similar to: app_mp3 with bri-stuff.0.1.0RC4a does not work

Hi there, I am using bri-stuff.0.1.0-RC2k and it seems that things didn't become better. I have got lots of dropouts on the IAX2 link (no matter if jitter buffers are enabled). Further the MP3Player application does not playback streams like http://somestreamserver/somestream. It stops saying: -- Executing MP3Player("SIP/27870-ba4f",

Hi, how can I setup asterisk to use the number presentation bits on the isdn side to suppress the number presentation? We need to transmit the subscriber number for billing purposes via ISDN whether or not the user wants to hide his/her number. Is there any way to do this? Deti

Hi there, I tried to use Voicemail from a PRI interface but it didn't work because pressing a key on the ISDN phone just caused Q931_IE_KEYPAD_FACILITY messages which are normally handled by a bri-stuffed libpri. Unfortunately a wrong if condition stops keypad messages from being passed to the channel driver. The patch attached to this mail removes the 2 lines from the code. Did I

Hi, ices does not work very reliable here in my place. I am using the latest CVS sources and I get up to three times a day something like that: Program received signal SIGSEGV, Segmentation fault. [Switching to Thread 2051 (LWP 8160)] 0x08051103 in resample_clear (s=0x40ef9338) at audio.c:142 142 if (s->buffers[c]) (gdb) bt #0 0x08051103 in resample_clear

Hi, I added metadata support for the stdinpcm module of ices. The metadata have to be submitted via file (as stdin is already used for the strem). Please apply this patch to the current cvs sources. <p>Deti <p> Index: im_stdinpcm.c =================================================================== RCS file: /usr/local/cvsroot/ices/src/im_stdinpcm.c,v retrieving revision 1.8 diff

I'm trying to compile Zaptel driver with the ztdummy. I have no hardware cards from digium. I tried following steps: http://www.voip-info.org/wiki/view/Asterisk+Zaptel+Installation http://www.voip-info.org/wiki/index.php?page=Asterisk+timer+ztdummy I'm running : Linux version 2.4.27-2-386 (horms@tabatha.lab.ultramonkey.org) (gcc version 3.3.5 (Debian 1:3.3.5-13)) #1 Wed Aug 17

There is a patch to mplayer that allows it to suppress stdout messages and instead output pcm data to stdout. I managed to get it working with app_mp3.c and seems like it is working fine. All that was needed was a change in the execl line and a slight increase in timeout value. I have only done limited testing. If mplayer is able to replace mpg123 without issues, this opens up a whole lot of

I'm getting this message when I start Asterisk chan_capi.c:2205 capi_handle_msg: Command.Subcommand = 0x5.0x81 but when I try and recompile I get this chan_capi.c:60: error: `__use_AST_MUTEX_DEFINE_STATIC_rather_than_AST_MUTEX_INITIALIZER__' undeclared here (not in a function) any help would be greatly appreciated. -- Dave Cotton <dcotton@linuxautrement.com>

Hi all, according to http://sip.fontventa.com/content/view/15/44/ i have compiled the mpeg4ip libries without problem. After copying the app_mp4.c file into de Asterisk apps directory and changing the Makefile like. [...] app_sql_odbc.so: app_sql_odbc.o $(CC) $(SOLINK) -o $@ ${CYGSOLINK} $< ${CYGSOLIB} -lodbc app_mp4.so : app_mp4.o $(CC) $(SOLINK) -o $@ ${CYGSOLINK} $<

Hi all! I downloaded right mpg123, chabged path to mpg123 binary in app_mp3.c, rebuilt app_mp3.so, and got MusicOnHold to work. But MP3Player refuses to do properly: -- Accepting AUTHENTICATED call from x.x.x.x, requested format = 1024, actual format = 1024 -- Executing Answer("IAX2/maxhome@maxhome/3", "") in new stack -- Executing

Hi all, Here another guy working with ztdummy and having problems with music. MP3Player does not work for me both from a telephone entering through a passive ISDN-adapter as well as a SIP-client in the LAN. Ztdummy works with conferences. Here are my messages. I have seen several discussion threads in the mail-archive regarding this subject, but could not find an answer. Is this something I

Hello Folks, What could be producing the following warnings on console, after an installation from source (Asterisk 1.4.41): [May 12 21:36:54] WARNING[15344]: loader.c:434 load_dynamic_module: Error loading module 'res_musiconhold.so': /usr/lib/asterisk/modules/res_musiconhold.so: undefined symbol: cap_set_proc [May 12 21:36:54] WARNING[15344]: loader.c:434 load_dynamic_module:

This simple additional check hardens sshbuf against linking an sshbuf into itself as parent/child pair, which could lead to ref counting issues. Purely defensive measure. I am not aware that this could happen somehwere in the code by now. Okay? Index: sshbuf.c =================================================================== RCS file: /cvs/src/usr.bin/ssh/sshbuf.c,v diff -u -p -u -p -r1.19

Reordering calloc arguments silences gcc compiler warnings of latest versions. Spotted with OpenSSH-portable on a Linux system. Okay? Index: cipher.c =================================================================== RCS file: /cvs/src/usr.bin/ssh/cipher.c,v diff -u -p -u -p -r1.121 cipher.c --- cipher.c 17 May 2024 02:39:11 -0000 1.121 +++ cipher.c 13 Aug 2024 16:46:00 -0000 @@ -249,7 +249,7

Hi there, I am using bri-stuff.0.0.2 and maybe I misunderstood something but my HFC card is in bri_cpe_ptmp mode and gets routed about 80 MSNs. Some of them are not intended to be used by asterisk but every incoming call is accepted even if the default extension leads to Congestion: -- Accepting call from '28715' to '27849' on channel 1, span 2 -- Executing

Hi, I'm running the latest Asterisk (built last Saturday) and can't get mp3's to playback on my handsets (this includes music on hold). I setup a couple of extensions, 901 and 902 to playback an mp3 I loaded on, and the sample moh that is included with Asterisk. When I attempt to call either extension I don't hear any sound, and the following displays on the console:

Does anyone know whether the linux machine running * have to have a sound card on it in order for musiconhold to work for sip phones? I've tried about everything (including tons of google searching) to get it to work, and nothing. When a call is placed on hold between two C7960's, the CLI indicates: -- Executing Dial("SIP/3002-c418", "SIP/3000|20") in new stack

Hi, the below patch adds a Windows NT user right to the sshd_server user which will be important in an upcoming version of Cygwin. I have a preliminary implementation which solves the problem that native Windows processes don't recognize the user name correctly, if the user has logged in using public key authentication. The new mechanism requires the SeTcbPrivilege for the user which

The freezero call is practically the same as calling explicit_bzero followed by free. Okay? Index: sshbuf.c =================================================================== RCS file: /cvs/src/usr.bin/ssh/sshbuf.c,v diff -u -p -u -p -r1.19 sshbuf.c --- sshbuf.c 2 Dec 2022 04:40:27 -0000 1.19 +++ sshbuf.c 13 Aug 2024 16:54:20 -0000 @@ -183,10 +183,8 @@ sshbuf_free(struct sshbuf *buf)

This patch fixes two spelling mistakes in code comments, which means no functional change: still-extant -> still-existant the -> then Okay? Index: sshbuf.c =================================================================== RCS file: /cvs/src/usr.bin/ssh/sshbuf.c,v diff -u -p -u -p -r1.19 sshbuf.c --- sshbuf.c 2 Dec 2022 04:40:27 -0000 1.19 +++ sshbuf.c 13 Aug 2024 16:39:12 -0000 @@