search for: slinr

...t the codec Asterisk wants, even if I force it in sip.conf. Is there anywhere else I need to look to tell ZAP to use G.729 preferrably ? The translator is loaded... [codec_gsm.so] => (GSM/PCM16 (signed linear) Codec Translator) == Registered translator 'gsmtolin' from format GSM to SLINR, cost 1 == Registered translator 'lintogsm' from format SLINR to GSM, cost 4 [codec_g729a.so] => (Annex A/B (floating point) G.729/PCM16 Codec Translator) == G.729 Host-ID: 1b:a1:18:82:47:6f:a8:f7:33:de:7d:77:e8:1d:60:15:53:ec:49:aa == Found license 'G729-700241AB' providi...

...reason incoming and outgoing calls will ALWAYS use G.711a. Is there anywhere else I need to look to tell ZAP to use G.729 preferrably ? The translator is loaded... [codec_gsm.so] => (GSM/PCM16 (signed linear) Codec Translator) == Registered translator 'gsmtolin' from format GSM to SLINR, cost 1 == Registered translator 'lintogsm' from format SLINR to GSM, cost 4 [codec_g729a.so] => (Annex A/B (floating point) G.729/PCM16 Codec Translator) == G.729 Host-ID: 1b:a1:18:82:47:6f:a8:f7:33:de:7d:77:e8:1d:60:15:53:ec:49:aa == Found license 'G729-700241AB' providi...

..._g729b.c, Line 413 (load_module): Unable to initialize va stuff: -1 == Detected 4 licensed G.729 transcoders WARNING[16384]: File translate.c, Line 219 (calc_cost): Translator 'g729tolinb' does not produce sample frames. == Registered translator 'g729tolinb' from format G729A to SLINR, cost 99999 Error Opening channel:0 call va_g729_init_global(..) prior to open! WARNING[16384]: File codec_g729b.c, Line 179 (lintog729_new): No available g729 resources for channel 0 WARNING[16384]: File translate.c, Line 225 (calc_cost): Translator 'lintog729b' appears to be broken and wi...

...ning the codec correctly or doesn't support the same subset of speex codecs as openh323. (I'm using speex-1.0.1 with asterisk) thanks, Adam Hart [codec_speex.so] => (Speex/PCM16 (signed linear) Codec Translator) == Registered translator 'speextolin' from format SPEEX to SLINR, cost 2 == Registered translator 'lintospeex' from format SLINR to SPEEX, cost 47 == Parsing '/etc/asterisk/enum.conf': Found Asterisk Ready.

...A/B (floating point) G.729/PCM16 Codec Translator) == G.729 Host-ID: 07:53:aa:d3:e2:f2:bd:cc:27:60:9d:5f:da:eb:5d:e9:6e:09:a1:4e == Found license 'G729-253D0C86' providing 4 channels == Found total of 4 G.729 licenses == Registered translator 'g729tolin' from format G729A to SLINR, cost 5 == Registered translator 'lintog729' from format SLINR to G729A, cost 24 *CLI> show translation Translation times between formats (in milliseconds) Source Format (Rows) Destination Format(Columns) G723 GSM ULAW ALAW G726 ADPCM SLINR LPC10 G729...

Hi. I just tried updating asterisk and I guess I broke something. Here's the log: Unable to find a path from G729A to SLINR Unable to find a path from ULAW to G729A Any ideas on what I should try? I tried nuking all the zaptel stuff in the system and the source and started over agian. Also nuked the asterisk config files.... I saw this asked once before but there was no reply :-/ Thanks, Kevin ___________________...

...oder) == Registered translator 'alawtoulaw' from format ALAW to ULAW, cost 1 == Registered translator 'ulawtoalaw' from format ULAW to ALAW, cost 1 [codec_adpcm.so] => (Adaptive Differential PCM Coder/Decoder) == Registered translator 'adpcmtolin' from format ADPCM to SLINR, cost 1 == Registered translator 'lintoadpcm' from format SLINR to ADPCM, cost 1 [codec_alaw.so] => (A-law Coder/Decoder) == Registered translator 'alawtolin' from format ALAW to SLINR, cost 1 == Registered translator 'lintoalaw' from format SLINR to ALAW, cost 1 [co...

...(floating point) G.729/PCM16 Codec Translator) == G.729 Host-ID: 5f:a1:18:82:47:6f:a8:f7:33:4e:7d:77:e8:1d:60:15:53:ec:49:aa == Found license 'G729-700241AB' providing 5 channels == Found total of 5 G.729 licenses == Registered translator 'g729tolin' from format G729A to SLINR, cost 2 == Registered translator 'lintog729' from format SLINR to G729A, cost 12 I was hoping by letting it ring out, I would get a voice-mail message, but that doesn't work either... *CLI> Jul 13 11:29:08 WARNING[98310]: chan_sip.c:2696 process_sdp: No compatible codecs!...

...RNING[18631600]: chan_sip.c:2804 process_sdp: No compatible codecs! -- Got SIP response 488 "Not Acceptable Here" back from 209.250.147.116 show translation (I figure this has anything to do with it) shows that all paths are supported: G723 GSM ULAW ALAW G726 ADPCM SLINR LPC10 G729A SPEEX ILBC G723 - 4 2 2 3 2 1 4 13 35 19 GSM 15 - 2 2 3 2 1 4 13 35 19 ULAW 15 4 - 1 3 2 1 4 13 35 19 ALAW 15 4 1 - 3 2 1 4 13...

Hello! Asterisk "CVS-HEAD-08/06/04-14:55:13" won't start on two of three different Gentoo machines. This is the output of gdb: ultra asterisk # gdb /usr/sbin/asterisk GNU gdb 6.0 Copyright 2003 Free Software Foundation, Inc. GDB is free software, covered by the GNU General Public License, and you are welcome to change it and/or distribute copies of it under certain conditions.

...b.so] => (Annex B (floating point) G.729/PCM16 Codec Translator) == Detected 2 licensed G.729 transcoders WARNING[16384]: File translate.c, Line 219 (calc_cost): Translator 'g729tolinb' does not produce sample frames. == Registered translator 'g729tolinb' from format G729A to SLINR, cost 99999 == Registered translator 'lintog729b' from format SLINR to G729A, cost 95 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031023/e8309fe5/attachment.htm

...234 (listener): Select retured error: Interrupted system call == Detected 1 licensed G.729 transcoders WARNING[1074420608]: File translate.c, Line 219 (calc_cost): Translator 'g729tolinb' does not produce sample frames. == Registered translator 'g729tolinb' from format G729A to SLINR, cost 99999 == Registered translator 'lintog729b' from format SLINR to G729A, cost 140 Can someone shed some light on this please. Alex.

...When calling from that Planet phone to anybody, everthing is fine. But when calling from anybody to that Planet phone, I get a mashine gun noise and the following msg in asterisk log: NOTICE[262161]: File channel.c, Line 1091 (ast_read): Dropping incompatible voice frame on H323:0 of format SLINR since our native format has changed to ULAW Both, the Planet phone and the asterisk oh323 channel, have G.711A as preferred codecs. For me, it seems, that the planet phone does not follow that hint, when receiving a call. But shouldn't asterisk be capable of understanding SLINR? Any help is...

...255 listener: Select retured error: Interrupted system call == Detected 4 licensed G.729 transcoders Jan 21 17:31:58 WARNING[1074411264]: translate.c:219 calc_cost: Translator 'g729tolinb' does not produce sample frames. == Registered translator 'g729tolinb' from format G729A to SLINR, cost 99999 == Registered translator 'lintog729b' from format SLINR to G729A, cost 164 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040121/4ac2f16c/attachment.htm

...] => (Annex B (floating point) G.729/PCM16 Codec Translator) == Detected 2 licensed G.729 transcoders Apr 21 20:52:15 WARNING[16384]: translate.c:213 calc_cost: Translator 'g729tolinb' does n t produce sample frames. == Registered translator 'g729tolinb' from format G729A to SLINR, cost 99999 == Registered translator 'lintog729b' from format SLINR to G729A, cost 43 == Parsing '/etc/asterisk/enum.conf': Found Asterisk Ready. *CLI> Apr 21 20:52:17 NOTICE[131081]: chan_sip.c:5880 sip_poke_noanswer: Peer 'santoext' s now UNREACHABLE! Apr 21 20:52...

...code. I am using HiSax driver and chan_modem_i4l. When disconnecting I get the following in Asterisk's log file but there are no messages when connecting. There are no such problems when using voicemail nor when dialling SIP phones. Apr 23 11:22:23 NOTICE[1097748]: Unable to find a path from SLINR to UNKN Apr 23 11:22:23 WARNING[1097748]: Unable to restore read format on 'Modem[i4l]/ttyI5' Apr 23 11:22:23 NOTICE[1097748]: Unable to find a path from UNKN to SLINR Apr 23 11:22:23 WARNING[1097748]: Unable to restore write format on 'Modem[i4l]/ttyI5' Running v1-0_stable * from...

...(floating point) G.729/PCM16 Codec Translator) == G.729 Host-ID: 5f:a1:18:82:47:6f:a8:f7:33:4e:7d:77:e8:1d:60:15:53:ec:49:aa == Found license 'G729-700241AB' providing 5 channels == Found total of 5 G.729 licenses == Registered translator 'g729tolin' from format G729A to SLINR, cost 2 == Registered translator 'lintog729' from format SLINR to G729A, cost 12 *CLI> Jul 13 11:29:08 WARNING[98310]: chan_sip.c:2696 process_sdp: No compatible codecs! -- Executing Dial("SIP/67.23.212.25-0814f830", "SIP/334|20") in new stack -- C...

...ything on the net. I installed asterisk using bristuff-0.0.2 without any errors, but when I start asterisk with "asterisk -vvvc" I get following error: [codec_ilbc.so] => (iLBC/PCM16 (signed linear) Codec Translator) == Registered translator 'ilbctolin' from format ILBC to SLINR, cost 127 Segmentation fault Removing codec_ilbc.so from /usr/lib/asterisk/modules shows up the next error: [codec_lpc10.so] => (LPC10 2.4kbps (signed linear) Voice Coder) == Registered translator 'lpc10tolin' from format LPC10 to SLINR, cost 63 Segmentation fault Ok, just remove...

...to record some sound the application will not finish. There is a sound file, but it is empty (0 bytes). "Record(${FILE}:gsm|10|30|skip)" is used in the dial plan. After hangup the following error messages show up: NOTICE[]: channel.c:1683 ast_set_read_format: Unable to find a path from SLINR to UNKN WARNING[]: app_record.c:287 record_exec: Unable to restore read format on 'ALSA/default' Kernel 2.6.5 and Asterisk CVS 04-97-23 are installed. Some test using arecord and aplay shows that microphone and speakers are working. Any help would be appreciated. -- Stefan Tichy &lt...

...so] => (Transfer) == Registered application 'Transfer' [app_setcidnum.so] => (Set CallerID Number) == Registered application 'SetCIDNum' [codec_ilbc.so] => (iLBC/PCM16 (signed linear) Codec Translator) == Registered translator 'ilbctolin' from format ILBC to SLINR, cost 7 == Registered translator 'lintoilbc' from format SLINR to ILBC, cost 38 [format_h263.so] => (Raw h263 data) == Registered file format h263, extension(s) h263 == Parsing '/etc/asterisk/enum.conf': Found Asterisk Ready. *CLI> -- Executing Dial("SIP/1800-6...