not set correct
values on RTP instance when "auto" DTMF is used
(Reported...
2020 Nov 19
0
Asterisk 16.15.0 Now Available
...RISK-29099 - res_musiconhold: Realtime MOH only loads a
single entry
(Reported by laszlovl)
* ASTERISK-29051 - res_pjsip_sdp_rtp: Does not set correct
values on RTP instance when "auto" DTMF is used
(Reported
by Sebastian Damm)
* ASTERISK-29091 - Crash when ast_translator_build_path fails
(Reported by Jasper van der Neut)
* ASTERISK-28311 - dsp: ast_dsp_silence_noise_with_energy wrong
judgment of frame format
(Reported by ���������)
* ASTERISK-29085 - func_curl: Segmentation fault when using
CURL after setting httpheader CURLOPT
(Reported by P...
2020 Nov 19
0
Asterisk 18.1.0 Now Available
...d by Jean Aunis - Prescom)
* ASTERISK-26424 - app_voicemail: Undocumented behavior from
VMSayName
(Reported by Eric Smith)
* ASTERISK-29097 - res_pjsip_config_wizard: Crash when freeing
string when failing to add extension
(Reported by Vieri)
* ASTERISK-29091 - Crash when ast_translator_build_path fails
(Reported by Jasper van der Neut)
* ASTERISK-29051 - res_pjsip_sdp_rtp: Does not set correct
values on RTP instance when "auto" DTMF is used
(Reported
by Sebastian Damm)
* ASTERISK-29099 - res_musiconhold: Realtime MOH only loads a
single entry...
2010 Jul 05
1
Problems with ulaw/g729 translation
....0 build 56125) trought local network,
using ulaw codec.
Sometimes, I got messages like:
[Jul 1 15:26:16] WARNING[26483]: chan_sip.c:5514 process_sdp: Unsupported
SDP media type in offer: image 65344 udptl t38
And then a lot of messages like:
[Jul 1 15:27:00] WARNING[26549]: translate.c:274 ast_translator_build_path:
No translator path from alaw to unknown
That's stopping the phone system. When I got the messages, I can't make or
receive calls. Then, a few minutes later (or when I stop and start
asterisk), the phone system back to work again.
Some confs and system status:
sip.conf:
[1050] ; THA...
2004 Nov 20
1
Asterisk dead but pid file exists - gdb asterisk core.13089
...ols for
/usr/lib/asterisk/modules/app_addon_sql_mysql.so
#0 0x001d8a55 in _int_malloc () from
/lib/i686/libc.so.6
(gdb) bt
#0 0x001d8a55 in _int_malloc () from
/lib/i686/libc.so.6
#1 0x001d7a23 in malloc () from /lib/i686/libc.so.6
#2 0x00b4c363 in gsm_new () at codec_gsm.c:63
#3 0x08061d7d in ast_translator_build_path (dest=64,
source=1) at translate.c:110
#4 0x08061008 in ast_set_read_format
(chan=0xbf2108a0, fmts=-1088314640) at channel.c:1738
#5 0x02b3b5f0 in socket_read (id=0x8805cb8, fd=17,
events=1, cbdata=0x0) at chan_iax2.c:5310
#6 0x080523c0 in ast_io_wait (ioc=0x8804588,
howlong=32129) at io.c:267
#...