Re: [asterisk-users] Faxing: Anyone have a compiled executable?
At 23:04 1/16/2010, Tilghman Lesher wrote: That's incorrect. module show shows only those modules which are currently loaded. BTW, there is also the command module show like fax, which is much easier than typing out the whole module name, may show you more modules than you were aware of, and might be extremely helpful by showing you other related modules that are already loaded. Thanks, guys. ~~~ CLI module show like fax Module Description Use Count 0 modules loaded CLI module show like zt Module Description Use Count 0 modules loaded CLI module show like zap Module Description Use Count app_zapateller.so Block Telemarketers with Special Informa 0 1 modules loaded ~~~ No joy. Read this and recompiled Asterisk: http://ibot.rikers.org/%23asterisk/20090618.html.gz Got these messages: WARNING WARNING WARNING Your Asterisk modules directory, located at /usr/lib/asterisk/modules contains modules that were not installed by this version of Asterisk. Please ensure that these modules are compatible with this version before attempting to run Asterisk. app_fax.so app_saycountpl.so chan_ooh323.so format_mp3.so Read something else and found this in: /var/log/asterisk/messages [Jan 17 01:28:16] NOTICE[2479] loader.c: 145 modules will be loaded. [Jan 17 01:28:16] WARNING[2479] loader.c: Error loading module 'app_fax.so': libspandsp.so.2: cannot open shared object file: No such file or directory [Jan 17 01:28:17] WARNING[2479] res_smdi.c: No SMDI interfaces are available to listen on, not starting SMDI listener. [Jan 17 01:28:19] WARNING[2479] loader.c: Error loading module 'app_fax.so': libspandsp.so.2: cannot open shared object file: No such file or directory [Jan 17 01:28:19] WARNING[2479] loader.c: Module 'app_fax.so' could not be loaded. [Jan 17 01:28:19] ERROR[2479] chan_dahdi.c: Unable to load zapata.conf [Jan 17 01:28:20] NOTICE[2479] chan_ooh323.c: - --- *** IMPORTANT NOTE *** --- --- This module is currently unsupported. Use it at your own risk. --- - Does libspandsp.so.2 need to be copied to someplace else? # find / -name libspandsp.so.2* /usr/src/asterisk/spandsp/spandsp-0.0.6/src/.libs/libspandsp.so.2.0.0 /usr/src/asterisk/spandsp/spandsp-0.0.6/src/.libs/libspandsp.so.2 /usr/local/lib/libspandsp.so.2.0.0 /usr/local/lib/libspandsp.so.2 Do I need a zapata.conf if I am using ztdummy? # find / -name zapata.conf # Any other ideas? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Faxing: Anyone have a compiled executable?
Hallo there! I had my own experience get RxFax/TxFax successful running with spandsp. I only got spandsp-0.0.4 running, because on newer package, there aren't created some needed libraries (don't remember the right one this moment) *find /usr -iname \*spandsp\** shows me following output: /usr/lib/libspandsp.so /usr/lib/pkgconfig/spandsp.pc /usr/lib/libspandsp.so.0.0.2 /usr/lib/libspandsp.a /usr/lib/libspandsp.so.2 /usr/lib/libspandsp.so.0 /usr/lib/libspandsp.so.2.0.0 /usr/lib/libspandsp.la and debian-server*CLI *core show application RxFAX* debian-server*CLI -= Info about application 'RxFAX' =- [Synopsis] Receive a FAX to a file [Description] RxFAX(filename[|caller][|debug]): Receives a FAX from the channel into the given filename. If the file exists it will be overwritten. The file should be in TIFF/F format. The caller option makes the application behave as a calling machine, rather than the answering machine. The default behaviour is to behave as an answering machine. Uses LOCALSTATIONID to identify itself to the remote end. LOCALHEADERINFO to generate a header line on each page. Sets REMOTESTATIONID to the sender CSID. FAXPAGES to the number of pages received. FAXBITRATE to the transmition rate. FAXRESOLUTION to the resolution. Returns -1 when the user hangs up. Returns 0 otherwise. debian-server*CLI *core show application TxFAX* debian-server*CLI -= Info about application 'TxFAX' =- [Synopsis] Send a FAX file [Description] TxFAX(filename[|caller][|debug]): Send a given TIFF file to the channel as a FAX. The caller option makes the application behave as a calling machine, rather than the answering machine. The default behaviour is to behave as an answering machine. Uses LOCALSTATIONID to identify itself to the remote end. LOCALHEADERINFO to generate a header line on each page. Sets REMOTESTATIONID to the receiver CSID. Returns -1 when the user hangs up, or if the file does not exist. Returns 0 otherwise. Regards Am 17.01.2010 09:11, schrieb Doug: At 23:04 1/16/2010, Tilghman Lesher wrote: That's incorrect. module show shows only those modules which are currently loaded. BTW, there is also the command module show like fax, which is much easier than typing out the whole module name, may show you more modules than you were aware of, and might be extremely helpful by showing you other related modules that are already loaded. Thanks, guys. ~~~ CLI module show like fax Module Description Use Count 0 modules loaded CLI module show like zt Module Description Use Count 0 modules loaded CLI module show like zap Module Description Use Count app_zapateller.so Block Telemarketers with Special Informa 0 1 modules loaded ~~~ No joy. Read this and recompiled Asterisk: http://ibot.rikers.org/%23asterisk/20090618.html.gz Got these messages: WARNING WARNING WARNING Your Asterisk modules directory, located at /usr/lib/asterisk/modules contains modules that were not installed by this version of Asterisk. Please ensure that these modules are compatible with this version before attempting to run Asterisk. app_fax.so app_saycountpl.so chan_ooh323.so format_mp3.so Read something else and found this in: /var/log/asterisk/messages [Jan 17 01:28:16] NOTICE[2479] loader.c: 145 modules will be loaded. [Jan 17 01:28:16] WARNING[2479] loader.c: Error loading module 'app_fax.so': libspandsp.so.2: cannot open shared object file: No such file or directory [Jan 17 01:28:17] WARNING[2479] res_smdi.c: No SMDI interfaces are available to listen on, not starting SMDI listener. [Jan 17 01:28:19] WARNING[2479] loader.c: Error loading module 'app_fax.so': libspandsp.so.2: cannot open shared object file: No such file or directory [Jan 17 01:28:19] WARNING[2479] loader.c: Module 'app_fax.so' could not be loaded. [Jan 17 01:28:19] ERROR[2479] chan_dahdi.c: Unable to load zapata.conf [Jan 17 01:28:20] NOTICE[2479] chan_ooh323.c: - --- *** IMPORTANT NOTE *** --- --- This module is currently unsupported. Use it at your own risk. --- - Does libspandsp.so.2 need to be copied to someplace else? # find / -name libspandsp.so.2* /usr/src/asterisk/spandsp/spandsp-0.0.6/src/.libs/libspandsp.so.2.0.0 /usr/src/asterisk/spandsp/spandsp-0.0.6/src/.libs/libspandsp.so.2 /usr/local/lib/libspandsp.so.2.0.0 /usr/local/lib/libspandsp.so.2 Do I need a zapata.conf if I am using ztdummy? # find / -name zapata.conf # Any other ideas? --
Re: [asterisk-users] Faxing: Anyone have a compiled executable?
On Sun, Jan 17, 2010 at 11:10:25AM +0100, IT-Connect wrote: Hallo there! I had my own experience get RxFax/TxFax successful running with spandsp. I only got spandsp-0.0.4 running, because on newer package, there aren't created some needed libraries (don't remember the right one this moment) *find /usr -iname \*spandsp\** shows me following output: /usr/lib/libspandsp.so /usr/lib/pkgconfig/spandsp.pc /usr/lib/libspandsp.so.0.0.2 /usr/lib/libspandsp.a /usr/lib/libspandsp.so.2 /usr/lib/libspandsp.so.0 /usr/lib/libspandsp.so.2.0.0 /usr/lib/libspandsp.la ... [Jan 17 01:28:16] NOTICE[2479] loader.c: 145 modules will be loaded. [Jan 17 01:28:16] WARNING[2479] loader.c: Error loading module 'app_fax.so': libspandsp.so.2: cannot open shared object file: No such file or directory What is the output of: ls -l /usr/lib/libspandsp.so* ldd /usr/lib/modules/app_fax.so ldd /usr/lib/libspandsp.so.2 -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Faxing: Anyone have a compiled executable?
Am 17.01.2010 11:28, schrieb Tzafrir Cohen: What is the output of: ls -l /usr/lib/libspandsp.so* lrwxrwxrwx 1 root root 19 18. Sep 19:40 /usr/lib/libspandsp.so - libspandsp.so.0.0.2 lrwxrwxrwx 1 root root 19 18. Sep 19:40 /usr/lib/libspandsp.so.0 - libspandsp.so.0.0.2 -rwxr-xr-x 1 root root 1401295 18. Sep 19:40 /usr/lib/libspandsp.so.0.0.2 lrwxrwxrwx 1 root root 19 18. Sep 19:31 /usr/lib/libspandsp.so.2 - libspandsp.so.2.0.0 -rwxr-xr-x 1 root root 1566819 18. Sep 19:31 /usr/lib/libspandsp.so.2.0.0 ldd /usr/lib/modules/app_fax.so O.k., I think, you use another application for app_fax? I've only *app_rxfax.so* and *app_txfax.so* and shows me following output: *ldd /usr/lib/asterisk/modules/app_rxfax.so* linux-gate.so.1 = (0xb77e3000) libspandsp.so.0 = /usr/lib/libspandsp.so.0 (0xb772e000) libpthread.so.0 = /lib/i686/cmov/libpthread.so.0 (0xb7715000) libc.so.6 = /lib/i686/cmov/libc.so.6 (0xb75ba000) libm.so.6 = /lib/i686/cmov/libm.so.6 (0xb7594000) libtiff.so.4 = /usr/lib/libtiff.so.4 (0xb753f000) /lib/ld-linux.so.2 (0xb77e4000) libjpeg.so.62 = /usr/lib/libjpeg.so.62 (0xb751f000) libz.so.1 = /usr/lib/libz.so.1 (0xb750a000) *ldd /usr/lib/asterisk/modules/app_txfax.so* linux-gate.so.1 = (0xb78b9000) libspandsp.so.0 = /usr/lib/libspandsp.so.0 (0xb7805000) libpthread.so.0 = /lib/i686/cmov/libpthread.so.0 (0xb77ec000) libc.so.6 = /lib/i686/cmov/libc.so.6 (0xb7691000) libm.so.6 = /lib/i686/cmov/libm.so.6 (0xb766b000) libtiff.so.4 = /usr/lib/libtiff.so.4 (0xb7616000) /lib/ld-linux.so.2 (0xb78ba000) libjpeg.so.62 = /usr/lib/libjpeg.so.62 (0xb75f6000) libz.so.1 = /usr/lib/libz.so.1 (0xb75e1000) * ldd /usr/lib/libspandsp.so.2* linux-gate.so.1 = (0xb7858000) libtiff.so.4 = /usr/lib/libtiff.so.4 (0xb7753000) libm.so.6 = /lib/i686/cmov/libm.so.6 (0xb772d000) libc.so.6 = /lib/i686/cmov/libc.so.6 (0xb75d2000) libjpeg.so.62 = /usr/lib/libjpeg.so.62 (0xb75b3000) libz.so.1 = /usr/lib/libz.so.1 (0xb759e000) /lib/ld-linux.so.2 (0xb7859000) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to escape the Pound-Char in Callfiles
Hello, I'm using Asterisk 1.6.2.0 and I like to call extension #8 from callfile. Unfortunately the #-char ist interpreted just as comment. I got a Invalid file contents in /var/spool/asterisk/outgoing/callfile, deleting from asterisk. When I try to escape with \ oder use quotes, I got: \#8,1 failed so falling back to exten 's' or #8,1 failed so falling back to exten 's' TIA, Dominik -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to escape characters in Dialplan
Hello,
I'm using Asterisk 1.6.2.0 and I like to use escape characters with SendText,
because I can just delete the message from my phone (Thomson Speedtouch
ST2030) display by sending a return-char (\n).
But \n is not escaped: I tried already:
exten = 222, n, SendText(\n)
exten = 222, n, SendText(\n)
exten = 222, n, SendText('\n')
exten = 222, n, SendText(`\n`)
So how can I use escape characters in dialplan?
TIA,
Dominik
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Re: [asterisk-users] Faxing: Anyone have a compiled executable?
I accidentally answered someone else. So here goes again: On Sun, Jan 17, 2010 at 02:11:17AM -0600, Doug wrote: [Jan 17 01:28:16] NOTICE[2479] loader.c: 145 modules will be loaded. [Jan 17 01:28:16] WARNING[2479] loader.c: Error loading module 'app_fax.so': libspandsp.so.2: cannot open shared object file: No such file or directory That's the problem. Does libspandsp.so.2 need to be copied to someplace else? # find / -name libspandsp.so.2* /usr/src/asterisk/spandsp/spandsp-0.0.6/src/.libs/libspandsp.so.2.0.0 /usr/src/asterisk/spandsp/spandsp-0.0.6/src/.libs/libspandsp.so.2 /usr/local/lib/libspandsp.so.2.0.0 /usr/local/lib/libspandsp.so.2 What is the output of: ls -l /usr/local/lib/libspandsp.so* ldd /usr/lib/asterisk/modules/app_fax.so ldd /usr/lib/libspandsp.so.2 Do I need a zapata.conf if I am using ztdummy? No, you don't need it if you merely want to use DAHDI as a timing source. Not sure about using it in app_meetme, though. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Faxing: Anyone have a compiled executable?
On 01/17/2010 06:10 PM, IT-Connect wrote: Hallo there! I had my own experience get RxFax/TxFax successful running with spandsp. I only got spandsp-0.0.4 running, because on newer package, there aren't created some needed libraries (don't remember the right one this moment) Spandsp only creates one library, unless you build it with the test suite enabled. With the test suite it will build spandsp-sim as well as spandsp. I don't remember any version of spandsp that entirely failed to build or install any important components. *find /usr -iname \*spandsp\** shows me following output: /usr/lib/libspandsp.so /usr/lib/pkgconfig/spandsp.pc /usr/lib/libspandsp.so.0.0.2 That one might be spandsp-0.0.4 /usr/lib/libspandsp.a /usr/lib/libspandsp.so.2 /usr/lib/libspandsp.so.0 /usr/lib/libspandsp.so.2.0.0 but that one is something newer. /usr/lib/libspandsp.la You have at least two versions versions of spandsp on your system. Do you have any more installed in directories like /usr/local/lib? Were these installed from RPMs or DEBs, or were they built and installed by you? If you built them the installed header files will come from the last version you installed. If they came from RPMs or DEBs, you should be able to find which set of headers is installed by checking which development package is installed. Regards, Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cross compiling Asterisk, Dahdi..
On Sun, 17 Jan 2010, Tzafrir Cohen wrote: On Sat, Jan 16, 2010 at 03:54:44PM +, Gordon Henderson wrote: Is there a proper, documented way to cross compile DAHDI and Asterisk for a processor/system other than the one you're currently typing on? Now.. I have been doing this for some time, but it's been really frustrating every time I change/upgrade, etc. I've just tried to compile DAHDI for an AMD Geode system on my development system which is Intel Atom. Building the kernel is easy - been doing that for years, but DAHDI is just hard and does the wrong thing. So I start by hardwiring HOTPLUG to no because my target device doesn't support nor need it. HOTPLUG is a slightly misleading name. If it is enabled, it means firmware loading from userspace is enabled in the kernel. If so, drivers for some digium cards will not include the firmware inside them. Most system I know support firmware loading. If you don't use those cards, those drivers will simply be smaller (as they don't include the firmware blobs). In short: leave this one for automatic detection. None of the cards I use require firmware loading. OK - I build very precise and specific systems. Call me old fashioned if you like, but I compile a kernel with the exact requirements for my systems - no hotplug, no udev, no modules - just a precise kernel and a cut-down installation of my own devising, but it's based on Debian. The hotplug check fails on my systems - not sure why, but I've always had to force it to no (as advised by the comments in the makefile!). Then setting KVERS to be the correct thing, Hmm... I'm not really sure if KVERS is still used (if you explicitly set KSRC, that is). Comment in the Makefile under: dahdi-linux-complete-2.2.0.2+2.2.0/linux # If you want to build for a kernel other than the current kernel, set KVERS So I set KVERS in the environment. That is the only way to get the build process to find the modules directory for my target kernel (cross compiled on my build system). So # echo $KVERS 2.6.32.3-dsx-geode and this is picked up by the Makefile, but I really want -march=geode and the only way I've found to get this is to edit Kbuild directly. Kbuild should do that for you. Or rather: if you used that for building the kernel, it should also be used for DAHDI. If this doesn't work, I suspect your kernel tree is misconfigured. I doubt it, but am willing to check - it's a vanilla kernel off kernel.org, compiled as per the instructions - the way I've been doing it for ever. I use make menuselect, then select the options I want. Module loading is enabled. Make the kernel (make bzImage), then make modules and make modules_install. Not that there are any modules, but it populates /lib/modules/2.6.32.3-dsx-geode with the right stuff. Reminder: to make Kbuild print the full build lines, use: make V=1 Actually, that's handy. I was grepping the output of 'ps' to find the gcc command lines to see what it was doing and make sure it was picking up -march=geode (And comment out all the modules I really don't want to build like torisa, xpp, etc. Even then it still barfed on the VPMADT032 loader, so I just commented that whole section out. What error did you get? WARNING: voicebus_get_pci_dev [/var/space/local/src/dsx/dahdi-linux-complete-2.2.0.2+2.2.0/linux/drivers/dahdi/dahdi_vpmadt032_loader.ko] undefined! And there are a few more like that. Maybe the warnings can be ignored, but my guess is that it's looking for a loader in a kernel that's not configured for hotplug. Now, at install time, it's fiddling with system files on my build box that it really should not be touching at all - output from make: [ `id -u` = 0 ] /sbin/depmod -a 2.6.32.3-dsx-geode || : install -d /etc/udev/rules.d build_tools/genudevrules /etc/udev/rules.d/dahdi.rules build_tools/genudevrules: line 3: udevinfo: command not found build_tools/genudevrules: line 7: udevadm: command not found install -m 644 drivers/dahdi/xpp/xpp.rules /etc/udev/rules.d/ for hdr in kernel.h user.h fasthdlc.h wctdm_user.h dahdi_config.h; do \ install -D -m 644 include/dahdi/$hdr /usr/include/dahdi/$hdr; \ done rmdir: failed to remove `/usr/include/zaptel': No such file or directory make: [install-include] Error 1 (ignored) I don't use udev on my build system, nor my target systems so why is it bothering... But I feel there really ought to be a means to tell it that it's not building for the local system, so don't fiddle with local files... You don't use udev at all? Not at all. In this case those static device files will actually have to be created on the target system. Yes, and I don't have a problem with that. I note you didn't really include the commands you used. I don't think the commands I used are actually that relevant here, but since you asked: make vi and a few other, cp, mv, and maybe tar to unpack things. As I said above, it's
Re: [asterisk-users] Cross compiling Asterisk, Dahdi..
On Sun, 17 Jan 2010, Tzafrir Cohen wrote: On Sat, Jan 16, 2010 at 07:00:26AM -1000, Jean-Denis Girard wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi Gordon Gordon Henderson a écrit : Is there a proper, documented way to cross compile DAHDI and Asterisk for a processor/system other than the one you're currently typing on? Here is what I'm doing for building dahdi modules on my x86_64 system, for geode target. In dahdi linux directory: make KVERS=2.6.33-rc3-git3-sysnux KSRC=/home/jdg/RPM/BUILD/linux Then install in /tmp/dahdi: make DESTDIR=/tmp/dahdi ARCH=i386 KVERS=2.6.33-rc3-git3-sysnux KSRC=/home/jdg/RPM/BUILD/linux install-modules Is an explicit ARCH needed? It shouldn't have been there in the first place. The ARCH is caculated by Kbuild from your config (in the kernel tree) and there should be no need to provide it (at least as of dahdi 2.2). Likewise: is KVERS really needed in that line? When your building on one platform (lets say Intel Atom) for a kernel running on a different platform, (e.g. ARM) The build process can't get the kernel version by any other means, so KVERS is needed at compile time to let the makefiles find the target kernel, and from there, it can find the target kernel source tree. And I'd like to think setting ARCH wasn't needed - as I'd like to think the build process can infer the target architecture from the kernel source tree, but it doesn't seem to be able to. Until I explicity set it in the KBuild file, the process was compiling for the Atom. Of-course myself and Jean-Denis could both be doing something wrong... However, we both seem to have a method that works for us - it's elegant enough - but it would be better if the make process recognised it was building for a system that's not the one it's being compiled on and didn't noodle local files on the build system. Gordon-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cross compiling Asterisk, Dahdi..
On Sun, Jan 17, 2010 at 11:37:43AM +, Gordon Henderson wrote: On Sun, 17 Jan 2010, Tzafrir Cohen wrote: On Sat, Jan 16, 2010 at 03:54:44PM +, Gordon Henderson wrote: Is there a proper, documented way to cross compile DAHDI and Asterisk for a processor/system other than the one you're currently typing on? Now.. I have been doing this for some time, but it's been really frustrating every time I change/upgrade, etc. I've just tried to compile DAHDI for an AMD Geode system on my development system which is Intel Atom. Building the kernel is easy - been doing that for years, but DAHDI is just hard and does the wrong thing. So I start by hardwiring HOTPLUG to no because my target device doesn't support nor need it. HOTPLUG is a slightly misleading name. If it is enabled, it means firmware loading from userspace is enabled in the kernel. If so, drivers for some digium cards will not include the firmware inside them. Most system I know support firmware loading. If you don't use those cards, those drivers will simply be smaller (as they don't include the firmware blobs). In short: leave this one for automatic detection. None of the cards I use require firmware loading. OK - I build very precise and specific systems. Call me old fashioned if you like, but I compile a kernel with the exact requirements for my systems - no hotplug, no udev, no modules - just a precise kernel and a cut-down installation of my own devising, but it's based on Debian. The hotplug check fails on my systems - not sure why, but I've always had to force it to no (as advised by the comments in the makefile!). Then setting KVERS to be the correct thing, Hmm... I'm not really sure if KVERS is still used (if you explicitly set KSRC, that is). Comment in the Makefile under: dahdi-linux-complete-2.2.0.2+2.2.0/linux # If you want to build for a kernel other than the current kernel, set KVERS So I set KVERS in the environment. That is the only way to get the build process to find the modules directory for my target kernel (cross compiled on my build system). So # echo $KVERS 2.6.32.3-dsx-geode and this is picked up by the Makefile, but I really want -march=geode and the only way I've found to get this is to edit Kbuild directly. Kbuild should do that for you. Or rather: if you used that for building the kernel, it should also be used for DAHDI. If this doesn't work, I suspect your kernel tree is misconfigured. I doubt it, but am willing to check - it's a vanilla kernel off kernel.org, compiled as per the instructions - the way I've been doing it for ever. I use make menuselect, then select the options I want. Module loading is enabled. Make the kernel (make bzImage) FWIW, 'make modules_prepare' should be good enough for building (or at least: test-building) modules. And takes less time. , then make modules and make modules_install. Not that there are any modules, but it populates /lib/modules/2.6.32.3-dsx-geode with the right stuff. Reminder: to make Kbuild print the full build lines, use: make V=1 Actually, that's handy. I was grepping the output of 'ps' to find the gcc command lines to see what it was doing and make sure it was picking up -march=geode (And comment out all the modules I really don't want to build like torisa, xpp, etc. Even then it still barfed on the VPMADT032 loader, so I just commented that whole section out. What error did you get? WARNING: voicebus_get_pci_dev [/var/space/local/src/dsx/dahdi-linux-complete-2.2.0.2+2.2.0/linux/drivers/dahdi/dahdi_vpmadt032_loader.ko] undefined! And there are a few more like that. Maybe the warnings can be ignored, but my guess is that it's looking for a loader in a kernel that's not configured for hotplug. If you build a custom kernel anyway, maybe the simplest approach would be to copy the dahdi files onto the kernel tree and build it there. drivers/dahdi/Kconfig has: config DAHDI_VOICEBUS tristate VoiceBus(tm) Interface Library depends on PCI default DAHDI An example entry for a card that uses it: config DAHDI_WCTDM24XXP tristate Digium Wildcard VoiceBus analog card Support depends on DAHDI DAHDI_VOICEBUS default DAHDI Thus if the instructions from Kconfig are applied, you'll default not to build any PCI driver without any further effort. Sadly it is not applied when you build DAHDI as modules. You'll still have to create the static device files. See http://svn.debian.org/viewsvn/pkg-voip/dahdi-linux/trunk/debian/make_static_nodes Now, at install time, it's fiddling with system files on my build box that it really should not be touching at all - output from make: [ `id -u` = 0 ] /sbin/depmod -a 2.6.32.3-dsx-geode || : install -d /etc/udev/rules.d build_tools/genudevrules /etc/udev/rules.d/dahdi.rules build_tools/genudevrules: line 3:
Re: [asterisk-users] How to escape characters in Dialplan
Somewhere \n needs to be converted into utf8 new line. Asterisk should
do this for you but it doesnt.
Try opening the dialplan in hex mode and insert hex code for utf8 new
line where the line break should be.
Peter
On 17 jan 2010, at 12.09, Dominik wrote:
Hello,
I'm using Asterisk 1.6.2.0 and I like to use escape characters with
SendText,
because I can just delete the message from my phone (Thomson
Speedtouch
ST2030) display by sending a return-char (\n).
But \n is not escaped: I tried already:
exten = 222, n, SendText(\n)
exten = 222, n, SendText(\n)
exten = 222, n, SendText('\n')
exten = 222, n, SendText(`\n`)
So how can I use escape characters in dialplan?
TIA,
Dominik
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Re: [asterisk-users] Changing ring cadence on FXS lines
On Fri, Jan 15, 2010 at 04:52:15PM +, Noah I. Engelberth wrote: Is there a way I can change the ring cadence on FXS lines on a system using a Digium Wildcard TDM2400 card? I recently deployed a new phone system, and the customer has a few POTS phones in areas where they don't have data network services, so we're using the FXS lines to provide dialtone at those outbuildings. The old phone system would ring those phones with a short ring-short ring-pause cadence, which sounds louder to the users than Asterisk's default long ring-pause cadence. I tried setting a cadence line in chan_dahdi.conf and restarting Asterisk, and typing dahdi show cadences in the CLI after the restart showed my custom cadence, but the phones were still ringing long ring-pause. Can someone point me in the direction of what I'm doing wrong? Look for the word 'cadence' in the sample chan_dahdi.conf . Also, use DAHDI/1r3 instead of DAHDI/1 for custom ring cadence no. 3. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] receive text
Is there any code that I can cut/paste that will allow me to receive an SMS text on Asterisk? and, where can I capture the incoming text? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Faxing: Anyone have a compiled executable?
On 01/17/2010 04:11 PM, Doug wrote: At 23:04 1/16/2010, Tilghman Lesher wrote: That's incorrect. module show shows only those modules which are currently loaded. BTW, there is also the command module show like fax, which is much easier than typing out the whole module name, may show you more modules than you were aware of, and might be extremely helpful by showing you other related modules that are already loaded. Thanks, guys. ~~~ CLI module show like fax Module Description Use Count 0 modules loaded CLI module show like zt Module Description Use Count 0 modules loaded CLI module show like zap Module Description Use Count app_zapateller.so Block Telemarketers with Special Informa 0 1 modules loaded ~~~ No joy. Read this and recompiled Asterisk: http://ibot.rikers.org/%23asterisk/20090618.html.gz Got these messages: WARNING WARNING WARNING Your Asterisk modules directory, located at /usr/lib/asterisk/modules contains modules that were not installed by this version of Asterisk. Please ensure that these modules are compatible with this version before attempting to run Asterisk. app_fax.so app_saycountpl.so chan_ooh323.so format_mp3.so Read something else and found this in: /var/log/asterisk/messages [Jan 17 01:28:16] NOTICE[2479] loader.c: 145 modules will be loaded. [Jan 17 01:28:16] WARNING[2479] loader.c: Error loading module 'app_fax.so': libspandsp.so.2: cannot open shared object file: No such file or directory [Jan 17 01:28:17] WARNING[2479] res_smdi.c: No SMDI interfaces are available to listen on, not starting SMDI listener. [Jan 17 01:28:19] WARNING[2479] loader.c: Error loading module 'app_fax.so': libspandsp.so.2: cannot open shared object file: No such file or directory [Jan 17 01:28:19] WARNING[2479] loader.c: Module 'app_fax.so' could not be loaded. [Jan 17 01:28:19] ERROR[2479] chan_dahdi.c: Unable to load zapata.conf [Jan 17 01:28:20] NOTICE[2479] chan_ooh323.c: - --- *** IMPORTANT NOTE *** --- --- This module is currently unsupported. Use it at your own risk. --- - Does libspandsp.so.2 need to be copied to someplace else? # find / -name libspandsp.so.2* /usr/src/asterisk/spandsp/spandsp-0.0.6/src/.libs/libspandsp.so.2.0.0 /usr/src/asterisk/spandsp/spandsp-0.0.6/src/.libs/libspandsp.so.2 /usr/local/lib/libspandsp.so.2.0.0 /usr/local/lib/libspandsp.so.2 spandsp follows the normal default behaviour for application using the autotools, but this behaviour can be a nuisance for some people. By default it puts the library in /usr/local when you do a make install. On many machines this directory exists, but is not in the runtime library search path. It is, however, in the build search path, so programs build OK, but do not run. Build spandsp with ./configure --prefix=/usr or add /usr/local/lib to your library search list. Do I need a zapata.conf if I am using ztdummy? # find / -name zapata.conf # Any other ideas? Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to escape characters in Dialplan
2010/1/17 Dominik d0m1...@geekmail.de
Hello,
I'm using Asterisk 1.6.2.0 and I like to use escape characters with
SendText,
because I can just delete the message from my phone (Thomson Speedtouch
ST2030) display by sending a return-char (\n).
But \n is not escaped: I tried already:
exten = 222, n, SendText(\n)
exten = 222, n, SendText(\n)
exten = 222, n, SendText('\n')
exten = 222, n, SendText(`\n`)
So how can I use escape characters in dialplan?
This 187362 added this feature for 1.4.
Obviously, this fix was not ported to 1.6.X.
I would also be very pleased, if it could be done.
TIA,
Dominik
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[asterisk-users] Dial String command after audio background
exten = s,1,Answer()
exten = s,n,Background(astcc-please-enter-your)
exten = s,n,Background(zip-code)
exten = s,n,WaitExten(5)
exten = s,n,Read(NUMBER,,5)
exten = s,n,SayDigits(${NUMBER})
exten = 22042,n,Dial(SIP/sipvendor/111,120,A(ginger3))
exten = 22601,n,Dial(SIP/sipvendor/111,120,A(ginger3)) ;
x/ winchester
exten = 21230,n,Dial(SIP/sipvendor/111,120,A(ginger3)) ;
Mobile/Baltimore
I want to background to play please enter your zip code
Then say the digits pressed (5 digits)
Then map the five digits to an extension as shown to engage a Dial string
Examples above are not working.
Do I need an Answer() entry first for each zip code (extension)?
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Re: [asterisk-users] Dial String command after audio background
Am 17.01.2010 18:39, schrieb Thomas Perron:
exten = s,1,Answer()
exten = s,n,Background(astcc-please-enter-your)
exten = s,n,Background(zip-code)
exten = s,n,WaitExten(5)
exten = s,n,Read(NUMBER,,5)
exten = s,n,SayDigits(${NUMBER})
you might want to add a GoTo(${NUMBER},1)
as well as start your other extensions with
exten = 22042,1,Dial(SIP/sipvendor/111,120,A(ginger3))
then
exten = 22042,n,Dial(SIP/sipvendor/111,120,A(ginger3))
I want to background to play please enter your zip code
Then say the digits pressed (5 digits)
Then map the five digits to an extension as shown to engage a Dial string
Examples above are not working.
Because your're staying in the s extension - you need to switch to another
extension by using (for example, since there are other ways...)
goto.
Do I need an Answer() entry first for each zip code (extension)?
Nope - just give each a real id or label (instead of n) so you can address
them via goto.
Timm
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Re: [asterisk-users] Cross compiling Asterisk, Dahdi..
On Sun, 17 Jan 2010, Tzafrir Cohen wrote: I doubt it, but am willing to check - it's a vanilla kernel off kernel.org, compiled as per the instructions - the way I've been doing it for ever. I use make menuselect, then select the options I want. Module loading is enabled. Make the kernel (make bzImage) FWIW, 'make modules_prepare' should be good enough for building (or at least: test-building) modules. And takes less time. OK. If you build a custom kernel anyway, maybe the simplest approach would be to copy the dahdi files onto the kernel tree and build it there. I can do that? fx: typing, copying, fliddling... OK - Didn't know this - I have to edit drivers/Kconfig to have it included, but that looks intersting... If I could compile a module-less kernel that would use dahdi_dummy when no TDM400 card is fitted that would be nice... drivers/dahdi/Kconfig has: config DAHDI_VOICEBUS tristate VoiceBus(tm) Interface Library depends on PCI default DAHDI An example entry for a card that uses it: config DAHDI_WCTDM24XXP tristate Digium Wildcard VoiceBus analog card Support depends on DAHDI DAHDI_VOICEBUS default DAHDI Thus if the instructions from Kconfig are applied, you'll default not to build any PCI driver without any further effort. Sadly it is not applied when you build DAHDI as modules. OK. You'll still have to create the static device files. See http://svn.debian.org/viewsvn/pkg-voip/dahdi-linux/trunk/debian/make_static_nodes Done that, thanks. I don't use udev on my build system, nor my target systems so why is it bothering... But I feel there really ought to be a means to tell it that it's not building for the local system, so don't fiddle with local files... That one probably needs addressing as well, I guess. But for how many people... Of-course if I can built it as part of a kernel build then these don't get done as they're in the top-level Makefile... Might be nice to know how many others do this sort if thing? I guess the blackfin people do - ARM? I'm about to get a Nokia N900, and I know I can install gcc if it's not there already, but I somehow don't fancy compiling on the device itself, and I'm also looking at some other ARM boards too. I think being able to run Asterisk on my next mobile phone is sort of neat - anyone ported it to Andriod yet? Cheers, Gordon -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] help with picking out a digium card.
Hey all, We have been using a TDM400 card at work to provide our IVR. We we have upgraded our server and now require the same capability, but on a card that goes into a PCI Express. Any suggestions would be greatly appreciated. oh, and it has to work with the zaptel drivers for linux. thanks all. sk -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] help with picking out a digium card.
Some rack-mount servers I've encountered have an option to have the older-style PCI slots available in at least some slots. If you're really just using four FXS/FXO ports, it's unlikely you need very much horsepower, and you could use an older system for the foreseeable future. If you really need FXS/FXO, but want new non-PCI hardware, you might be better off considering an asterisk appliance that would convert FXS/FXO to SIP and let your new gear do the SIP, or just configure asterisk directly on that appliance. You would probably save power consumption versus a new server or even the old server currently in use. On Sun, Jan 17, 2010 at 3:25 PM, shawn bright sh...@skrite.net wrote: Hey all, We have been using a TDM400 card at work to provide our IVR. We we have upgraded our server and now require the same capability, but on a card that goes into a PCI Express. Any suggestions would be greatly appreciated. oh, and it has to work with the zaptel drivers for linux. thanks all. sk -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial String command after audio background
veilen danke timm
cheers
tom
On Sun, Jan 17, 2010 at 2:10 PM, Timm Korte
korte-ast-us...@easycrypt.de wrote:
Am 17.01.2010 18:39, schrieb Thomas Perron:
exten = s,1,Answer()
exten = s,n,Background(astcc-please-enter-your)
exten = s,n,Background(zip-code)
exten = s,n,WaitExten(5)
exten = s,n,Read(NUMBER,,5)
exten = s,n,SayDigits(${NUMBER})
you might want to add a GoTo(${NUMBER},1)
as well as start your other extensions with
exten = 22042,1,Dial(SIP/sipvendor/111,120,A(ginger3))
then
exten = 22042,n,Dial(SIP/sipvendor/111,120,A(ginger3))
I want to background to play please enter your zip code
Then say the digits pressed (5 digits)
Then map the five digits to an extension as shown to engage a Dial string
Examples above are not working.
Because your're staying in the s extension - you need to switch to another
extension by using (for example, since there are other ways...)
goto.
Do I need an Answer() entry first for each zip code (extension)?
Nope - just give each a real id or label (instead of n) so you can address
them via goto.
Timm
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Re: [asterisk-users] help with picking out a digium card.
Hey thanks. the IVR server we are using is also the web server and database server. My employer wants everything to run off of the same box, then duplicate that box as a backup. The server is already here and, alas, only the new PCI express slots. THnaks for taking your time on this for me. Still kinda new at this. sk On Sun, Jan 17, 2010 at 3:03 PM, David Backeberg dbackeb...@gmail.comwrote: Some rack-mount servers I've encountered have an option to have the older-style PCI slots available in at least some slots. If you're really just using four FXS/FXO ports, it's unlikely you need very much horsepower, and you could use an older system for the foreseeable future. If you really need FXS/FXO, but want new non-PCI hardware, you might be better off considering an asterisk appliance that would convert FXS/FXO to SIP and let your new gear do the SIP, or just configure asterisk directly on that appliance. You would probably save power consumption versus a new server or even the old server currently in use. On Sun, Jan 17, 2010 at 3:25 PM, shawn bright sh...@skrite.net wrote: Hey all, We have been using a TDM400 card at work to provide our IVR. We we have upgraded our server and now require the same capability, but on a card that goes into a PCI Express. Any suggestions would be greatly appreciated. oh, and it has to work with the zaptel drivers for linux. thanks all. sk -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] help with picking out a digium card.
You can also go with external FXO gateways, e.g. as AudioCodes Mediatrix, etc. This way you can avoid IRQ issue with standard cards. From: David Backeberg dbackeb...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sun, January 17, 2010 1:03:32 PM Subject: Re: [asterisk-users] help with picking out a digium card. Some rack-mount servers I've encountered have an option to have the older-style PCI slots available in at least some slots. If you're really just using four FXS/FXO ports, it's unlikely you need very much horsepower, and you could use an older system for the foreseeable future. If you really need FXS/FXO, but want new non-PCI hardware, you might be better off considering an asterisk appliance that would convert FXS/FXO to SIP and let your new gear do the SIP, or just configure asterisk directly on that appliance. You would probably save power consumption versus a new server or even the old server currently in use. On Sun, Jan 17, 2010 at 3:25 PM, shawn bright sh...@skrite.net wrote: Hey all, We have been using a TDM400 card at work to provide our IVR. We we have upgraded our server and now require the same capability, but on a card that goes into a PCI Express. Any suggestions would be greatly appreciated. oh, and it has to work with the zaptel drivers for linux. thanks all. sk -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] receive text
On Sun, Jan 17, 2010 at 7:34 AM, Thomas Perron thomas.per...@gmail.comwrote: Is there any code that I can cut/paste that will allow me to receive an SMS text on Asterisk? and, where can I capture the incoming text? See chan_mobile in the asterisk-addons... For certain cell phones there is a facility there to pass an SMS on thru the phone to Asterisk. You do it all via dialplan apps in chan_mobile.c, you'll see apps MobileSendSMS(device,dest,message), which allows you to send an SMS message via the dialplan, thru the bluetooth attached phone. To get an SMS, you have to have a cellphone bluetooth attached, and capable of passing sms messages. When it reports to Asterisk via the bluetooth connection, that an SMS message was recieved, Asterisk will try to run the sms extension, with the channel variables SMSSRC and SMSTXT channel variables set to the appropriate values. In the dialplans you can turn this into an email, an announcement, a text-to-speech (via festival or Cepstral or whatever), or whatever your needs or imagination can supply. I've asked around a while back, and the only phone capable of such sms capabilities was one running the Symbian os, iirc, and that means Nokia, I guess, and Erickson, and a few others... according to the Wikipedia, it's a pretty popular smart phone OS. Hmmm, wonder if the google Android can handle this? Anyway, another non-hardware solution might be to use an internet SMS gateway (for 10 cents/msg in low volume), to send/receive SMS also... murf -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Steve Murphy ParseTree Corp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] receive text
you can also use fix phone SMS sending and receiving if your provider allow sending and receiving SMS over the fixed phone line (.using FXO/FXS interface) the other way is to use kamalio (send and receive SMS through serial interface) this can be a good solution but I think this can not be used production since serial interface is a synchronized interface. So if you loose the synchronisation then you can not send and receive SMS currently what I use a multitech SMS server, which is easy to integrate but not free. If you want more information about this SMS server let me know. regards Mickael 2010/1/18 Steve Murphy m...@parsetree.com On Sun, Jan 17, 2010 at 7:34 AM, Thomas Perron thomas.per...@gmail.comwrote: Is there any code that I can cut/paste that will allow me to receive an SMS text on Asterisk? and, where can I capture the incoming text? See chan_mobile in the asterisk-addons... For certain cell phones there is a facility there to pass an SMS on thru the phone to Asterisk. You do it all via dialplan apps in chan_mobile.c, you'll see apps MobileSendSMS(device,dest,message), which allows you to send an SMS message via the dialplan, thru the bluetooth attached phone. To get an SMS, you have to have a cellphone bluetooth attached, and capable of passing sms messages. When it reports to Asterisk via the bluetooth connection, that an SMS message was recieved, Asterisk will try to run the sms extension, with the channel variables SMSSRC and SMSTXT channel variables set to the appropriate values. In the dialplans you can turn this into an email, an announcement, a text-to-speech (via festival or Cepstral or whatever), or whatever your needs or imagination can supply. I've asked around a while back, and the only phone capable of such sms capabilities was one running the Symbian os, iirc, and that means Nokia, I guess, and Erickson, and a few others... according to the Wikipedia, it's a pretty popular smart phone OS. Hmmm, wonder if the google Android can handle this? Anyway, another non-hardware solution might be to use an internet SMS gateway (for 10 cents/msg in low volume), to send/receive SMS also... murf -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Steve Murphy ParseTree Corp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] jitterbuffer and PLC
On 16/01/10 12:56 AM, nak...@02.246.ne.jp wrote: Hi, I have a question about jitterbuffer and PLC. Do you get the same results if you use: iax2 test losspct x Where x is the loss percent you'd like to test? -- Cheers, Matt Riddell Managing Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Will SIP connection stop automaticlly when detect no voice between the channel after a period of time?
hi, in my test, i noticed that sip connection will hangup automaticlly when no speaks between the channel. about half a minute. is this the asterisk inner mechanism or is my configuration error? Thanks! messages on the cli as follow: -- SIP/1003-001d is ringing -- SIP/1003-001d answered SIP/1004-001c -- Stopped music on hold on SIP/1004-001c [Jan 18 10:08:42] WARNING[17022]: app_queue.c:3268 try_calling: The device state of this queue member, Zhang Jianming, is still 'Not in Use' when it probably should not be! Please check UPGRADE.txt for correct configuration settings. -- Packet2Packet bridging SIP/1004-001c and SIP/1003-001d -- Executing Playback(SIP/1004-001c, vm-goodbye) [Jan 18 10:09:13] WARNING[17022]: file.c:764 ast_readaudio_callback: Failed to write frame -- SIP/1004-001c Playing 'vm-goodbye' (language 'en') [Jan 18 10:09:13] WARNING[17022]: app_playback.c:440 playback_exec: ast_streamfile failed on SIP/1004-001c for vm-goodbye == Spawn extension (95040654321, 1, 2) exited non-zero on 'SIP/1004-001c' [Jan 18 10:09:18] NOTICE[16700]: chan_sip.c:16209 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 1003 -- Best regards, Sucan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Faxing: Anyone have a compiled executable?
At 04:28 1/17/2010, Tzafrir Cohen wrote: On Sun, Jan 17, 2010 at 11:10:25AM +0100, IT-Connect wrote: Hallo there! I had my own experience get RxFax/TxFax successful running with spandsp. I only got spandsp-0.0.4 running, because on newer package, there aren't created some needed libraries (don't remember the right one this moment) *find /usr -iname \*spandsp\** shows me following output: /usr/lib/libspandsp.so /usr/lib/pkgconfig/spandsp.pc /usr/lib/libspandsp.so.0.0.2 /usr/lib/libspandsp.a /usr/lib/libspandsp.so.2 /usr/lib/libspandsp.so.0 /usr/lib/libspandsp.so.2.0.0 /usr/lib/libspandsp.la ... [Jan 17 01:28:16] NOTICE[2479] loader.c: 145 modules will be loaded. [Jan 17 01:28:16] WARNING[2479] loader.c: Error loading module 'app_fax.so': libspandsp.so.2: cannot open shared object file: No such file or directory What is the output of: ls -l /usr/lib/libspandsp.so* # ls -l /usr/lib/libspandsp.so* ls: /usr/lib/libspandsp.so*: No such file or directory # find / -name libspandsp* /usr/src/asterisk/spandsp/spandsp-0.0.6/src/libspandsp.2005.vcproj /usr/src/asterisk/spandsp/spandsp-0.0.6/src/libspandsp.la /usr/src/asterisk/spandsp/spandsp-0.0.6/src/libspandsp.2008.sln /usr/src/asterisk/spandsp/spandsp-0.0.6/src/libspandsp.2005.sln /usr/src/asterisk/spandsp/spandsp-0.0.6/src/libspandsp.2008.vcproj /usr/src/asterisk/spandsp/spandsp-0.0.6/src/libspandsp.dsp /usr/src/asterisk/spandsp/spandsp-0.0.6/src/.libs/libspandsp.la /usr/src/asterisk/spandsp/spandsp-0.0.6/src/.libs/libspandsp.lai /usr/src/asterisk/spandsp/spandsp-0.0.6/src/.libs/libspandsp.so.2.0.0 /usr/src/asterisk/spandsp/spandsp-0.0.6/src/.libs/libspandsp.so /usr/src/asterisk/spandsp/spandsp-0.0.6/src/.libs/libspandsp.a /usr/src/asterisk/spandsp/spandsp-0.0.6/src/.libs/libspandsp.so.2 /usr/src/asterisk/spandsp/spandsp-0.0.6/debian/libspandsp-doc.install /usr/src/asterisk/spandsp/spandsp-0.0.6/debian/libspandsp6.install /usr/src/asterisk/spandsp/spandsp-0.0.6/debian/libspandsp-dev.install /usr/local/lib/libspandsp.la /usr/local/lib/libspandsp.so.2.0.0 /usr/local/lib/libspandsp.so /usr/local/lib/libspandsp.a /usr/local/lib/libspandsp.so.2 Do I need a symbolic link? ldd /usr/lib/modules/app_fax.so # ldd /usr/lib/modules/app_fax.so ldd: /usr/lib/modules/app_fax.so: No such file or directory # find / -name app_fax.so /usr/src/asterisk/app_fax/app_fax.so /usr/lib/asterisk/modules/app_fax.so # ldd /usr/lib/asterisk/modules/app_fax.so linux-gate.so.1 = (0x0069f000) libspandsp.so.2 = not found libtiff.so.3 = /usr/lib/libtiff.so.3 (0x001eb000) libc.so.6 = /lib/libc.so.6 (0x003b3000) libjpeg.so.62 = /usr/lib/libjpeg.so.62 (0x0097a000) libz.so.1 = /usr/lib/libz.so.1 (0x00955000) libm.so.6 = /lib/libm.so.6 (0x0011) /lib/ld-linux.so.2 (0x006ec000) # find / -name libspandsp.so.2 /usr/src/asterisk/spandsp/spandsp-0.0.6/src/.libs/libspandsp.so.2 /usr/local/lib/libspandsp.so.2 What is the proper location for libspandsp.so.2? ldd /usr/lib/libspandsp.so.2 # ldd /usr/lib/libspandsp.so.2 ldd: /usr/lib/libspandsp.so.2: No such file or directory # ldd /usr/local/lib/libspandsp.so.2 linux-gate.so.1 = (0x00353000) libtiff.so.3 = /usr/lib/libtiff.so.3 (0x005f1000) libm.so.6 = /lib/libm.so.6 (0x004be000) libc.so.6 = /lib/libc.so.6 (0x0011) libjpeg.so.62 = /usr/lib/libjpeg.so.62 (0x00256000) libz.so.1 = /usr/lib/libz.so.1 (0x00bb5000) /lib/ld-linux.so.2 (0x006ec000) I would have hoped that when compiling the files would end up where the depending programs expect them to be. How to fix now, please? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Faxing: Anyone have a compiled executable?
At 09:12 1/17/2010, Steve Underwood wrote: spandsp follows the normal default behaviour for application using the autotools, but this behaviour can be a nuisance for some people. By default it puts the library in /usr/local when you do a make install. On many machines this directory exists, but is not in the runtime library search path. It is, however, in the build search path, so programs build OK, but do not run. Build spandsp with ./configure --prefix=/usr or add /usr/local/lib to your library search list. OK, /etc/ld.so.conf now looks something like: include ld.so.conf.d/*.conf /usr/local/lib Update: ldconfig -v /usr/local/lib: libspandsp.so.2 - libspandsp.so.2.0.0 /lib: libgcc_s.so.1 - libgcc_s-4.1.2-20080825.so.1 etc. Reboot. Joy: *CLI module show like fax Module Description Use Count app_fax.so FAX Application based on SpanDSP 0 1 modules loaded Thanks a bunch, guys! Now let's see if I can actually receive a fax... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to play the voicemail recorded?
Hi,all i want to hear the voicemail recorded, but when hear if you want to play message , press 3, after i press 3 i only hear that that's the time the file recorded. not the content. do you know how to hear content of voicemail fle? debug message: == Parsing '/var/spool/asterisk/voicemail/95040654321/3/INBOX/msg0001.txt': Found -- SIP/1003-0058 Playing 'vm-received' (language 'en') -- SIP/1003-0058 Playing 'digits/at' (language 'en') -- SIP/1003-0058 Playing 'digits/2' (language 'en') -- SIP/1003-0058 Playing 'digits/30' (language 'en') -- SIP/1003-0058 Playing 'digits/9' (language 'en') -- SIP/1003-0058 Playing 'digits/p-m' (language 'en') -- SIP/1003-0058 Playing '/var/spool/asterisk/voicemail/95040654321/3/INBOX/msg0001' (language 'en') -- Best regards, Sucan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] What's customer_id mean?
hi ,all I do'nt know exactly what customer_id mean? while if i have password i could visit the voicemail box. CREATE TABLE voicemail_users ( uniqueid int(11) NOT NULL auto_increment, customer_id int(11) NOT NULL default '0', context varchar(50) NOT NULL default '', mailbox int(5) NOT NULL default '0', password varchar(4) NOT NULL default '0', fullname varchar(50) NOT NULL default '', email varchar(50) NOT NULL default '', pager varchar(50) NOT NULL default '', stamp timestamp(14) NOT NULL, PRIMARY KEY (uniqueid), KEY mailbox_context (mailbox,context) ) TYPE=MyISAM; -- Best regards, Sucan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] iaxmodem / hylafax receive problem
Kingsley Tart wrote: Jan 14 12:44:49.39: [ 3403]: -- [9:AT+FRH=3\r] Jan 14 12:44:56.39: [ 3403]: -- [0:] Jan 14 12:44:56.39: [ 3403]: MODEM Empty line Jan 14 12:44:56.39: [ 3403]: MODEM TIMEOUT: waiting for v.21 carrier Jan 14 12:44:56.39: [ 3403]: -- data [1] Jan 14 12:44:56.39: [ 3403]: -- [2:OK] iaxmodem cannot hear any fax signaling in the call. Jan 14 12:44:56.39: [ 3403]: -- [9:AT+FRS=7\r] Jan 14 12:45:26.39: [ 3403]: MODEM TIMEOUT: reading line from modem Jan 14 12:45:26.39: [ 3403]: MODEM Timeout Jan 14 12:45:26.39: [ 3403]: Failure to receive silence (synchronization failure). Jan 14 12:45:26.39: [ 3403]: -- data [1] Jan 14 12:45:26.41: [ 3403]: -- [2:OK] However, there is *some* kind of audio on the call. It would seem that this test call is producing some kind of long-duration bad audio sounds which are not detectable by the modem as fax. Thanks, Lee. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to play the voicemail recorded?
Sorry. I can hear now. last time i have not record successfully. 2010/1/18 Zhang Shukun bit...@gmail.com: Hi,all i want to hear the voicemail recorded, but when hear if you want to play message , press 3, after i press 3 i only hear that that's the time the file recorded. not the content. do you know how to hear content of voicemail fle? debug message: == Parsing '/var/spool/asterisk/voicemail/95040654321/3/INBOX/msg0001.txt': Found -- SIP/1003-0058 Playing 'vm-received' (language 'en') -- SIP/1003-0058 Playing 'digits/at' (language 'en') -- SIP/1003-0058 Playing 'digits/2' (language 'en') -- SIP/1003-0058 Playing 'digits/30' (language 'en') -- SIP/1003-0058 Playing 'digits/9' (language 'en') -- SIP/1003-0058 Playing 'digits/p-m' (language 'en') -- SIP/1003-0058 Playing '/var/spool/asterisk/voicemail/95040654321/3/INBOX/msg0001' (language 'en') -- Best regards, Sucan -- Best regards, Sucan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
