search for: func_speex

I'm seeing Asterisk crashes with the following frame at func_speex.c:188: (gdb) p *frame $6 = {frametype = AST_FRAME_VOICE, subclass = {integer = 0, format = 0xe2f9e20, frame_ending = 0}, datalen = 0, samples = 640, mallocd = 1, mallocd_hdr_len = 232, offset = 64, src = 0x2ac07413e7f8 "siren14tolin32", data = {ptr = 0x3cab9378, uint32 =...

On 5/26/23 01:15, Fourhundred Thecat wrote: > how do I fix this? > What do I have to do to "register" denoise ? confbridge.conf states: "Requires func_speex to be built and installed." I am guessing you have not fulfilled that requirement. Doug -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20230526/d475d21c/attachment.html>

Hi, I'm trying to use AGC in combination with Asterisk 1.8 and an odd telephone which is very loud when used with a headset and more quiet when used "normal". Regarding to the documentation, AGC should be available since * 1.6 - but every time I want to set it, the CLI tells me: -- Executing [0160xxxxxxx at intern:2] Set("SIP/intern-xxx-000000d2",

Hi, I have followed steps which were mentioned on forum and given below. Still couldn't get speex working. On test calls getting error "chan_sip.c: sip_call: No audio format found to offer." # yum install speex # yum install speex-devel # cd /usr/src/asterisk # make clean # make # service asterisk stop # make install # service asterisk start Also, it is not

...in terms of samples, not the > data payload itself. I've also commented on your original issue in > regards to the siren codecs that it should NULL out the data pointer > itself. That is more commonly used. But I don't think that it would have helped in either case, this one or in func_speex.c, because neither tests for a null data pointer either. Can you explain the difference between "datalen" and "samples" in this context, shouldn't they always be related by a (small) linear factor? Should I open a JIRA issue for this as well? Can you suggest ways of searc...

Hello, when I call my conference, I see this error in my logs: ERROR: Function DENOISE not registered here is snippet from extensions.conf ... same => n,Set(CONFBRIDGE(user,announce_join_leave)=yes) same => n,Set(CONFBRIDGE(bridge,record_conference)=yes) same => n,Set(CONFBRIDGE(bridge,record_file)=/home/asterisk/file.wav) same => n,ConfBridge(1000) same =>

Another crash with a packet: $10 = {frametype = AST_FRAME_VOICE, subclass = {integer = 0, format = 0x12c62170, frame_ending = 0}, datalen = 0, samples = 640, mallocd = 1, mallocd_hdr_len = 324, offset = 64, src = 0x2ad290064a08 "siren14tolin32/speex", data = {ptr = 0x80893318, uint32 = 2156475160, pad = "\030\063\211\200\000\000\000"}, delivery = { tv_sec =

Is there any possibility of DAHDI supporting Automatic gain control on TDM ports? I'm having issues at a couple of offices where calls made to local numbers are fine but a when a calls from or goes to a large percentage of long-distance or 1-800 numbers the person at the remote end cannot hear the person in my office. Boosting the gains in zapata.conf (I'm still using 1.4.21) to 8

...r??ger) * ASTERISK-26951 - chan_sip: ACK with SDP does not update a direct media bridge (Reported by Jean Aunis - Prescom) * ASTERISK-26930 - pjproject/Makefile.rules for pjsip 2.6 build fails for non-SSE2 instrunction Linux (Reported by abelbeck) * ASTERISK-26926 - func_speex: Crash caused by frame with no datalen (Reported by Richard Kenner) * ASTERISK-26929 - pjsip: Add database tables for RLS (Reported by Joshua Colp) * ASTERISK-26953 - Asterisk crash if hep.conf have some missing parameters (Reported by Joel Vandal) * ASTERISK-...

...K-26908 - res_pjsip: The ChanIsAvail causes a res_pjsip session to be leaked. (Reported by Richard Mudgett) * ASTERISK-25823 - SIGSEGV, Segmentation fault. - ../sysdeps/x86_64/strlen.S: No such file or directory. (Reported by Andreas Kr??ger) * ASTERISK-26926 - func_speex: Crash caused by frame with no datalen (Reported by Richard Kenner) * ASTERISK-26951 - chan_sip: ACK with SDP does not update a direct media bridge (Reported by Jean Aunis - Prescom) * ASTERISK-26930 - pjproject/Makefile.rules for pjsip 2.6 build fails for non-SSE2 i...

...r??ger) * ASTERISK-26951 - chan_sip: ACK with SDP does not update a direct media bridge (Reported by Jean Aunis - Prescom) * ASTERISK-26930 - pjproject/Makefile.rules for pjsip 2.6 build fails for non-SSE2 instrunction Linux (Reported by abelbeck) * ASTERISK-26926 - func_speex: Crash caused by frame with no datalen (Reported by Richard Kenner) * ASTERISK-26929 - pjsip: Add database tables for RLS (Reported by Joshua Colp) * ASTERISK-26953 - Asterisk crash if hep.conf have some missing parameters (Reported by Joel Vandal) * ASTERISK-...

...* ASTERISK-26959 - dial: Allow topology of dialing channel to influence dialed channel (Reported by Joshua Colp) * ASTERISK-25823 - SIGSEGV, Segmentation fault. - ../sysdeps/x86_64/strlen.S: No such file or directory. (Reported by Andreas Kr??ger) * ASTERISK-26926 - func_speex: Crash caused by frame with no datalen (Reported by Richard Kenner) * ASTERISK-26964 - res_pjsip_session: Wrong From on reinvite when request and To URI differ (Reported by Yasin CANER) * ASTERISK-26930 - pjproject/Makefile.rules for pjsip 2.6 build fails for non-SSE...

...* ASTERISK-26959 - dial: Allow topology of dialing channel to influence dialed channel (Reported by Joshua Colp) * ASTERISK-25823 - SIGSEGV, Segmentation fault. - ../sysdeps/x86_64/strlen.S: No such file or directory. (Reported by Andreas Kr??ger) * ASTERISK-26926 - func_speex: Crash caused by frame with no datalen (Reported by Richard Kenner) * ASTERISK-26964 - res_pjsip_session: Wrong From on reinvite when request and To URI differ (Reported by Yasin CANER) * ASTERISK-26930 - pjproject/Makefile.rules for pjsip 2.6 build fails for non-SSE...

...- [ASTERISK-25823 <https://issues.asterisk.org/jira/browse/ASTERISK-25823>] - SIGSEGV, Segmentation fault. - ../sysdeps/x86_64/strlen.S: No such file or directory. (Reported by Andreas Krüger) - [ASTERISK-26926 <https://issues.asterisk.org/jira/browse/ASTERISK-26926>] - func_speex: Crash caused by frame with no datalen (Reported by Richard Kenner) - [ASTERISK-26964 <https://issues.asterisk.org/jira/browse/ASTERISK-26964>] - res_pjsip_session: Wrong From on reinvite when request and To URI differ (Reported by Yasin CANER) - [ASTERISK-26930 <https://iss...