Re: [asterisk-users] Running configure from subdirectory of source tree
That's not something that is likely to be supported. Any configure script in the tree will be run via the top-level build process, as needed. Is there some reason you think you need to run the other configure scripts yourself? On 03/05/2014 08:54 AM, Gianluca Merlo wrote: Hello everyone, I would like to seek your advice regarding a build (or rather configure) problem I am running into. For reference, tests are all relative to a build from a 1.8.26.0 tarball, on Debian Wheezy. I would like to understand if it is possible, and if any of you have tried, to build Asterisk from a subdirectory of the source tree, i.e., from a clean source tree # mkdir my-build-directory # cd my-build-directory # ../configure # make I lack a proper amount of knowledge on the matter, but I think that this should be legit with a common autotools build toolchain. Tests suggest that (at least in my case) this is not working with configure: error: cannot find install-sh, install.sh, or shtool in `pwd` ../`pwd` Looking in the configure process in detail, the failure seem to follow the checks (/bin/sh -x output) + for ac_dir in '`pwd`' '$srcdir/`pwd`' + test -f /home/gian/src/asterisk-1.8.26.0/my-build-directory/install-sh + test -f /home/gian/src/asterisk-1.8.26.0/my-build-directory/install.sh + test -f /home/gian/src/asterisk-1.8.26.0/my-build-directory/shtool + for ac_dir in '`pwd`' '$srcdir/`pwd`' + test -f ..//home/gian/src/asterisk-1.8.26.0/my-build-directory/install-sh + test -f ..//home/gian/src/asterisk-1.8.26.0/my-build-directory/install.sh + test -f ..//home/gian/src/asterisk-1.8.26.0/my-build-directory/shtool It looks to me that despite checking `pwd` leads to a correct behaviour, checking ../`pwd` is not correct. I seem to understand that this behaviour was introduced in configure.ac http://configure.ac at r259848, by adding AC_CONFIG_AUX_DIR(`pwd`) The log for the commit reports r259848 | qwell | 2010-04-28 22:32:14 +0200 (Wed, 28 Apr 2010) | 9 lines Merged revisions 259847 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 r259847 | qwell | 2010-04-28 15:30:21 -0500 (Wed, 28 Apr 2010) | 1 line Add AC_CONFIG_AUX_DIR to configure script, so systems without install can use install-sh from our source dir. Isn't the default behaviour for autoconf enough (http://www.gnu.org/software/automake/manual/html_node/Optional.html)? Can this be considered as a bug in Asterisk's the build system, preventing an otherwise working build scenario (i.e. configuring and building in a subdirectory of the source tree)? Thank you in advance for your help Gianluca -- _ -- 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] RHEL6 packages - SRTP support?
The packages currently do not support SRTP. On 06/03/2013 10:56 AM, Daniel Pocock wrote: I tried installing the Asterisk 11 RHEL6 packages from packages.asterisk.org I followed this guide: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages The SRTP support appears to be missing though. I notice libsrtp was not automatically installed as a dependency, and no srtp module exists under /usr/lib64/asterisk/modules Is it still necessary to do a source build every time SRTP is needed? Or is the srtp module distributed in some other rpm? -- _ -- 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] RHEL6 packages - SRTP support?
On 06/03/2013 12:03 PM, Daniel Pocock wrote: I tried building manually from the source RPM Before running rpmbuild, I installed libsrtp-devel and I notice that res_srtp.so is generated during the build However, the rpmbuild fails for other reasons (see the other email I sent to the list about mISDNutils-devel and other spec file errors) Can you confirm the exact procedure you recommend for rpmbuild on a CentOS6 system rpmbuild --rebuild --without tds --without misdn somepackage.src.rpm -- _ -- 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 Log rotate not working
On 05/21/2013 10:19 AM, Ahmed Munir wrote:
Hi,
Last year, I installed Asterisk 10.4.2 and enabled logrotate on daily
basis which was working perfect. Now in couple of months back, the
logrotate feature is not working at all but simply appending the logs
in 'messages' file. Listing down down the configuration for logrotate
below;
/var/log/asterisk/messages {
missingok
rotate 5
daily
postrotate
/usr/sbin/asterisk -rx 'logger reload' /dev/null 2 /dev/null
endscript
}
As asterisk is running by user: root so no need set asterisk
permissions 'create 0640 asterisk asterisk' in above configuration.
Please advise so I can resolve this issue.
I believe you want to execute logger rotate, rather than logger reload.
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Re: [asterisk-users] What is bootstrap.sh for ? Possible bug in 11.3.0 ?
On 05/07/2013 05:13 AM, Olivier wrote: 2013/5/7 Matthew Jordan mjor...@digium.com mailto:mjor...@digium.com 2. It appears as if you're running a modified version of Asterisk, in which case all bets are off. This works fine on the Linux build agents, which is what we use to build the tarballs on downloads.asterisk.org http://downloads.asterisk.org. So, no, I don't think there's a bug in the shell script. I can reproduce this behaviour at will on a fresh new untouched asterisk 11.3.0 install on a debian squeeze (see ASTERISK-21760 https://issues.asterisk.org/jira/browse/ASTERISK-21760) Would you say that for a given asterisk version, included configure script should match the one generated by bootstrap.sh ? The bootstrap.sh script is run by developers after making changes that require regenerating the configure script. It isn't needed on an unpatched installation. Having said that - the generated configure script will very rarely match the one provided in the source, since it will change (sometimes significantly) with differing versions of autoconf, et al. -- _ -- 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] Moving User Agent To Remote Location
On 01/03/2013 02:23 PM, Markus Weiler wrote: Am 03.01.2013 21:21, schrieb Nick Khamis: Oh that's so smart!!! So, if I did not misunderstand you, for this one call, have: rtpstart=10004 rtpend=1008 do you mean 1_000_8 ? Markus I think he means 10007. -- _ -- 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] Call Termination Provider Madness
On 10/03/2012 10:46 AM, Eric Wieling wrote: A port is not a door if there is nothing listening on the port. Open ports are not a security issue. Stuff running on open ports are. Do you have some external software listening on those ports when there isn't an active call? Asterisk isn't listening on them. -- _ -- 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] How do you convert your prompts to an asterisk-friendly format?
On 08/28/2012 10:04 AM, Andrew Latham wrote: On Tue, Aug 28, 2012 at 11:00 AM, Johan Wilfer li...@jttech.se wrote: 2012-08-28 16:44, Andrew Latham skrev: Try this to test with http://www.digium.com/en/products/ivr/audio-converter.php and compare your output first... Interesting. Didn't know about this. It's good for testing, but I would like to automate it. Is the source-code open or available? Yep, check out repotools for that http://svn.asterisk.org/svn/repotools/sound_tools/scripts/ I don't know whether those scripts are what is actually used on the digium.com website, but they are what we use to create the various Asterisk sounds packages on http://downloads.asterisk.org/. -- _ -- 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] How do you convert your prompts to an asterisk-friendly format?
On 08/28/2012 10:32 AM, Danny Nicholas wrote: Does the .c program compile stand-alone or as an add-on? g++ check_sounds.c check_sounds.c: In function âint main(int, char**)â: check_sounds.c:152: error: invalid conversion from âvoid*â to âdirent**â check_sounds.c:154: error: invalid conversion from âvoid*â to âdirent*â There may be issues building it with g++. I just added a basic Makefile, so you should be able to `svn update` and `make`. -- _ -- 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] Problems installing DPMA
On 06/12/2012 02:56 PM, Danny Dias wrote: Hi, I'm just trying to install the DPMA on my Asterisk. I already made the upgrade from Asterisk 1.8.5 to Asterisk 1.8.11-cert2. This is what i did: /mv /usr/lib/asterisk/modules /usr/lib/asterisk/modules-185 / *compiling Asterisk-Cert2 1.8.11* /./configure make make install make config / Afther that i register the DPMA license, and finally copied the *res_digium_phone.so* to //usr/lib/asterisk/modules / When i try to load the module on asterisk console this is what i get /*CLI module load res_digium_phone.so Unable to load module res_digium_phone.so Command 'module load res_digium_phone.so' failed. / With /tail -f /var/log/asterisk/message / /[Jun 11 18:53:26] WARNING[2554] loader.c: Error loading module 'res_digium_phone.so': libavahi-client.so.3: cannot open shared object file: No such file or directory [Jun 11 18:53:26] WARNING[2554] loader.c: Module 'res_digium_phone.so' could not be loaded. / Hope you can help Questions like this should usually be directed to Digium support. Your issue can be fixed by installing the package containing libavahi-client. On CentOS: yum install avahi on Debian/Ubuntu: apt-get install libavahi-client3 -- _ -- 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] Cannot get Digium Phones back into service after changing sip device name.
On 06/05/2012 10:23 AM, Chet W. Stevens wrote: During testing with the Digium phones I have run into a problem where I try to make a change to the sip device name. I make the device name change in sip.conf then make the matching change to the lines in res_digium_phone.conf. I then do 'sip reload' and 'module reload res_digium_phone.so'. I then end up with phones that I cannot bring into service no matter what I have tried. They act normally as far as seeing the configuration server, allowing me to enter the key, select the user, and then I receive the Error fetching config from proxy. message. I occasionally receive the following error in the CLI: NOTICE[5941]: phone_users.c:1626 token_set_last_info: Phone at '' reconfigureed. Another phone located at 'sip:10.72.65.114:5060' took over the config. I have tried Reset to Factory Defaults on the phones, I have tried clearing the keys from the Asterisk database, I have tried 'sip unregister', I have tried restarting Asterisk and rebooting. I cannot seem to get these test phones back into service. I am only in testing now where the phones are literally at arms reach and I am really nervous what can happen when we go into production. I really hope that I did something wrong in the process and that the phones are not really this fragile. My test system is currently running these versions: Asterisk 1.8.11-cert2 x86_32 DPMA Module: 1.8.11_1.0.1-x86_32 Digium Phone Firmware: 1_0_5_46476 Your help on this is really appreciated. Thank you. The first step would be to contact Digium technical support. They would be happy to assist you with any issues you're having. -- _ -- 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] Deleting OLD Voicemails
On 05/22/2012 04:54 PM, Danny Dias wrote: There are 4 files for each voicemail: msg.gsm msg.txt msg.wav msg.WAV That is perfectly normal. The .txt file is metadata that contains things like caller ID and duration. Asterisk will also save voicemails into every format you have specified in voicemail.conf. -- _ -- 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] Flashphoner
On 04/27/2012 01:39 PM, Don Kelly wrote: What flavor are flashphoner minties? --Don Dailing flavored. What else? -- _ -- 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] Compiling asterisk with mysql support
On 03/06/2012 12:31 PM, Ron Bergin wrote: Mathew, Each of those odbc modules are unavailable i.e., marked with XXX I even deleted the asterisk build directory and started over, but had the same results. What prereqs do I need besides these: mysql.i386 5.0.95-1.el5_7.1installed mysql-connector-odbc.i386 3.51.26r1127-1.el5 installed mysql-devel.i3865.0.95-1.el5_7.1installed mysql-server.i386 5.0.95-1.el5_7.1installed unixODBC.i386 2.2.11-7.1 installed unixODBC-devel.i386 2.2.11-7.1 installed libtool-ltdl-devel should be a dependency for unixODBC-devel in CentOS, but it is not. You'll need to install that and re-run ./configure. -- _ -- 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] Group write permissions /etc/asterisk/.
On 03/06/2012 03:44 PM, Karl Fife wrote: It's not a question of whether the default directory permissions are appropriate. I agree with those. What we're talking about here is what happens during updates to an existing directory. I can't see any rationale for changing the group permissions. If the group permissions differ from the installation defaults, it is because the sysadmin needed them to be different in order to implement one or more methods of extensibility / interoperability that make Asterisk so powerful. Absolutely, it would make sense for the installer to check to be sure it has SUFFICIENT permissions to operate properly, but it is a huge leap of faith to assume that it's appropriate to simply delete certain group permissions. Users only in the owner's group if they belong there, no?? The upshot is that ever since upgrading to 1.8 we have to re-re-re-reset the group directory permissions to make things work, and that just seems insane to me if that is a design choice, not a regression. -Karl It should only set them if the directory does not exist. If it's changing them, something is very seriously broken. -- _ -- 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] Group write permissions /etc/asterisk/.
On 03/06/2012 04:24 PM, Patrick Lists wrote: On 06-03-12 23:07, Karl Fife wrote: Yep. That's what's happening. I'll file a bug. AFAICT it's not a bug but the way RPM works. Regards, Patrick He didn't suggest that he was talking about RPMs. If that's the case, then I take back everything I said. -- _ -- 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] RPM availability -- [asterisk-announce] Asterisk 1.8.9.3 Now Available
On 03/05/2012 06:34 AM, Eric Germann wrote: Does anyone have an idea on when 1.8.9.3 might show up in the RPM repositories? Thanks! EKG They should be available now. -- _ -- 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] RPM availability -- [asterisk-announce] Asterisk 1.8.9.3 Now Available
On 03/05/2012 01:49 PM, Eric Germann wrote: Will a 1.8.10.0 build be imminent or should we go ahead and push this in to production with testing? Thanks! EKG ~20 minutes -- _ -- 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] RPM availability -- [asterisk-announce] Asterisk 1.8.9.3 Now Available
On 03/05/2012 06:00 PM, Lefteris Zafiris wrote: Some packages seem to lag behind, eg asterisk18-addons-mysql is compiled against 1.8.7: asterisk18-addons-core-1.8.7.0-2_centos5 asterisk18-addons-mysql-1.8.7.0-2_centos5 Is this a problem with the repo? Are these packages obsolete/unmaintained or have been replaced by others? They've been replaced. The latest packages are in new repositories and are now more appropriately named. See http://packages.asterisk.org/centos/5/asterisk-1.8/ as an example. Also, as of Asterisk 1.8, the -addons RPMs are now built from the same SRPM as the rest of the asterisk RPMs. -- _ -- 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] Group write permissions /etc/asterisk/.
On 03/05/2012 06:22 PM, Karl Fife wrote: I notice that the installation of Asterisk 1.8.8 thru 1.8.10 (probably earlier versions too) remove the group write permissions from /etc/asterisk/. which is different than 1.4. And 1.6. Is this expected behavior? If so, what's the rationale? If not, I'll submit a bug report if someone hasn't beaten me to it. -K The difference comes from using `install` rather than `mkdir`. mkdir defaults to a+rwx (777) - umask (likely 002 on your system), whereas install defaults to the much more sane u+rwx,g+rx,o+rx (755). I don't know if I would call it a bug since the switch to install was intentional, but I wouldn't say it's necessarily expected either. I don't really have a strong opinion either way though. If anything, I might be inclined to argue that 750 (or 770) would be more appropriate. -- _ -- 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 Asterisk 10 available in Digium Repository? I doesn't show up
On 02/26/2012 06:22 PM, Patrick Lists wrote: On 25-02-12 19:47, Jason Parker wrote: yum and rpm do not support downgrades. Incorrect. There is yum downgrade. See man yum. yum downgrade is extremely broken. It fails, often, potentially leaving a system in an unrecoverable state. That is not to mention how poorly conceived the concept is. Consider what would happen if a package upgraded some resource to a non-backwards-compatible version. It is completely unsupported on the Digium repositories. Please don't try it - I will not help fix it. -- _ -- 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 Asterisk 10 available in Digium Repository? I doesn't show up
yum and rpm do not support downgrades. You can try using `yum shell` to uninstall one version and install another version in one transaction, but you'll have to go it alone. On 02/25/2012 11:49 AM, Ast Coder wrote: Thanks Jason. One more question: Is there anyway to go back on an Asterisk version when using the repository? For example, Asterisk 1.8.9.2 is available now. But I want to use 1.8.9.1. Can I downgrade somehow? I want to test NAT bug issue. Thanks On Thu, Feb 23, 2012 at 11:15 AM, Jason Parker jpar...@digium.com mailto:jpar...@digium.com wrote: On 02/23/2012 10:09 AM, Ast Coder wrote: Hi, I have followed instruction on https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages#AsteriskPackages-Prerequisites to add Digium Asterisk repositories but doing a, yum search asterisk only shows me Asterisk 1.4, 1.6, and 1.8. There is no Asterisk 10 and yum install asterisk10 fails. Am I missing something? or Asterisk 10 is just no available in binary? Thanks, There are now repositories for each major version of Asterisk, which have to be explicitly enabled to use them. `yum update` to get to the latest of everything, then do `yum update --enablerepo=asterisk-10`. Asterisk 10 will be installed, and that repository will be enabled permanently. I'll add that information to the wiki shortly. -- _ -- 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 Asterisk 10 available in Digium Repository? I doesn't show up
On 02/23/2012 10:09 AM, Ast Coder wrote: Hi, I have followed instruction on https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages#AsteriskPackages-Prerequisites to add Digium Asterisk repositories but doing a, yum search asterisk only shows me Asterisk 1.4, 1.6, and 1.8. There is no Asterisk 10 and yum install asterisk10 fails. Am I missing something? or Asterisk 10 is just no available in binary? Thanks, There are now repositories for each major version of Asterisk, which have to be explicitly enabled to use them. `yum update` to get to the latest of everything, then do `yum update --enablerepo=asterisk-10`. Asterisk 10 will be installed, and that repository will be enabled permanently. I'll add that information to the wiki shortly. -- _ -- 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] Praking lot issues.
On 02/21/2012 02:55 PM, Bryant Zimmerman wrote: Ok I now have the basics for dynamic parking working but for some reason when a caller calls in and is parked with a transfer the return call dials the sip peer of the caller and not hte peer of the last party that parked the call. Anyone have any ideas on this? Also when a call is transfered to a parking space. the caller hears the space number. How can I stop that as well? Thanks Bryant See https://issues.asterisk.org/jira/browse/ASTERISK-19322 -- _ -- 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] Streaming musiconhold via mpg123
On 02/21/2012 05:34 PM, Stephen Brown wrote: application=/usr/bin/mpg123 -q -s -f 8192 --mono -r 8000 /var/lib/asterisk/sounds/music/Rolling In The Deep.mp3 Probably unrelated to your issue, but you're going to want to quote that filename. -- _ -- 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 1.8.9.0 Now Available
On 01/30/2012 11:06 AM, Eric Germann wrote: We mirror off http://packages.asterisk.org to a staging server, then update from there. When will this show up on packages.asterisk.org? Thanks! EKG The RPMs should be there in a few minutes. -- _ -- 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] 1.6 and 1.8
On 12/28/2011 03:10 PM, Danny Nicholas wrote: Can somebody point me to an explanation from Kevin or Tzafir or someone else up the food chain explaining the differences/benefits of 1.6/1.8 vs 1.4/10.0? Every branch (1.0, 1.2, 1.4, 1.6.0, 1.6.1, 1.6.2, 1.8, 10) of Asterisk contains new features that previous branches did not have. Many of these changes are documented in http://svnview.digium.com/svn/asterisk/branches/10/CHANGES Each branch of Asterisk has a lifecycle, which is documented at https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions. As you can see, 1.8 and 10 are the currently supported branches. 1.4 and 1.6.2 are in security maintenance mode, which means that the only issues that will be fixed are security issues. They will both be EOL in April 2012, and will no longer receive any updates. Short version: If you aren't already using Asterisk 1.8 or higher, you really should be - and soon. -- _ -- 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] What version to upgrade to...?
On 12/12/2011 09:26 AM, Danny Nicholas wrote: I'm wondering if the bind 161 as root statement is a mis-statement or if not, maybe somebody like Tzafir can explain why since none of the other Asterisk binds require root access (this message is still in 10.0-rc3). This is accurate. Non-root users cannot bind ports =1024. There are ways around it, however. See setcap/CAP_NET_BIND_SERVICE at http://www.kernel.org/doc/man-pages/online/pages/man7/capabilities.7.html I haven't looked at the Asterisk code, but there may be changes necessary to disable that check, if this is enabled. -- _ -- 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] Standard UIDs, especially for asterisk?
On 11/15/2011 09:58 AM, Tony Mountifield wrote: I see on my CentOS systems that certain users for particular subsystems have standardised UIDs and GIDs. For example mysql=27, ntp=38, sshd=74. My two questions are: 1. Is there a list of these standard assignments somewhere? Googling did not turn up anything for me. 2. Are there standard values of UID and GID reserved for the asterisk user, if used for running Asterisk as non-root.? Cheers Tony There are no standard UID/GIDs for things. They are just system users that have no login shell. They are given lower IDs than normal user accounts (on redhat systems, see -r option to useradd) so that they can be easily distinguished. -- _ -- 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] Standard UIDs, especially for asterisk?
On 11/15/2011 10:42 AM, Tony Mountifield wrote: Yes, I was hoping to use such a system user and group for asterisk, which would not conflict with any other system package I might install in the future, by virtue of being reserved for asterisk. There shouldn't be any conflict either way. (Properly written) packages don't specify a UID to use - they just get created sequentially, so the next available ID is used. -- _ -- 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] Astricon: GPG Key signing event
On 10/20/2011 05:16 PM, Paul Belanger wrote: Greetings, If you are planning on attending Astricon, please take the time to attend the GPG key signing event. More information can be found on the wiki page[1]. [1] https://wiki.asterisk.org/wiki/display/~pabelanger/Astricon+2011+Key+signing+event I fail at wikis and don't know how to add comments (perhaps a permissions thing, with it being in your private space, Paul?). I just wanted to note that if you already have a keypair, you will need to have access to a copy of your *private* key in order to be able sign somebody else's key at the event. -- _ -- 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] How to use menuselect.makeopts?
On 10/18/2011 09:52 PM, Luke Hamburg wrote: I think this might actually be a bug. https://issues.asterisk.org/jira/browse/ASTERISK-18137 It is indeed a bug, but it's not the bug you referenced. This issue only exists in 1.8.8.0-rc1. It has been fixed for 1.8.8.0-rc2 which will be released this morning. -- _ -- 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 Centos RPM packages question
On 10/17/2011 02:22 PM, Ioan Indreias wrote: Hello, Trying to upgrade (from asterisk18-1.8.6.0-1) to the latest RPM version from Asterisk repo I found that asterisknow-version is needed by package asterisk18-core-1.8.7.0-2 How could this be explained? Best regards, Ioan The asterisknow-version package contains the repository files (see /etc/yum.repos.d/) for the repositories on packages.asterisk.org and packages.digium.com. Installing this should have been in the setup instructions. The repository layout has changed significantly, and people that didn't install this package would have been stuck on old versions of Asterisk. We opted to add this package as a dependency (as it should have always been one) to resolve that issue. Short version: This is intentional. -- _ -- 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] Core show translation 4000ms
On 09/30/2011 09:53 AM, Tony Mountifield wrote: In article 4e85d19f.4090...@digium.com, Kevin P. Fleming kpflem...@digium.com wrote: This is why the output was changed to microseconds from milliseconds; in the older version, the lowest number that should be shown was 1 millisecond, even if the actual amount of time consumed was 10 microseconds (or less). The 1 numbers in the output from the older could easily have been 0.02, which would be closer to the output from the new version. Maybe, but that still doesn't explain why there is a factor of 2000 between some conversions and others. And 4001, 4002 and 4003 are remarkably like a big round number plus a tiny offset! I would agree with the OP that the values shown look suspicious and would bear some investigating... I believe the way it gets calculated was also changed a bit. You'll commonly see numbers that are near multiples of 1000. If I'm not mistaken these are the duration of a context switch (or several context switches), which means that with this output, you can guess that his kernel is probably compiled with CONFIG_HZ_250. -- _ -- 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] AsteriskNow install addons despite license conflict with FFA and G.729
On 07/19/2011 01:02 PM, Michael wrote: On Tue, Jul 19, 2011 at 3:34 PM, Kevin P. Fleming kpflem...@digium.com mailto:kpflem...@digium.com wrote: You don't need to install asterisk-addons to be able to store CDRs; you need them to be able to store CDRs in MySQL specifically. If you choose another database that doesn't have licensing restrictions that interfere with usage of non-GPL modules, then you'll be fine. Asterisk 1.6.2.19 includes CDR modules for PostgreSQL and FreeTDS (Microsoft SQL Server), and also generic ODBC support which can be used to connect to MySQL if you wish. Doesn't FreePBX CDR page/engine require MySQL CDRs? Yes, but you don't have to use cdr_mysql to insert into a MySQL database. The cdr_odbc module works just fine for that. -- _ -- 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] Jabber / facebook chat?
On 05/17/2011 07:18 AM, Stefan Gofferje wrote: On 04/17/2011 02:13 AM, Stefan Gofferje wrote: has anyone managed to establish an XMPP connection to the facebook Jabber servers? I'd like to send messages on missed calls vie FB. I finally figured it out. For facebook chat to work you have to use usetls = no usesasl = yes Chat.facebook.com offers TLS but it seems, it's incompatible to res_jabber. To clarify, does that mean that you were able to successfully use facebook chat with sasl? -- _ -- 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] question on digium repo
On 05/16/2011 08:36 AM, Jerry Geis wrote: I an running centos 5. I added this to the digium.repo file in /etc/yum.repos.d directory. [digium-current] name=CentOS-$releasever - Digium - Current baseurl=http://packages.digium.com/centos/$releasever/current/$basearch/ enabled=1 gpgcheck=0 #gpgkey=http://packages.digium.com/RPM-GPG-KEY-Digium Then I did yum install asterisk14 addons | 951 B 00:00 base | 2.1 kB 00:00 base/primary_db | 2.2 MB 00:03 digium-current | 1.1 kB 00:00 digium-current/primary | 33 kB 00:00 digium-current 260/260 extras | 2.1 kB 00:00 extras/primary_db | 244 kB 00:00 updates | 1.9 kB 00:00 updates/primary_db | 544 kB 00:01 Setting up Install Process No package asterisk14 available. What did I miss? jerry You missed the Asterisk repo. Replace all instances of digium.com with asterisk.org (and then Digium with Asterisk). packages.digium.com is Digium modules, such as FaxForAsterisk, whereas packages.asterisk.org is Asterisk, DAHDI, libpri, etc. -- _ -- 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] lead time for RPM's?
On 05/12/2011 02:46 PM, Jason Parker wrote: I'll make it a point to respond to this email when the new builds are available. These builds are now available. -- _ -- 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] lead time for RPM's?
On 05/12/2011 02:40 PM, Cassius Smith wrote: Hi all Usually I build asterisk from source, but recently have been doing a couple of test installations with packages from the Digium repository. About how long does it take to get from new release announcement into the Digium RPM repository? Specifically 1.8.4 CentOS hasn't made it to the rpm repository yet. Cassius In most cases, we'll have RPMs built and available before the release notifications go out. However, we are currently in the process of rebuilding our build servers, so it has been delayed a few days. I expect that builds will be available in the next day or so. I'll make it a point to respond to this email when the new builds are available. -- _ -- 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] Cannot install dahdi-linux on (old) PAE kernel.
On 05/06/2011 01:30 PM, Bob Beers wrote:
Not sure if this will work, but I'd try adding, before line 86:
#Workaround for PAE
%if %{paevar} == PAE
Provides: kmod-dahdi-linux
%endif
Can't actually test it myself, sorry.
- Bob
You'd probably want to modify the kmodtool that comes with it, to just always
provide kmod-dahdi-linux.
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Re: [asterisk-users] Some errors
On 03/15/2011 12:34 PM, Fellipe Paes wrote: why I can't use _. in my dialplan? Because it matches everything. In this case, it's matching the 'h' exten. So when the call gets hung up, it goes to _. and does what you're seeing. -- _ -- 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] REFER and dialplan broken (as documented inchan_sip.c on line 11951)
On 02/23/2011 12:43 PM, vip killa wrote: I recognize all the options given yet as I explained before they are not viable. I do not have the resources to pay someone, I do not have the expertise to fix this issue because according to an asterisk developer any fix in that area would be deeply architectural in nature... what other options are there? Option 3 was wait for someone else with the skills and/or money necessary to fix it. Demanding that somebody fix an issue will not work in any community, open source or otherwise. You'll only be labeled a nuisance and ignored. -- _ -- 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] asterisk18 rpm issues
On 02/02/2011 02:14 PM, Frank Liu wrote: Hi there, Per the instruction from http://www.asterisk.org/downloads/yum , I setup the yum repository on my Centos 5 x86_64 machine and did a yum install asterisk18 asterisk18-configs then I startup the asterisk (with no changes to config) just to see if it runs, but see below errors in the /var/log/asterisk/messages: [Jan 31 11:41:40] WARNING[2899] loader.c: Error loading module 'res_pktccops': /usr/lib/asterisk/modules/res_pktccops.so: cannot open shared object file: No such file or directory [Jan 31 11:41:40] WARNING[2899] loader.c: Error loading module 'chan_mgcp.so': /usr/lib/asterisk/modules/chan_mgcp.so: undefined symbol: ast_pktccops_gate_alloc I checked the system and can't find the file /usr/lib/asterisk/modules/res_pktccops.so at all. I double checked the rpm file downloaded by yum and res_pktccops.so is not in any rpms. Asterisk should still load fine with this warning. chan_mgcp wouldn't work, but that isn't used very often. I will take a look at it. -- _ -- 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] Top Posting
On 01/19/2011 12:18 AM, randulo wrote:
Although there's no requisite mention of ${Horrible_Dictator}, can't
we pretend there was, call a Godwin and kill this subject?
That would fall under Quirk's Exception: Intentionally invoking Godwin's Law to
attempt to kill a thread is rarely successful. :)
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Re: [asterisk-users] Asterisk 1.8.2 and digium yum repositories
On 01/19/2011 04:41 AM, Ishfaq Malik wrote: Hi Does anyone have any idea how long it will take for the new release of asterisk 1.8 to make it to the digium yum repositories? Thanks Ish They've been there since yesterday afternoon. It's possible that you hit the repository before the packages were there, causing the refresh timer to be extended (the default is probably 24 hours - but I'd have to check). If they still aren't showing up for you, you can run `yum clean metadata; yum update` -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unexpected dialplan match
On 12/20/2010 11:35 AM, Daniel Tryba wrote:
I was wondering why *...@default should match '_*[0-9a-zA-Z].*0.'
in 1.6.13. Who is making the parse error, * or me?
CLI dialplan show *...@default
'_*[0-9a-zA-Z].*0.' =
1. NoOp(${EXTEN}) [pbx_config]
2. Set(accountcode=${CUT(EXTEN,*,2)}) [pbx_config]
3. Set(extension=${CUT(EXTEN,*,3)}) [pbx_config]
4. Set(CDR(accountcode)=${accountcode}) [pbx_config]
7. ResetCDR() [pbx_config]
8. ...
'.' stops further matching. Your extension ends up being (effectively)
shortened to _*[0-9a-zA-Z].
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Re: [asterisk-users] DAHDI on VMWARE
On 12/02/2010 02:03 PM, Danny Nicholas wrote: Hi gang, We are moving our computers from a cluster of physical machines to a VMWARE server with virtual machines. We investigated and are looking to replace our TDM400P/TDM410P with AEX410P cards. Can we run asterisk with the DAHDI drivers from one of the Virtual machines or is DAHDI going to have to be a native process on the “REAL” machine? Thanks Danny Nicholas VMware has no type of PCI-passthrough feature that I'm aware of. There are virtualization environments that do, but the added overhead is going to make things extremely unreliable. -- _ -- 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] IAX authentication oddity - Known issue? Fixed?
On 07/28/2010 11:32 AM, Tilghman Lesher wrote: They permit what packets will even reach user2 It should also be pointed out that the config option is permit, and not allow. -- _ -- 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] Voice prompts
On 07/19/2010 01:23 PM, Danny Nicholas wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of mattias Sent: Monday, July 19, 2010 1:16 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Voice prompts Have now installed a swedish prompt set In /var/lib/asterisk/sounds/se I run elastix And set Language=se in /etc/asterisk/sip.conf But not work \-\- \-\- Show your CLI output so we see that you are getting correct playback Here's a QD snippet to let you do a verification Exten = 1234,1,answer Exten = 1234,n,Set(CHANNEL(language)=se) Exten = 1234,n,playback(tt-monkeys) Exten = 1234,n,playback(vm-goodbye) Exten = 1234,n,hangup Danny, When bottom posting, something you should keep in mind is that a -- on a line by itself causes most email clients to consider anything below it a signature (a sane client will lighten the text, and it won't appear when you hit reply). It would make things much nicer if you were to also remove that part of the signature on your replies. -- _ -- 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] centos 5 rpm pacakges (add asterisk16-xmpp module)
On 07/15/2010 08:16 AM, Vasiliy G Tolstov wrote: Hello. Who can add asterisk16-xmpp module to packages.asterisk.org or build asterisk with support xmpp and update packages? Thank You. This is something we've been considering for a while. It should make its way onto the list shortly. -- _ -- 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 problems with kernel 2.6.32
On 05/26/2010 08:00 PM, cov...@ccs.covici.com wrote: From another thread, I blacklisted netjet and now things are working. But I wonder what is going on here and where did netjet come from -- it doesn't look like an dahdi module to me. It comes from mISDN. It is a very badly misbehaving module. IIRC, it wildcards a large portion of tigerjet PCI IDs. -- _ -- 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] include sip configuration from another file in sip.conf
On 05/12/2010 01:03 PM, Robert Wagner wrote: Hi, when i include a sip configuration from another file in my sip.conf using #include /etc/asterisk/sip-sipgate.conf everything seems to be working. The peer is listed when i execute sip show peers and Status is OK. But the peer is not listed using sip show registry. I need to place the register = ... in the sip.conf to make it work. Is this working as expected or is it a bug? Working as expected. When you #include a file, the #include line is replaced with the contents of the file. Meaning your register line is likely being placed inside the previous context. -- _ -- 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] mISDN installation via yum
Michael Nausch wrote: HI, I tried to install asterisk and mISDN via http://www.asterisk.org/downloads/yum My machine is running with kernel-2.6.18-164.15.1.el5.i686 Packages for that kernel version were missing. That was an oversight and has been corrected. A `yum update` should be enough to solve this for you. -- _ -- 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] Change in menuselect handling of sound files (in 1.6.1.X)
Olivier wrote:
Hi,
Between 1.6.1.9 and 1.6.1.18, handling of menuselect has changed in such
a way that I cannot script non-english sound files downloading anymore.
The following used to work (unattended) with 1.6.1.9 (for instance):
cd /usr/src/asterisk-${ASTERISK_VERSION}
./configure
make menuselect.makeopts
echo MENUSELECT_CORE_SOUNDS=CORE-SOUNDS-EN-GSM CORE-SOUNDS-FR-GSM
menuselect.makeopts.defaults
make USER_MAKEOPTS=menuselect.makeopts.defaults menuselect.makeopts
make
make install
Now, with 1.6.1.18, CORE-SOUNDS-FR-GSM is not downloaded anymore.
I quickly compared both Makefile contents but it's too complex for me.
How should I change my script to download sounds files ?
Regards
Remove this line:
make menuselect.makeopts
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Re: [asterisk-users] asterisk 1.6.1/1.6.2 binary packages
Pablo Ruiz wrote: Hello, Does anyone know when we will see asterisk 1.6.1 (and/or 1.6.2) binary packages at packages.asterisk.org http://packages.asterisk.org? Greets. Packages for 1.6.2 will be available Real Soon Now. It's near the top of my short list. They exist, and are sitting in a(n internal) testing repository. Mostly, I just need to make sure upgrades go smoothly. -- _ -- 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 1.6.1/1.6.2 binary packages
bruce bruce wrote: Thanks for the update Jason, How do the upgrades work if v1.6.0 is already install and one wants to upgrade to 1.6.2 (once it's available)? yum upgrade asterisk* ??? Thanks It should be as easy as a `yum update`. That's the goal, anyways. -- _ -- 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 as a skinny/sccp client?
Brian J. Murrell wrote: I wonder if Asterisk's skinny/sccp channel driver could be used as a client to register with a Cisco PBX. That is, along with a SIP client, say, have Asterisk and said SIP client stand in for a Cisco phone, or an IP Communicator. Anyone done this? Cheers, b. No, this isn't currently possible. I did ponder this for a while, but my conclusion was that the effort required to do so would far outweigh any benefit you'd gain from it. Cisco has been moving to SIP for a very long time. There aren't any phone features that Asterisk could emulate that would make this any better than SIP (or even anything approaching parity). -- _ -- 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 RPM's
Jay Vocaire wrote: Thanks for researching this for me. If you actually look at the link you sent me, you will see that the latest is: asterisk16-core-1.6.0.21-1_centos5.x86_64.rpm 20-Jan-2010 15:45 11M So, we come back to my original question: is there a reason for the delay on getting the RPM's out? Btw- I am doing yum update, it seems to agree with the above, that the latest RPM is .21. Thanks. -Jay Usually the RPMs are available at the same time new source tarballs are released. This time, that was not the case. Updated packages are available now, however. To force a refresh of repository information and upgrade, you can run `yum clean metadata; yum update`. -- _ -- 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] Cannot built kmod-dahdi-linux for PAE kvariant from SRPM
stephen.hindma...@bt.com wrote: rpmbuild --bb ~/localrpms/SPECS/dahdi-linux-kmod.spec snip error: Failed build dependencies: kernel-devel = 2.6.18-164.11.1.el5 is needed by dahdi-linux-kmod-2.2.1-1_centos5.2.6.18_164.11.1.el5.i386 Add a --target=i686 to your rpmbuild line. -- _ -- 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] Unable to open pid file '/var/run/asterisk/asterisk.pid': No such file or directory
Brian wrote: Each time the server is rebooted Asterisk duly deletes the manually created /var/run/asterisk directory - quite why it does this I just don't know - perhaps it is a bug? Your assumption is incorrect. Some Linux distributions will empty /var/run/ on boot, just as they do with /tmp/. I do believe you're right, however, in suggesting that there is a bug in Asterisk. It appears that Asterisk creates /var/run/asterisk/ during install and assumes that it will always exist. Some of the sample init scripts (Debian) create that directory before starting Asterisk. This should be done in all of them (or in Asterisk itself, maybe?). Please report an issue on http://issues.asterisk.org/ -- _ -- 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] Setting outgoing callerid on when using a PRI
Doug Lytle wrote: Dave Fullerton wrote: Note num and not number I don't know if that was a change from 1.4 to 1.6 or if Doug mistyped it. Not a mistype. I've been using number all along, but looking at the docs shows that I've been incorrect. It must concatenate the number down to num. Looks like I've got a little modifying to do this evening: core show function CALLERID livonia*CLI -= Info about function 'CALLERID' =- [Syntax] CALLERID(datatype[,optional-CID]) [Synopsis] Gets or sets Caller*ID data on the channel. [Description] Gets or sets Caller*ID data on the channel. The allowable datatypes are all, name, *num*, ANI, DNID, RDNIS. Uses channel callerid by default or optional callerid, if specified. Doug The documentation is correct, but the way the check really works, is that it reads the first 3 chars and matches it to num. This means that num, number, and numnumnumIloveapplesauce would all technically match. ___ -- 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 setting up IMAP_STORAGE on CentOS 5
Noah I. Engelberth wrote: I’ve been spending the day trying to get IMAP_STORAGE on my test box, to evaluate for production, but I’m having no luck getting uw-imap to build. I’ve tried installing it from an upstream package, but Asterisk still isn’t finding it to compile –with-imap. My google searches have turned up very little for documentation on dependencies, gotchas, etc for either item, so I’m hoping someone here can help me get IMAP set up for my Asterisk box. You should be able to just `yum install libc-client-devel` on CentOS. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Extra Sounds Missing on 1.6.1.6 install
Mark Hulber wrote: It looks like there's a problem with the location or naming of the Extra SLN16 sounds: This has already been fixed in the 1.6.1 branch. It should make its way into the next releases. See 1.6.1 revision 212386. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New CentOS 5 repo: dahdi, asterisk, freepbx RPMs
D Tucny wrote: %changelog [snip] awesomeness here [/snip] I'm speechless. This is far beyond what I could have possibly hoped for. It is also extremely accurate. Thank you very much for this. I'll be sure to keep this (and others) up to date in the future. ___ -- 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] New CentOS 5 repo: dahdi, asterisk, freepbx RPMs
D Tucny wrote: 2009/3/26 John Morris aster...@zultron.com mailto:aster...@zultron.com Hi, Axel. Axel Thimm wrote: How about merging in your changes/improvements/new packages with ATrpms (and automatically later into rpmrepo.org http://rpmrepo.org)? That way we won't have further fragmentation and a larger user base to test bits (which will be distributed in stable, testing etc repos). Of course I'd love to contribute my changes to ATrpms. Some of the small changes I made, such as adding OSLEC to the DAHDI RPMs, might be nice for ATrpms users. I'll whip up some patches against the ATrpms sources. My problem with ATrpms, though, is that the RPMs make use of many custom macros that make them unbuildable outside the ATrpms environment. I understand that might be necessary for RPMs like DAHDI that build kernel modules for several versions of several distros, where vanilla specfile code would get hairy. (I think we had this discussion a couple of years ago on the ATrpms ML.) Since I don't have to worry about multiple versions of multiple distros in my environment, I prefer to use vanilla specfile that will rebuild on anyone's CentOS 5 system. Alternatively, there's also the RPMS at http://packages.asterisk.org/centos/ which seem to have a nice spread of options available, including 1.4/1.6 packages, are pretty nicely modularised and seem to be kept pretty fresh... They do however seem to have some issues that your RPMS (and Axel's) don't (e.g. why wouldn't an init file be included? and where's the changelog?)... Perhaps it would be useful to help the digium packager build some better packages... That would also help with reducing fragmentation, if there were decent quality 'official' packages available then it would save the time and effort Axel and the rpmrepo.org http://rpmrepo.org folks too as they could in theory base any extras on those packages rather than needing to maintain the entire set... d As the author of the RPMs at http://packages.asterisk.org/ (as well as http://packages.digium.com/), and the maintainer of the repositories, I wanted to respond to this. I would love it if some of this were to happen. I am very familiar with Axel and ATrpms - he has proven countless times that he knows what he's doing when it comes to this sort of thing. Getting help/advice from somebody like him would be extremely beneficial. As far as basing the ATrpms (or others) packages on the AsteriskNOW packages, if that is something that Axel (or others) wanted to do, I would be more than willing to help with whatever is needed. On a somewhat related, and very interesting note - I found out yesterday that the latest trixbox beta is using these RPMs (without even needing to rebuild them, in some cases). Hopefully that means I'm doing something right. D, the two issues you brought up are valid. For the Asterisk RPMs, I honestly don't know why there isn't an init script - I actually thought there was one. FreePBX is what starts Asterisk in AsteriskNOW, so it was easily overlooked. It will be there in future builds. As far as the changelog, it was one of those things that I intentionally left out for a while, and I kept meaning to do it later. Really, it's because I'm not sure what should go into an RPM changelog (I'd love to hear from anybody that has any insight into that). As always, if anybody has any ideas, suggestions, criticism, or any other type of feedback, I'd be happy to hear from you. ___ -- 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] Bounty- CDR Bug Fix
Robert Broyles wrote: I saw some of the heat about the $20 bounty earlier. So I don't want to put a low bounty out. Quote me a bounty, and I'll see if I can get it approved by management. :-) I'm in need of getting this bug fixed. Bug has all of the details, but basically 1.4.22 broke it all. I've waited as long as I can - hoping the bug would 'resolve itself' - but now I'm putting a bounty out on it. http://bugs.digium.com/view.php?id=13691 There is already a patch on this bug that requires testing... If you have feedback, please respond on the bug so that it can get committed for inclusion into future releases. If the patch works, I'm sure murf would accept a good rootbeer. :) ___ -- 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] Bounty- CDR Bug Fix
Tilghman Lesher wrote: On Wednesday 04 March 2009 10:24:16 Robert Broyles wrote: By the way, I'm more than happy to send murf a case of rootbeer (or real beer assuming he's legal :-P ) if this bug and/or related bugs can be resolved soon. :-) Murf is plenty legal; he simply doesn't consume alcohol. This, of course, has nothing to do with my original point. It was more along the lines of no need to pay a bounty - it may already be fixed. :) There was another patch uploaded to that bug several weeks ago that I believe supersedes the original patch(es). That is what I was suggesting testing. The comments on the bug explain it. ___ -- 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] AsteriskNOW 1.5 upgrade from 1.4 to 1.6
Jason Lixfeld wrote: This link (http://blogs.digium.com/2008/10/13/asterisknow-15-beta-available-more-coming-soon/ ) seems to indicate that in order to upgrade AsteriskNOW v1.5 from Asterisk 1.4 to 1.6, it's as easy as installing an upgrade package. Does anyone know where to find that upgrade package? If it doesn't yet exist, what is the process for upgrading? Haven't figured out quite how I want to do this yet, but this is what has worked for me in testing (you may need to modify this slightly to add asterisk addons, if you're using it). Run `yum shell`, then in that shell, execute: install asterisk16-core asterisk16 remove asterisk14 asterisk14-core ts solve ts run remove asterisk14-core ts solve ts run If it went properly, it won't try to remove anything like FreePBX (it will prompt you before it does anything, so you can say 'No' if it tries). ___ -- 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] AsteriskNOW 1.5 - app_voicemail_imapstorage.so won't talk to IMAP server
This should now be fixed. If you want to force an update, you can do something like `yum clean metadata; yum update` Jason Parker wrote: It apparently isn't built with IMAP support. That would be a bug in my packaging. I'll see what I can do with it. Jason Lixfeld wrote: I'm having some issues getting app_voicemail_imapstorage to talk to my IMAP server. From imapstorage.txt, I've got the voicemail.conf configured properly, but if I leave a voicemail for extension , I see no indication that the module is trying to reach the IMAP server. What am I missing? # voicemail.conf [general] imapserver=172.16.17.2 [default] = ,,,,imapuser=joe|imappassword=joespassword # full.log | grep -i voicemail [Nov 11 18:11:37] VERBOSE[13681] logger.c: -- Reloading module 'app_voicemail_imapstorage.so' (Comedian Mail (Voicemail System)) [Nov 11 18:11:37] VERBOSE[13681] logger.c: == Parsing '/etc/asterisk/ voicemail.conf': [Nov 11 18:11:37] DEBUG[13681] config.c: Parsing /etc/ asterisk/voicemail.conf [Nov 11 18:11:37] DEBUG[13681] app_voicemail_imapstorage.c: VM Temperary Greeting Reminder Option disabled globally [Nov 11 18:11:37] DEBUG[13681] app_voicemail_imapstorage.c: VM CID Info before msg disabled globally [Nov 11 18:11:37] DEBUG[13681] app_voicemail_imapstorage.c: Send Voicemail msg disabled globally [Nov 11 18:11:37] DEBUG[13681] app_voicemail_imapstorage.c: ENVELOPE before msg enabled globally [Nov 11 18:11:37] DEBUG[13681] app_voicemail_imapstorage.c: Duration info before msg enabled globally [Nov 11 18:12:14] VERBOSE[13893] logger.c: -- Executing [EMAIL PROTECTED] exten-vm:15] NoOp(SIP/-097abc68, Voicemail is ) in new stack [Nov 11 18:12:14] VERBOSE[13893] logger.c: -- Executing [EMAIL PROTECTED] exten-vm:17] NoOp(SIP/-097abc68, Sending to Voicemail box ) in new stack [Nov 11 18:12:14] VERBOSE[13893] logger.c: -- Executing [s- [EMAIL PROTECTED]:1] NoOp(SIP/-097abc68, BUSY voicemail) in new stack [Nov 11 18:12:14] DEBUG[13893] pbx.c: Launching 'VoiceMail' [Nov 11 18:12:14] VERBOSE[13893] logger.c: -- Executing [s- [EMAIL PROTECTED]:3] VoiceMail(SIP/-097abc68, [EMAIL PROTECTED]|b) in new stack [Nov 11 18:12:14] DEBUG[13893] app_voicemail_imapstorage.c: Before find_user [Nov 11 18:12:14] DEBUG[13893] app_voicemail_imapstorage.c: /var/spool/ asterisk/voicemail/default//busy doesn't exist, doing what we can [Nov 11 18:12:26] DEBUG[13893] app.c: Locked path '/var/spool/asterisk/ voicemail/default//INBOX' [Nov 11 18:12:26] DEBUG[13893] app.c: Unlocked path '/var/spool/ asterisk/voicemail/default//INBOX' [Nov 11 18:12:26] DEBUG[13893] app.c: play_and_record: None, /var/ spool/asterisk/voicemail/default//tmp/t4mOcQ, 'wav49|wav' [Nov 11 18:12:26] VERBOSE[13893] logger.c: -- x=0, open writing: / var/spool/asterisk/voicemail/default//tmp/t4mOcQ format: wav49, 0x9844ad0 [Nov 11 18:12:26] VERBOSE[13893] logger.c: -- x=1, open writing: / var/spool/asterisk/voicemail/default//tmp/t4mOcQ format: wav, 0x98219f0 [Nov 11 18:12:33] DEBUG[13893] app.c: Locked path '/var/spool/asterisk/ voicemail/default//INBOX' [Nov 11 18:12:33] DEBUG[13893] app.c: Unlocked path '/var/spool/ asterisk/voicemail/default//INBOX' ___ -- 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] AsteriskNOW 1.5 - app_voicemail_imapstorage.so won't talk to IMAP server
It apparently isn't built with IMAP support. That would be a bug in my packaging. I'll see what I can do with it. Jason Lixfeld wrote: I'm having some issues getting app_voicemail_imapstorage to talk to my IMAP server. From imapstorage.txt, I've got the voicemail.conf configured properly, but if I leave a voicemail for extension , I see no indication that the module is trying to reach the IMAP server. What am I missing? # voicemail.conf [general] imapserver=172.16.17.2 [default] = ,,,,imapuser=joe|imappassword=joespassword # full.log | grep -i voicemail [Nov 11 18:11:37] VERBOSE[13681] logger.c: -- Reloading module 'app_voicemail_imapstorage.so' (Comedian Mail (Voicemail System)) [Nov 11 18:11:37] VERBOSE[13681] logger.c: == Parsing '/etc/asterisk/ voicemail.conf': [Nov 11 18:11:37] DEBUG[13681] config.c: Parsing /etc/ asterisk/voicemail.conf [Nov 11 18:11:37] DEBUG[13681] app_voicemail_imapstorage.c: VM Temperary Greeting Reminder Option disabled globally [Nov 11 18:11:37] DEBUG[13681] app_voicemail_imapstorage.c: VM CID Info before msg disabled globally [Nov 11 18:11:37] DEBUG[13681] app_voicemail_imapstorage.c: Send Voicemail msg disabled globally [Nov 11 18:11:37] DEBUG[13681] app_voicemail_imapstorage.c: ENVELOPE before msg enabled globally [Nov 11 18:11:37] DEBUG[13681] app_voicemail_imapstorage.c: Duration info before msg enabled globally [Nov 11 18:12:14] VERBOSE[13893] logger.c: -- Executing [EMAIL PROTECTED] exten-vm:15] NoOp(SIP/-097abc68, Voicemail is ) in new stack [Nov 11 18:12:14] VERBOSE[13893] logger.c: -- Executing [EMAIL PROTECTED] exten-vm:17] NoOp(SIP/-097abc68, Sending to Voicemail box ) in new stack [Nov 11 18:12:14] VERBOSE[13893] logger.c: -- Executing [s- [EMAIL PROTECTED]:1] NoOp(SIP/-097abc68, BUSY voicemail) in new stack [Nov 11 18:12:14] DEBUG[13893] pbx.c: Launching 'VoiceMail' [Nov 11 18:12:14] VERBOSE[13893] logger.c: -- Executing [s- [EMAIL PROTECTED]:3] VoiceMail(SIP/-097abc68, [EMAIL PROTECTED]|b) in new stack [Nov 11 18:12:14] DEBUG[13893] app_voicemail_imapstorage.c: Before find_user [Nov 11 18:12:14] DEBUG[13893] app_voicemail_imapstorage.c: /var/spool/ asterisk/voicemail/default//busy doesn't exist, doing what we can [Nov 11 18:12:26] DEBUG[13893] app.c: Locked path '/var/spool/asterisk/ voicemail/default//INBOX' [Nov 11 18:12:26] DEBUG[13893] app.c: Unlocked path '/var/spool/ asterisk/voicemail/default//INBOX' [Nov 11 18:12:26] DEBUG[13893] app.c: play_and_record: None, /var/ spool/asterisk/voicemail/default//tmp/t4mOcQ, 'wav49|wav' [Nov 11 18:12:26] VERBOSE[13893] logger.c: -- x=0, open writing: / var/spool/asterisk/voicemail/default//tmp/t4mOcQ format: wav49, 0x9844ad0 [Nov 11 18:12:26] VERBOSE[13893] logger.c: -- x=1, open writing: / var/spool/asterisk/voicemail/default//tmp/t4mOcQ format: wav, 0x98219f0 [Nov 11 18:12:33] DEBUG[13893] app.c: Locked path '/var/spool/asterisk/ voicemail/default//INBOX' [Nov 11 18:12:33] DEBUG[13893] app.c: Unlocked path '/var/spool/ asterisk/voicemail/default//INBOX' ___ -- 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] dahdi service start
Jerry Geis wrote: wct4xxp: sh: /sbin/ztcfg: No such file or directory FATAL: Error running install command for wct4xxp [FAILED] Hmm.. Something in /etc/modprobe.conf, /etc/modules.conf, or /etc/modprobe.d/? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7970, CTLSEPmac.tlv
Jason Parker wrote: I just wanted to post this so that it was out there and Googleable. Hopefully it will save other people a bit of time. If you have a Cisco phone (I was testing with a 7970, though presumably it would affect 7960 and others as well) that is looping trying to fetch the CTL tlv file - it may be because you are using Debians 'tftpd' (should be netkit-tftpd...*cough*hey, Debian developers*cough*) package, which is apparently not RFC 783 (tftp) compliant with file not found responses. The whopping 18 page RFC states that Error Code should be 0x00,0x01 for file not found errors, but netkit-tftpd returns 0x00,0x00 which is Not defined - causing the phone to ignore it and request the file again a few seconds later. Solution: Switch to any other tftpd. The moment I switched to tftpd-hpa or atftpd, the phone stopped looping, picked up the SEPmac.cnf.xml file, and immediately registered to Asterisk. Hopefully in the future Debian will rename, remove, or fix this package so it is no longer the default tftpd. Responding to myself... When I initially sent this, I had made several false assumptions. The biggest of which, was that the 'tftpd' package in Debian was no longer maintained (upstream hadn't made a release in 8 years, and Debian hadn't made a release in 3 years - I think it was a fairly reasonable one). Well, the maintainer of this package, Alberto, emailed me to let me know that somebody pointed him to this post, and that less than 24 hours later, he had fixed this bug (I've confirmed this) and made a new release - 0.17-16 - which is currently in Sid, and will hopefully be put into Lenny. This can be downloaded from http://packages.debian.org/search?keywords=tftpd Also, as Alberto correctly pointed out - I violated one of the most important rules of Open Source Software. If I may quote him: You had perfectly traced the problem, you perfectly described it, god! you even gave a reference to the RFC. You had the perfect bug report, but it was never going to make it to me arrrggg :) Such a great loss!! I failed to complete one critical step - reporting a bug. It ended up working out, but only because somebody else took the time to report the bug. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cisco 7970, CTLSEPmac.tlv
I just wanted to post this so that it was out there and Googleable. Hopefully it will save other people a bit of time. If you have a Cisco phone (I was testing with a 7970, though presumably it would affect 7960 and others as well) that is looping trying to fetch the CTL tlv file - it may be because you are using Debians 'tftpd' (should be netkit-tftpd...*cough*hey, Debian developers*cough*) package, which is apparently not RFC 783 (tftp) compliant with file not found responses. The whopping 18 page RFC states that Error Code should be 0x00,0x01 for file not found errors, but netkit-tftpd returns 0x00,0x00 which is Not defined - causing the phone to ignore it and request the file again a few seconds later. Solution: Switch to any other tftpd. The moment I switched to tftpd-hpa or atftpd, the phone stopped looping, picked up the SEPmac.cnf.xml file, and immediately registered to Asterisk. Hopefully in the future Debian will rename, remove, or fix this package so it is no longer the default tftpd. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Remote Support
Philipp Kempgen wrote: I would suggest screen ( http://en.wikipedia.org/wiki/GNU_Screen ). screen doesn't solve the security aspect of your question though. Grüße, Philipp Kempgen Actually, it could. What I've done before, is give out an unprivileged account on the box (or some intermediate gateway box). Once they log in, you ask them to run screen (as the unprivileged user) to connect to a session you've created, then proceed to login as root yourself. If they disconnect their screen session, they leave your root terminal. You can also kill the screen session at any time. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel Install Error
Steve Totaro wrote: This looks like it may be your problem. http://bugs.digium.com/view.php?id=9592 (0070069) qwell - administrator 09-06-07 17:05 Closing. The simple solution here is to just comment out the #define USE_RTC in ztdummy.c. The ztxen module does not appear to be needed. Thanks, Steve Totaro Just to clarify for those that don't want to read through the bug notes.. That bug was a feature enhancement that added support for xen, through a new module named ztxen. The only difference in this new module vs ztdummy, was that it removed the RTC code. In order to mimic this, and get a proper ztdummy on xen, all somebody needs to do is comment out the single #define USE_RTC line in ztdummy.c ___ -- 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] gotoif syntax error
מוישי ברעוודה wrote:
Asterisk is reporting the following error:
[Apr 15 16:58:32] WARNING[14759] ast_expr2.fl: ast_yyerror(): syntax
error: syntax error, unexpected ':', expecting $end; Input:
: Always
^
here is the dialplan:
exten = OUT,1,Gotoif($[$[${DB(AMPUSER/${ARG1}/recording)} :
out=([^|]+)] = Always]?r,1)
exten = GROUP,1,Gotoif($[$[${DB(AMPUSER/${ARG1}/recording)} :
out=([^|]+)] = Always]?r,1)
exten = IN,1,Gotoif($[$[${DB(AMPUSER/${ARG1}/recording)} :
in=([^|]+)] Always]?r,1)
The error is for the last line (IN,1). Funny thing is that asterisk
doesnt report any error for the first line (OUT,1)
Because OUT is correct. IN is missing a =, as in = Always
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Re: [asterisk-users] Zaptel 1.2.25 and 1.4.10 released
Brent Davidson wrote: Do they mean 1.4.20 instead of 1.4.10? If not, then this message was seriously delayed :-D -Brent Zaptel, not Asterisk. :) 1.4.10 is correct. ___ -- 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] Menuselect?
Continuing the top-posting madness... For future reference (and for the archives), you could have done `make dist-clean` and re-run configure, rather than remove the directory. Kyle Gibbons wrote: All, Thank you very much for your help, I have solved the problem. After installing ncurses-devel I had to completely delete the zaptel directory(I know I was asking about Asterisk, but I was having the same problem and of course was starting with zaptel install). I tried doing make dirclean, but even that did not work. Once I completely deleted the directory, I untarballed the file again, ran ./configure and make menuselect and now everything runs properly. Thanks again for your help! On Tue, Mar 25, 2008 at 9:38 PM, Kyle Gibbons [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Okay, I think I have found part of the problem. When I do make menuselect at the end it reads Install ncurses to use menu interface!. I already have ncurses and ncurse-devel installd so I am perplexed as to why this is coming up. Any thoughts? On Tue, Mar 25, 2008 at 9:06 PM, Kyle Gibbons [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi, I am running asterisk 1.4.18.1 http://1.4.18.1 when I do make menuselect it makes it and the last line is menuselect changes NOT saved! and then it goes back to the prompt On Tue, Mar 25, 2008 at 6:00 AM, Grygoriy Dobrovolskyy [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: 1.2 has no menuselect 1.4: ./configure make menuselect (and you get into it after this command automatically) 2008/3/25, Rob Hillis [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]: Only if you're trying to compile Asterisk 1.2. Asterisk 1.4 also has the menuselect configuration, though for most applications you don't really need to fiddle with it. James M Kupernik wrote: There actually is no menuselect, its just a simple ./configure make make install in that order Hope that helps James -- James M Kupernik Network Engineer VoodooVox, Inc. [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] Office: 646.710.3127 Mobile: 413.446.5974 - Original Message - *From:* Kyle Gibbons mailto:[EMAIL PROTECTED] *To:* asterisk-users@lists.digium.com mailto:asterisk-users@lists.digium.com *Sent:* Monday, March 24, 2008 9:47 PM *Subject:* [asterisk-users] Menuselect? Hi, I hope this is not too much of a noob question. I am trying to compile Asterisk and I cannot figure out how to get into the Menuselect menu. I do #make clean #./configure #make menuselect, but I cannot figure out how to actually get into the menu select interface. I am running CentOS 5. I have done quite a bit of searching on Google and have not come up with anything. Also, I am reading the Asterisk book from O'reilly and it does not seem to explain this. Please forgive this probably simple question as I am new to Linux and semi-new to Asterisk. Thank you in advance for your help. -- All the best, Kyle bobert5064.deviantart.com http://bobert5064.deviantart.com ___ -- 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] Connecting a Rolm CBX to Asterisk via T1?
Joshua Kinard wrote: -Original Message- You probably mean a T100P? The single E1/T1 card? Been a few years but I remember seeing the NMI Errors on a HP DL380 (the Intel dual Xeon model). Nah, it's classified as a D110P, although the driver says TE110P. And I checked to make sure I had the onboard jumper rigged for T1 (open), not E1 mode (closed). There's another, unidentified jumper on the board too, but I'm not sure what it's for. The D110P is a clone card, which is *not* made/sold/endorsed/etc by Digium. I would suggest getting a newer card, which would not exhibit these types of issues. You will save yourself many headaches in the future. ___ -- 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] It's about time! -- Digium PCI-Express Cards
Matt wrote: Just noticed this today: Digium TE220B Dual Span T1/E1 PCI Express Card with Octasic DSP-based Echo Cancellation Module http://www.voipsupply.com/product_info.php?products_id=3352 It's about time Digium got on the ball and made PCI-e cards. What are people's experiences with this card? Anyone know if there are plans for a PCI-e analog card for FXO use? Digium already makes PCI Express analog cards - AEX800 and AEX2400. -- Jason Parker Digium ___ -- 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] func_odbc - trouble
Dirk Enrique Seiffert wrote: I guess this libtool-ltdl-1.5.22-6.1 ... which is installed. Thanks Enrique I believe you're looking for libtool-ltdl-dev(el) -- Jason Parker Digium ___ -- 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] 7970 CTLFile.tlv?
You don't need the .tlv file. It's optional, and will be skipped if it cannot be found. Your problem is elsewhere. I've found that the 7970s are very finicky. I've never had luck with the SEPMAC.cnf.xml - only XmlDefault.cnf.xml (case may vary - check your tftp logs) Matthew Rubenstein wrote: I've got a Cisco 7970 that's not completing its network registration to Asterisk. The Registering message stays on the screen (with the moving time wheel). After a few minutes, the onscreen message flashes Updating CTL then Loading..., then the status messages update with: No valid CAPF server File Not Found: CTLFile.tlv No CTL installed SEPMAC.cnf.xml (where MAC is the phone's MAC addr minus :s) before repeating the cycle (forever). Where can I get a CTLFile.tlv , or remove the requirement for it? Or is there another way to fix this problem? TIA. Asterisk v1.2.9.1-BRIstuffed-0.3.0-PRE-1q SCCP firmware Load File: TERM70.7-0-1-0s App Load ID: Jar70.2-9-0-117.sbn JVM Load ID: CVM70.2-0-0-112.sbn OS Load ID: cnu70.2-7-4-134.sbn Boot Load ID: 7970_64060118.bin -- Jason Parker Digium ___ --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] Happy Birthday Asterisk
Philip Prindeville wrote: [...] There were earlier experimental versions of IP, but v4 got it right. and v6 will get it even more right. ;) -- Jason Parker Digium ___ --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] No timezone in Voicemail email?
Jason Martin wrote: Hello, I'm using Asterisk 1.4.14, and I've noticed that the emails that are sent out when a user gets a voicemail don't have the timezone set in the header, so they're appearing in the user's email clients at the wrong time. Has anyone else seen this? I didn't find any bug reports or other info with Google. This is already fixed in 1.4.15. -- Jason Parker Digium ___ --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] Soundcard necessary on an asterisk server to get output of playback()??
I think Lacy means rub the mouthpiece of the phone - to make sound (blowing into it should yield the same result) Lacy Moore wrote: My quick guess would be that it's a timing issue. You didn't mention whether you are using a Zaptel device or ztdummy. I know this sounds like I'm being a smart***, but I'm not... try this... rub the mouthpiece of the file while the sound file is playing and see if you hear any of the file. If so, I would definitely say you have a timing issue. On Dec 3, 2007 12:01 PM, Stefan Guenther [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi, I' still fighting the problem, that I can talk from one SIP phone to another, but I can't hear the output of the playback or similar applications: exten = 202,1,ANSWER() exten = 202,2,PLAYBACK(tt-monkeys) exten = 202,3,HANGUP() When I dial 202, asterisk show the following on the cli: -- Executing [ [EMAIL PROTECTED]:1] Answer(SIP/user1-0827ebe8, ) in new stack -- Executing [EMAIL PROTECTED]:2] Playback(SIP/user1-0827ebe8, tt-monkeys) in new stack -- SIP/user1-0827ebe8 Playing 'tt-monkeys' (language 'de') Yes, the file tt-monkeys exist in /var/lib/asterisk/sounds and the subdirectory de. No, there is no error message even if turn on debugging. :-( Besides this strange behaviour, I was wondering whether the asterisk server needs an soundcard to send the output of e.g. the playback application to the phone. BTW, this is asterisk 1.4.13 I would be really happy, if someone has an idea how to solve this problem. Thanks in advance, Stefan -- in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Geschaeftsfuehrer Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93 http://www.in-put.de http://www.in-put.de/ Schulungen Installationen Beratung Support Voice-over-IP-Loesungen ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- http://www.api-digital.com--/ asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users -- Lacy Moore Somewhere I wish I wasn't ___ --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 -- Jason Parker Digium ___ --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] asterisk ODBC dependencies
Robert McNaught wrote: ... Anyone know the secret to the dependencies? Robert McNaught It's case sensitive. I believe RH uses unixODBC as the package name. You also need the development package of that. -- Jason Parker Digium ___ --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] ztdummy and BackGround
I've seen something similar happen before, and it was due to having drivers for cards loaded that were not in the system. Try removing all modules (including ztdummy), then loading ztdummy. Atis Lezdins wrote: On 11/2/07, Tony Plack [EMAIL PROTECTED] wrote: We are going to implement MeetMe, but this should still work right? I had similar issues with 1.4.12 just one time (also topmost zaptel at that date). We recompiled everything, and it suddenly worked. But we had to revert to 1.4.10 because of some crashes - so no more data on this. If you can repeate this, i suggest you registering a bug in bugs.digium.com, i could add meetoo there :D Regards, Atis Tony Plack wrote: I have an interesting issue. I am running Asterisk 1.4 (SVN branch latest) and same with Zaptel. If I load ztdummy, my audio in BackGround (or Playback) cannot be heard. If I rmmod ztdummy and restart Asterisk, Background works. What am I missing? What things are you using that requires zaptel timing (ztdummy)? ___ --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 -- Jason Parker Digium ___ --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] [Fwd: Internal LAN echo problem]
See response in-random-lined. David Gomillion wrote: On 10/24/07, *David Gomillion* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: On 10/24/07, *Steve Totaro* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Let me screw this thread up by top posting now. Could echo be caused by late packets if jitterbuffer is on or something or would that just cause lag? Thanks, Steve So, does this qualify as an in-line reply, or a top post? Maybe it's a medium post ;) If both calls were in the LAN, chances are good that the phones will have re-invited to go around the SIP server. If that's the case, then it shouldn't be a problem. Now, if dial options, recording, or SIP settings prevent reinvites, then this might be part of the problem. Sorry, I need to clarify my own post. By part of the problem, I mean magnifying the effect. The real problem is the handset leaking, probably too much sidetone. Anyway, the more the delay, the more noticeable this echo will usually be. kevin bergner wrote: On 10/24/07, Eric ManxPower Wieling [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Jonn Taylor wrote: Eric ManxPower Wieling wrote: Any echo you hear on pure IP calls is caused by the endpoint phone. You cannot do ANYTHING about it on Asterisk. Jonn Taylor wrote: Any ideas ? Jonn Original Message Subject:[asterisk-users] Internal LAN echo problem Date: Wed, 24 Oct 2007 08:34:32 -0500 From: Jonn R Taylor [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] Will the madness never end? Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com mailto:asterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com mailto:asterisk-users@lists.digium.com Hi all, I have an internal echo problem on my LAN only. I replaced the LAN switch with a new linksys 2024 with QOS and seemed to help but not fix the problem. Any ideas? Here in my setup - Dell PE6400 Dual 700, Asterisk 1.2.24 /FreePBX, 2-NIC cards, one with a public ip and one with an internal ip, one PAP2, one SPA3102 and 2 BT101. I know that bt's are cheap that are known for echo problem in the handset. I have one remote user that never has a problem. I have a remote test server at home connect via IAX with no problems, also a PAP2 with no problem. External faxing from the rest of the world via our voip provider is working great. One strange thing that I noticed is that we can not fax to our iaxmodem, ATA --- iaxmodem, but works perfect ATA --- rx_fax. Not sure why either. That does not make sense. I can any one of these ata's or phones and connect them to the public ip side and they work fine. It can make sense or not make sense, but you cannot have echo on a pure VoIP call unless the endpoints introduce it. ___ --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 http://lists.digium.com/mailman/listinfo/asterisk-users i have seen this when the headset volume is too high and simply lowering the volume addressed the problem as others have said an echo is simply not possible ___ --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 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 -- Jason
Re: [asterisk-users] Sangoma vs Digium: (was Re: ping too)
Lee Howard wrote: The report appears to have been reaped from Mantis, but I was involved with a contribution from OpenVOX for zaptel, and from my perspective it looked like the Digium staff involved killed it and never gave any indication that the contribution would be accepted. I assume you are referring to issue 7742 - http://bugs.digium.com/view.php?id=7742 The OpenVox tech (MiaoLin) said to Tzafrir (who does not work for Digium) that he needed to make changes to the patch. As was stated on the bug when it was closed, once those changes are made, we would certainly add the patch. If you have the updated patch with the changes he said were needed, please do reopen the bug. -- Jason Parker Digium ___ --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] Cisco 7940G licensing with asterisk
Matthew Rubenstein wrote: I just got SIP firmware images from Cisco for installation on 7970G. Cisco requires you buy a SmartNet account (about $15, no other dependencies apply) that entitles you to download a SIP firmware image file from their protected support website. The 7970G now needs a different image than the other 79xx phones, but the same rules apply to all of them. Those rules do not require any other license or other restriction, once you have legitimately obtained and installed the firmware on the phone, to use the phones with Asterisk (or any other 3rd party system). Of course, to use the phones with Cisco's CallManager product, you must have a licensed copy of the CallManager product, with all the other restrictions and fees that come with it. FWIW, the procedure of buying that SIP image from Cisco was a nightmare. I had to buy the SmartNet account from a reseller which did nothing to ensure that I completed the download transaction that was the stated purpose (as they described it to me) of buying the license. Then navigating to the license I needed, among the many versions and revisions, was confusing and opaque. The SmartNet account took days to send to me, and didn't work for the required access when it arrived. Cisco consumed an entire workweek to deliver the license that didn't unlock the website, then of course ignored requests for support through the weekend (into which their late delivery forced my request to be made). When I finally got Cisco to respond, they did deliver a knowledgeable and honest support tech who stuck with me until I had everything I needed to proceed. Though every stated maximum turnaround time for every phase in the process was exceeded, sometimes by many multiples. But since the image can be used only with a Cisco phone, which must (ultimately) be bought from Cisco, the kafkaesque procedure is intolerable. The image should be a one-click download that charges your credit card and comes with a SmartNet account, if they absolutely must charge the $15. In a sane world, the SIP image wouldn't have any restrictions, a free download that people could just email each other (or its URL), because its distribution would market Cisco phones. But probably Cisco knows that the SIP image lets (free) Asterisk compete with its proprietary CallManager, so they make it both a revenue source, and as complicated as possible. The way I understand it, that $15 doesn't actually even give you the right to use the SIP firmware. It only gives you the right to access the download area. The whole model is silly, at best. -- Jason Parker Digium ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] OT: Samsung Sprint CDMAoIP
C F wrote: AFAIK, the calls are free when you use it thru that device. Sprint however charges $15 a month per phone or $30 for family plan. While I agree that sprint should pay me for this, it's not as bad. T-mobile on the other hand, does the same thing with wifi enabled phones, it doesn't cost extra, and is completely free. If you're referring to T-Mobile's [EMAIL PROTECTED] service, it's actually $20 per month, per line on the account (unless it's changed very recently). As far as how it works on T-Mobile, I recently had some questions answered by them about that.. They use UMA over wifi, and it does automatic switching between the wifi and the gsm towers (ie; your call stays up). Quote from the tech I talked to: [EMAIL PROTECTED] does not use a VoIP protocol, as the voice data is transferred from the Internet directly to our UMA Gateway and then through our regular Mobile Switching Centers. Pretty interesting stuff. -- Jason Parker Digium ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] g729 on 1.4.10.1
Scott Moseman wrote: The gateway is transcoding the PSTN into g729 and passing it to Asterisk. The Asterisk never sees the PSTN from the outside. I have watched the INVITE requests, they contain a request for a g729 only call. But the INVITE to the phone does not include g729. However, as previously stated, I did get a g729 phone to talk to another g729 phone. So I assume that means pass-through *can* work, but something is not working right? Thanks, Scott If you have anything in Asterisk trying to handle the audio, you cannot pass it through. For instance, if you are trying to record the call in ulaw, or trying to playback prompts that aren't available in g729. -- Jason Parker Digium ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] Followme app_followme
Kevin Kiely wrote: When using app_followme, I am receiving the following warnings on the console. We are calling the followme app with no options for additional voice announcements. Is anyone else experiencing this issue with 1.4.11? -- Executing [EMAIL PROTECTED]:1] FollowMe(SIP/101206006-b72223d8, 101206002) in new stack [Sep 13 12:10:37] WARNING[1447]: file.c:563 ast_openstream_full: File /var/spool/asterisk/followme.1189699837.464 does not exist in any format [Sep 13 12:10:37] WARNING[1447]: file.c:813 ast_streamfile: Unable to open /var/spool/asterisk/followme.1189699837.464 (format 0x4 (ulaw)): No such file or directory -- SIP/101206006-b72223d8 Playing 'followme/pls-hold-while-try' (language 'en') Kevin, You have already posted 2 duplicate bugs reports which were closed and a very clear answer was given as to why. I honestly do not know how much more clear I can make this. Yes, it was a problem in 1.4.11. However, this has ALREADY been fixed in svn. It will be in the next release. If you would like to have this fix, you can run the latest version of svn branch 1.4. -- Jason Parker Digium ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] Which cause less CPU usage: GSM or wav??
It will automatically pick the best recording for the current codec, so if you are in ulaw, it will choose the ulaw prompt. Barton Fisher wrote: Thanks Guys... ulaw it is. One more question if you don't mind. If a phase recorded as both .wav and .ulaw in the same folder, which will asterisk pick using Playback(), Read() and Background() since you can't specify the file extension in the command? I thought I change my script to begin recording new messages in ulaw instead of converting them all to ulaw at once. So it's possible to have two prompts with both file extension at a time Bart -- Jason Parker Digium ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] TDM400P (TDM22P) and aux power.
Joe Acquisto wrote: I need to ask, to refresh, is the aux power connector on the TDM400P card *only* to power the ringer on any analog phones/devices on the system? Can I still use this board, to terminate POTS lines and use all SIP Phones? Due to circumstances, I end up with a 1u server that has no aux power connectors available. I have to use this server, so am considering abandoning the analog phones and using all SIP. IIRC, the aux power *is* only to power ringers. joe a. Correct, it is to provide the ringing voltage on the FXS modules. For systems without internal molex connectors available, there is another option. Digium has created an externally powered supply that can be used with these cards. http://www.digium.com/en/products/hardware/analogpwr.php -- Jason Parker Digium ___ --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] Cisco 7960 Won'
Dan Austin wrote: Shawn wrote: I'm having a wierd problem with a Cisco 7960 (sccp2) and asterisk (1.4.2) If the call that I'm trying to make goes through, everything works fine. But if there's any sort of error (like me messing around in my extensions.conf, etc). I can't get the connection to drop. ie: If I get the conjestion tone and hang up the phone, I can do a sccp show channels I can see that the channel is still in use (even after several minutes). If I pick up the phone to attempt to make another call, I get an error that it can't put the current call on hold to start the new call. What am I missing? An upgrade. The sccp channel in early 1.4 had quite a number of problems, and it was completely broken in 1.4.3 to 1.4.6 Any version after 1.4.7 should work better, with the latest being the best choice. Dan Well, he's also using chan_sccp, so no amount of upgrading is going to help with that. In my opinion (and I think Dan and several others would agree), chan_skinny is far more stable (and active...) than chan_sccp. -- Jason Parker Digium ___ --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] G729 copy protection
Bruce McAlister wrote: Bruce McAlister wrote: Jul 19 14:11:23 WARNING[28243]: codec_g729.c:481 load_module: Failed to initialize G.729 copy protection! Hi, Could anyone from Digium please shed some light on the build environment for the solaris 10 g729 codec? Was it build on Solaris or OpenSolaris? Are there any specific versions of libraries required? I'm still having this issue, and still cannot get the codec working. I've had a few tips/pointer from Joe at Solaris VoIP, but now we need to know a little more about the build environment to see if we can actually get this codec working. i have tried to run the codec with asterisk 1.2.17, 1.2.20. 1.2.24, 1.4.4, 1.4.10, 1.4.10.1 and 1.4.11, they all fail with the same messages. Asteris has been built on Solaris 10 Update 3 patched up as of friday last week. Our focaus now is to try and get the codec working with asterisk 1.4.x on Solaris 10. I've also tried i386, i586 - pentium4 32bit, opteron 32bit, on physical Opteron 285's and intel Xeon (Nacona's), all faile with the same message. The codec version is v32. This message comes up whether I have a valis g729 license from Digium or not, I have tried both. In either case, I would assume that codec would at least load, and a show g729 at the cli would work with and without a license. Has anyone been able to test this codec with asterisk? Any tips/suggestions would be greatly appreciated. Thanks Bruce Bruce, Please see my response to some of these questions on July 23rd. http://lists.digium.com/pipermail/asterisk-users/2007-July/192473.html I'm not entirely certain of what libraries we statically link in, but if you see any problems with the output of `ldd codec_g729.so`, those will of course need to be installed. -- Jason Parker Digium ___ --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] 1.4 Branch -- which revision
Jay Milk wrote: I'm finally migrating from the 6/7/05 CVS version to 1.4. Had quite a run, I have to admit. Asterisk itself only segfaulted once or twice, but the dns issues have been bothering me. And the box just needs to go. Everything is going on a Ubuntu 6.06TLS server, that's been perfectly stable. I had 1.4.1 installed and running, but not configured. Yesterday I upgraded to 1.4.11, which went smoothly... alas, I really wanted chan_mobile. I attempted to use asterisk 1.4.11 with addons-branch 1.4, but that didn't do it. So it looks like I'd have to use the 1.4 branch for both asterisk and addons. What's the recommended revision here? I don't need bleeding edge (obviously), I just need it stable with chan_mobile and not too much else. Thanks! chan_mobile isn't in asterisk-addons in 1.4 - only trunk. You'll likely have to backport it... (it was developed against 1.4, so the diff from trunk is probably trivial) -- Jason Parker Digium ___ --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] Chan_mobile and Asterisk SVN-branch-1.4-r80183 compile error
Administrator TOOTAI wrote: Hi all, I receive this error while compiling chan_mobile: gcc -g -c -fPIC -fPIC -o chan_mobile.o chan_mobile.c chan_mobile.c: In function `mbl_load_config': chan_mobile.c:1745: erreur: trop d'arguments pour la fonction « ast_config_load » make[1]: *** [chan_mobile.o] Erreur 1 make[1]: Leaving directory `/usr/src/asterisk-addons' Does anyone know what's the problem? You're trying to use a module written for trunk on 1.4. -- Jason Parker Digium ___ --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] Chan_mobile and Asterisk SVN-branch-1.4-r80183 compile error
Administrator TOOTAI wrote: Jason Parker a écrit : Administrator TOOTAI wrote: Hi all, I receive this error while compiling chan_mobile: gcc -g -c -fPIC -fPIC -o chan_mobile.o chan_mobile.c chan_mobile.c: In function `mbl_load_config': chan_mobile.c:1745: erreur: trop d'arguments pour la fonction « ast_config_load » make[1]: *** [chan_mobile.o] Erreur 1 make[1]: Leaving directory `/usr/src/asterisk-addons' Does anyone know what's the problem? You're trying to use a module written for trunk on 1.4. Ah, ok, sorry. Was thinking it's for 1.4 trunk ;-) There is no such thing as 1.4 trunk. -- Jason Parker Digium ___ --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
