Re: [asterisk-users] Sound files
On Tue, May 08, 2018 at 04:48:21PM -0400, Dovid Bender wrote: > Hi, > > It is my understanding that while Hebrew is supported by Asterisk the sound > files are not shipped with it as they are no longer being maintained. They were never official Asterisk sound files. Their license is likewise less free (e.g. regarding modifications). > Can > anyone advise on what's needed to maintain a specific sound package? We are > considering to support Hebrew and possibly Yiddish. For Yiddish I guess you also need to add support for the syntax. I must admit to not knowing Yiddish well enough. I suppose you need a different syntax for it. Look for the string "he" or "de" (with quotes) in main/asterisk.c and add similar functions. Also, regarding https://wiki.asterisk.org/wiki/display/AST/Asterisk+Sounds+Submission+Process Any reason not to accept CC-BY-SA-4 for new submissions? Rationale: for the CC licenses ver. 3 there are multiple variants for several countries. For CC ver. 4 there is only a single international version. -- Tzafrir Cohen +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Alias for country in indications.conf
Also, On Mon, Apr 23, 2018 at 04:08:58PM +1000, Patrick Wakano wrote: > Hello list, > Hope you all doing fine! > I've tried to use the 'alias' directive in the indications.conf file but > apparently it doesn't work > It looks like maybe this feature was removed, because old sample for the > indications.conf file have example using the alias parameter, but newer > samples don't have it anymore also I couldn't find any ticket saying > this parameter was deprecated Anyway when trying to use it, it doesn't > work. Anyone aware of some change related to this? > I am using Asterisk 13.6.0 and have this in indications.conf: > [uk] > *alias = gb* Given that aliases don't work, you can alternatively use: ;;; [uk] description= ringcadence= ... [gb](uk) ;;; -- Tzafrir Cohen +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Strange problem with PRI on 64-bit?
On Wed, Apr 04, 2018 at 11:28:33AM +, Tony Mountifield wrote: > In article > , > Richard Mudgett wrote: > > > > The libpri makefile doesn't install things for 64 bit systems in the right > > place [1] without your help. You'll need to specify where to install the > > library on the command line for your system: > > > > sudo make install libdir=/usr/lib64 > > > > > > Richard > > > > [1] https://issues.asterisk.org/jira/browse/PRI-100 > > Ah, thanks. I did in fact discover the following 64-bit libraries were > installed into /usr/lib instead of /usr/lib64: > > 1. From DAHDI, libtonezone.so dahdi-tools 2.11 now uses autoconf. It still installs to /usr/lib or is it an older version? > > 2. From LibPRI, libpri.so > > 3. From Asterisk, libasteriskssl.so > > I found that running "ldconfig" caused them all to be discovered: > > [root@bridge05 ~]# ldd /usr/sbin/asterisk > linux-vdso.so.1 => (0x7ffc77ff9000) > libasteriskssl.so.1 => /usr/lib/libasteriskssl.so.1 > (0x7efeae1d4000) > libc.so.6 => /lib64/libc.so.6 (0x7efeade4) > libxml2.so.2 => /usr/lib64/libxml2.so.2 (0x7efeadaed000) > libz.so.1 => /lib64/libz.so.1 (0x7efead8d7000) > libm.so.6 => /lib64/libm.so.6 (0x7efead653000) > libsqlite3.so.0 => /usr/lib64/libsqlite3.so.0 (0x7efead3c4000) > libssl.so.10 => /usr/lib64/libssl.so.10 (0x7efead158000) > libcrypto.so.10 => /usr/lib64/libcrypto.so.10 (0x7efeacd73000) > libdl.so.2 => /lib64/libdl.so.2 (0x7efeacb6f000) > libpthread.so.0 => /lib64/libpthread.so.0 (0x7efeac952000) > libtinfo.so.5 => /lib64/libtinfo.so.5 (0x7efeac731000) > libresolv.so.2 => /lib64/libresolv.so.2 (0x7efeac517000) > /lib64/ld-linux-x86-64.so.2 (0x7efeae3d6000) > libgssapi_krb5.so.2 => /lib64/libgssapi_krb5.so.2 (0x7efeac2d3000) > libkrb5.so.3 => /lib64/libkrb5.so.3 (0x7efeabfec000) > libcom_err.so.2 => /lib64/libcom_err.so.2 (0x7efeabde8000) > libk5crypto.so.3 => /lib64/libk5crypto.so.3 (0x7efeabbbc000) > libkrb5support.so.0 => /lib64/libkrb5support.so.0 (0x7efeab9b1000) > libkeyutils.so.1 => /lib64/libkeyutils.so.1 (0x7efeab7ae000) > libselinux.so.1 => /lib64/libselinux.so.1 (0x7efeab58f000) > [root@bridge05 ~]# ldd /usr/sbin/dahdi_cfg > linux-vdso.so.1 => (0x7fff6cbaa000) > libtonezone.so.2 => /usr/lib/libtonezone.so.2 (0x7f862f74a000) > libpthread.so.0 => /lib64/libpthread.so.0 (0x7f862f52d000) > libm.so.6 => /lib64/libm.so.6 (0x7f862f2a9000) > libc.so.6 => /lib64/libc.so.6 (0x7f862ef15000) > /lib64/ld-linux-x86-64.so.2 (0x7f862f97e000) > [root@bridge05 ~]# ldd /usr/lib/asterisk/modules/chan_dahdi.so > linux-vdso.so.1 => (0x7ffe8b1df000) > libtonezone.so.2 => /usr/lib/libtonezone.so.2 (0x7f54adde4000) > libpri.so.1.4 => /usr/lib/libpri.so.1.4 (0x7f54adb68000) > libpthread.so.0 => /lib64/libpthread.so.0 (0x7f54ad94b000) > libc.so.6 => /lib64/libc.so.6 (0x7f54ad5b7000) > libm.so.6 => /lib64/libm.so.6 (0x7f54ad333000) > /lib64/ld-linux-x86-64.so.2 (0x7f54ae2d3000) > [root@bridge05 ~]# > > So I assumed that all should be ok, otherwise the executables would fail to > run > (I initially discovered this when dahdi_cfg couldn't find libtonezone). > > Would there be any subtle issues with the 64-bit libraries being loaded > from /usr/lib instead of /usr/lib64? > > Should Asterisk and DAHDI builds also be updated to use /usr/lib64 when > building on a 64-bit OS? Or the build instructions? dahdi-tools: not AFAIK. -- Tzafrir Cohen +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with DAHDI
:28] WARNING[3428]: cel_tds.c:557 load_module: cel_tds module > had config problems; declining load > [Feb 15 18:43:28] NOTICE[3428]: cel_custom.c:97 load_config: No mappings > found in cel_custom.conf. Not logging CEL to custom CSVs. > [Feb 15 18:43:29] ERROR[3428]: codec_dahdi.c:820 find_transcoders: Failed to > open /dev/dahdi/transcode: No such file or directory > Asterisk Ready. > > it does not seems to be normal, but I can't understand why /dev/dahdi/channel > does not exists... > I installed the Paket asterisk-dahdi, of course... If /dev/dahdi/channel itself does not exist, it means that the kernel-level support is not loaded (or not even configured). It is generally from the kernel module dahdi, which is an out-of-tree one. If you really want it, you may need to run: m-a a-i dahdi But do you really have a DAHDI device? -- Tzafrir Cohen +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] chan_oss.c: Unable to register channel type 'OSS'
On Thu, Feb 15, 2018 at 08:38:03AM +, Luca Bertoncello wrote: > Zitat von Tzafrir Cohen : > > Hi, > > > Off-topic: any reason you don't use chan_alsa? > > This was the "Armbian installation", I didn't configured it extra... > > > Are you sure you quote the error message right? > > Copy+Paste... ;) > > But I searched a little bit and I really don't think, I need this module... > As I undestand, I just need it, if I want to call/answer call using the > console, and I really don't need this... > > Or I understood wrong? Yes. It is useful if you want to call using a local sound device. Consider editing /etc/asterisk/modules.conf and disable ('noload =>') chan_oss.so . -- Tzafrir Cohen +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] chan_oss.c: Unable to register channel type 'OSS'
Hi, On Thu, Feb 15, 2018 at 07:45:00AM +, Luca Bertoncello wrote: > Hi list! > > Currently I use Asterisk 1.8.30.0 on an OpenWRT-Switch. > Now I want to change to Asterisk 13.14.1 on a Banana PI (with Armbian/Debian > 9). > Well, I copied the configuration and changed what needed, so basically, it > works, at least with my tests. Off-topic: any reason you don't use chan_alsa? > > But when Asterisk will be started, in the message log I get this error: > > [Feb 15 08:40:15] ERROR[3971] chan_oss.c: Unable to register channel type > 'OSS' > > Unfortunately I cannot find WHY Asterisk was unable to register a channel > type "OSS". > And then: do I need this? On the old Asterisk I didn't had that... Huh? chan_oss registers the channel type 'Console' . I just tried it on a rpi with a somewhat similar deb, and chan_oss had no problem loading and registering channel type Console. I'm not aware of a patch to do so in Debian asterisk package and I don't suppose Armbian maintain their own Asterisk package. Are you sure you quote the error message right? Maybe the error message is: Unable to register channel type 'Console' which may be because either chan_alsa or chan_console was already loaded? Of those three: * I suppose chan_alsa should work better than chan_oss * chan_console has nice support of multiple devices, but is completely broken in your specific version[1] [1] https://issues.asterisk.org/jira/browse/ASTERISK-27426 , and thanks Sean Bright -- Tzafrir Cohen +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to add SNMP support to packaged asterisk on Debian stretch
On Fri, Feb 09, 2018 at 03:55:40PM +0100, Olivier wrote: > Hello, > > If I'm not mistaken SNMP support is missing in Debian Stretch packaged > Asterisk while this support is present in either Jessie or Buster (looking > at [1] or equivalent pages). > > Is it something that can be worked around or shall I fear a major obstacle > when re-packaging my own asterisk package for Stretch ? >From the changelog (linked fro https://packages.debian.org/stretch/asterisk-modules ) * Disable SNMP support for now libsnmp-dev pulls in libssl1.0-dev, which is not coinstallable with libssl-dev needed by Asterisk and all other dependencies I don't remember how difficult it would be to work around. In the worst case, try backporting libsnmp from Buster. -- Tzafrir Cohen +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] opus from git : install questions
On Sun, Feb 04, 2018 at 03:15:02PM -0500, sean darcy wrote: > > On 13.9.0 > https://github.com/traud/asterisk-opus > > The README: > > Alternatively, you can use the Makefile of this repository to create just > the shared libraries of the modules. That way, you do not have to (re-) make > your whole Asterisk. > > The Makefile generates: > codecs/codec_opus_open_source.so > formats/format_ogg_opus_open_source.so > formats/format_vp8.so > res/res_format_attr_opus.so See, e.g. the Debian package asterisk-opus: https://packages.debian.org/source/sid/asterisk-opus That package builds the opus modules as stand-alone modules out of the tree of asterisk. It has a build-time dependency on the binary package asterisk-dev. The list of files in it: /usr/lib/asterisk/modules/codec_opus_open_source.so /usr/lib/asterisk/modules/format_ogg_opus_open_source.so /usr/lib/asterisk/modules/format_vp8.so IIRC res_format_attr_opus.so in Asterisk proper is by now good enough and needs no patching. > > Without any of the patches the asterisk build generates: > > codec_opus.so > > format_ogg_opus.so Not by default. > > res_format_attr_opus.so > > Questions: > > Should the *_opus_open_source.so be in modules with the *opus.so libraries > from the asterisk build ? If not, do I build asterisk without selecting opus > ? If they are in modules, how does asterisk know which library to use ? You can either copy them to the Asterisk build directory (I do that in another Asterisk package I maintain). The build system will just pick them up and use them. Alternatively, as suggested in the README above, build them outside of the Asterisk tree (you'll need to point them to the asterisk source tree, or at least to the installed asterisk header files and copy the resulting .so files to the astmoddir (e.g. /usr/lib/asterisk/modules). > > The res_format_attr_opus.so library ? > > And another question, to enable PLC do I need to apply the patch even if I'm > installing the libraries directly ? Which patch specifically? I'm not sure I follow. > > Thanks for all the work getting opus to work on asterisk. -- Tzafrir Cohen +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Installation instructions for Opus are incorrect - maybe?
On Sat, Jan 27, 2018 at 05:25:48PM +, Jonathan H wrote: > Hmm, as it's free and open, I wonder why opus isn't a core codec? Digium seems to have some concerns regarding patent issues with Opus. As such it will not distribute Opus. Rather, it will only distribute a binary build of such a codec, that comes with a number of strings attached, even thought a price tag for the end user is not one of them. I am not a lawyer. I don't know of any specific secret information Digium's lawyers may know about specific patents that may be an issue here. I'll just point out that many companies, including the one I work for the company I work for see no issue with using it and find no patent issue here. As such we use an external patch. See https://github.com/traud/asterisk-opus . Packages of Asterisk in any major distribution include this patch or equivalents. (Same should go for wp8 and wp9, and av1 in the future, if you care about video) -- Tzafrir Cohen +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] remote Asterisk console
On Tue, Jan 16, 2018 at 11:05:01AM +0100, Paul Neuwirth wrote: > Hello group, > > what is the preferred method to connect to asterisk cli over network? I > need to run asterisk cli commands remotely. As others have mentioned: the manager interface is normally better for running over network. The manager interface also has an action calld 'Command' that runs a CLI command. In fact, contrib/scripts/astcli uses it to allow providing a remote console. Permissions needed for your manager user: For most things just: write=command To also be able to originate calls: write=command,originate To also be able to restart / reload: write=command,system > Sharing the unix socket through NFS, if that's working? No. > Or any other approaches, despite using SSH or rlogin, rsh. SSH: should work, sure. However, it means you ssh to root at the remote host. Better set a key with 'command' explicitly set in authorized_keys for this. Rlogin, rsh: seriously? Anybody still uses those? Not only are they way less secure than SSH, they are also way less conveninet than any decent SSH implementation. Anyway, as mentioned before: you should probably use AMI. -- Tzafrir Cohen +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can't compile Asterisk on Fedora server
On Wed, Jan 10, 2018 at 10:28:42AM -0500, Tech Support wrote: > > > All; > > I have a Fedora 26 server that I am trying to compile > asterisk-certified-13.13-cert6 on. However, I'm getting the following > errors. I'm also having a tough time trying to compile Dahdi. I'm not sure > what I'm missing, but if anyone else is running Fedora, I'd really > appreciate any help at all. > > Thanks Much; > > John V. > > > > make[1]: Leaving directory > '/usr/src/asterisk-certified-13.13-cert6/menuselect' > >[CC] tcptls.c -> tcptls.o > > tcptls.c: In function 'tcptls_stream_close': > > tcptls.c:401:20: error: dereferencing pointer to incomplete type 'SSL {aka > struct ssl_st}' > > if (!stream->ssl->server) { > > ^~ When I want to bisect something, I know that if I need to go back far enough, I need ./configure --without-ssl # :-( Asterisk 13.14.0 includes basic OpenSSL 1.1.0 support. I have no idea if anybody wants to backport it. Look at the log of main/tcptls.c in branch 13 in git to see the relevant patches. I suppose hopefully they'll apply cleanly. -- Tzafrir Cohen +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can't compile Asterisk on Fedora server
On Wed, Jan 10, 2018 at 10:28:42AM -0500, Tech Support wrote: > I'm getting the following > errors. I'm also having a tough time trying to compile Dahdi. I'm not sure > what I'm missing, Two recent issues: https://issues.asterisk.org/jira/browse/DAHLIN-356 An issue with kernel >= 4.13 . Now fixed in git (patch by Jean-Denis Girard). https://issues.asterisk.org/jira/browse/DAHLIN-359 An issue with kernel >= 4.15 . There is a patch there that fixes building but is not tested. Please test and report. -- Tzafrir Cohen +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] General Kernel practices on CentOS
On Wed, Dec 20, 2017 at 05:26:14PM +0200, Tzafrir Cohen wrote: > On Wed, Dec 20, 2017 at 10:20:11AM -0500, Eric Wieling wrote: > > > > That only applies to DAHDI, not Asterisk. > > > > I add exclude=*kernel* to /etc/yum.conf so the kernel doesn't get upgraded > > accidentally and break DAHDI. > > The Ubuntu (though not the Debian) dahdi package [irrelevant stuff > follows] Oops. Sorry for the irrelevant reply. I confused this with the other thread. -- Tzafrir Cohen +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] General Kernel practices on CentOS
On Wed, Dec 20, 2017 at 10:20:11AM -0500, Eric Wieling wrote: > > That only applies to DAHDI, not Asterisk. > > I add exclude=*kernel* to /etc/yum.conf so the kernel doesn't get upgraded > accidentally and break DAHDI. The Ubuntu (though not the Debian) dahdi package uses dkms and thus will automatically build as you install (install? boot to the?) new kernel. -- Tzafrir Cohen +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk and Hyper-V
On Mon, Dec 18, 2017 at 10:22:33AM +, Kseniya Blashchuk wrote: > To be honest we are a bit afraid to set 100% )), but we have tried to set > 90% - no luck. I have also tested with 4.8 and 4.11 kernels - same results. > I will try with Centos 6 and kernel 3.10 to check if something changes. > VSwitch shows 1-2% load on the interfaces, and this host is not overloaded > at all, so I don't think the VM has some lack of resources. What indication do you have that the problem is with the kernel or within the system? If you call from Asterisk to itself (with no networking involved), is there still distortion? Consider making a conference of several local channels (Echo, Playback, and whatever), and record whatever channel. Ubuntu has a "lowlatency" kernel. Does it matter if you use that variant? -- Tzafrir Cohen +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] General Kernel practices on CentOS
On Wed, Dec 20, 2017 at 03:30:50PM +0500, Abdul Basit wrote: > Olivier > > If you installed asterisk from source, you need to recompile it after > kernel version upgrade. > > This will compile & install asterisk modules with latest installed kernel > sources. Asterisk needs nothing from the specific version (certainly not with respect to minor version changes). -- Tzafrir Cohen +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can't install package asterisk-dbgsym on Stretch [SOLVED]
On Fri, Dec 15, 2017 at 02:35:14PM +0100, Olivier wrote: > Hi, > > 2017-12-14 16:28 GMT+01:00 Tzafrir Cohen : > > > 2. Is correct to understand that to get DONT_OPTIMZE, BETTER_BACKTRACE > > and > > > so on options compiled in, I must recompile anyway ? > > > > Right. DONT_OPTIMZE has a considerable performance impact. > > > How would you roughly evaluate this performance impact ? > I don't want to CREATE issues with performance penalties but I think I > can't afford to see Asterisk segfaults (a customer of mine has this with > PJSIP) within having required data to open tickets. DONT_OPTIMZE means building with hardly any compiler optimizations. Last time I asked at #asterisk-dev they agreed with me it is not sensible to ship such binaries. > > I never > > considered BETTER_BACKTRACE and its performance impact. Is it > > independent of DONT_OPTIMZE? > > I just did. Indeed it seems to be independent. I fail to see any real run-time impact of this if not explicitly used. I must have missed something, as otherwise it would have been surely enabled by default. What did I miss? -- Tzafrir Cohen +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Explain how to maintain a compiled from source Asterisk instance ?
On Fri, Dec 08, 2017 at 04:08:02PM +0100, Olivier wrote: > Hello, > > When compiling Asterisk from source, the classical ./configure, make and > make install commands are issued. > > > If a vulnerabilty is found within Asterisk code, then Asterisk source code > is patched and depending on what files were touched parts or all of above > commands need to be re-issued. > > What should be done for Asterisk runtime dependencies ? > > Which of the following sentences is or are correct ? > > 1. All Asterisk runtime dependencies are delivered as .so files. Is this > correct ? In standard Linux distributions, yes. > 2. I don't need to re-configure, re-compile or even re-start asterisk when > a such .so file is updated Hopefully, not reconfigure / recompile. Generally on properly-maintained distributions those libraries will change their name (SONAME) if they have non backward-compatible changes. In such a case you need to rebuild. This is why there's a number after the .so . As mentioned before, you need to restart, because otherwise you still use the original version of the library. -- Tzafrir Cohen +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can't install package asterisk-dbgsym on Stretch
On Fri, Dec 08, 2017 at 06:11:47PM +0100, Olivier wrote: > Hello, > > On a fresh Debian Stretch setup, I have: > $ cat /etc/apt/sources.list.d//dbgsym.list > deb http://debug.mirrors.debian.org/debian-debug/ stretch-debug main > > # apt-get update > ... > # apt-get install asterisk gdb > > # apt-get -s install asterisk-dbgsym > ... > asterisk-dbgsym : Depends: asterisk (= 1:13.14.1~dfsg-2+deb9u1) but > 1:13.14.1~dfsg-2+deb9u2 should be installed You seem to mix two versions. apt-cache policy asterisk apt-cache policy asterisk-dbgsym > > (above output is translated from the output I had) (Hint: LC_ALL=C ) > > > 1. Is this a bug in debian-debug repo ? If positive, should I file a bug > report ? rmadison shows me now that the versions of asterisk is 1:13.14.1~dfsg-2+deb9u2 in both the stable and the stable-debug repos. > > 2. Is correct to understand that to get DONT_OPTIMZE, BETTER_BACKTRACE and > so on options compiled in, I must recompile anyway ? Right. DONT_OPTIMZE has a considerable performance impact. I never considered BETTER_BACKTRACE and its performance impact. Is it independent of DONT_OPTIMZE? -- Tzafrir Cohen +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Adding custom commands to AMI
On Sun, Nov 12, 2017 at 04:45:45PM +, Antony Stone wrote: > Hi. > > https://www.voip-info.org/wiki/view/Asterisk+manager+API says that "There are > a finite (but extendable) set of actions available to the client, determined > by > the modules presently loaded in the Asterisk engine." > > Can anyone point me at some appropriate documentation for adding custom > commands to the AMI to extend the available actions? Generally: write your own asterisk module (in C), build and install it. -- Tzafrir Cohen +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How tu run runtests.py on Debian Stretch ?
On Tue, Oct 31, 2017 at 12:07:57PM +0100, Olivier wrote: > Hello, > > I'm giving asterisk-testsuite package a try on a fresh Debian Stretch setup. > > I've got this: > # /usr/share/asterisk-testsuite/runtests.py > Traceback (most recent call last): > File "/usr/share/asterisk-testsuite/runtests.py", line 24, in > from asterisk.version import AsteriskVersion > ImportError: No module named asterisk.version > > 1. Which documentation shall I look for ? > I've found [1] > > > 2. How shall I run runtests.py ? As asterisk user ? As root ? asterisk-tests-run That said, that package needs more work. -- Tzafrir Cohen +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dahdi get latest
On Wed, Oct 18, 2017 at 01:35:34PM -1000, Jean-Denis Girard wrote: > Le 18/10/2017 à 02:11, Jerry Geis a écrit : > > I am trying to use dahdi complete 2.11.1 with a 4.13 kernel. - NOT > > working for know reasons. > > I tried applying two patches but still get compile errors. AHHH! > > > > How do I just use git to get the latest with the fixes > > > > This command did not work - I still get the errors. > > git clone git://git.asterisk.org/dahdi/linux > > <http://git.asterisk.org/dahdi/linux> dahdi-linux > > Hi Jerry, > > Maybe you missed this patch: > https://issues.asterisk.org/jira/browse/DAHLIN-356 > > Or you can try my fork: > git clone -b next https://github.com/sysnux/dahdi-linux.git Thanks for the fix! As mentioned in that bug report, that patch missed a few minor things. See the -v3 patch there. -- Tzafrir Cohen +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk.conf ignored?
On Fri, Jun 30, 2017 at 03:15:21PM +0200, Stefan Viljoen wrote: > Hi all > > I'm trying to limit the maximum concurrent calls on my Asterisk to try and > mitigate another problem I posted about earlier. > > I've edited /etc/asterisk/asterisk.conf > > And uncommented this line, and put a value of 60 in there: > > maxcalls = 60 > > in an effort to limit my Asterisk to 60 simultaneous calls. > > I did a > > core reload > > in the CLI after doing that. > > Any idea why my running instance totally ignores this setting? I still goes > right ahead and services unlimited numbers of simultaneous calls - we have > 90 extensions or so and it will happily service 90 simultaneous calls in > spite of asterisk.conf clearly stating I suppose asterisk.conf is not read on a reload. IIUC it is read before the rest of the configuration. Another small thing that makes it slightly different: In any other configuration file you can either have '#include relative/path' or '#include /absolute/path' to include files. A relative path would be relative to $astetcdir. However when reading asterisk.conf, $astetcdir is not set yet, and thus '#include relative/path' would generally not work as expected. -- Tzafrir Cohen +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] phpagi packages
Hi all, We packaged phpagi for Centos 7 and Debian 8 (though nothing version-specific in those packages, I suppose). Packaging: http://git.xorcom.com/cgit/rpm/phpagi.git/ Packages: * RPM: http://updates.xorcom.com/servers/ombutel/ * Deb: should soon be in http://updates.xorcom.com/servers/spark/ That said, packaging there has indeed been rather trivial. The files are taken from https://github.com/welltime/phpagi/ that has some 20 so commit commits on top of the version from https://phpagi.sourceforge.net/ (sadly that git repository started from an import of the files and does not preserve the history of the Subversion repository[1]). There is a separate fork from Sangoma / FreePBX (https://github.com/sangoma/phpagi/ ), but it has changed more things and thus AGIs written to use the original will fail. I just wonder how many people do use this, given the staggering rate of changes in it. [1] That includes some 20 or so commits of its own, and begins with an import of the files from CVS, ignoring the history in the CVS repository. -- Tzafrir Cohen +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CentOS7: How to debug SEGV when asterisk starts with autoload=yes ?
On Mon, Jun 12, 2017 at 10:36:21AM +0200, Olivier wrote: > Hello, > > I was tasked to install Asterisk 13.16.0. from source on a CentOS7 platform. > > For that purpose, I used an unmaintened script of mine, written 10 monthes > ago, and I was surprised to get segmentation violations whenever I ran > "asterisk -cvvv -U asterisk". > > Usually, my /etc/asterisk/modules.conf file includes "autoload=yes" setting. > > Basically, I see two alternative methods: > > 1. leave "autoload=yes and remove modules one by one in modules.conf. > When segmentation violations stops, then focus on latest disabled module. Never a good idea. Disable half for starters. Then disable/enable half of the remaining, etc. > > 2.set "autoload=no and add modules one by one in modules.conf. > When segmentation violations starts, then focus on latest enabled module. 3. strace -eopen asterisk -U asterisk -c IIRC the message that Asterisk prints for loading a module is only printed after the module is loaded, and hence you need strace. -- Tzafrir Cohen +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CM for menuselect choices
On Fri, May 05, 2017 at 11:21:20AM -0400, Richard Kenner wrote: > I'd like to be able to save the choices made in menuselect in a way > that they can be tracked in a CM system and applied to a later release > of Asterisk using an automated tool like Ansible. What's the best > way to do that? Use menuselect's command line (--enable and --disable). Note that this requires an extra build stage: $MAKE menuselect.makeopts ./menuselect/menuselect \ --enable foo \ --disable bar \ # Alternatively, patch the sources of Asterisk to have foo enabled and bar disabled. This should be simple if you maintain your own stack of patches anyway. Examples: --- a/addons/res_config_mysql.c +++ b/addons/res_config_mysql.c @@ -24,7 +24,6 @@ /*** MODULEINFO mysqlclient - no extended ***/ and: --- a/sounds/sounds.xml +++ b/sounds/sounds.xml @@ -10,7 +10,6 @@ core - yes core @@ -246,7 +245,6 @@ - yes core You can also add 'defaultenabled' to set the default, if needed. menuselect.makeopts is not a file to keep as it is a generated file that is overly verbose and breaks all too often (on a change of version. And also potentially on a change of configure options?). I tried in the past to replace menuselect. In my replacement it had a simple configuration file (build_tools/conf) that cuild be easily hand edited. See menuselect/contrib/menuselect-dummy . However, it takes effort to keep it up-to-date with menuselect, and I never bothered for quite some time. -- Tzafrir Cohen +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] log incoming calls without answering
On Thu, Apr 20, 2017 at 05:51:59PM -0300, Fabio Moretti wrote: > Il 20/04/2017 17:32, kevin.lar...@pioneerballoon.com ha scritto: > > > > This gets kinda Rube Golberg-ish, but convert the incoming analog line > > to sip, route it through asterisk and have asterisk do its thing > > before converting it back to analog to send to the phone. Only problem > > is you get a lot of extra hardware involved in the mix to make it > > work. It will be a lot of expense and trouble, so you need to make > > sure that whatever part you want asterisk to play is worth that > > effort. Also, I wouldn't touch a fax line in this manner. > > > > If you could give a bit more info on what you want asterisk to do, we > > could maybe give better advice on how to solve your problem. > > Hi Kevin, > > I've already proposed your solution (is the most reasonable) but they > have more than 60 analogs lines (no faxes) and some of them terminate in > appliances like alarms, etc, so the solution must not touch in any way > the connection between the line and his termination: doing a analog to > digital conversion, passing it to asterisk and the convert it back to > analog is prone to problems (what if asterisk crashes? or if a gateway > fail?). > I can split the existing lines (there are no complex things like adsl or > digital signaling), convert the branches to digital and terminate then > into an asterisk machine, so any failure will not affect the old > circuit, but of course I've to configure asterisk to ONLY LOG calls and > nothing more. > > This is what they want: > - line 1 ring > - line 1 is splitted in two, the first branch (let's say the "analog" > branch) go to an analog phone, that rings > - the second branch go through a gateway and then to asterisk > - asterisk log (with an AGI for example) "line 1 rings at from " Simple dialplan. Depending on the type of caller ID system, you may need to wait a few seconds (in case the caller ID is sent after the first ring). Thus, assuming you have a DAHDI device, your dialplan is: exten => s,1,Wait(5) ; check how much and if waiting is needed same => s,n,NoOp(Caller ID is ${CALLERID(num)} on DAHDI channel ${CHANNEL(dahdi_channel)}) And move on to report from there. If you also need to report the total time of the call: that might be possible if the remote side reverses polarity of the channels on call start and end. Information about it is currently only reported in debug messages by chan_dahdi. So it is possible (given polarity reversal), but tricky. -- Tzafrir Cohen +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Backport of Stretch's asterisk.service file into Jessie: successful start not detected by systemd
On Thu, Apr 20, 2017 at 12:02:36PM +0200, Olivier wrote: > Hello, > > I've been tasked to enable automatic Asterisk restart on failure on a > Jessie platform (running latest Asterisk 13.15.0). > > I build a dedicated Jessie VM on which I installed Asterisk from source. > I configured a couple of files in /etc/asterisk directory. > I positively checedk that with simple config, Asterisk could sucessively > start using an /etc/init.d/asterisk file and sysv-systemd compatibility > tools. > > Then I copied a /etc/systemd/system/asterisk.service file with the > following content: [snip] > [Service] > Type=notify [snip] > After creating above asterisk.service file, running "systemctl > daemon-reload" and "systemctl start asterisk", I could observe: > - asterisk is starting OK, > - asterisk prints to its log file a line such as: > [Apr 20 11:23:22] VERBOSE[770] asterisk.c: Asterisk Ready. > > but, at the same time, I got: > april 20 11:23:22 jessievm asterisk[770]: [Apr 20 11:23:22] NOTICE[770]: > app_queue.c:9095 reload_queues: No call queueing config fil > april 20 11:24:48 jessievm systemd[1]: asterisk.service start operation > timed out. Terminating. > april 20 11:24:48 jessievm asterisk[770]: > april 20 11:24:48 jessievm systemd[1]: Failed to start Asterisk PBX. 'Type=notify' means that the service is only considered ready once is notifies systemd so through a special socket, using sd_notify(3). You told systemd that Asterisk will notify it when ready, but Asterisk never did. Thus systemd decided that service has failed. Make sure you have libsystemd-dev installed. Alternatively, remove that line (to keep the default Type=simple). When built with systemd support, asterisk updates systemd about its status using sd_notify(). This means that it is only considered as started when it has loaded all the modules. This can make startup ordering simpler. BTW: if anybody wants, you can use sd_notify to set the status (as shown in 'systemctl status asterisk.service') to an arbitrary string. Basically just use ast_sd_notify() anywhere in the code. It becomes a no-op if there's no systemd support. -- Tzafrir Cohen +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk as non root
On Wed, Apr 19, 2017 at 04:44:39PM +0300, Atux Atux wrote: > hello there. i am running debian 8 in my swerver and i would like to run > asterisk as non root. The Asterisk package included with Debian already does that. Why not have a look at it? > i did follow the > https://www.voip-info.org/wiki-Asterisk+non-root without any success. when > i issue > root@PBX: ~ $ asterisk -U asterisk -G asterisk The options -U and -G are for the case of running Asterisk as root and having Asterisk change user and group afterwards. There are a number of options that only work that way (real-time priority, special socket permissions, IIRC). Alternatively you can use other mans to change to that user (--chuid or start-stop-daemon or User: and Group: in a systemd service file, or whatever). And then you don't need those options. > Privilege escalation protection disabled! > See https://wiki.asterisk.org/wiki/x/1gKfAQ for more details. Read that text. But it is irrelevant for your situation. > Unable to access the running directory (Permission denied). Changing to '/' > for compatibility. /root is not accessible by the user asterisk. This is mostly harmless, but not if you want to have core files (see also -g) and maybe a few other minor things. > Asterisk already running on /var/run/asterisk/asterisk.ctl. Use 'asterisk > -r' to connect. Because you already ran that command before. Or already have the system copy of asterisk running. Or whatever. Reading error messages helps. -- Tzafrir Cohen +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PBX selection
On Mon, Apr 17, 2017 at 10:57:27PM +0800, Speed Boy wrote: > Hi all, I'm new to VoIP, now we have a project that needs a > PBX with client APPs. > In our team we have argument for choosing PBX. By so far, we > have following candidates: > > A: Open source > > 1) Asterisk PBX (http://www.asterisk.org) (with longest > history that almost every one knows it, now the last version using the > PJSIP stack) > 2) FreeSwitch (http://www.freeswitch.org) (A lot people > recommended it to us) > > > B: Commercial > > 1) Vodia PBX (http://www.vodia.com). It comes from SNOM, now > acquired by a HongKong company now > 2) PortSIP PBX (http://www.portsip.com/portsip-pbx). It > also includes VoIP SDK, WebRTC and offer rebranding app for free. > > My boss prefers the Open Source PBX since they are free, > but our CTO prefers the commercial editions, according to > whom the business PBX has better support, and the > performance is good, and easy to use - considering our team > all are new to VoIP/PBX. I answered elsewhere[1]. I'll just note one important point from my reply: Asterisk and FreeSwitch are not PBXs. They are telephony servers. One application you can build using them is a PBX. You can either program it yourself or use an existing one (e.g.: FreePBX for Asterisk). It's not clear from your question which of the two you need. To me personally the real advantage of open source is not the cost. It is the ability to tweak, and the control you retain. Right now you are new to VoIP. But that will soon change. [1] http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/2017-April/019929.html -- Tzafrir Cohen +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] restart system from extension
On Thu, Apr 06, 2017 at 08:16:34PM +0300, Atux Atux wrote: > hi. i would like to be able to reboot the system from my extension. is that > possible? if yes, how? System('sudo /sbin/reboot') You need to allow that in a sudoers file, of course. This may or may not be a good idea. There are a host of other methods to permit unplivilidged users / processes to run do specific priviliged actions. -- Tzafrir Cohen +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Commit dialplan & other config. in memory to disk?
On Thu, Apr 06, 2017 at 09:54:25AM +, Nathan Anderson wrote: > 'lo, > > So yesterday, one of our clients had the misfortune of having the disk that > their Asterisk config (*.conf) was stored on take a dirt nap. Of course, > Asterisk was still running at the time, and everything continued to work > (except for voicemail, which was stored on the same disk) right up until I > shut down Asterisk to investigate what was going on. Because the disk was > dead, though, I couldn't start Asterisk back up after that, and OF COURSE the > backups were not firing off correctly so now we are faced with regenerating > the config again (including dialplan) from scratch. > > In the future, if I were to ever run into a similar situation, is there any > way to request or instruct Asterisk to write the current dialplan that is in > memory and other important config files (e.g., users.conf) to disk in a > *different* location than where it originally read them from when it started > up? I could have saved myself a crap-ton of work if this were possible... I'm not sure this works. But it's worth a shot: (bind-?)mount a writable file system at /etc/asterisk . Be sure to umount it quickly enough after the write. -- Tzafrir Cohen +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Questions regarding asterisk-opus package in Debian Stretch repo and Opus in general
On Wed, Mar 29, 2017 at 06:04:49PM +0200, Olivier wrote: > Is there any relation between this external patch and the binary mentioned > in [2] > [2] http://blogs.digium.com/2016/09/30/opus-in-asterisk/ > > The later one mentions a binary-only distribution to comply with legal > constraints. No, it is not. This package is in Debian's main archive, which tells you it is not based on any binary blob. Opus is widely implemented in software, including free software (Firefox, Chromium, Linphone, Jitsi and a host of others). See also https://en.wikipedia.org/wiki/Opus_(audio_format)#Software My understanding is that to Digium's best legal advice, there are still patent issues with the Opus codec. Even though many others disagree (as evident from above) and I also happen to disagree. But I certainly am not the one who runs Digium. And the powers that be there probably decided that whatever patent issues there are, have merit and need to be mitigated. -- Tzafrir Cohen +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Questions regarding asterisk-opus package in Debian Stretch repo and Opus in general
On Wed, Mar 29, 2017 at 05:18:18PM +0200, Olivier wrote: > Hello, > > After reading [1] (in french), I would be very happy if I could get answers > to: > > 1. Does this 13.7+20161113-3 package version has any relation with > asterisk's version it complements ? Current asterisk version in repo is > 13.14.0. Does this 13.7 complies with it ? The opus codec was used as an external patch. It looked ugly and thus a separate package was preffered. Its version number is not directly related to Asterisk. It has originally been split from the Debian packaging of Asterisk, and starting from the same version number allowed easier upgrading. There is no version number for the upstream code (the patch). > > 2. From package description, is this package enough or not to allow > transcoding with G711 ? > For instance, in the following situation: > SIP Phone < Opus > Asterisk < G111 ---> ITSP Technically Asterisk codecs translate to/from (typically) linear and Asterisk combines codecs to do whatever transcoding needed. So the codec does not transcode directly to G.711. But Asterisk can transcode between opus and G.711. > > 3. Can you share here any personal field experience with this codec, for > home worker use case ? > Is there a better user experience with Opus than with G729 or G711 ? > > 4. Does it work on ARM boxes (Raspberry, ...) ? Should work just the same. > > > [1] https://packages.debian.org/stretch/asterisk-opus -- Tzafrir Cohen +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Running asterisk as non-root
On Tue, Mar 21, 2017 at 08:54:22PM -0400, Jerry Geis wrote: > I found a page that assisted in running asterisk as non-root. > > My question is why is the default install not more friendly in this > manner??? It requires either creating a user or making sure such a user exists. That's not a firendly thing to do in a typical 'make install'. It is quite the expected thing on just about any packaging of Asterisk, because there are relatively standard methods of creating users in the context of a specific distribution. > > I mean basically the instructions changed the permissions to allow GROUP > write, so 775 instead of 755 and changed the astrun to be a directory That assumes USER or GROUP exists. > > Why doesn't the default install do this? Seems logical to me. I installed > 11.25.1 If you're still not convinved, Patches are welcomed (for the master branch). -- Tzafrir Cohen +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to install and configure Dahdi from Debian Stretch repo ?
On Tue, Mar 21, 2017 at 05:18:50PM +0100, Steinwendtner wrote: > Hello Tzafrir, > > Am 2017-03-21 um 11:23 schrieb Tzafrir Cohen: > > On Tue, Mar 21, 2017 at 09:36:21AM +0100, Olivier wrote: > > > > I'm still having some questions: > > > > 1. I can't find any /etc/init.d/dahdi file in my newly built system so > > "service dahdi status" (or systemctl status dahdi) fails with: > > Unit dahdi.service could not be found. > > Shall I worry about this ? > > No. As I mentioned, that is meaningless. > So it is no longer possible to make changes on dahdi parameters > without rebooting the machine ? There should be no need for a reboot. Long live your system. > > I think in the early days it was possible something like this: > > make a change in /etc/dahdi/system.conf What kind of change? Technically it would only take running 'dahdi_cfg' to apply that change. However that may not be enough. Depending on the specific change. > asterisk -rx "module unload chan_dahdi.so" > service dahdi stop > service dahdi start > asterisk -rx "module load chan_dahdi.so" It only got faster. No need to deconfigure anything in Asterisk (This happens on its own. At least on Asterisk >= 13). No need to unload and re-load any modules (And why would you? Just to run dahdi_cfg?). But anyway, if merely running dahdi_cfg is not enough, try: dahdi_span_assignment remove dahdi_span_assignment add # or: auto This should also configure DAHDI and register spans with Asterisk (again: >= 13). If there are still problems with asterisk: asterisk -rx "dahdi restart" -- Tzafrir Cohen +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to install and configure Dahdi from Debian Stretch repo ?
On Tue, Mar 21, 2017 at 09:36:21AM +0100, Olivier wrote: > Thanks to Tzafrir help, Now I have an Asterisk-Dahdi system installed from > Stretch repository. > This system has a Digium HX8 card with BRI modules, Asterisk 13.14.0, Dahdi > 2.11.1. > BRI spans appears as up and active but I've not tested them yet: > CLI> pri show spans > PRI span 1/0: Up, Active > PRI span 2/0: Up, Active > PRI span 3/0: Up, Active > > > I'm still having some questions: > > 1. I can't find any /etc/init.d/dahdi file in my newly built system so > "service dahdi status" (or systemctl status dahdi) fails with: > Unit dahdi.service could not be found. > Shall I worry about this ? No. As I mentioned, that is meaningless. There are various commands that show the status at various levels: lsmod | grep ^wctdm24xxp dahdi_hardware | grep 'wctdm24xxp\+' dahdi_span_assignments list lsdahdi asterisk -rx 'pri show spans' What exactly do you consider as "status of dahdi"? > > 2. Where can Dahdi 2.11.1 Changelog file be found ? > In http://downloads.asterisk.org/pub/telephony/ ? git log #? The tarball has no changelog and thus it is not packaged. Sorry. -- Tzafrir Cohen +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Having problem getting Asterisk to work on CentOS 7
On Tue, Mar 14, 2017 at 02:46:19PM -0400, Ron Wheeler wrote: > https://docs.fedoraproject.org/en-US/Fedora/11/html/Security-Enhanced_Linux/sect-Security-Enhanced_Linux-Working_with_SELinux-Enabling_and_Disabling_SELinux.html > > If disabling Selinux solves your problem, then your problem may be related > to Selinux. > If it does not change yout problem, you may want to look elsewhere. > > It seems that a lot of things do not work with Selinux or have > no instructions about how to make them work with Selinux that it almost > seems like a useless feature. Many things work well, once properly configured. Looking at the exact error (again, audit.log) is the first step. Once upon a time Asterisk used to be able to run with SELinux: https://issues.asterisk.org/jira/browse/ASTERISK-3088 The problem may be missing a profile for Asterisk. Or the fact that it interacts too much with other services? I'll have to give it a shot. At least for a stand-alone Asterisk. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Having problem getting Asterisk to work on CentOS 7
On Tue, Mar 14, 2017 at 06:03:33PM +0100, Jean Aunis wrote: > Hello, > > Did you disable selinux ? It usually causes troubles when starting asterisk > as a service. You can do this with : setenforce 0 (this will not totally > disable selinux, but switch it to a permissive mode). Generally before advising that, check if this is the error: tail -f /var/log/audit/audit.log and try the command. Is there any open bug for a security policy for Asterisk? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Having problem getting Asterisk to work on CentOS 7
On Tue, Mar 14, 2017 at 05:09:17PM +, Dan Cropp wrote: > Thank you Tzafrir. > > I had been using different users in earlier attempts to make this work. > Decided to try everything where root is the only user, simply to verify it's > working. > > For problem 2, where asterisk is writing to the log but doesn't seem to > receive the SIP packets even though tcpdump indicates they are making it to > the box on 5060, I am starting asterisk while logged in as root. > /usr/sbin/asterisk -dddc > > > For problem 1, where it seems to be stuck when running as a service, I simply > reboot the machine. Then I log it as root and notice it's not writing to the > log. > > When running it as a service (after restart). Here is what the output from > strace -p $PID_OF_ASTERISK > > [root@localhost ~]# strace -p 1470 pkill? nice? That is not asterisk. Are you sure you got the right process? Maybe you got safe_asterisk instead? If it is safe_asterisk: 1. That script is pointless now that you have systemd. Replace it with a simple systemd unit (hint: Restart=on-failure gets you most of the way there). Isn't there one already included with Asterisk by now? 2. Use the option -f of strace to see the exact error message. What is error status 34 of asterisk? ERANGE? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Having problem getting Asterisk to work on CentOS 7
On Tue, Mar 14, 2017 at 04:45:38PM +, Dan Cropp wrote: > After that, I modify the asterisk conf files for a couple pjsip endpoints and > turn on debugging and verbosity. Copying settings from another box which is > working. > > I am seeing two different issues > > First, when I restart the box, the asterisk process is present. However, > it's not writing anything to the log files so it seems to be stuck. Any idea > why running Asterisk as a service after the make config would not seem to > fully start up? You mentioned starting asterisk manually. Do you do that with the proper permissions (assuming you do run Asterisk as a non-root user, as you should). Any chance asterisk does not have write permissions to the log files? Failing that, try strace. strace -p $PID_OF_ASTERISk Maybe also add -f and / or -o if there's too much output. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to install and configure Dahdi from Debian Stretch repo ?
On Tue, Mar 14, 2017 at 02:58:07PM +0100, Olivier wrote: > 2017-03-14 13:08 GMT+01:00 Tzafrir Cohen : > > > On Tue, Mar 14, 2017 at 11:10:57AM +0100, Olivier wrote: > > > Hello, > > > > > > After all these years installing from source, I'm giving Dahdi package > > > installation a try on a recent Stretch box. > > > > > > Google over the web, I didn't find too many doc on this topic. > > > > > > 1. Is this one [1] up-to-date ? > > > > Not exactly. > > > > > Reading Stretch I would say a single asterisk-dahdi would be enough to > > > install asterisk dahdi and libpri. > > > > Almost. Except the kernel modules. For those: > > > > apt install dahdi-source > > # also install module-assistant, if it wasn't installed already > > m-a a-i dahdi > > # Should also install linux-headers-`uname -r`, IIRC > > > > > > > > 2. On my box, the following fails. What would you suggest ? Re-base > > > everything on Jessie ? > > > apt-get install linux-headers-`uname -r` > > > > That should not happen. What is the output of uname -r #? > > > > # uname -r > 4.8.0-2-686-pae > > # apt-get install linux-headers-`uname -r` > ... > E: Couldn't find any package by glob 'linux-headers-4.8.0-2-686-pae' > > # apt-cache show linux-headers-4.8 > ... > N: Couldn't find any package by glob 'linux-headers-4.8' > ... > > # apt-cache show linux-headers-4.9 > Package: linux-headers-4.9.0-2-686 > Source: linux > ... > > It looks like linux-headers-4.8 are currently missing in Stretch repo > though currently installed kernel is 4.8. > This issue seems quite independant from asterisk, anyway. You probably started working on this box a while ago. Streetch's current kernel is 4.9.0-2 (it changed shortly before the freeze). It is now frozen and won't change (barring a really good reason). > > > I started all over with a Jessie box. I recommend to go back to Stretch and just upgrade the kernel. In other words: keep the software up-to-date. > Relating to [1], I could positively run: > # m-a a-i dahdi > ... > > But # dahdi_genconf required a reboot to run OK. Certainly not. See below. > Though this worries me as I need to script the whole install process, I can > leave it aside at the moment. > > > > > > > > > 3. How is dahdi started-stopped in Stretch ? (I can't find any > > > /etc/init.d/dahdi file after apt-get install asterisk-dahdi). > > > > Started: should be automatically at boot. > > > Shall I find an /etc/init.d/dahdi file or equivalent ? > If positive which command produces this file ? > I would expect dahdi-linux or dahdi packages to install these file. > > > > Waht hardware device do you > > have? > > > > # dahdi_hardware > pci::04:05.0 wctdm24xxp+ d161:8007 HA8- Traditionally dahdi has been shipped with a modprobe.d configuration file to blacklist all of the PCI cards. This was because the order in which they were loaded was not well-defined. This is irrelevant on a system with a single device. And anyway, irrelevant in a system configured (as is in the Debian packages) with auto_assign_spans=0 . In such a system, devices register automatically, but their spans are not assigned span and channel numbers unless they are configured in /etc/dahdi/assigned-spans.conf . So try: dahdi_span_assignment auto dahdi_genconf Another reason that the init file was removed is because there is no need to assign DAHDI spans before asterisk is started: if the channels are configured with Asterisk, they will appear when assigned (using a script in /usr/share/dahdi/span_config.d/). -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to install and configure Dahdi from Debian Stretch repo ?
On Tue, Mar 14, 2017 at 11:10:57AM +0100, Olivier wrote: > Hello, > > After all these years installing from source, I'm giving Dahdi package > installation a try on a recent Stretch box. > > Google over the web, I didn't find too many doc on this topic. > > 1. Is this one [1] up-to-date ? Not exactly. > Reading Stretch I would say a single asterisk-dahdi would be enough to > install asterisk dahdi and libpri. Almost. Except the kernel modules. For those: apt install dahdi-source # also install module-assistant, if it wasn't installed already m-a a-i dahdi # Should also install linux-headers-`uname -r`, IIRC > > 2. On my box, the following fails. What would you suggest ? Re-base > everything on Jessie ? > apt-get install linux-headers-`uname -r` That should not happen. What is the output of uname -r #? > > 3. How is dahdi started-stopped in Stretch ? (I can't find any > /etc/init.d/dahdi file after apt-get install asterisk-dahdi). Started: should be automatically at boot. Waht hardware device do you have? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which tool to automatically restart Asterisk ?
On Mon, Feb 27, 2017 at 06:00:30PM +0500, Tahir Almas wrote: > Sorry , I forget it for another monitoring tool monit that we have > used in our production systems to restart asterisk in case of asterisk > crash or halt. [snip] Some notes regarding the asterisk monit configuration: > check process asterisk with pidfile /var/run/asterisk/asterisk.pid > group asterisk > start program = "/bin/bash -c 'ulimit -n 16386 && /etc/init.d/asterisk > start'" If you use systemd, this ulimit will have no effect: when you restart a service, it is restarted from a separate systemd context (cgroup) and not directly under your own. It would generalyl be a good idea not to embed such settings in your scripts and rather put them in a proper configuration file. What happens in you happen to run '/etc/init.d/asterisk restart'? It seems that all's well, until you're suddenly out of file descriptors. > stop program = "/etc/init.d/asterisk stop" > if does not exist for 2 cycles then restart > if failed port 5060 type udp protocol SIP > and target "011@127.0.0.1" maxforward 10 > for 2 cycles then restart > if failed host 127.0.0.1 port 5038 with timeout 15 seconds for 2 cycles > then restart > if 5 restarts within 5 cycles then timeout Nice. Also: what happens when you run 'core stop now' from within asterisk? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which tool to automatically restart Asterisk ?
On Mon, Feb 20, 2017 at 11:36:24AM -0300, Victor Villarreal wrote: > Hi, Oliver. > > Maybe something like this (add this script to your crontab): > > 8<-- > > #!/bin/bash > # > # File: asterisk-watchdog.sh > # Date: 2015.05.26 > # Build:v1.0 > # Brief:Secuencia para monitorizar procesos. > # > # ${PATH}: Variable de entorno con las rutas a los ejecutables. > PATH=/bin:/sbin:/usr/bin:/usr/sbin > > # ${DAEMON}: Demonio a monitorizar. > DAEMON="asterisk" > > # ${MSG}: Cuerpo del mensaje a enviar por mail. > MSG="$(date '+%F %T'): ${DAEMON} se ha caido!" > > pidof ${DAEMON} > /dev/null 2>&1 > > [ $? -ne 0 ] && { echo ${MSG}; service ${DAEMON} start; } > > exit 0 Both Debian 8 and Centos 7 have systemd. Systemd gives you this type of monitoring almost for free (see previous reply). Using cron is generally not a good idea here: 1. No way to stop Asterisk when you need it. 2. If Asterisk has failed, it may take up to a minute to restart it. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] compiling asterisk-14.3.0-rc2
On Sun, Feb 12, 2017 at 05:56:31PM +0100, Goke Aruna wrote: > Thanks. > The configure run successfully. > > but I got the warning below.. > > checking for the ability of -lsrtp to be linked in a shared object... no > configure: WARNING: *** > configure: WARNING: *** libsrtp could not be linked as a shared object. > configure: WARNING: *** Try compiling libsrtp manually. Configure libsrtp > configure: WARNING: *** with ./configure CFLAGS=-fPIC --prefix=/usr > configure: WARNING: *** replacing /usr with the prefix of your choice. > configure: WARNING: *** After re-installing libsrtp > configure: WARNING: *** configure script. > configure: WARNING: *** > configure: WARNING: *** If you do not need SRTP support re-run configure > configure: WARNING: *** with the --without-srtp option. If you excplicitly configure --with- and the configure script fails to find , it fails. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] compiling asterisk-14.3.0-rc2
On Sun, Feb 12, 2017 at 05:24:16PM +0100, Goke Aruna wrote: > hi all, > > can someone help? I have centos 6.8 trying to install asterisk 14.3.0-rc2 > on it with options as stated below - > ./configure --with-crypto --with-ssl --with-srtp=/usr/local/lib > --with-jansson=/ --with-pjproject-bundled Did it end successfully? > > when I tried to run "make menuselect". i get the error below. > > > Makefile:109: makeopts: No such file or directory > > The configure script must be executed before running 'make'. > Please run "./configure". > > make: *** [makeopts] Error 1 -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using g729 now that patents have expired
On Tue, Feb 07, 2017 at 04:55:02PM -0500, Carlos Rojas wrote: > Hi > > You can uses: > > http://asterisk.hosting.lv/ For the record: the coded module from there can be built with the library bcg729, and I believe it should be perfectly legal to use now (Unlike the Intel IPP, that will require a specific license due to copyright considerations). However the author is not seem willing to submit that code for inclusion into Asterisk. So if anybody wants to write a G.729 codec using that library: feel free to do so. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial() from the console?
On Wed, Jan 11, 2017 at 07:31:31AM -0500, Doug Lytle wrote: > >>> On Jan 11, 2017, at 7:20 AM, Thufir Hawat hawat.thu...@gmail.com wrote: > > >>> Can I dial directly from the asterisk console with the Dial() application? > > > console dial number@context Note, however, that it is a different thing. It uses a specific device: the "console". That is: the speaker and microphone of the system Asterisk is running on. For that to be available you need to have one of chan_alsa, chan_console or chan_oss loaded and working. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk installation script on CentOS7 with systemd [SOLVED]
On Mon, Dec 19, 2016 at 05:10:42PM +0100, Olivier wrote: > Thanks for the tip: > changing to permissive mode made it ! > > Using methods suggested in [1], do you think its possible and worth the > effort to configure SELinux to work with Asterisk/Systemd in Enforcing mode > ? > > [1] https://wiki.centos.org/HowTos/SELinux I think it should be possible. IIRC I once gave it a shot and was mildly successful, but eventually gave up due to issues related to interaction with Apache. If you do run into a problem, I wonder what it is. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk installation script on CentOS7 with systemd
On Mon, Dec 19, 2016 at 03:54:47PM +0100, Olivier wrote: > Hello, > > For a new project, I'm adapting existing installation script to CentOS 7. > I must admit I don't understand how to adapt things to systemd. > > Here are my questions: > > 1. I don't see any systemd sub-directory in asterisk-13.13.1/contrib. > Do you think such directory and matching Makefile target could be useful ? > > 2. Should /run/asterisk directory creation be left to systemd or done by > installation script before running "systemctl start asterisk" ? > > 3. I edited the following /etc/systemd/system:asterisk.service file: > [Unit] > Description=Asterisk PBX and telephony daemon. > After=network.target > > [Service] > Type=forking > PIDFile=/var/run/asterisk/asterisk.pid Remove those two (or get latest version with sd_notify support, make sure it works, and use 'Type=notify') > Environment=HOME=/var/lib/asterisk > WorkingDirectory=/var/lib/asterisk > ExecStart=/usr/sbin/asterisk -vvvgF -U asterisk -G asterisk -C Drop -F as well > /etc/asterisk/asterisk.conf > #ExecStart=/usr/sbin/asterisk -vvvgF -C /etc/asterisk/asterisk.conf > ExecStop=/usr/sbin/asterisk -rx 'core stop now' I'm trying to think if this is needed. Anything wrong with just letting systemd kill asterisk and all of its child precesses? > ExecReload=/usr/sbin/asterisk -rx 'core reload' Also, IIRC: User=asterisk -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how are channels numbers assigned
On Thu, Dec 08, 2016 at 02:24:14PM -0200, Ethy H. Brito wrote: > > Hi All > > I have this system here where: > > # dahdi_hardware > pci::07:04.0 wctdm24xxp+ d161:8005 Wildcard TDM410P > pci::07:09.0 wcte11xp+e159:0001 Digium Wildcard TE110P T1/E1 Board > > channels are > 0-31 for TE110P (1-31) > and > 32-35 for TDM410P > > I want to insert a new TE110P. > How will these new channels be assigned? > What is the dahdi_genconf logic for these assignments? Your version of dahdi is 2.7.0, which is a bit of a shame. Slightly newer versions introduced span assignment: you can specify in /etc/dahdi/assigned_spans.conf to which span number and channel number will be used by each device. dahdi_genconf can generate a default one for you, but see also 'dahdi_span_assignment dump'. Which means you would not depend on a specific modue loading order. That aside, dahdi_genconf just uses whatever dahdi gives it. With automatic span assignments, the order of spans (and channels) is set by the order in which the kernel modules happened to load. > > Is there a way to force these assignments to prevent they to change if I > insert/remove any card from the system? > > chan_dahdi.conf is a bit confusing to me. > What makes the configuration for one channel be finished and start the > configuration for the next channel? > > Would you point me any docs on this matter? The best I have so far is the README of dahdi-tools. A copy of it is in http://docs.tzafrir.org.il/dahdi-tools/ > > Asterisk is 11.7.0 > Dahdi is 2.7.0 > Ubuntu 14.04.5 LTS -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What to do when changing from one asterisk version to another ?
On Thu, Dec 08, 2016 at 06:23:15PM +0100, Olivier wrote: > Hello, > > I'm compiling Asterisk from source on Debian systems. > > I'm currently writing a script I'm planning to launch when upgrading from > one Asterisk version to another one within the same class (from 13.4.0 to > 13.12.0 or from 13.12.0 to 13.8.0, for instance). > > Reading [1], I thought the following would work: > cd /usr/src/asterisk-13.4.0 > ./configure > make > make install > ... > cd /usr/src/asterisk-13.4.0 > make dist-clean > > After running above commands, /usr/sbin/asterisk and > /usr/lib/asterisk/modules/*.so files still exist. > I would expect both /usr/sbin/asterisk and /usr/lib/asterisk/modules/*.so > filesto be removed so that if I newly installed asterisk instance wouldn't > inherit uncontrolled files. If you package the result in a deb instead of directly installing it, you can make sure it is completely removed upon package removal. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] failing to start asterisk on centos7
On Sat, Dec 10, 2016 at 08:24:09PM +0200, christopher kamutumwa wrote: > ive installed asterisk but below is what am getting proces gets > killed.please help > > [root@localhost sounds]# asterisk -c > Asterisk 13.13.1, Copyright (C) 1999 - 2014, Digium, Inc. and others. > Created by Mark Spencer > Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for So now we have asterisk somewhere in the path. [snip] > Asterisk Dynamic Loader Starting: > == Parsing '/etc/asterisk/modules.conf': Found > [Dec 10 20:21:06] NOTICE[16058]: loader.c:1446 load_modules: 263 modules > will be loaded. > Killed Looks odd. My first instict would be to try to run this under strace. > [root@localhost sounds]# asterisk -c > -bash: /usr/sbin/asterisk: No such file or directory It's still in a cache for bash, right? But it's no longer there. Somebody deleted it? Or rather (Just a guess) - someone has been messing with the contents of the disk directly? Disk problems? > [root@localhost sounds]# asterisk -r > -bash: /usr/sbin/asterisk: No such file or directory > [root@localhost sounds]# asterisk > -bash: /usr/sbin/asterisk: No such file or directory > [root@localhost sounds]# -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Can't start with the default configs
On Sun, Dec 04, 2016 at 08:00:45PM +0100, Mr Dini wrote: > Hi, > > I tried to run the make progdocs, but the first time, it said, I have no > doxygen installed. So I compiled the latest release and reconfigure the > asterisk. And after it, ut sucessfully started to build the docs. But it > took a lot of time, So finally I aborted the process... > > Is there a way to disable doc creating? The --disable-xmldoc is enough? I must be missing something here. Isn't 'progdoc' an optional target? The relevant XML documentation is doc/core-en_US.xml, which is generated unconditionaly in the build and using awk, shell and make. Another note: it is installed to the astdatadir. Which is indeed by default var/lib/asterisk, but may have different values (build time or set at run time in asterisk.conf). -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk install challenge
On Tue, Dec 06, 2016 at 06:35:07PM +0200, christopher kamutumwa wrote: > am new to asterisk and am trying to "make all". or dahdi install but in > only reach this stage below on centos 6.8 . Any idea how to resolve or > bypass this > > > configure: creating ./config.status > ./configure: line 18858: cannot create temp file for here-document: No such > file or directory > configure: error: write failure creating ./config.status > make: *** [all] Error 1 Not really sure. But maybe you're out of disk space? Can you create a file? df -h . Failing that: permission issues? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] bash: asterisk: command not found
On Wed, Dec 07, 2016 at 09:23:30AM +, k...@mayten.sch.bme.hu wrote: > On 2016-12-07 09:13, Steve Howes wrote: > >On 07/12/16 04:56, christopher kamutumwa wrote: > >>Ive installed asterisk 14.2 on centos 6.8 but i am not able to start it > >>below is what am executing and those are the errors anything am doing > >>wrong? > > > >It doesn't look like it is installed to me... Check the install > >actually worked etc. I've never had to do any path changes or anything > >for asterisk on centos so I suspect it just isn't there... > > > > it could be either of two things. > > 1) asterisk not being in the $PATH, try launching it by using an absolute > path Right. Next time become root with 'su -' and not 'su'. Make sure /usr/sbin is in your PATH. > 2) it is, and you have it installed, but you have a 64 bit binary on a 32 > bit OS. the error message is a match for this scenario. The binaries would still be in /usr/sbin . -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dahdi_scan
On Mon, Nov 14, 2016 at 07:36:40AM -0500, Jerry Geis wrote: > >dahdi_scan (and lsdahdi) shows data reported by the DAHDI kernel modules > >themselves. lspci shows the PCI device. > > >What is the output from dahdi_hardware ? It should show if there's a > >module handling this device, and also which module. > > > lsdahdi > > lspci | grep Digium > 03:05.0 Ethernet controller: Digium, Inc. Wildcard TE122 single-span > T1/E1/J1 card (rev 11) > > dahdi_hardware > pci::03:05.0 wcte12xp-d161:8001 Wildcard TE122 So as you can see, there's no kernel module that handles this device. Any change if you run: modprobe wcte12xp #? Do you have dahdi-linux installed? Properly? If you think it is, what is the output of: lsmod | grep dahdi modinfo dahdi uname -r find /lib/modules/`uname -r` -name dahdi.ko find /lib/modules -name dahdi.ko -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dahdi_scan
On Fri, Nov 11, 2016 at 01:18:48PM -0500, Jerry Geis wrote: > lspci | grep Digium > 03:05.0 Ethernet controller: Digium, Inc. Wildcard TE122 single-span > T1/E1/J1 card (rev 11) > > dahdi_scan dahdi_scan (and lsdahdi) shows data reported by the DAHDI kernel modules themselves. lspci shows the PCI device. What is the output from dahdi_hardware ? It should show if there's a module handling this device, and also which module. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What's the smallest, lightest Asterisk you can build? Does size even matter?
On Tue, Nov 01, 2016 at 11:00:10PM +, Jonathan H wrote: > All I need is PJSIP, ulaw, alaw, wav, astdb and all the dialplan functions. > > I don't need any other DB layer, I have no hardware, and I was wondering > what the smallest build possible was. > > I experimented, but everything relied on other things. And then I > wondered... is there actually any point? Is there anything to be gained? > > Will it matter more when there are lots of concurrent calls, or should I > just not worry, leave all the options in makemenu, make it easy on myself > and build the full thing each time? Do you worry about build time? Dependencies? Will you need to often build things? If not, just build everything in (unless it involves much effort). And avoid loading modules you don't need. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Wildcard AEX800 digium card asterisk configuration
On Thu, Oct 13, 2016 at 06:40:05PM +0200, christopher kamutumwa wrote: > hello, > > i recently purchased a Wildcard AEX800 digium card. Ive installed > asterisk 13 and all prerequistses on ubuntu serv14.04 LTS. Dahi is the > driver am using; ive configured all but when i call from PSTN through > fxo port an not getting anything in logs or to extensions. below are > my config > > please help > chan dahdi.conf > > [trunkgroups] > > ; No trunk groups are needed in this configuration. > > [channels] > #include /etc/asterisk/dahdi-channels.conf Does this file exist? If not, remove this line. If not, what does is its contents? > ; The channels context is used when defining channels > > using the > ; older deprecated method. Don't use this as a section > > name. > > ;[phone](!) > ; > ; A template to hold common options for all phones. > ; > usecallerid = yes > hidecallerid = no > callwaiting = yes > threewaycalling = yes > transfer = yes > echocancel = yes > echocancelwhenbridged = yes > ;immediate = no > rxgain = 0.0 > txgain = 0.0 > ;FXS Modules > group = 1 > echocancel = yes > signalling = fxo_ks Hint: Just use 'signalling = auto' and avoid the confusion in chan_dahdi.conf . However, this is not the source of your problem. > context = Internal > channel = 1-4 > > ;FXO Modules > group = 2 > echocancel = yes > signalling = fxs_ks > context = Incoming > channel = 5-8 > root@ubuntu:/etc/asterisk# lsdahdi > ### Span 1: WCTDM/0 "Wildcard AEX800" (MASTER) > 1 FXO RED > 2 FXO RED > 3 FXO RED > 4 FXO RED > 5 FXS > 6 FXS > 7 FXS > 8 FXS The channels were created (at the kernel level). Kernel level signalling was given to them by dahdi_cfg. However they are not in use by chan_dahdi. Maybe dahdi-channels.conf doesn't exist and this causes problems? Anyway, in the asterisk command-line run: module unload chan_dahdi.so module load chan_dahdi.so And see what errors you get. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016 http://www.asterisk.org/community/astricon-user-conference New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 11.23 with libmysqlclient20 on Debian 8
On Mon, Oct 03, 2016 at 07:54:23PM -0300, Victor Villarreal wrote: > Hi List! > > I'm facing a problem while compiling Asterisk-11 on a Debian 8 server. > > The mysql-server version installed is 5.7 and come from the official mySQL > community repo for Debian. For the record, we ubild both asterisk 11 (last version: 11.21.2) and 13 (13.11.2) for Debian Stable using the distro-provided MySQL packages. > > After compile, install and execute Asterisk, the comman "lsof -p `pidof > asterisk` | grep mysql" don't produce any output. Like if confgure script > don't found the mysql lib. Is Are there any mysql-related module loaded? Start with e.g. ldd /usr/lib/asterisk/module/cdr_mysql.so > > With libmysqlclient18 every is Ok. How can I use libmysqlclient20 with > Asterisk ? Were the relevant modules built? Do you use direct MySQL support? ODBC? Maybe you forgot to install development headers? > > Thanks in advance, and best regards. > > root@nodo1:/usr/src/asterisk-11.23.0# ls -lh /usr/lib/x86_64-linux-gnu/ | > grep mysql > -rw-r--r-- 1 root root 5,7M ago 25 09:37 libmysqlclient.a > lrwxrwxrwx 1 root root 20 ago 25 09:37 libmysqlclient.so -> > libmysqlclient.so.20 This one typically comes from the development headers package, so it's probably installed. > lrwxrwxrwx 1 root root 24 ago 25 09:37 libmysqlclient.so.20 -> > libmysqlclient.so.20.3.2 > -rw-r--r-- 1 root root 4,2M ago 25 09:37 libmysqlclient.so.20.3.2 > -rw-r--r-- 1 root root 18K ago 25 09:37 libmysqlservices.a -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016 http://www.asterisk.org/community/astricon-user-conference New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [SOLVED] Re: Feature Request: what about "core stop panic" ?
On Wed, Sep 07, 2016 at 01:41:55PM -0600, George Joseph wrote: > On Wed, Sep 7, 2016 at 11:03 AM, Olivier wrote: > > My system shows: > > # ps aux | grep asteri > > asterisk 429 7.3 2.4 59468 25088 ?Ssl 18:47 0:03 > > /usr/sbin/asterisk -U asterisk -G asterisk -g > > ... > > # sysctl kernel.core_pattern > > kernel.core_pattern = core > > > > Since "core" is a relative file name, the file will be in whatever the > working directory is for the process. You may have to hunt it down. For > better debugging, you might want to set core_pattern to something like > " /tmp/core-%e-%t". That way all core files will have a name like > "/tmp/core-asterisk-1473164587.7705". "man core" should give you more info > on constructing the file name. But looking at the process of asterisk may help: ls -l /proc/$PID_OF_ASTERISK/cwd -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016 http://www.asterisk.org/community/astricon-user-conference New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [SOLVED] Re: Feature Request: what about "core stop panic" ?
On Tue, Sep 06, 2016 at 06:37:52AM -0600, George Joseph wrote: > On Tue, Sep 6, 2016 at 1:55 AM, Olivier wrote: > > Where should core file be created when Asterisk is run as a daemon by > > asterisk user and group ? > > Is there a setting I can use to specify the directory used (so that we can > > make sure appropriate ownership is set) ? > > > > "$ sysctl kernel.core_pattern" will show you where core files are written. > For Asterisk to produce the core file, it has to be started with the '-g' > option so make sure your asterisk.service file is adding the option. Specifically, if the first character of core_pattern is '!', the rest should be an executable, to which the core file is handled. IIRC Centos7 had something of that type installed by default. On Debian Stable you have the package corekeeper (or maybe also systemd-coredump from backports). I haven't tried any of those. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016 http://www.asterisk.org/community/astricon-user-conference New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need ISDN call generator
On Mon, Aug 29, 2016 at 05:26:17PM +0430, Hooman Fazaeli wrote: > On 2016-08-29 12:28, Eric Klein wrote: > >Hi Hooman, > > > >What you probably want is a PRI PBX running Asterisk. > > > >You should either plan to build your own (with the cards you need) or get > >one of the low cost options: > > > > * Allo.com has their Mega PBX with 1 PPR port > > (http://allo.com/megapbx-line.html) > > * Pika Tech has the Warp PBX with BRI > > (http://www.pikatech.com/warp-telephony-all-in-one/) > > > >Both ship world wide. There are other brands and models out there and > >depending on where you are located they may be better options based on > >shipping etc. > > > >Hope this helps, > >Eric > > Thanks. But I am not looking for a SOHO PBX, but a product specifically built > to generate ISDN load for > PBX (asterisk) test purposes (like this: > http://www2.vconsole.com/8-Port-T1/E1/PRI-Bulk-Call-Generator-p-28.html) > <http://www2.vconsole.com/8-Port-T1/E1/PRI-Bulk-Call-Generator-p-28.html> > > I have done my googling homework and already know of a few options. I asked > in this list > to benefit from people's experiences and recommendations, those who have > used/worked with such > devices or have setup systems for the sole purpose of ISDN call/load > generation. > Why don't you start with something simple as: channels() { lsdahdi | awk '/^ /{print $1}' # | head -n whatever } for chan in `channels`; do asterisk -rx "channel originate DAHDI/$chan/something application Echo" done Alternatively, if you have a convenient way to control the number of SIP calls coming from a certain trunk, redirect the all to DAHDI/g0 . Simple load is very simple to generate with Asterisk. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016 http://www.asterisk.org/community/astricon-user-conference New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] auto_assign_spans=0 [was: Re: DAHDI on CentOS 7]
On Mon, Aug 15, 2016 at 08:36:43AM -0400, Jerry Geis wrote: > What is needed to get DAHDI to start up correctly on CentOS 7 and systemd... > I am using DAHDI-linux-complete 2.11.1 > > I saw mention in my search that it has been fixed after 2.11.1 but cannot > find > what the fix is. > > Thanks, > > Jerry There are some good reasons for that. But I'll start with recommending to set auto_assign_spans=0 . All of our packages have a patch such as: --- a/drivers/dahdi/dahdi-base.c +++ b/drivers/dahdi/dahdi-base.c @@ -7405,7 +7405,7 @@ int dahdi_assign_device_spans(struct dahdi_device *ddev) return 0; } -static int auto_assign_spans = 1; +static int auto_assign_spans = 0; static const char *UNKNOWN = ""; Or simpler, applicable at runtime (module load time): echo options dahdi auto_assign_spans=0 >>/etc/dahdi/modprobe.d/dahdi.conf For starters, this means you no longer need to disableautomatic module loading for PCI cards (which was not needed for most systems anyway), and that on its own avoids most of the need for the init script. It also avoids unloading the modules at shutdown. Which is a good thing. Do that, and report what problems you have. Because I don't seem to have problems with my packages (on Centos 7 and Debian Jessie). -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DAHDI Dynamic span as INT device identificator
On Thu, Aug 04, 2016 at 12:54:45PM -0300, Rafael dos Santos Saraiva wrote: > Hi > > Is it possible assign an INT identification to dynamic device in DAHDI? > Example: > > This is device: DYN/eth/eth1/04:74:A1:00:0A:AE/0 > I want call this as span 1 > > I saw the assigned-spans.conf and aparently can be this. > > Thanks in advance. See what I wrote earlier on the asterisk-dev list: http://lists.digium.com/pipermail/asterisk-dev/2016-April/075486.html Generally I don't like the way dsystem.conf is used both for defining synamic spans and for configuring existing spans. I explained there how to separte the two. This also makes it simpler to interact with span assignment. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 13 High CPU usage
On Sun, Jul 24, 2016 at 06:39:19PM +0100, ian gilmour wrote: > The following bash 1-liner may be useful... > > while true; do top -Hbc -p `pgrep asterisk` -n 1 && asterisk -rx "core show > threads"; sleep 1; done Just for the kicks: ps --no-headers -L -o lwp,cp --sort lwp `pidof asterisk` \ | join -1 1 -2 2 - <(asterisk -rx 'core show threads' | sort -k2) \ | sort -n -k 2 Notes: 1. Bashism. <(...) requires bash. 2. Join counts on a non-numeric sort. Maybe there's an extrs sort needed for ps, as I guess it sorts numerically. 3. I tried puting this all in watch: watch -n1 -d '...' or watch -n1 -d bash -c "...", but I got an error message: "sh: 1: Syntax error: "(" unexpected" (sh? shouldn't it be bash?) But at this point I have already exceeded the allocated time slot. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 13 High CPU usage
On Fri, Jul 22, 2016 at 12:02:43AM +0100, Chirag Desai wrote: > I am not sure where to start looking in order to debug the CPU usage by > asterisk and would very much appreciate some guidance. If you run 'top', the basic information would be to show per-CPU information (press '1'). Another thing to look at: press 'H' to get per-thread entries. Do you have many many threads each taking a small part of a core, or a few threads taking lots of CPU time? I believe that the PID (process/thread ID) you see in top is also the second item in each line in the output of 'core show threads'. So this could give you some clues regarding the CPU hogs you see in top. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] rasberry pi
On Wed, Jul 06, 2016 at 01:10:23AM -0700, Thufir wrote: > I'm debating between a cloud PBX or, perhaps, rasberry pi. For a SOHO, > maybe three hardphones, rasberry pi would suffice? I would be amazed, but, > if so, great. Just a reminder: The original Raspberry Pi uses a SoC with an ARM core that doesn't support the ARMv7 architecture (though it does have floating-point support). Thus most ARM-based distributions will not work on it, and you'll typically need to run Raspbian on it. Nowadays, unless you have an older one or you buy a rpi-zero, this is not what you use. Nowadays we have rPi2 and rPi3. They are quad-core (though generally only the first core can handle interrupts. So when looking at top, be sure to press '1' for per-core information). The rPi3 should be able to run arm64, but this seems mostly theoretical (not sure if it is supported in practice) and you get a system slightly faster and mostly compatible with the rPi2. For such a small system, I think that even the original rPi would do. The rPi2 will certaily do. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Pet project: one step Asterisk compile on Centos 7
Off-topic: On Tue, Jun 14, 2016 at 11:02:17AM +0200, Lenz Emilitri wrote: > Project located at https://github.com/l3nz/CompileAsteriskPBX >From the build script: # build Asterisk cd $TARGET_DIR/$ASTVERSION ./configure --libdir=/usr/lib64 cd $TARGET_DIR/$ASTVERSION/menuselect make menuselect cd $TARGET_DIR/$ASTVERSION make menuselect-tree ./menuselect/menuselect \ #[snip menuselect parameters] # we want mp3's ./contrib/scripts/get_mp3_source.sh make (This is not intended to criticise Lenz) All of this magic (specifically the bits before the menuselect) is needed for a proper build? * Why is an explicit libdir packameter needed? What break if it is not passed? * Why is there a need to build menuselect manually? What about 'make menuselect'? Do we need a Makefile target to build menuselect but not run it? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Pet project: one step Asterisk compile on Centos 7
On Tue, Jun 14, 2016 at 06:22:05PM +0200, Lenz Emilitri wrote: > 2016-06-14 17:44 GMT+02:00 Tzafrir Cohen : > > > > 1. Asterisk basically has such a script inside. > > It is - as you say - inside. This is outside and does the download for you. > > > 2. Asterisk has an RPM package. An RPM package is exactly a reproducible > > build (listing dependecies, and such). > > It's true. They are very interesting, especially if you are a > historian of software. > http://packages.asterisk.org/centos/ > > If you need something less, say, "vintage", you may need to compile it > yourself. > > > 3. You are reinventing RPM. Badly. Do you people really want to run: > >- As root > >- A huge blob nobody can inspect > >- that is executable, and hence has tons of places to add nice hooks > > in? > > > > Learn how to use rpmbuild. > > I personally happen to have shipped RPMs for about 10 years now. But > building a RPM might be overkill if you are deploying a test, > throwaway box, or just once for a Docker image. Of course I would not > use this as an RPM substitute, and if I were to use something like > this I'd fork it or at least read it (it is maybe 20 lines). YMMV. > > And IIRC there is more places to ship "nice hooks" into a binary you > ship as an RPM than in a shell script that does what you would > manually from the terminal! RPM is one of the common formats to describe building of a package. There are other such similar formats, but if you're on Centos, RPM is already included with the platform, and hence it makes sense to use it. An RPM package is an archive (technically: cpio. But you don't get direct access to that) with a small dictionary attached in front of it. That dictionary specifies the name, version, revision, and many other fields. There are basically two types of RPM packages: source and binary. Binary ones ar ethe products of builds. Source ones describe how to build, and thus they are the ones that interest us. A source RPM package has in its archive a spec file, sources (one or more) and patches (zero or more). The spec includes instructions how to build a binary package. So if you don't have any patches, the one thing you need to maintain is ths spec file. Everything else (including the source tarball) could be automated using the standard RPM build toolchain. It would also be nice if you could verify the signature on the tarball[0]. I personally prefer git-buildpackage-rpm that allows keeping the "tarballs" in git. You can either use tags directly to get tarballs from, or use pristine tar to store the delta between a tarball and a tag. But right now this is not a standard tool on Centos systems, and you'll have to install it on your own[1]. The standard RPM+Git toolchain in Fedora seems to be fedpkg[2], but I think it would take quite some tweaking to make it work with your own seperate tarball repository. I don't think it is useful for your case, but maybe someone more familiar than me can give an answer. If you really want to avoid conaminating the build system with all of those packages, do a chroot build inside mock (supported by the above two, or directly). A bit slower, and a longer debugging cycle, but keeps your system clean. It has support for caching downloaded packages and such. So in short, one proposed method: 1. Install git-buildpackage-rpm 2. Clone the package asterisk 3. Create a branch called, say, packaging, with: rpm/asterisk.spec debian/gbp.conf # Why Debian. that's the way it is cat <debian/gbp.conf [DEFAULT] packaging-branch = packaging upstream-branch = master upstream-tag = '%(version)s' packaging-dir = rpm EOF 4. Run: gbp buildpackage-rpm -ba # or also: --git-mock --git-dist=epel-7 If you use mock, mock pulls all build-dependencies on its own. If you don't, then you should probably use yum-builddep (from package eyum-utils. But I never tried it). Oh, and what if you just want rpm for getting the build dependencies installed and for running ./configure, 'make' (and menuselect) with the right magical arguments, and then you'll run 'make install' on your own? Use -bc instead of -ba / -bb. You'll end up with a built source tree. su[do] make install, and you're done. [0] On Debian uscan only aquired this ability relatively recently. When I wrote a similar script a decade ago[4] I just verified the md5 checksum to verify that I can safely avoid re-downloading. [1] RPM support is already mostly merged to the main git-buildpackage project: https://honk.sigxcpu.org/piki/projects/git-buildpackage/ . Some extra features may be available in https://github.com/marquiz/git-buildpackage-rpm [2] See https://fedoraproject.org/wiki/Package_maintenance_guide?rd=
Re: [asterisk-users] Pet project: one step Asterisk compile on Centos 7
On Tue, Jun 14, 2016 at 11:02:17AM +0200, Lenz Emilitri wrote: > Hi all, > I thought I'd share I script I made (based on some of Leif's works) > that lets you download, compile and install Asterisk all in one go; > and then removed the dev tools used. > > We use it quite a bit to provision systems using Ansible, but it is > easier than remembering everything every time even if you are using a > shell. > > At the moment I have scripts for Centos 7 and Asterisk 13, but plan to > port them to other versions of Asterisk as there is a need to do so. > Contribs welcome! > > Project located at https://github.com/l3nz/CompileAsteriskPBX 1. Asterisk basically has such a script inside. 2. Asterisk has an RPM package. An RPM package is exactly a reproducible build (listing dependecies, and such). 3. You are reinventing RPM. Badly. Do you people really want to run: - As root - A huge blob nobody can inspect - that is executable, and hence has tons of places to add nice hooks in? Learn how to use rpmbuild. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM800 just receive calls, but not make
On Wed, May 18, 2016 at 02:29:30PM -0300, Vitor Mazuco wrote: > Hello everyone > > I have a TDM 800 on an Ubuntu Server > > he's just getting call normally, but when I call any number by this > board, it is silent and not make the call. > > look at the log > > Executing [629886874@ramais:1] Dial("SIP/2000-000e", > "DAHDI/6-1/29xxx,60,tT") in new stack > [May 18 14:21:31] WARNING[4332][C-000d]: chan_dahdi.c:13433 > dahdi_request: Unknown option '-' in '6-1/29886874' This should have given you a hint. There is simply no need for this '-1'. In your dialplan you need to have: Dial(DAHDI/6/${THENUMBER}) instead of: Dial(DAHDI/6-1/${THENUMBER}) -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk admin interface
On Mon, May 16, 2016 at 04:54:10PM -0700, John Kiniston wrote: > You could explore using ARI with it's Push configuration. > > https://wiki.asterisk.org/wiki/display/AST/ARI+Push+Configuration With all due respect, this answer is basically like answering "use a text editor to edit the dialplan". The OP asked for an Asterisk administration interface. Right now the only proper answer given (that did not involve writing one) was FreePBX. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Russian and French sounds
On Wed, May 11, 2016 at 06:05:49AM -0400, Dovid Bender wrote: > Hi, > > Does anyone know who did the prompts for French and Russian for Asterisk? I > need some custom prompts. You can find CREDIT files inside the tarballs (the trick is: ls | grep -v '$' or: ls [A-Z]*) $ cat asterisk-core-sounds-ru-gsm/CREDITS-asterisk-core-ru-1.4.27 core-sounds-ru v1.1 Provided by: http://www.ivrvoice.ru To order new files recorded with the same voice please sent your request to i...@pbxware.ru Acknowledgments: Max Litnitskiy - idea and team building, Andrew Roman - director of recording, Alex Litnitskiy - testing and bug fixing, Denis Gaidamak - assistant of producer of initial release. $ cat asterisk-core-sounds-fr-gsm/CREDITS-asterisk-core-fr-1.4.27 Recorded by: June Wallack (http://www.junewallack.com) Translated into French by: Clod Patry Kristopher Lalletti June Wallack Financial Contributions by: Digium, Inc. (http://www.digium.com) Unlimitel (http://www.unlimitel.ca) BGM Informatique (www.bgm.qc.ca) -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DAHDI press button get fast busy
On Thu, May 12, 2016 at 03:04:02PM -0600, Greg Woods wrote: > My DAHDI phones were broken since a recent power outage (which required a > reboot). For some reason the Asterisk or DAHDI configuration is messed up > somewhere (probably from an update that was applied before the reboot?). > > I am using a Digium TDM410 card. > > At first, nothing worked at all, but I discovered that if I loaded things > manually (e.g. modprobe dahdi; modprobe wctdm24xxp, then go into asterisk > -r and "module unload chan_dahdi; module load chan_dahdi") then at least > the lights on the card come on and the phones half work. Inbound calls work > fine, but if I try to dial out, as soon as I press a numeric button on the > phone, I get an immediate fast busy, so outbound calls don't work. Outbound > calls from a SIP phone off the same server work fine (the outbound line is > an IAX link to a VOIP provider). > > I have been Googling this for nearly an hour without finding any reference > to this problem. Anybody have any idea where I can look to debug this, or a > guess as to what might be wrong with the configuration? Sounds like you did not set a proper dialplan for them. I assume the phone is on dahdi channel 1. If so, please provide the output of: context=`asterisk -rx 'dahdi show channel 1' | awk '/Context: / {print $2}' asterisk -rx "dialplan show $context" -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Proper way to start Asterisk on CentOS 7? (Carlos Chavez)
On Mon, May 09, 2016 at 08:46:36AM +0200, Stefan Viljoen wrote: > Hi Carlos > > I have experienced something similar starting Asterisk 1.8.32.3 on Centos 7 > on commodity / whitebox hardware. > > The problem was that Asterisk was starting "too quickly" in the systemd > startup sequence, before the required services it needs to run were up and > ready. > > I eventually came up with this systemd script to start it, and this now > starts Asterisk on all our our new Centos 7-using deployments. > > Maybe you can try it out...? This starts our Asterisk 1.8 instances without > problems on Centos 7 on boot: > > -- > > [Unit] > Description=Asterisk > After=network.target > After=network-online.target > After=startup.service > After=mysql.service > Wants=mysql.service IIRC, mysql is socket activated, and hence services don't need to have any explicit dependency on it Remove those two. > > [Service] > Type=idle Huh? Stick the the default: Type=basic. No need to mention this one. > User=root > Group=root Those two are superfluous. And anyway, are wrong. Asterisk is a network-facing daemon, and as such should run with it # (Be sure to create that one) User=asterisk > ExecStart=/usr/sbin/asterisk -f > ExecStop=/usr/sbin/asterisk -rx 'core stop now' > ExecReload=/usr/sbin/asterisk -rx 'core reload' Right. But see my note below about Restart. > TimeoutSec=300 Not sure what this is for. > > LimitCORE=infinity > LimitNOFILE=8096 > Restart=always Restart=on-failure With your current configuration: asterisk -rx 'core stop now' becomes a restart. > RestartSec=4 I used a bit less. > > [Install] > WantedBy=multi-user.target > > -- > > I place this in /etc/systemd/system as "asterisk.service" and then install > it via > > systemctl enable asterisk.service Fine. I placed mine as part of a package, and hence it is /usr/lib/systemd/system/asterisk.service Here it is: [Unit] Description=Asterisk PBX Documentation=man:asterisk(8) Wants=network-online.target After=network-online.target [Service] ExecStart=/usr/sbin/asterisk -g -f -U asterisk ExecReload=/usr/sbin/asterisk -rx 'core reload' Restart=on-failure RestartSec=1 WorkingDirectory=/var/lib/asterisk # Extra settings: # If you want to set them, you can generate the file # /etc/systemd/system/asterisk.service that has the following lines # (without the comments) #include /lib/systemd/system/asterisk.service #[Service] # # and following those two lines add directives or override existing # directives. Some extra directives that may be useful: # You can run a script to clean up after asterisk. An example script is # included in contrib/scripts/asterisk_cleanup. #ExecStopPost=/path/to/script #Nice=0 #UMask=0002 #LimitCORE=infinity #LimitNOFILE= # safe_asterisk runs Asterisk in a virtual console. This allows easy # access to the asterisk command-line without logging it. on the other # hand, it allows anyone with physical access to the console full access # to Asterisk. To enable this console, unrem the following lines and add # '-c' to the ExecStart line above: #TTYPath=/dev/tty9 #StandardInput=tty #StandardOutput=tty #StandardError=tty # For more information on what these parameters mean see: # # http://0pointer.de/public/systemd-man/systemd.service.html # http://0pointer.de/public/systemd-man/systemd.exec.html [Install] WantedBy=multi-user.target -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ubuntu 14 Warning
On Mon, May 02, 2016 at 10:04:54AM -0600, George Joseph wrote: > This morning, 2 of us noticed that running contrib/scripts/install_prereq > on a fresh Ubuntu 14 system actually removed critical packages like > network-manager, openssh-server, perl, git, and a bunch of others. It > appears that the culprit is the libsnmp-dev package. It's default conflict > resolution solution is to uninstall conflicting packages and the alternate > solution is to correctly upgrade the conflicting packages. Since > install_prereq doesn't give you the opportunity to choose the solution, it > does the uninstall and you're left with an unusable system. Can you please remove any '-y's from the apt-get / aptitude command-lines and try reproducing it? Also maybe use aptitude instead of apt-get. > > Why this is happening now is unclear. To work around this, you can either > run "apt-get update" and "apt-get upgrade" manually, or run "apt-get > install libsnmp-dev" manually and say "no" to the first solution and "yes" > to the second. Then run install_prereq. Could you please provide a more complete log? See /var/log/apt/history.log or term.log in the same directory. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] my dahdi dont'n start
Package: dahdi-linux Version: 1:2.10.2~dfsg-1 Severity: minor Add support to automatic loading of the base module 'dahdi', as it is e.g. needed for cases where DAHDI is needed for app_meetme in Asterisk (and in such a case won't be pulled by hardware devices auto-detected). This is a regression from the version of dahdi-linux in Wheezy. On Tue, Apr 26, 2016 at 11:13:08AM -0500, Richard Mudgett wrote: > Administrator TOOTAI: You must have DAHDI running when using meetme because > DAHDI does the audio mixing for the conference. There's no such thing as "DAHDI running". What Meetme needs is the basic DAHDI module loaded. It also loads the built-in timer (which was originally in a separate module: dahdi_dummy, and bfore that ztdummy). On Debian systems, if you want a module loaded at startup, you can add it to a file under /etc/modules-load.d . So what you need to do is: echo dahdi > /etc/modules-load.d/dahdi.conf And you're done. (Also reporting this as a bug so I will not forget it) -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] my dahdi dont'n start
On Fri, Apr 29, 2016 at 09:38:10AM +0200, Mamadou NGOM wrote: >Hello, > >I have not resolved my problem.I renamed my dahdi file "mv dahdi.bash >dahdi " in the directory /etc/init.d, but it doesn'nt work yet. > >the same error after the command /etc/init.d/dahdi start >-bash: /etc/init.d/dahdi: /bin/sh^M: bad interpreter: No such file or >directory This is a DOS-formatted text file. A '#!' at the beginning of the line is handed over to the Linux kernel, which takes the rest of first line (up to the first newline character) as the interpreter of the script. This first line here is '/bin/sh\r', because this file is a DOS-formatted text file, with lines that end in '\r\n' instead of '\n' as in UNIX text files. sed -i -e 's/\r$//' /etc/init.d/dahdi And then again, do you really need a DAHDI init script? See my previous message: http://lists.digium.com/pipermail/asterisk-users/2016-April/288968.html -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Debian 8.4 : dahdi startup scripts ?
Thanks for your reply, On Mon, Apr 18, 2016 at 06:25:30PM +0200, Bertrand LUPART - Linkeo.com wrote: > Hello, > > > >> I just made a asterisk / dahdi fresh install on Debian 8.4, and ended up > >> with the following packages : > […] > >> However, i can't find any dahdi startup script, neither init.d neither > >> systemd fashion. > >> > >> $ sudo dpkg -l|grep -Ei 'dahdi'|cut -d " " -f3 |xargs dpkg -L|grep -E > >> 'init.d|system' > >> /usr/share/doc/dahdi/examples/system.conf.sample.gz > >> /etc/init.d > >> > >> > >> Am I missing something? > > > > Yes. README.Debian[1], included as /usr/share/doc/dahdi/README.debian . > > > > README.Debian (.gz, if it's long enough) is a file for some extra > > documentation by the packager. See the section about "Automatic startup" > > there. > > > > The only real use the DAHDI init script has is for interactive module > > loading / unloading. Modprobe is not useful on its own, as DAHDI > > includes a set of modules (especially if you use Astribanks. There a bit > > of extra magic besides the recursive rmmod was added). > > > > Furthermore, there is really no point is removing of modules at system > > shutdown. It only wastes time (and some CPU cycles, if you care about > > power) and increases the chances of exposing bugs at that time. > > This startup / shutdown scheme apply to nominal cases where you don't need to > reboot your system :) > > Building a brand new box may require you to restart dahdi a bunch of times: > "no span" error [1], > [1] > http://asteriskfaqs.org/2015/01/15/asterisk-users/dahdi_genconf-fails-with-empty-configuration-no-spans.html But as you can see from my replies there, the modules were already loaded. All that was needed was to get the spans assigned. Unloading the modules and loading them wouldn't have mattered. My original plan was that if you have a blank configuration (no /etc/dahdi/system.conf and no /etc/dahdi/assigned-spans.conf), the spans would be generated and configured automagically at the DAHDI level. Part of this works: If you have no /etc/dahdi/system.conf, when a new span is created, dahdi_genconf creates configuration for it: http://git.asterisk.org/gitweb/?p=dahdi/tools.git;a=blob;f=hotplug/span_config.d/10-dahdi-cfg;hb=HEAD#l25 Originally this was also the case with a lack of assigned-spans.conf. See the last hunk here: http://git.asterisk.org/gitweb/?p=dahdi/tools.git;a=commitdiff;h=1292ea90789aa20bff5a533141086f6ecf4f82df > digium firmware not downloading [2], forgetting Europe requires E1 line mode, > new telco uncertain if using crc4 … > More, in case your telco is failing, he'll often ask you to reset on your > side first. Restarting dahdi feels way more convenient than rebooting a > distant system. Again, in many cases you'll need to just use: dahdi_span_assignment remove dahdi_span_assignment add This also runs dahdi_cfg on the span. Thus it already does a partial reset. Not sure how much of a reset. If a full reset is needed, I included the module removal and load functionality in the seperate script. > > Red a bunch of READMEs of the packages installed before posting, not sure i > red this one. Maybe a "Manual startup / shutdown" section would be welcome ;) Will add. > > > > IIRC the Asterisk package in Jessie lacks the extra script to get the > > configured DAHDI channels into Asterisk[2]. You may need to drop it > > yourself to /usr/share/dahdi/span-config.d > > > > If things still don't work, please do report it here on on the BTS. > > This file is missing on my side (1:2.10.0.1-1), but asterisk seems to be > aware of the channels. Not sure how. > > Thank you for your answer, It's missing because it should not be part of dahdi (I removed it explicitly in the packaging but forgot to add it to Asterisk in time for Jessie). The DAHDI package should not assume Asterisk is installed (at least not if it's so simple to fix). The directory is theree for hooks to be added. If you have your own interesting action, feel free to add it. Just make it short. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Debian 8.4 : dahdi startup scripts ?
On Tue, Apr 12, 2016 at 04:36:58PM +0200, Bertrand LUPART - Linkeo.com wrote: > Hello, > > > I just made a asterisk / dahdi fresh install on Debian 8.4, and ended up with > the following packages : > > $ sudo dpkg -l|grep -Ei 'dahdi|asterisk|libpri' > ii asterisk 1:11.13.1~dfsg-2+b1 > amd64Open Source Private Branch Exchange (PBX) > ii asterisk-config1:11.13.1~dfsg-2 all > Configuration files for Asterisk > ii asterisk-core-sounds-en1.4.22-1 all > asterisk PBX sound files - US English > ii asterisk-core-sounds-en-gsm1.4.22-1 all > asterisk PBX sound files - en-us/gsm > ii asterisk-dahdi 1:11.13.1~dfsg-2+b1 > amd64DAHDI devices support for the Asterisk PBX > ii asterisk-modules 1:11.13.1~dfsg-2+b1 > amd64loadable modules for the Asterisk PBX > ii asterisk-moh-opsound-gsm 2.03-1all > asterisk extra sound files - English/gsm > ii asterisk-voicemail 1:11.13.1~dfsg-2+b1 > amd64simple voicemail support for the Asterisk PBX > ii dahdi 1:2.10.0.1-1 > amd64utilities for using the DAHDI kernel modules > ii dahdi-firmware-nonfree 2.10.0-1 all > DAHDI non-free firmware > ii dahdi-linux1:2.10.0.1~dfsg-1 all > DAHDI telephony interface - Linux userspace parts > ii dahdi-modules-3.16.0-4-amd64:amd64 1:2.10.0.1~dfsg-1+3.16.7-ckt25-2 > amd64DAHDI modules for Linux (kernel 3.16.0-4-amd64) > ii dahdi-source 1:2.10.0.1~dfsg-1 all > DAHDI telephony interface - source code for kernel driver > ii libpri1.4 1.4.15-1 > amd64Primary Rate ISDN specification library > > > However, i can't find any dahdi startup script, neither init.d neither > systemd fashion. > > $ sudo dpkg -l|grep -Ei 'dahdi'|cut -d " " -f3 |xargs dpkg -L|grep -E > 'init.d|system' > /usr/share/doc/dahdi/examples/system.conf.sample.gz > /etc/init.d > > > Am I missing something? Yes. README.Debian[1], included as /usr/share/doc/dahdi/README.debian . README.Debian (.gz, if it's long enough) is a file for some extra documentation by the packager. See the section about "Automatic startup" there. The only real use the DAHDI init script has is for interactive module loading / unloading. Modprobe is not useful on its own, as DAHDI includes a set of modules (especially if you use Astribanks. There a bit of extra magic besides the recursive rmmod was added). Furthermore, there is really no point is removing of modules at system shutdown. It only wastes time (and some CPU cycles, if you care about power) and increases the chances of exposing bugs at that time. IIRC the Asterisk package in Jessie lacks the extra script to get the configured DAHDI channels into Asterisk[2]. You may need to drop it yourself to /usr/share/dahdi/span-config.d If things still don't work, please do report it here on on the BTS. [1] http://sources.debian.net/src/dahdi-tools/1:2.10.0.1-1/debian/README.Debian/ [2] http://git.asterisk.org/gitweb/?p=dahdi/tools.git;a=blob;f=hotplug/span_config.d/50-asterisk;hb=HEAD -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Recommendations for free virtual server tech and Asterisk?
On Sat, Apr 09, 2016 at 08:13:29PM +0700, Ikka Tirtawidjaja wrote: > https://sysadminman.net/blog/2011/asterisk-virtualization-openvz-or-vmware-3004 > > > OpenVZ provides operating system-level virtualization where the underlying > hardware runs a kernel that is shared by all of the virtual machines. > > Where OpenVZ gets a bad name is that it’s very easy to provision many more > VPSs on a physical server than that server can really handle. This means > lots of virtual machines all trying to use the CPU, ram, network etc on the > underlying server, resulting in bottlenecks. This might not be a problem on > a webserver. If a web server takes half a second longer to display a web > page because the server is overloaded then maybe nobody will notice. > However, if your VOIP packets are delayed for half a second then you will > definitely notice! > > Probably the most crucial fact about running Asterisk on a VPS though is > who you are sharing the underlying hardware with, and how well the server > is managed. Even if there are only a few other virtual servers on that > server and they are allowed to abuse the resources available then you will > likely get a bad VOIP experience. This can definitely be the case where > Asterisk is installed on a general purpose VPS. That's odd logic, IMHO. OpenVZ (and likewise LXC, docker, systemd nspawn, rkt and the likes) are containers: they seperate userspace while running on the same kernel. This greatly reduces the overhead of virtualization compared to full-system virtualization, such as KVM, VirtualBox or VMWare. The reduced overhead naturally means that service providers are able to overbook their service much more efficiently with openvz and the likes than with vmware and the likes. But this is irrelevant if you run the server. There is a different issue of separation, but it is parallel to performance. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] opus : patches for FEC and PLC useful ?
On Mon, Apr 04, 2016 at 11:39:07AM +0200, Ludovic Gasc wrote: > We're testing this branch for a while, not with the latest commits. > > For now, it works, however, time to time audio quality issues with > transcoding, but I don't know yet where is the problem. > > We'll test with the latest commits. > > BTW, it isn't included in the main because potential patent license issues. Those two patches are not on the opus code and hence this is irrelevant to them (regardless of what I personally think about the patent concerns). -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is possible to use FXO Digium card like a Fax modem?
On Wed, Mar 30, 2016 at 01:36:16PM +0100, A J Stiles wrote: > On Wednesday 30 Mar 2016, Vitor Mazuco wrote: > > Hi! > > > > Is possible to use X100p TDM400P, Tdm410p, Tdm400, A400p, Ax400p or > > any others digium card FXO for use Fax modem? > > Yes, in theory it is entirely possible to use an FXO card driven by software > as a modem (and indeed, this is exactly what Winmodems do); although you > will have to do all the hard work of generating the outgoing tones, and > decoding the incoming tones, yourself. This is a highly non-trivial task, > and > there is almost certain to be a better way than this of achieving whatever > you > want. With the sole exception of the X100P, which, IIRC, was basically an existing fax/modem card. If you dig deep enough you may find existing drivers for it. Assuming what is now called X100P is close enough to what was originally that modem. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OPUS support in Asterisk 13
On Thu, Mar 24, 2016 at 07:09:07PM +, Chirag Desai wrote: > Tzafrir, does your update support pass through only or transcoding too? "My" (actually: Meetecho's) patch includes transcoding support. As already mentioned: Asterisk already supports Opus pass-through. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OPUS support in Asterisk 13
On Thu, Mar 24, 2016 at 09:36:40AM -0500, Matt Fredrickson wrote: > On Thu, Mar 24, 2016 at 9:09 AM, Chirag Desai wrote: > > Hi all, > > > > Sorry if this has been asked before. I searched a lot, but found conflicting > > answers, so hoping for some clarification. > > > > My question is does Asterisk 13 support OPUS? If so which version exactly? > > Sort of - it supports OPUS pass through officially, but does not > support OPUS transcoding. I'm not certain when pass through support > went in though, but I believe it was prior to cutting of the 13 > branch. > > There are some unofficial patches that add OPUS support to Asterisk > but I cannot point you to which ones are best to use, unfortunately. I still find it odd that OPUS is not considered safe enough for Asterisk even though it has been used in various programs such as Firefox and Chromium/Chrome. Not very handy if anybody wants to use Asterisk for WebRTC. Anyway, the up-to-date one I have is: http://anonscm.debian.org/cgit/pkg-voip/asterisk.git/tree/debian/patches/opus.patch?h=debian/1%2513.7.2_dfsg-2 based on https://github.com/seanbright/asterisk-opus , which has rotted a bit (but took only a minor bit of tweaking). -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1000 analogue lines with asterisk
On Fri, Feb 19, 2016 at 10:52:34AM +1300, Daniel Harper wrote: > What about leaving the old PBX in place and trunking it via ISDN to the > asterisk server. > > We use rhino 24 channel bank but are 2U for rhino + 1U for patch panel. > (RJ21 cable so might be able to use existing ones if they are RJ21) > > Used USB xorcoms a while back, things may of changed but if one is down and > reboot server then asterisk doesn't come up. Not with recent versions of DAHDI. Assuming you set auto_assign_spans=0, DAHDI channel and span numbers are mapped in /etc/dahdi/assigned-spans.conf the order of their startup is irrelevant: the identification of channels attached to either serial number or connector, and not to semi-random initialization order. And this is why Asterisk could afford itself not to fail if DAHDI channels are missing. Originally it failed because missing devices meant that a device may be missing and thus existing channels may be misplaced. With a recent enough version of Asterisk, and with ignore_failed_channels = true in chan_dahdi.conf (the default as of 13. Makes sense also if you only have a single DAHDI device to begin with) Asterisk will not fail if channels are missing: they wil just come later. And indeed Asterisk can now allow DAHDI channels to be created after initialization, and thus Asterisk does not need to wait for all DAHDI devices to initialize before starting. (This is not specific to Xorcom hardware and should generally apply to any DAHDI hardware. I'm not exactly sure how this interacts with dynamic spans and interested to hear reports, probably in a different thread) -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error compiling dahdi on CentOS 7
On Wed, Feb 24, 2016 at 03:55:09PM -0600, Carlos Chavez wrote: > I am having a problem trying to compile > dahdi-linux-complete-2.11.0+2.11.0 on a CentOS 7.2 server. Version 2.10.2 > compiles fine. Is there a new dependency for 2.11.0 that was not required > for previous versions? Here are some of the errors I get: > > INSTALL > /usr/src/dahdi-linux-complete-2.11.0+2.11.0/linux/drivers/dahdi/dahdi.ko > Can't read private key > INSTALL > /usr/src/dahdi-linux-complete-2.11.0+2.11.0/linux/drivers/dahdi/dahdi_dynamic.ko > Can't read private key > INSTALL > /usr/src/dahdi-linux-complete-2.11.0+2.11.0/linux/drivers/dahdi/dahdi_dynamic_eth.ko > Can't read private key I'm not sure what this is. However, > > /usr/bin/install: cannot stat ‘./dahdi_registration.8’: No such file or > directory > /usr/bin/install: cannot stat ‘./xpp_sync.8’: No such file or directory > /usr/bin/install: cannot stat ‘./lsdahdi.8’: No such file or directory > /usr/bin/install: cannot stat ‘./xpp_blink.8’: No such file or directory > /usr/bin/install: cannot stat ‘./dahdi_genconf.8’: No such file or directory > /usr/bin/install: cannot stat ‘./dahdi_hardware.8’: No such file or > directory > /usr/bin/install: cannot stat ‘./twinstar.8’: No such file or directory This one shouldl be fixed in dahdi-tools 2.11.1-rc1 . -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error making dahdi linux compete 2.11.0
On Mon, Feb 15, 2016 at 05:28:14PM +0200, Tzafrir Cohen wrote: > On Mon, Feb 15, 2016 at 02:15:58PM +, Ryan, Travis wrote: > > Getting the some errors making dahdi 2.11.0. > > > > Seems same as listed here > > http://forums.asterisk.org/viewtopic.php?f=1&t=96455 > > > > In that link they say to use 2.10.2 but that's from December. Is there a > > fix yet for this? > > My bad, I forgot to push the fix for that. > > http://git.asterisk.org/gitweb/?p=dahdi/tools.git;a=commit;h=e1da7b528467a8f8f82058993b2e01333677ee39 > > You may need to run autoreconf after applying this patch to rebuild the > Makefile (though IIRC the makefile will be re-generated by running > 'make'). And while I'm at it: http://git.asterisk.org/gitweb/?p=dahdi/tools.git;a=commit;h=6057ef25e984a2c7f8327b872233ba610b9aabe6 Another thing to note: you should install pkg-config (though practically you only need it to detect libusb, that is: for building Astribank-related utilities). If you installed libusb(1)-dev but it is still not detected, maybe it is because pkg-config is not installed. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error making dahdi linux compete 2.11.0
On Mon, Feb 15, 2016 at 02:15:58PM +, Ryan, Travis wrote: > Getting the some errors making dahdi 2.11.0. > > Seems same as listed here http://forums.asterisk.org/viewtopic.php?f=1&t=96455 > > In that link they say to use 2.10.2 but that's from December. Is there a fix > yet for this? My bad, I forgot to push the fix for that. http://git.asterisk.org/gitweb/?p=dahdi/tools.git;a=commit;h=e1da7b528467a8f8f82058993b2e01333677ee39 You may need to run autoreconf after applying this patch to rebuild the Makefile (though IIRC the makefile will be re-generated by running 'make'). -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Compile error with libpri 1.4.15
On Tue, Feb 02, 2016 at 11:57:08AM -0500, Jerry Geis wrote: > make[4]: Entering directory > `/home/silentm/LayeredSolutions/digium/dahdi-linux-complete-2.11.0+2.11.0/tools/xpp' > /usr/bin/install: cannot stat ‘./dahdi_registration.8’: No such file or > directory > /usr/bin/install: cannot stat ‘./xpp_sync.8’: No such file or directory > /usr/bin/install: cannot stat ‘./lsdahdi.8’: No such file or directory > /usr/bin/install: cannot stat ‘./xpp_blink.8’: No such file or directory > /usr/bin/install: cannot stat ‘./dahdi_genconf.8’: No such file or directory > /usr/bin/install: cannot stat ‘./dahdi_hardware.8’: No such file or > directory > /usr/bin/install: cannot stat ‘./twinstar.8’: No such file or directory Those should have been generated by pod2man. Look for '%.8: %' in xpp/Makefile.am . So it seems that a check for pod2man needs to be added to the configure script. For now, generate a dummy pod2man binary somewhere in your path. Or just install perl-doc or whatever it is called. For instance: ln -s /bun/true /usr/local/bin/pod2man # Results in empty man pages I stress that this is a TEMPORARY WORKAROUND until the configure script is adapted. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dahdi on systemd (CentOS 7)
On Tue, Feb 02, 2016 at 10:07:42AM -0500, Jerry Geis wrote: > It doesnt appear dahdi is starting up under systemd and CentOS 7.2 > > What should I look for? > > find /etc/systemd | grep -i dahdi > > find nothing. /etc/systemd is for services installed by the system administrator. You should have also looked under /usr/lib/systemd . Alternatively: systemctl status dah # completes to: systemctl status dahdi.service Which will shows me that dahdi.service was generated from /etc/init.d/dahdi . Or: systemctl list-units | grep dahdi > > How does dahdi startup under systemd ? Right now with the same init.d script. That said, I'd like to avoid using it. If you switch to automatic span assignment (http://docs.tzafrir.org.il/dahdi-linux/#_span_assignments - auto_assign_spans=0), you don't really need the DAHDI init script. In fact, using it may become confusing, as under some circumstances the dahdi "service" may be in the wrong state for you (and unloading modules at poweroff is pointless). Thus, in the spirit of the parallel boot of systemd, my general recommendation is not to have any dahdi service, and just let spans load and get initialized separately. This also means you no longer need to start DAHDI before Asterisk. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trying to compile DAHDI on Pidora 2014 (RPi)
On Fri, Nov 13, 2015 at 04:01:33PM -0600, Carlos Chavez wrote: > I just purchased an Amfeltec USB-FXO adapter and am trying to compile > DAHDI 2.10 on a Raspberry PI running Pidora 2014 R3. I have all the > dependencies but I get an error and cannot finish. Is it even possible to > compile DAHDI for the ARM plataform? Here is the error I am getting: > > root@astpi dahdi-linux-complete-2.10.2+2.10.2]# make > make -C linux all > make[1]: Entering directory > `/usr/src/dahdi-linux-complete-2.10.2+2.10.2/linux' > make -C drivers/dahdi/firmware firmware-loaders > make[2]: Entering directory > `/usr/src/dahdi-linux-complete-2.10.2+2.10.2/linux/drivers/dahdi/firmware' > make[2]: Leaving directory > `/usr/src/dahdi-linux-complete-2.10.2+2.10.2/linux/drivers/dahdi/firmware' > make -C > /lib/modules/3.12.26-1.20140808git4ab8abb.rpfr20.armv6hl.bcm2708/build > SUBDIRS=/usr/src/dahdi-linux-complete-2.10.2+2.10.2/linux/drivers/dahdi > DAHDI_INCLUDE=/usr/src/dahdi-linux-complete-2.10.2+2.10.2/linux/include > DAHDI_MODULES_EXTRA=" " HOTPLUG_FIRMWARE=yes modules DAHDI_BUILD_ALL=m > make[2]: Entering directory > `/usr/src/kernels/3.12.26-1.20140808git4ab8abb.rpfr20.armv6hl.bcm2708' > /usr/src/dahdi-linux-complete-2.10.2+2.10.2/linux/drivers/dahdi/Kbuild:124: > CPU Architecture 'arm' does not support VPMADT032 or HPEC. Skipping. > CC [M] > /usr/src/dahdi-linux-complete-2.10.2+2.10.2/linux/drivers/dahdi/dahdi-base.o > In file included from > /usr/src/dahdi-linux-complete-2.10.2+2.10.2/linux/drivers/dahdi/dahdi-base.c:68:0: > /usr/src/dahdi-linux-complete-2.10.2+2.10.2/linux/include/dahdi/kernel.h:63:5: > warning: "LINUX_VERSION_CODE" is not defined [-Wundef] > #if LINUX_VERSION_CODE >= KERNEL_VERSION(2,6,19) > ^ Something seems to be wrong with your kernel headers. Perhaps you need to install a matchig kernel-devel package. What is the output of: ls -l /lib/modules/3.12.26-1.20140808git4ab8abb.rpfr20.armv6hl.bcm2708/build ? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk 13 systemd
On Sat, Nov 07, 2015 at 09:34:33AM +0100, Ludovic Gasc wrote: > I've some Asterisk 13 on production, it's a custom compilation + I've > retrieved systemd configuration file from asterisk Debian package of > unstable. > > After a small adaptation, I've no issue like you, however, I use Debian > Jessie. Speaking of those (that use the pending review for a systemd unit): https://bugs.debian.org/801629 -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] oslec echo cancellation
On Thu, Oct 22, 2015 at 11:46:38AM -0200, Ethy H. Brito wrote: > On Thu, 22 Oct 2015 15:06:59 +0300 > Tzafrir Cohen wrote: > > > On Wed, Oct 21, 2015 at 01:33:27PM -0200, Ethy H. Brito wrote: > > > > > > Hi > > > > > > Who should insert dahdi_echocan_oslec.ko module in Ubuntu 14.04? > > > > dahdi_echocan_oslec should be built by default. What you may miss is the > > 'echo.ko' (OSLEC) kernel module. > > Ooopss. > > I did not make myself clear. > > All modules are in place and working - *if* I "modprobe" them manually. > > The problem is "service dahdi start" does not load echo+dahdi_echocan_oslec. > > I have a production system that does it. I just do not know how. > > After copying this system to another machine and upgrade it (apt-get > dist-upgrade), echo+dahdi_echocan_oslec does not get inserted anymore and I am > wondering why. The echo canceller modules get loaded automatically when needed: if you have the line 'echocanceller=foo,12' in /etc/dahdi/system.conf and run dahdi_cfg, dahdi_cfg will attempt to configure the echo canceller 'foo' for channel 12. The DAHDI core (kernel code) will notice it has no such echo canceller and try to load a module caller 'dahdi_echocan_foo'. So, what do you have in /etc/dahdi/system.conf ? What is the output of lsdahdi? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] oslec echo cancellation
On Wed, Oct 21, 2015 at 01:33:27PM -0200, Ethy H. Brito wrote: > > Hi > > Who should insert dahdi_echocan_oslec.ko module in Ubuntu 14.04? dahdi_echocan_oslec should be built by default. What you may miss is the 'echo.ko' (OSLEC) kernel module. Try: /sbin/modinfo echo Do you have it available? The https://notabug.org/tzafrir/dahdi-linux-extra repository is where I maintain a copy of DAHDI with OSLEC and from it I create the OSLEC patch for the DAHDI package in Debian. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Xorcom T1 to PRI
Hi, This seems to be an issue of terminology, On Thu, Sep 24, 2015 at 05:08:24PM -0500, Jeff LaCoursiere wrote: > > Hi, > > I have a client that has a 24 channel voice T1 that I have been using e&m > signalling over for a number of years. The local telco finally got an ISDN > switch and wants to move them to PRI. I didn't see this as a big problem - > I've done a few others on this particular Caribbean island without issues, > but this would be the first time with a Xorcom unit involved. > > We tried to do it tonight and failed. I suspect there is an issue at their > central office where they patched the line away from an old dms-100 and to > the new ISDN switch - the tech said he couldn't see anything at the line > layer, though I showed no alarms. Regardless, something odd happened that I > am hoping for some advice on. We reverted to the T1 and will try again next > week. > > The odd bit is this - lsdahdi showed all 24 channels as "T1" instead of > "PRI". The output of lsdahdi at another site where I had already done this > conversion (which uses a Sangoma card) looks like this: > > root@astsouth:/etc/dahdi# lsdahdi > ### Span 1: WPT1/0 "wanpipe1 card 0" (MASTER) ESF/B8ZS > 1 PRIClear (In use) (EC: WANPIPE_HWEC - INACTIVE) > 2 PRIClear (In use) (EC: WANPIPE_HWEC - INACTIVE) > 3 PRIClear (In use) (EC: WANPIPE_HWEC - INACTIVE) > 4 PRIClear (In use) (EC: WANPIPE_HWEC - INACTIVE) > 5 PRIClear (In use) (EC: WANPIPE_HWEC - INACTIVE) > 6 PRIClear (In use) (EC: WANPIPE_HWEC - INACTIVE) > 7 PRIClear (In use) (EC: WANPIPE_HWEC - INACTIVE) > 8 PRIClear (In use) (EC: WANPIPE_HWEC - INACTIVE) > 9 PRIClear (In use) (EC: WANPIPE_HWEC - INACTIVE) > 10 PRIClear (In use) (EC: WANPIPE_HWEC - INACTIVE) > 11 PRIClear (In use) (EC: WANPIPE_HWEC - INACTIVE) > 12 PRIClear (In use) (EC: WANPIPE_HWEC - INACTIVE) > 13 PRIClear (In use) (EC: WANPIPE_HWEC - INACTIVE) > 14 PRIClear (In use) (EC: WANPIPE_HWEC - INACTIVE) > 15 PRIClear (In use) (EC: WANPIPE_HWEC - INACTIVE) > 16 PRIClear (In use) (EC: WANPIPE_HWEC - INACTIVE) > 17 PRIClear (In use) (EC: WANPIPE_HWEC - INACTIVE) > 18 PRIClear (In use) (EC: WANPIPE_HWEC - INACTIVE) > 19 PRIClear (In use) (EC: WANPIPE_HWEC - INACTIVE) > 20 PRIClear (In use) (EC: WANPIPE_HWEC - INACTIVE) > 21 PRIClear (In use) (EC: WANPIPE_HWEC - INACTIVE) > 22 PRIClear (In use) (EC: WANPIPE_HWEC - INACTIVE) > 23 PRIClear (In use) (EC: WANPIPE_HWEC - INACTIVE) > 24 PRIHardware-assisted HDLC (In use) (EC: WANPIPE_HWEC - > INACTIVE) > > Bah, the scrollback has already erased the output from the box in question, > but essentially it showed the same as above with "T1" in the second column > for all channels instead of "PRI". Right. For Sangoma cards, lsdahdi can't tell if the port is E1 or T1 and thus calls it "PRI". Note that "PRI" here is a poor name that refers to the port type itself and not to the signalling in it (which don't have to be ISDN). Suggestions? Patches? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Tdm4010p 4 port card
On Fri, Jul 10, 2015 at 11:37:20AM -0400, Tom Judge wrote: > > Hi running asterisk 13.x and dahdi-linux-complete-2.10.2+2.10.2. > > It looks like the kernel drivers lod but in asterisk console dahdi > show anything not working. Trina to use a TDM410P pci card. Is this > just too old and extinct card? I guess not. > Any suggestions gratefully apprecuated. Please provide the output from the following commands (in the Linux command line): dahdi_span_assignments list lsdahdi egrep '^(#include|channel)' /etc/asterisk/chan_dahdi.conf rasterisk -x 'dahdi show channels' -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users