Re: [asterisk-users] 11.18.0 patch against 11.17.0 running version failed to apply
On Thu, Jul 09, 2015 at 12:28:15AM +0200, Administrator TOOTAI wrote: zone-s:/usr/src/asterisk-11.18.0# patch --dry-run -p1 ../asterisk-11.18.0-patch patching file .version Hunk #1 FAILED at 1. 1 out of 1 hunk FAILED -- saving rejects to file .version.rej patching file ChangeLog Hunk #1 FAILED at 1. 1 out of 1 hunk FAILED -- saving rejects to file ChangeLog.rej The next patch would delete the file asterisk-11.18.0-rc1-summary.html, which does not exist! Assume -R? [n] Apply anyway? [n] Skipping patch. 1 out of 1 hunk ignored The next patch would delete the file asterisk-11.18.0-rc1-summary.txt, which does not exist! Assume -R? [n] Apply anyway? [n] Skipping patch. 1 out of 1 hunk ignored patching file asterisk-11.18.0-summary.html patching file asterisk-11.18.0-summary.txt As you can see, patch is against -rc1 not 11.17.0 ... The content of files has changed. patch refuses to change from an unfamiliar content. Either edit the patch file and remove .version (edit the version manually) or edit the patch file and edit the version form 11.7.0 to 11.8.0 . The content of the files you refer to is normally insignificant to the behaviour of Asterisk. Just remove them from the patch and be done with it. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] howto copy a voicemail message to another machine ?
On Tue, Jun 16, 2015 at 11:35:26AM -0400, sean darcy wrote: My asterisk server is in the cloud. Figuring out how to send an email is too much brain damage. So i can't use the email feature that's built into voicemail. Why? If you can ssh between them, you should also be able to route mail between them. Be that using a custom sendmail command which uses ssh in it. What I want to do is execute a remote command with the voicemail as an argument. The remote machine command would email the message. I'm thinking of: same =n,VoiceMail(vm,u) same =n,System(ssh myserver emailVM '_THE_VOICEMAIL_MESSAGE_') What variables can I use for _THE_VOICEMAIL_MESSAGE_ Or is this better done with externcmd in voicemail.conf ?: externcmd = ssh myserver emailVM '_THE_VOICEMAIL_MESSAGE_' But same question. And can externcmd take arguments ? rsync the whole mailbox. Should be quick enough. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] small homebrew pbx
On Mon, Jun 15, 2015 at 04:56:31PM +1000, Tim Groeneveld wrote: On Mon, 15 Jun 2015 16:46:13 +1000 [Lucio] wrote Hello all, I'm new here and I'm interested in building a small PBX with asterisk at home. I have one single PSTN line and ethernet cabling in place. I already have fairly decent PC that I can use (AMD FX 8350 16GB of RAM and RAID 10 SATA disks). I make and receive 10 calls a day on average. You could have used a system that is a bit less powerful. But anyway, I guess that those are not that expensive nowadays. For that kind of load, even a much smaller server would do. I want 4 IP phones connected to the ethernet network. When there is a incoming call, all phones must ring and the first that takes the call makes the others stop ringing, but lets them available for internal calls. Given the requirements above, what's a cheap but working PCIe card / USB adapter I could buy for this kind of PBX? Do I need things like echo cancellation? Do I need FXS ports? You will need a FXS port. I would recommend setting up something like Cisco SPA3102. This is a slight confusion. The SPA3102 has both an FXS and an FXO port. What you need is an FXO port - a port to connect to the PSTN as a phone. An FXS port allows you to connect an analog phone. It is something you could have used for local extensions. But you already have IP pohnes. The SPA3102 can be found cheap on Ebay, and will be easy to setup in Asterisk. http://www.infoworld.com/article/2633694/data-modeling/your-pstn-and-you--linksys-spa-3102-and-asterisk.html Once the FXS is set up, it's just a matter of adding a ring group/pickup group: http://edoceo.com/exemplar/asterisk-call-groups Again, I guess you meant the FXO port of the device. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] I'm not able to install asterisk in AWS cloud
On Mon, Apr 13, 2015 at 05:43:28PM +0530, ajahar mohd wrote: Hi Akhilesh, Here is another fix, getting the error, that: make[1]: *** No rule to make target `../main/modules.link’, needed by `asterisk’. Stop. make: *** [main] Error 2 when compile asterisk To get around this, just delete following line in file makeopts.embed_rules EMBED_LDSCRIPTS+=../main/modules.link Source: http://showmyroutes.com/wordpress/?p=500 Two conclusions: 1. Why would you use module embedding? I don't think this gives you any benefits on a system with a proper linker. 2. When you ask a question about a failed build, please include something closer to a full build log. Specifically, a few more lines of context before the actual error. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Recommended changes to the binary packaging system
On Sun, Apr 12, 2015 at 08:22:44AM -0400, Aaron Hunter wrote: I think the way Digium has structured the binary packages could use a major change. I rely on the binary packages rather than compiling by source because my systems are managed by an automated CM tool (I use Ansible but those using Chef or Puppet will face the same problems) and for security reasons. I use the CentOS packages. If you don't like the existing packages, build your own packages. Some issues with the current package structure are: - It installs unneeded repositories. The asterisknow package install 12 repositories, of which I use at most 4 (2 asterisk and 2 digium) - It changes which repositories are enabled and disabled. - It overwrites .repo files - Packages are not signed - It overwrites /etc/issue which is a security violation (albeit a minor one). - It installs packages I don't need such as the dahdi ones.( Maybe some of these are needed for a minimal system, I could be wrong.) Unless something changed recently, there's a separate package asterisk-dahdi. Don't install it if you don't want it. - It requires the --enablerepo=x in the yum command line The conflict between the Digium repositories and epel is a problem for me as well but since I can't determine what the actual cause is (probably package naming issues) I won't include it in my list. Do include an example output of a conflict so we can have an idea of the potential problem. These problems break the automated management of my system and cause security concerns. Instead of the complex current system I would recommend something more simple: - Remove asterisknow or at least make it optional - Allow users to install a minimally functioning asterisk from the asterisk-x and asterisk-current repos only. Any additional modules needed should be installed separately, including those from the commercial digium repositories. - Sign the packages and enable gpgcheck - Don't overwrite system files or current .repo files What's the problem? Have you considered providing your own asterisk.conf with an alternative astetcdir? In other words: install asterisk .repo files, yum install asterisk, install config files, done. yum install asterisk means it installs a pre-defined set of modules. But you preffered to have a more modular packaging. If others on this list also use automated tools to manage their systems I'd like to hear how you handle the installation and maintenance of asterisk. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] switching from SIP to Skype..or not
On Thu, Mar 12, 2015 at 10:04:08AM -0400, Andres wrote: On 3/12/15 9:39 AM, Ron Wheeler wrote: Your characterization may be true but Skype works much better than SIP when it comes to sound quality. SIP is not to blame for this. Its the audio codec being used. Skype has spend a great deal of effort with their SILK codec by making it highly tolerant of packet loss and jitter. The same cannot be said for the standard codecs Asterisk uses. Opus was co-developed by Skype and could be used with Asterisk (if support to it was added). -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New Asterisk build
On Fri, Mar 06, 2015 at 11:33:55AM -0800, Ira wrote: Hello Asterisk, Back in 2009 I built a small Intel Atom based computer running Centos 5 for my asterisk system. 5 phones, 2 people 1 POTs line and six or so SIP numbers. So basically no load. I'm feeling like it's time to build another machine. It's probably silly, but it's been six years and I can't upgrade the OS which is falling behind. I'd likely just put it on a Raspberry Pi or something like that, but I need the one POTS line and all I have for that at the moment is a Digium card for a PCI slot. Are there any current thoughts on this? Very few ARM boards have anything close to PCI. Some of them: * nitrogen6x - A good board, well supported[1] but on the expensive range (~250$ (?)). Has a PCI-express slot. * Mira-box - has an internal mini-PCI slot. Not sure if it's good for you. Cheaper than the above, but still 150$ or so. If you look for a low-power ARM board, would you consider giving up the requirement of a PCI slot? Also note that many ARM boards don't have SATA. If you want to write to SD or to USB, please test performance of writing to see that Asterisk works well with it. Though quite a few (even of the cheaper ones) have SATA. [1] Raspbian uses those boards for their build servers. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New Asterisk build
On Fri, Mar 06, 2015 at 03:34:42PM -0500, John Novack SCII wrote: Find a HPT5720 with expansion chassis on eBay for under $50, load AstLinux ( instructions at AstLinux.org ) Move your Digium card and your confs , fix up any differences from your older version of Asterisk to the fairly current version 11 currently available with AstLinux. Use the GUI to edit and mage the system, as AstLinux has a somewhat different directory structure than you may be familiar with You should be up and running in a couple of hours, have a low power 20 watts, fanless flash based system that will just work in a real case. I would expect a decent ARM system to use roughly 2-5W. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] System() command refuses to execute bash script
Hi, Some notes, Don't run Asterisk as root. But also: On Mon, Mar 02, 2015 at 04:44:48PM +0200, Stefan Viljoen wrote: Hi all I got this solved. Turns out the script WAS executing, but I forgot that apparently you need to follow cron rules in any BASH scripts executed via System() from an Asterisk dialplan. E. g. all paths must be fully and absolutely specified, there are no relative path references available. So I changed the the file wireless.sh which was: #!/bin/bash touch wireless-executed to #!/bin/bash touch /root/wireless-executed A variant on: cd `dirname $0` can help in such cases. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] System() command refuses to execute bash script
On Mon, Mar 02, 2015 at 11:15:26AM -0500, Tech Support wrote: I'm surprised that you didn't have to specify the full path to the 'touch' command. When writing AGI scripts, I always do something like $touch = which( 'touch' ). I guess it's over kill. John You should generally not need a path to commands in /bin / /usr/bin . If 'which touch' returned it, it's in the path. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is Asterisk a Linux only system?
On Thu, Feb 12, 2015 at 10:38:33AM -0600, David M. Lee wrote: Unfortunately, I doubt the Python test suite would run on non-Linux. I don’t even bother trying to run it on Ubuntu; I have a CentOS VM specifically for running the test suite to avoid platform problems. I have a package of the test suite with some of its dependencies. In managed to make it into Jessie. It still misses quite a few things. It would be a good start running it there. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is Asterisk a Linux only system?
On Thu, Feb 12, 2015 at 09:25:39AM -0500, D'Arcy J.M. Cain wrote: I know that it runs on other systems but do other ports get the same attention? I have been running it on a NetBSD server for about a year now and while it mostly works it just crashes from time to time with no explanation or core dump. Use the option -g to get core dumps. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is Asterisk a Linux only system?
On Thu, Feb 12, 2015 at 03:09:02PM +, Justin Sherrill wrote: I would love to run Asterisk on a BSD system. I do not know of any developers actively working on Asterisk on a BSD platform, though my knowledge isn't comprehensive. That developer could be you. Run. Report bugs. Try troubleshooting. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] When are /proc/dahdi files created
On Wed, Feb 04, 2015 at 11:54:12PM +, Michelle Dupuis wrote: Can someone tell me when the /proc/dahdi files are created for spans? Are they created when asterisk starts (or the asterisk init script) - if not what script creates them? /proc/dahdi is created when dahdi is loaded. Each span N that is loaded creates /proc/dahdi/N . See the README: http://docs.tzafrir.org.il/dahdi-linux/#_procfs_interface_proc_dahdi and while we're at it: the following sections about the sysfs interface. /proc and /sys are generated by the kernel. No script generates them. A script may trigger the load of those modules. Those modules will be loaded typically in the dahdi init script, if not loaded earlier by hotplug (if the DAHDI hardware driver is not blacklisted). -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dahdi_genconf fails with Empty configuration - no spans
On Sat, Jan 17, 2015 at 09:31:33AM +0100, Bertrand LUPART - Linkeo.com wrote: However, dahdi_genconf keeps finding no span: What am i missing? It looks like your driver is loaded correctly. My guess would be maybe the dahdi-tools is packaged as an older version that doesn't know about the newer te435 card. You could hand craft the config file using the info in the card's manual http://www.digium.com/sites/digium/files/quad-span-digital-card-user-manual.pdf The error is no spans. What is the output of: dahdi_span_assignment list # /usr/sbin/dahdi_span_assignments list 1:-:-[1TE435F_-_DM06134700043_-_A_-_20131202] @PCI_Bus_10_Slot_01 /sys/devices/pci:00/:00:1c.0/:0a:00.0/pci::0a:00.0 2:-:-[1TE435F_-_DM06134700043_-_A_-_20131202] @PCI_Bus_10_Slot_01 /sys/devices/pci:00/:00:1c.0/:0a:00.0/pci::0a:00.0 3:-:-[1TE435F_-_DM06134700043_-_A_-_20131202] @PCI_Bus_10_Slot_01 /sys/devices/pci:00/:00:1c.0/:0a:00.0/pci::0a:00.0 4:-:-[1TE435F_-_DM06134700043_-_A_-_20131202] @PCI_Bus_10_Slot_01 /sys/devices/pci:00/:00:1c.0/:0a:00.0/pci::0a:00.0 Interesting. Does the files /etc/dahdi/assigned-spans.conf exist? Anyway, try: dahdi_span_assignments auto dahdi_genconf -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dahdi_genconf fails with Empty configuration - no spans
On Thu, Jan 15, 2015 at 12:58:26PM -0600, Russ Meyerriecks wrote: On Thu, Jan 15, 2015 at 2:05 AM, Bertrand LUPART - Linkeo.com bertrand.lup...@linkeo.com wrote: However, dahdi_genconf keeps finding no span: What am i missing? It looks like your driver is loaded correctly. My guess would be maybe the dahdi-tools is packaged as an older version that doesn't know about the newer te435 card. You could hand craft the config file using the info in the card's manual http://www.digium.com/sites/digium/files/quad-span-digital-card-user-manual.pdf The error is no spans. What is the output of: dahdi_span_assignment list -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] On Fedora, kernel update resets /var/run/asterisk owner to root.root
On Tue, Dec 02, 2014 at 04:00:25PM -0500, sean darcy wrote: On 12/02/2014 02:46 PM, Jeffrey Ollie wrote: On Tue, Dec 2, 2014 at 1:22 PM, sean darcy seandar...@gmail.com wrote: Or do I find a new place to put asterisk.pid? Also, if you use the native systemd unit file, you no longer need a PID file, although you still need /run/asterisk to store the control socket. So systemd is taking over the galaxy. No. /run has become a tmpfs directory. /var/run has been a tmpfs in Debian (tmpfs or deleted at startup) even before systemd. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Strange Issue: asterisk deleted
On Wed, Nov 26, 2014 at 10:37:49AM +, Antoine Megalla wrote: Hi, I am struggling with a very strange issue I have been facing for the past week;I have a fresh install of CENTOS 5.11 and I have installed asterisk 1.8.32 form sources.The asterisk installation went fine but as soon as I start asterisk executable it loads everything and then after the Ready line the process gets killed and when I try to run it again i get: /usr/sbin/asterisk : command not found I cleaned the source and re-installed asterisk and again the same thing happened again !!!I downloaded asterisk versions 1.4, 11, 12 and compiled them from sources and installed them (make install) and amazingly, the same thing happened to all of them: I do a make then make install and as soon as I start asterisk the process is killed and the executable removed from /usr/sbin. If you suspect that something is being run from the asterisk process or one of its children, run it under 'strace -f' and look for hints (e.g.: 'unlink') in the generated log. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] bristuff-0.4.0-RC4-xr7
On Mon, Oct 20, 2014 at 10:58:58PM +0100, Ray Image wrote: I am following the guide here: http://updates.xorcom.com/astribank/bristuff/1.4/bristuff-current/INSTALL.html Err... what would you do that? Bristuff (and anything Zaptel) is left for the record and historical value. But there's practically nothing useful there. I wouldn't waste my time on it. That said, for installing bristuff on a CentOS5.11 box (stock 2.6.18-398.el5.i686). Whilst building zaptel I get the error as follows: CC [M] /usr/src/bristuff-0.4.0-RC4-xr7/zaptel-1.4.12.9.svn.r4649/kernel/xpp/card_bri.o In file included from /usr/src/bristuff-0.4.0-RC4-xr7/zaptel-1.4.12.9.svn.r4649/kernel/xpp/xpd.h:26, from /usr/src/bristuff-0.4.0-RC4-xr7/zaptel-1.4.12.9.svn.r4649/kernel/xpp/card_bri.c:29: /usr/src/bristuff-0.4.0-RC4-xr7/zaptel-1.4.12.9.svn.r4649/kernel/xpp/xdefs.h:157: error: conflicting types for ‘bool’ include/linux/types.h:36: error: previous declaration of ‘bool’ was here In file included from Yeah, see how this is resolved in later versions of DAHDI. Or just remove that typedef of bool. If it's provided by the kernel, DAHDI doesn't need to provide it. /usr/src/bristuff-0.4.0-RC4-xr7/zaptel-1.4.12.9.svn.r4649/kernel/xpp/xpd.h:31, from /usr/src/bristuff-0.4.0-RC4-xr7/zaptel-1.4.12.9.svn.r4649/kernel/xpp/card_bri.c:29: include/linux/device.h:407: error: expected identifier or ‘(’ before ‘const’ make[4]: *** [/usr/src/bristuff-0.4.0-RC4-xr7/zaptel-1.4.12.9.svn.r4649/kernel/xpp/card_bri.o] Error 1 make[3]: *** [/usr/src/bristuff-0.4.0-RC4-xr7/zaptel-1.4.12.9.svn.r4649/kernel/xpp] Error 2 make[2]: *** [_module_/usr/src/bristuff-0.4.0-RC4-xr7/zaptel-1.4.12.9.svn.r4649/kernel] Error 2 make[2]: Leaving directory `/usr/src/kernels/2.6.18-398.el5-i686' make[1]: *** [modules] Error 2 make[1]: Leaving directory `/usr/src/bristuff-0.4.0-RC4-xr7/zaptel-1.4.12.9.svn.r4649' make: *** [all] Error 2 Can anyone help please? Thanks in advance. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PBX hacked: why hundred of calls to the same number ?
On Thu, Oct 02, 2014 at 07:52:34AM +0200, Rainer Piper wrote: Is the destination Number like Country Code +972? +972 59 xx(x) mobile - Jawall [moving to 7-digit subscriber numbers] source - http://www.wtng.info/wtng-972-il.html That page is slightly dated. +972 59 XXX are all the numbers in the Palestinian Authority (there are several providers besides Jawall). My SIP Proxy logs all the unauth. INVITEs and I found the a lot calls go to the Country code +972 xxx As a resident of +972 (+972-4), I'll just note that those hack attempts are typically related to PA numbers (+972-59) as rates there are higher. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dahdi problem with dahdi_genconf
On Mon, Sep 29, 2014 at 04:03:52PM +0200, Claudio ML wrote: Il 29/09/2014 15:57, Tzafrir Cohen ha scritto: On Mon, Sep 29, 2014 at 03:52:25PM +0200, Claudio ML wrote: Hi, It's the first time i try to configure an ISDN card with dahdi, so my experience is very poor (be kind ;)) My problem is with dahdi_genconf, when i start it it says: /usr/sbin/dahdi_span_assignments: Missing '/sys/bus/dahdi_devices/devices' (DAHDI driver unloaded?) What version of the DAHDI drivers is loaded? cat /sys/module/dahdi/version cat /sys/module/dahdi/version 2.5.0.1 Try running: dahdi_genconf system chandahdi # instead -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dahdi problem with dahdi_genconf
On Mon, Sep 29, 2014 at 03:52:25PM +0200, Claudio ML wrote: Hi, It's the first time i try to configure an ISDN card with dahdi, so my experience is very poor (be kind ;)) My problem is with dahdi_genconf, when i start it it says: /usr/sbin/dahdi_span_assignments: Missing '/sys/bus/dahdi_devices/devices' (DAHDI driver unloaded?) What version of the DAHDI drivers is loaded? cat /sys/module/dahdi/version -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ubuntu 14.04 LTS Asterisk and ISDN Cologne Chip
On Tue, Sep 23, 2014 at 11:30:41AM +0200, Claudio ML wrote: Hi to all, I am searching to make work an Asterisk, with an ISDN card with Cologne Chipset. Here is the lspci: 01:09.0 Network controller: Cologne Chip Designs GmbH ISDN network controller [HFC-PCI] (rev 02) Subsystem: Cologne Chip Designs GmbH ISDN Board Flags: bus master, medium devsel, latency 16, IRQ 5 I/O ports at c400 [size=8] Memory at fdefd000 (32-bit, non-prefetchable) [size=256] Capabilities: [40] Power Management version 1 I think i need the zaphfc module to make it work, but it is not included into the dahdi package, and i cant find it. The command dahdi_hardware says this: dahdi_hardware pci::01:09.0 zaphfc- 1397:2bd0 HFC-S ISDN BRI card But, how i can install the zaphfc module? The Debian (and Ubuntu) packages of dahdi-source / dahdi-dkms should include zaphfc. I'm not really sure how well it works. The code is maintained in a forked git tree of external drivers which I sort-of maintain (that is: make sure that they build when someone complains. I can't really test them). I'd love to see someone else properly maintaining that (or better: getting those drivers merged in the tree. But that's off-topic for this thread). https://gitorious.org/dahdi-extra/ -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dahdi CAPI migration
On Fri, Aug 08, 2014 at 03:34:44PM +0200, Patrick Laimbock wrote: On 08-08-14 10:09, Toney Mareo wrote: Hello Thank you for your response. I thought it could be easier moving the old card to the new machine and using the DAHDI driver. Unfortunately my first attempt for this failed. The card shows up in the original machine as: dahdi_hardware -v pci::00:00.0 wcb4xxp+ 1397:08b4 Junghanns QuadBRI ISDN card IIRC the wcb4xx module is correct for this card. Not only that, but the '+' means that this module handles this specific hardware. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] compiling dahdi and exporting it to another system
On Wed, Jul 30, 2014 at 10:29:19PM +0200, Anthony Azzopardi wrote: Hello asterisk-users, I need to compile dahdi and then export it to another system. I managed to do this with DESTDIR=/root/destDir, then make a tar file and extract in / of the other system. However the module is not loading and /dev/dahdi is not created. Are the modules available? What is the output of: modinfo dahdi If not available: On the target system, what is the output of: find /lib/modules/`uname -r` -name dahdi.ko If you find dahdi.ko: maybe you forgot to run depmod? If you don't find it: maybe you built it for an incorrect kernel version? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk in debian Wheezy 1.8.13.1 vs. Squeeze 1.8.23.1
On Wed, Jul 02, 2014 at 10:05:44PM +0200, Thomas wrote: Hello, in Squeeze Asterisk 1.8.23.1 is installed, Self-installed in Wheezy older version 1.8.13.1~dfsg1-3+deb7u3. From a package. With version 1.8.13.1 I have some problems so I would like to install version 1.8.23.1 used in Squeeze whats running fine for me. How I can do this? Install from source as in Squeeze? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RPM updates
On Mon, Jan 28, 2013 at 05:21:10PM +0200, Tzafrir Cohen wrote: On Mon, Jan 28, 2013 at 01:55:09PM +, Steven Howes wrote: Hi All, Who do I need to poke to get the yum repository / RPM files updated? The dahdi RPMs are not up to date with the CentOS kernel versions any more, it's making doing an installation a bit tricky due to dependancies, I'd rather not roll back / remove new kernels if I don't have to.. See also: http://git.tzafrir.org.il/?p=rpm/dahdi-linux.git;a=summary http://git.tzafrir.org.il/?p=rpm/dahdi-tools.git;a=summary A new set of package repositories is now available under http://git.xorcom.com/ , or specifically: http://git.xorcom.com/rpm/ . The packages there are in initial stages of packaging and thus not yet published as a repository. I built them using git-buildpackage-rpm, see: http://git.xorcom.com/?p=rpm/tools.git;a=blob;f=README.txt -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] issue installing voicemail imap support: imap_tk module missing
On Thu, Jun 12, 2014 at 10:45:37AM +0200, Bart Remmerie wrote: I'm using * ubuntu 14.04 LTS * asterisk 11.10 (from source) * imap-2007f (from source) Any reason you don't install uw-imap from ubuntu? libc-client2007e-dev or libc-client-dev . -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] issue installing voicemail imap support: imap_tk module missing
On Tue, Jun 03, 2014 at 10:26:26PM +0200, Bart Remmerie wrote: Does anybody know where imap_tk is supposed to be / where it comes from ? Is it a part of asterisk / imap / linux / ... I can't seem to find any references other than related to asterisk, but in asterisk I only can find it as a (unfortunately missing) dependency for imap support for voicemail... What distribution is it? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Terrible dahdi_test results
On Thu, May 15, 2014 at 12:28:44PM -0300, Mike Leddy wrote: Hi Russ, I rebooted the machine loading dahdi_dummy in /etc/modules before the /etc/init.d/dahdi. Unless you're using a relatively old version of dahdi, there's no separate module called dahdi_dummy. It is an alias to the main dahdi module (which you must have loaded, as all the card drivers depend on). -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk -rx, how expensive is it? Should you avoid spamming it?
On Fri, Apr 25, 2014 at 11:48:42AM +0100, A J Stiles wrote: On Thursday 24 Apr 2014, Mikael Fredin wrote: I will look into netcat as well, thank you There's not much to look into, really! It's just a command-line tool for connecting STDIN and STDOUT to a network socket. $ echo -e WIBBLE\nWIBBLE\nWIBBLE | nc somehost.co.uk 3245 will send WIBBLE WIBBLE WIBBLE to port 3245 of host somehost.co.uk , and display any response coming back on STDOUT. It really is that simple! Another example: $ echo -e USER fred\nPASS b00bies\nQUIT | nc pop3.myisp.co.uk 110 will quickly login to your POP3 server and then logout without retrieving any messages; this can be useful if you need to make a POP3 connection before you can send mail using SMTP. Which is a bit irrelevant if you want to connect to Asterisk through the unix-domain socket [/var]/run/asterisk/asterisk.ctl . -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk -rx, how expensive is it? Should you avoid spamming it?
On Thu, Apr 24, 2014 at 12:20:37PM +0200, Mikael Fredin wrote: Just like the subject sais - how expensive is it to execute a lot of these commands to keep track of different things in asterisk? I have avoided doing this because it feels a bit like a risk to spam the asterisk CLI this way, but is it really? CPU-wise it doesn't seem very expensive to do it 100 times a second (from a simple test I did), but is it possible it will affect the asterisk service in any other negative way? It feels very expensive. Part of it is because of starting a new instance of Asterisk. It will not load any module and such, but if you care about speed, you can use netcat (it takes some care). You'll also encounter some artificial delays in the response which make it feel more expensive. The main reason to avoid it is because its output is not intended for automated parsing. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk on OSX
The solution is obvious: On Fri, Apr 11, 2014 at 09:01:36AM +0200, Manu wrote: Hi, I used asterisk on Debian7 and it was good experience. Now, i'm using osx on mac mini. I'd like to install asterisk 12. Install Debian 7.0 on the system and proceed as before :-) -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] is g729 codec free? or under license???
On Wed, Apr 09, 2014 at 10:19:59AM +0800, Steve Underwood wrote: Hi Jeff, On 04/08/2014 12:13 PM, Jeff Brower wrote: Darrel- The G729 essential patents were *granted* in 1996, but applied for prior to June 8 1995. That means their lifespan is either 20 years from their application date, or 17 years from their grant date, whichever is greater (http://www.uspto.gov/main/faq/p120013.htm). Either way, they expire in 2014. -Jeff Where did you get the cutoff date of June 8 1995, and how does 20 years from that date lead to the last of the patents expiring in 2014? Nobody uses G.729. They use G.729A. The G.729A spec is somewhat later than the original G.729, but I don't know if there are any additional patents which specifically relate to Annex A. You could use G.729 instead, but it roughly doubles the compute needed. If it allows me to avoid the trolls: I'll pay that performance hit. In many caces there are CPU cycles to spare. But the licensing is a hard limit. There are various things on the web saying the last of the patents on G.723.1, which was around in draft form long before G.729, expires in 2014. However, there seem to be patents related to that codec which don't really expire until some time in 2015. Its really hard to find solid information. The ITU patent database rarely identifies the actual patents being claimed, so its damned hard to look them up. Nice. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unable to build DAHDI-Linux in mock chroot
On Fri, Mar 28, 2014 at 07:57:54PM -0500, Anthony Messina wrote: On Friday, March 28, 2014 07:43:48 PM Anthony Messina wrote: Unfortunately, after http://git.asterisk.org/gitweb/?p=dahdi/linux.git;a=commitdiff;h=6cebc1c1fb1 2cc0661f3810ef47ad33206b2e398 I am unable to build DAHDI-Linux in a mock chroot for packaging purposes. I believe this is related to the Makefile calling install_firmware with only 2 args, where install_firmware is a shell script with DESTDIR set to $3, which is empty. In this case, the DESTDIR evaluates to /usr/lib/hotplug/firmware, rather than buildroot_destdir/usr/lib/hotplug/firmware. make -C drivers/dahdi/firmware hotplug-install DESTDIR=/builddir/build/BUILDROOT/dahdi-linux-2.9.1-1.fc20.x86_64 HOTPLUG_FIRMWARE=yes make[1]: Entering directory `/builddir/build/BUILD/dahdi- linux-2.9.1/drivers/dahdi/firmware' mkdir -p /builddir/build/BUILDROOT/dahdi- linux-2.9.1-1.fc20.x86_64/usr/lib/hotplug/firmware mkdir -p /builddir/build/BUILDROOT/dahdi- linux-2.9.1-1.fc20.x86_64/lib/firmware Installing dahdi-fw-oct6114-032.bin to hotplug firmware directories install: cannot create regular file '/usr/lib/hotplug/firmware': No such file or directory make[1]: *** [hotplug-install] Error 1 make[1]: Leaving directory `/builddir/build/BUILD/dahdi- linux-2.9.1/drivers/dahdi/firmware' make: *** [install-firmware] Error 2 https://issues.asterisk.org/jira/browse/DAHLIN-337 Thanks for your report. I hope to get it fixed soon. I should note that this specific target does not belong in a proper chroot build, as it downloads from outside. How can I get those firmware files properly included? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk CLI Banner
On Sat, Mar 29, 2014 at 02:52:24PM -0400, Paul Belanger wrote:
On Fri, Mar 28, 2014 at 2:39 PM, Steve Edwards
asterisk@sedwards.com wrote:
On Fri, 28 Mar 2014, Richard Kenner wrote:
And this certainly may vary from jurisdiction to jurisdiction. For a
(quite dated at this point) discussion of this issue from a US perspective,
see
http://www.law.berkeley.edu/php-programs/faculty/facultyPubsPDF.php?facID=346pubID=157
The publication (43 pages) is dated 1988. The DMCA (1998) and subsequent
legislation may have changed the landscape.
My (ignorant) opinion -- just don't. Is it worth the effort to research? Is
it worth paying a lawyer to research it and give an opinion that may be
worth nothing until it is examined in court?
If you want to display something custom, how about a 'wrapper' script that
displays a file using 'curl' before handing off to Asterisk -- easier to
implement, easier to maintain, no legal BS to consider.
Or can you express your creativity by fiddling with ASTERISK_PROMPT?
If you really want to do it:
1) create a wrapper to asterisk -r
Or use cotrib/scripts/astcli (which uses the manager interface).
2) pipe the welcome message to /dev/null
3) ???
4) profit
you didn't modify Asterisk.
And you still have 'core show version' and 'core show
{license|copyright}'.
#include std/ianal.h
#include std/tinala.h
The license does not require you to show that specific notice. But it
requires you to show the copyright notice in the place where the user
would expect it. The users MUST be aware of their legal right. In a GUI
program, you'd put that notice in help = about and not necessarily in
the startup banner (even if you have one).
That said, I would not go changing it unless it had caused a real
technical difficulty.
--
Tzafrir Cohen
icq#16849755 jabber:tzafrir.co...@xorcom.com
+972-50-7952406 mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Spammer direct replying to those posting on the users list
[Intentionally ignoring the Reply-to header in this reply. And yes, this is on-list] On Tue, Mar 25, 2014 at 03:15:22PM +, A J Stiles wrote: On Tuesday 25 Mar 2014, Digium's Asterisk Development Team wrote: We apparently have a spam bot subscribed to the list and replying *directly* to anyone who posts on the list. The e-mail address I use for this mailing list is asterisk_l...@earthshod.co.uk ; so I used the following procmail recipe. This filters out anything being sent to that address *without* a Received: header mentioning lists.digium.com: :0 * ^To.*asterisk_list * !^Received.*lists.digium.com asterisk_unwanted What if I wanted to reply to one of your messages off-list? My message would end up in asterisk_unwanted. (when I am satisfied that it does not lose anything legitimate, I probably will change the last line to /dev/null .) Or even worse. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] gsm codec compile
On Mon, Mar 03, 2014 at 11:13:52AM -0800, Doug wrote: I was successful in compiling asterisk in raspbien except for the following error If I enable the gsm codec. It appears there is something in the Makefile n this directory that needs to be changed. Probably involving optimization. Not sure why it does not recognize the processor since it is one that is mentioned in the Makefile. Any help would be appreciated. make[2]: Entering directory `/usr/src/asterisk/codecs/gsm' [AS] src/k6opt.s - src/k6opt.o Assembler messages: Error: unknown architecture `armv6l' Error: unrecognized option -march=armv6l make[2]: *** [src/k6opt.o] Error 1 For the sake of the archives: Just use the system libgsm (make sure you have libgsm-dev installed). No point in using the bundled version. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Running configure from subdirectory of source tree
Hi Again, answering a bit late, On Wed, Mar 05, 2014 at 09:24:11PM +0100, Gianluca Merlo wrote: 2014-03-05 20:08 GMT+01:00 Jason Parker jason.par...@schmoozecom.com: 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. Hello Jason, I admit I naively have yet to consider whether the actual build could work or not, and assumed (for lack of knowledge) it could. By chance, is yours an assumption or you have personally experienced (or know by understanding its inner workings) that the top level make does not perform correctly if the sources lie elsewhere (in this case, in the parent directory)? Is there some reason you think you need to run the other configure scripts yourself? If your question is related to the origin of the need of configuring and building in a subdirectory, I am in need of building custom Debian packages for Asterisk, and my plan was to use a single source package to build several monolithic Asterisk binary packages, each with different configure options and patches to modify/add some functionalities (I currently compile and install them manually, and could use some Debian-magic-automation). Why is that, BTW? Debhelper offers a --builddirectory option which allows to easily perform such task, and I already successfully used it on more simple software using the autotool build toolchain, but essentially this works as in my manual example, thus failing. Thanks in advance for your advice. Does the rest of the build system support this? Anyway, the obvious workaround is (cd ..; ./configure) . If you can't use shell and must use an explicit command, add a wrapper script: #!/bin/sh cd .. exec ./configure $@ -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Not Starting after YUM Update
On Wed, Feb 12, 2014 at 10:44:42PM +0100, Aldo Bergamini wrote: Hi List, it feels silly, but here I am. My Asterisk box is useless, after running a long delayed yum update (Centos box). [snip] Starting Asterisk very verbosely seems to load the dialplan, but at some point I get a segmentation fault. This is new to me! […] edited […] chan_agent.so = (Agent Proxy Channel) == Registered custom function 'EXTENSION_STATE' func_extstate.so = (Gets an extension's state in the dialplan) == Registered application 'DAHDIBarge' app_dahdibarge.so = (Barge in on DAHDI channel application) == Registered custom function 'CALLERPRES' == Registered custom function 'CALLERID' func_callerid.so = (Caller ID related dialplan functions) [2014-02-12 22:40:07] NOTICE[13842]: codec_g729a.c:760 load_module: G.729A transcoding module version 1.6.0_3.1.4, Copyright (C) 1999-2009 Digium, Inc. [2014-02-12 22:40:07] NOTICE[13842]: codec_g729a.c:761 load_module: This module is supplied under a commercial license granted by Digium, Inc. [2014-02-12 22:40:07] NOTICE[13842]: codec_g729a.c:762 load_module: Please see the full license text supplied by the accompanying [2014-02-12 22:40:07] NOTICE[13842]: codec_g729a.c:763 load_module: register utility, or ask for a copy from Digium. Segmentation fault The problem seems to come after the callerid module loads: does this make sense? BTW: I do have a G729 pack of licenses (they were actually active without any problem before messing with the update).. What should the clever sysadmin do? Thanks in advance, Aldo Try: # standard asterisk command-line. No verbosity strace -eopen asterisk -U asterisk -c See which module was the one last loaded. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] What is dahdi.auto_assigned_spans and why should you care? (II)
. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] What is dahdi.auto_assigned_spans and why should you care?
-spans.conf
and then connect the new device. But maybe you did not remember to do
this in advance, or you're not sure what the ID is without connecting
the new device.
In that case:
1. connect the new device.
2. either:
dahdi_span_assignments auto
dahdi_genconf span
or, if you have anything special thereand want keep it:
sed -i -e 's/old-identifier/new-identifier/' /etc/dahdi/assigned-spans.conf
dahdi_span_assignments add
The configuration is already in place, so everything should work from
here.
Change Span and Channel Numbers
~~~
Naturally you're not limited to the above scenarios. What if you just
feel like using different channel and span numbers? All you need is:
dahdi_assigned_spans remove
vi /etc/dahdi/assigned-spans.conf
dahdi_assigned_spans add
dahdi_genconf # likely to get run here, as you changed span and channel
# numbers.
But you do need a proper text editor. And see (3) and (4) above.
Example: if you have two devices. you want to reverse the order in which
they show.
The configuration file I have is:
# Device: [usb:000156] @
# /sys/devices/pci:00/:00:10.4/usb1/1-1/xbus-00/astribanks:xbus-00
usb:000156 1:1:1
usb:000156 2:2:32
usb:000156 3:3:63
usb:000156 4:4:94
usb:000156 5:5:125
usb:000156 6:6:133
usb:000156 7:7:141
# Device: [usb:INT05668] @
# /sys/devices/pci:00/:00:10.4/usb1/1-3/xbus-01/astribanks:xbus-01
usb:INT05668 1:8:149
usb:INT05668 2:9:152
usb:INT05668 3:10:155
usb:INT05668 4:11:158
usb:INT05668 5:12:161
usb:INT05668 6:13:164
usb:INT05668 7:14:167
usb:INT05668 8:15:170
##
As you can see here, this system has two devices, each with several
spans. The device usb:000156 has 7 spans, of which 4 are E1 ports (a
range of 31 numbers[U]) and the other three have only 8 ports. The other
device, identified as usb:INT05668, has eight spans of its own. Their
numbers start from 8, and the channel numbers start from 149. Spans are
are BRIs (3 channels per span).
The long list of channels of the E1 spans is annoying, and I'd like it
to appear after the BRI ports.
Here's the first shot: move the span numbers:
vi () { perl -pi -e 'if (/^usb:000156/) {s/:([0-9]+):/:.($1+30).:/e}' $@;
}
This adds 30 to the span number of the first device. Thus they appear
first on lsdahdi. But the listing is Asterisk is by channel numbers.
Let's add 300 to its channel numbers:
vi () { perl -pi -e 'if (/^usb:000156/) {s/:([0-9]+)$/:.($1+300)/e}' $@; }
And now let's have both:
vi () { perl -pi -e 'if (/^usb:000156/) {s/:([0-9]+)$/:.($1+300)/e;
s/:([0-9]+):/:.($1+30).:/e}' $@; }
Note that it's easy to change the span numberes, changing channel
numbers is slightly trickier: each span has to have the required number
of channels.
Get vi Back
^^^
1. run the above in a subshell. Or:
1.1. unset vi
[Q] Reminder: there is a kernel-level module called DAHDI, and Asterisk
module called chan_dahdi.
[W] A span is logical groups of channels: a single digital port, all the
analog ports on a card, or whatever. Spans and channels are identified
by their numbers.
[E] http://docs.tzafrir.org.il/dahdi-linux/#_devices_bus
[R] http://docs.tzafrir.org.il/dahdi-linux/#_span_assignments
[T] http://docs.tzafrir.org.il/dahdi-tools/#_implicit
[Y] There are interesting things you could do if it were possible to
configure Asterisk more dynamically.
[U] Note, however, The fact that a port has a range of 31 channels does
not necessarily mean it has to be E1. Maybe a smaller span uses that
range.
--
Tzafrir Cohen
icq#16849755 jabber:tzafrir.co...@xorcom.com
+972-50-7952406 mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Starpy and Asterisk on different machines ? [SOLVED]
On Thu, Jan 16, 2014 at 04:58:14PM +0100, Olivier wrote: Thanks for replying. So as python-starpy requires asterisk in Debian Wheezy repo, for a Debian setup the alternatives are either : - to install it from source - tto build my own custom package removing this asterisk dependency (is it easy or even possible ?) Should be simple. - to use another solution such as pyst. - To provide Asterisk by a dummy package such as one built by equivs. See, e.g. https://wiki.debian.org/CreateDummyPackage -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to get Asterisk acting like a Multi-thread application?
On Sun, Jan 05, 2014 at 11:22:44AM +, s...@yahoo.com wrote: Hello All, By default, Astersik uses only one thread as processing jobs, Now I need to know how can it will be possible to configuring asterisk as a Multi-thread application? Asterisk is already multi-threaded. Asterisk generally uses one thread per channel with quite a few other helper threads. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to get Asterisk acting like a Multi-thread application?
On Sun, Jan 05, 2014 at 12:20:37PM +, s...@yahoo.com wrote: Hello, thanks for your reply. So if I have 10 active channels I have 10 threads ? How can I get the number of threads that Asterisk is now using or running under? On my system there are no active channels[1]. I have: # pidof asterisk 14386 # ls /proc/14386/task 14386 14391 14395 14399 14404 14409 14413 14417 14421 14425 14429 14388 14392 14396 14400 14406 14410 14414 14418 14422 14426 14430 14389 14393 14397 14402 14407 14411 14415 14419 14423 14427 14431 14390 14394 14398 14403 14408 14412 14416 14420 14424 14428 14432 And of coursse: asterisk -rx 'core show threads' [1] but it's still good enough to get Wesnoth hung: http://bugs.debian.org/734124 -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ctrl-W killing entire line, not just last word
On Wed, Dec 18, 2013 at 10:20:12PM +, dotnetdub wrote: 1.4 1.6 1.8 11.6.0 All compiled and all running on debian 6 or 7 In 1.4 1.6 1.8 there was no option to build with an external libedit. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk on Windows
On Tue, Dec 10, 2013 at 08:43:01PM -0500, Brian wrote: On Tue, 10 Dec 2013 23:02:45 +0200 Tzafrir Cohen tzafrir.co...@xorcom.com wrote: On Wed, Dec 04, 2013 at 02:12:41PM -0500, Ruddy Gbaguidi wrote: I never tought this is become a Linux vs Windows fight. We have been using asterisk on linux from a long time now and happy with it. But some of our customers who has windows in their environment want to use our call center software we developed on top of asterisk. So, the question was : Did anybody ever tried to isolate the asterisk SIP server/module and make it run under Windows ? Since, asterisk 12 is using pjsip (which is cross platform already), I tought it may be possible and wanted advices. I would love that every single customer switch to Linux and Ubuntu tomorrow morning but at the moment, that's not the case. There was an old half-working port of Asterisk to Cygwin which does run on Windows. It has not worked since at least 1.6.0 . That's just a unix-like interface which won't address the issues the OP has/had with running/configuring asterisk. IMHO it would probably be even more challenging. And IIRC the OP was looking for a non emulated solution anyway. It's not emulated. It uses a compatibility layer library. If emulation were such a major issue for you, I guess you'd never consider using a language such as Java, Perl or Python, where the program runs in its own virtual machine. And for others: the name is [MS-]Windows. Not 'wind-blows or whatever name you find for it. Please respect this list. If you don't have anything useful to add to the thread, please refrain from replying. I have to agree with the name calling part but the OP did imply that Windows was superior and that a Windows port would be profitable. You can't really expect to get away with that on a list devoted to an open source application without making a complete fool out of yourself. If it was a post regarding one of the many proprietary closed source applications/games without a native port to Linux/BSD/OS X then it would be a valid complaint. Having access to the source as well as liberal licensing terms which allow porting isn't a valid complaint and never will be. I did not reply to that troll. I replied an OP who said that MS-Windows was a requirement for his case. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dahdi show channels no such command
On Wed, Dec 11, 2013 at 09:22:13AM -0800, Joseph Towery wrote: Here is the output of lsdahdi: asteriskpbx@pbxmain:~$ sudo lsdahdi ### Span 1: WCTDM/0 Wildcard TDM800P (MASTER) 1 FXO FXSKS (EC: MG2 - INACTIVE) 2 FXO FXSKS (EC: MG2 - INACTIVE) RED 3 FXO FXSKS (EC: MG2 - INACTIVE) RED 4 FXO FXSKS (EC: MG2 - INACTIVE) RED 5 FXS FXOKS (EC: MG2 - INACTIVE) 6 FXS FXOKS (EC: MG2 - INACTIVE) 7 FXS FXOKS (EC: MG2 - INACTIVE) 8 FXS FXOKS (EC: MG2 - INACTIVE) For the record: this output shows that the DAHDI channels have been configured (they got the FXSKS signalling, which means it was applied by running dahdi_cfg) but they are not '(in use)', which means that no channel has been opened by Asterisk's chan_dahdi. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ctrl-W killing entire line, not just last word
On Wed, Dec 11, 2013 at 09:22:54PM +0200, Dotan Cohen wrote: Interestingly, I don't even seem to have editline installed and thus Asterisk is likely using its own packaged editline: You need libedit-dev, not libeditline-dev. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dahdi show channels no such command
On Wed, Dec 11, 2013 at 12:09:17PM -0800, Joseph Towery wrote: From: Tzafrir Cohen tzafrir.co...@xorcom.com To: asterisk-users@lists.digium.com Sent: Wednesday, December 11, 2013 1:37 PM Subject: Re: [asterisk-users] dahdi show channels no such command On Wed, Dec 11, 2013 at 09:22:13AM -0800, Joseph Towery wrote: Here is the output of lsdahdi: asteriskpbx@pbxmain:~$ sudo lsdahdi ### Span 1: WCTDM/0 Wildcard TDM800P (MASTER) 1 FXO FXSKS (EC: MG2 - INACTIVE) 2 FXO FXSKS (EC: MG2 - INACTIVE) RED 3 FXO FXSKS (EC: MG2 - INACTIVE) RED 4 FXO FXSKS (EC: MG2 - INACTIVE) RED 5 FXS FXOKS (EC: MG2 - INACTIVE) 6 FXS FXOKS (EC: MG2 - INACTIVE) 7 FXS FXOKS (EC: MG2 - INACTIVE) 8 FXS FXOKS (EC: MG2 - INACTIVE) For the record: this output shows that the DAHDI channels have been configured (they got the FXSKS signalling, which means it was applied by running dahdi_cfg) but they are not '(in use)', which means that no channel has been opened by Asterisk's chan_dahdi. Okay, I did run dahdi_cfg in an attempt to get the show channels working. Is this a problem? Certainly not. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk on Windows
On Wed, Dec 04, 2013 at 02:12:41PM -0500, Ruddy Gbaguidi wrote: I never tought this is become a Linux vs Windows fight. We have been using asterisk on linux from a long time now and happy with it. But some of our customers who has windows in their environment want to use our call center software we developed on top of asterisk. So, the question was : Did anybody ever tried to isolate the asterisk SIP server/module and make it run under Windows ? Since, asterisk 12 is using pjsip (which is cross platform already), I tought it may be possible and wanted advices. I would love that every single customer switch to Linux and Ubuntu tomorrow morning but at the moment, that's not the case. There was an old half-working port of Asterisk to Cygwin which does run on Windows. It has not worked since at least 1.6.0 . Feel free to try to fix it. I suspect it won't be easy. Patches would be welcomed, I guess (look at what odd fixes that were accepted to make Asterisk build and work on OS/X). And for others: the name is [MS-]Windows. Not 'wind-blows or whatever name you find for it. Please respect this list. If you don't have anything useful to add to the thread, please refrain from replying. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dahdi show channels no such command
On Thu, Dec 05, 2013 at 07:19:56PM -0800, Joseph Towery wrote: Here are the commands asteriskpbx@pbxmain:/etc/dahdi$ dahdi_cfg -vvv DAHDI Tools Version - 2.7.0.2 DAHDI Version: 2.7.0.1 Echo Canceller(s): MG2 Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 02) Channel 03: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 03) Channel 04: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 04) Channel 05: FXO Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 05) Channel 06: FXO Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 06) Channel 07: FXO Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 07) Channel 08: FXO Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 08) 8 channels to configure. Setting echocan for channel 1 to mg2 Setting echocan for channel 2 to mg2 Setting echocan for channel 3 to mg2 Setting echocan for channel 4 to mg2 Setting echocan for channel 5 to mg2 Setting echocan for channel 6 to mg2 Setting echocan for channel 7 to mg2 Setting echocan for channel 8 to mg2 asteriskpbx@pbxmain:/etc/dahdi$ pbxmain*CLI module show like chan_dahdi.so Module Description Use Count chan_dahdi.so DAHDI Telephony Driver w/PRI 0 1 modules loaded What's the output of lsdahdi? Note channels there that are open (by Asterisk). Try running in the asterisk CLI: module unload chan_dahdi.so module load chan_dahdi.so The 'load' command will probably fail and give you the real error. Though you should probably already have it in the logs. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] pipeast [was: Re: How to repeat pri show span and zap show channel commands]
Hi, Long ago, On Wed, Feb 21, 2007 at 09:32:26AM +0200, Tzafrir Cohen wrote: On Wed, Feb 21, 2007 at 07:56:18AM +0100, Olivier wrote: [snip] Any better idea than running an dedicated shell script with a asterisk -rx pri show span 1 line ? Here is my pipeast script: (while read line; do echo -n $line # trim the ending newline sleep 0.001 # generate different write(2)-s done) | socat - /var/run/asterisk/asterisk.ctl You can pipe whatever you want to that. e.g: while true; do echo pri show span 1 sleep 30 done | pipeast Note: socat is available as a debian package: http://packages.debian.org/socat . It has lately come to my attention that this method is broken (and has been, for quite some time. Sure enough. It does not use any published inteerface) and that there is actually some demand for using it. A version of pipeast that seems to work now is: #!/bin/sh while read line; do echo $line | tr '\n' '\0' sleep 0.001 done | nc -U /var/run/asterisk/asterisk.ctl Notes: 1. nc, or at least openbsd-netcat, can use unix domain sockets (-U). No need for socat. 2. that /var/run is slowly converting to /run , though the compatibility link /var/run will likely stay for quite some time. 3. If this script breaks in the future, you get to keep both pieces. But this is only guaranteed if it breaks to exactly two pieces. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Read Telnet Packet
On Fri, Oct 11, 2013 at 11:34:48AM +0530, akhilesh chand wrote: Dear All, I want to read telnet packet continuously whenever a new call is originated and store into a variable after that pass into window server. I have written a Perl script to read telnet packet but problem is that whenever I executed Perl script then got a telnet packet( mean Only when i execute Perl script) here I want to put scheduler,event or other technique whenever a new call will come Perl script automatically run. You can use a packet sniffer such as tcpdump or wireshark instead. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dahdi gains
On Tue, Aug 27, 2013 at 10:08:24AM +0200, Sil wrote: Hi, I'm trying to find the differences between the two CLI gain parameters of Dahdi : dahdi set swgain and dahdi set hwgain. When I change one of these parameters the output of : asterisk -rx dahdi show channel X | grep Gains don't show me any changes. Did dahdi show channel X shows HW or SW Gains ? Whan I set rxgain and txgain in my chan_dahdi.conf file, is it a HW or SW gain ? See http://lists.digium.com/pipermail/asterisk-dev/2013-August/062219.html -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 'n Dahdi on Sun Solaris
On Wed, Jun 12, 2013 at 12:32:40PM +0530, Chandrakant Solanki wrote: Hello All, I am trying to install Asterisk 1.8.13.0 dahdi-complete 2.5.1 libpri 1.4.13 version. Is it possible to install dahdi on Sun Solaris? I have searched so many, but don't found any help. Maybe. But dahdi-complete you're trying to install includes dahdi-linux which is drivers for Linux. What do you need DAHDI for? I am using SunOS solaris-server 5.11 11.1 i86pc i386 i86pc on Virtual Box. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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] Where is HAVE_NEWLOCALE set?
On Mon, Jun 10, 2013 at 04:06:27PM -0400, D'Arcy J.M. Cain wrote: I am trying to build Asterisk on a NetBSD system but I am running into two problems. The first only happens on an installation built from NetBSD HEAD. The config variable HAVE_NEWLOCALE is erroneously set during configure but this system does not have newlocale(). I can't seem to find where this gets set to true. What version of asterisk is it? It is set by the configure script (look for 'newlocale' in configure.ac). Interestingly a stable release of NetBSD does not have this issue although it still has the second issue which I will start a separate thread for. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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] DTLSv1_method on NetBSD
On Mon, Jun 10, 2013 at 04:10:23PM -0400, D'Arcy J.M. Cain wrote: This is the second issue I found while trying to install Asterisk on a NetBSD box. I can't load the rtp module because HAVE_OPENSSL_SRTP seems to be set. Is there some way to simply force this variab;e to be unset from a configuration variable? Do you have OpenSSL? Again, what version of Asterisk? What version of NetBSD? People may read this in the future, so please provide a version number rather than stable. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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] Sample config files installed to /etc
On Fri, Jun 07, 2013 at 11:53:57AM +0200, Daniel Pocock wrote: The sample config files in the Asterisk distribution and packages are really good for getting the demo up and running quickly, for example, to extend the demo to run behind a WebRTC proxy only required about 6 lines of extra code to define a peer in sip.conf and enable TCP The sample config files are mostly reference. They are also examples for some sorts. But they are the best documentation installed in a typical installation. However, I'm not sure that they should be installed by default by packages. Most package managers provide a way to diff the files and merge new config options that appear in a new release Rather: dpkg is rather unique with this feature, IIRC. However, because a lot of things have to be ripped out of the default config to harden it and disable the demo, a simple diff doesn't really help somebody upgrading to a new version, because usually they've altered the files quite dramatically I'd suggest that the config for the demo could be placed under /usr/share/asterisk/samples while the configs installed to /etc/asterisk should be fairly minimal s|/usr/share/asterisk|$astdatadir| Some folks like the value of /var/lib/asterisk for astdatadir. To complement that you'd need to have a simple way to revert back to the standard configuration. Do you think it's a good idea to have asterisk look at $astdatadir/configs/ first if the file in $astetcdir does not exist? Failing that, you could use something along the lines of #include /path/to/astdatadir/configs/origname.conf [content to override] This could be handy for the likes of cli_aliases.conf and say.conf where you really don't like to start from scratch. Do you think it would be a good idea to have a directive that will include a file from under $astdatadir/configs , so the above would become: ;; #includedef ali_aliases.conf [general](+) template = myown [myown](asterisk11) restart now = core stop now ;; The downside of this is that on upgrades you don't see the real changes. My own workaround at the moment involves tracking the released configs in a git repository and tracking my changes on a branch. However, working with the package manager diff output would help a lot more people and make it much more like other packages they are familiar with. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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] Problem with dahdi XPP driver?
debugging verbosity as a bitfield, to see general debugging set this to 1. To see RBS debugging set this to 32 (int) parm: deftaps:int parm: max_pseudo_channels:Maximum number of pseudo channels. (int) parm: hwec_overrides_swec:When true, a hardware echo canceller is used instead of configured SWEC. (int) parm: auto_assign_spans:If 1 spans will automatically have their children span and channel numbers assigned by the driver. If 0, user space will need to assign them via /sys/bus/dahdi_devices. (int) What can be the problem? Thanks, Matteo -- _ -- 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 -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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] offline builds - mp3 [patch]
On Tue, Jun 04, 2013 at 12:49:35PM +0200, Daniel Pocock wrote: As mentioned in the thread about MP3, I found that the rpmbuild process demands network access, e.g. to access the mp3 code in SVN. Some people need to build on isolated networks though I've attached a patch that allows the MP3 code to be placed in /tmp before the build starts, then svn will not be used during the build. If it finds /tmp/asterisk-contrib-mp3.tar.gz then it will be used instead of going to SVN I'm not sure if there are other build steps that access the network, this one was more obvious because I was trying to build on a fresh VM without any svn client I'm sure you're aware of: http://patch-tracker.debian.org/patch/series/view/asterisk/1:1.8.13.1~dfsg-3/mpglib -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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] offline builds - mp3 [patch]
On Tue, Jun 04, 2013 at 06:44:43PM +0200, Daniel Pocock wrote: On 04/06/13 18:37, Tzafrir Cohen wrote: On Tue, Jun 04, 2013 at 12:49:35PM +0200, Daniel Pocock wrote: As mentioned in the thread about MP3, I found that the rpmbuild process demands network access, e.g. to access the mp3 code in SVN. Some people need to build on isolated networks though I've attached a patch that allows the MP3 code to be placed in /tmp before the build starts, then svn will not be used during the build. If it finds /tmp/asterisk-contrib-mp3.tar.gz then it will be used instead of going to SVN I'm not sure if there are other build steps that access the network, this one was more obvious because I was trying to build on a fresh VM without any svn client I'm sure you're aware of: http://patch-tracker.debian.org/patch/series/view/asterisk/1:1.8.13.1~dfsg-3/mpglib The notes suggest that MP3 patent issues are a factor so I guessed that's why it is excluded from the tarball When building with rpmbuild the tarball is usually not unpacked manually, hence my own proposed patch looks in /tmp for the mp3 code - it could just as easily use your the patch from Debian as an input though, as long as it can be found in /tmp or some other predefined location. How would you do that in a proper chrooted build? The proper fix would be to applow to use a newer version of mpglib that is included with some distributions. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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] problem to install asterisk on vps digitalocean
On Tue, Jun 04, 2013 at 01:33:56PM -0600, troxlinux wrote: thnk Russ , I have seen the Makefile, but I see many KSRC, where exactly would put KVERS:=$(shell uname -r) endif ifndef KSRC ifneq (,$(wildcard /lib/modules/$(KVERS)/build)) KSRC:=/lib/modules/$(KVERS)/build else KSRC_SEARCH_PATH:=/usr/src/linux KSRC:=$(shell for dir in $(KSRC_SEARCH_PATH); do if [ -d $$dir ]; then echo $$dir; break; fi; done) endif endif make KSRC=/full/path/to/kernel/headers/directory Makefiles allow you to override variables in the command line: if you run 'make KSRC=whatever', the variable KSRC is defined, and hence make will not get into the part between 'ifndef KSRC' and the corresponding 'endif'. BTW: this is mentioned in the README of dahdi-linux: http://docs.tzafrir.org.il/dahdi-linux/#_kernel_source_headers -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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] Not able to build the chan_sip.c module
On Mon, May 27, 2013 at 04:09:06PM +0530, upendra wrote: hi, chan_local and res_crypto are building but the chan_sip is not building . installed openssl also but still the chan_sip not building. ./menuselect/contrib/menuselect-dummy -c ./menuselect/contrib/menuselect-dummy -m sip -v What's the output? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 Wed, May 22, 2013 at 02:54:46PM -0400, Ahmed Munir wrote: Jim, Cron and Logrotate already installed in my machine and already configured as the steps you enlisted. But still logrotate is not running. How can you tell that the logrotate cron job was run? At what time it was configured to run? Did you see its output in the logs? And please, do make some minimal effort to RTFM and answer questions on your own. Some tools for your disposal: rpm -ql logrotate | grep cron grep -i crom /var/log/messages Cron jobs which have failed and/or had an output send a message to the user who ran them (root, in your case). Is there a sendmail (sendmail, postfix, whatever) running on the system? If so, where does root's mail go to? Read it. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 Tue, May 21, 2013 at 11:29:37AM -0500, Jason Parker wrote:
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
}
I believe you want to execute logger rotate, rather than logger reload.
It is indeed 'logger reload'. logrotate handles the rotation. All it
needs is that Asterisk will re-open the log file to use the new files
after the old ones were moved.
--
Tzafrir Cohen
icq#16849755 jabber:tzafrir.co...@xorcom.com
+972-50-7952406 mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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] Installing Asterisk on Virtual Machine
On Tue, Apr 23, 2013 at 02:17:47PM +0530, Sandeep Raju wrote: @Hans, I just tried installing from pre-built packages (which has asterisk 1.8). Its working fine! :) only the compiled installed versions were giving me the error!.. PS: sorry for spamming with multiple mails.. Distro packages naturally disable BUILD_NATIVE. In the Debian package build rules: # Make sure the configure script gets an CFLAGS parameter. Otherwise # it will build with -march=native What is the minimal code that will get asterisk crash on your system when built with -march=native? It would b einteresting to make this an autoconf test (see the existing test for NATIVE on configure.ac). The bug report notes that this is a gcc issue, but I don't see any link to a gcc bug report anywhere. Here we have gcc 4:4.6.3-1ubuntu5 (right? That what I got from packages.ubuntu.com) still buggy. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 music on hold recommendations
On Tue, Apr 23, 2013 at 02:30:24PM +0200, Frederic Van Espen wrote: Hi all, I'm wondering what the recommendations are for using music on hold on asterisk. As far as I understood from various pages on the web and a response from the IRC channel, I am to avoid using mp3 files because of licensing and transcoding issues. correct? Short version: Not really. But just use the built in The earliest moh support Asterisk had was playing of MP3 files (or piping the output of an external command). Only later on native MoH was developed - playing any file Asterisk could play. At the time Digium licensed a set of mp3 files from FreePlay Music that could be freely used as MoH files with Asterisk. Later on a certain more subtle licensing issue came up and Digium chose to stop distributing those MoH files with Asterisk. They were replaced with a set of five files which are: * Longer * Better licensed (CC-BY-SA 3.0) * Available in all the required formats So the licensing issues in question are: * MP3 is patent-encumbered and some Linux distribution keep out even MP3 playing code (other only remove MP3 encoding code). * If you don't intend to play it to a MP3 channel, why waste CPU resources on transcoding it? The newer files are available in more convinient formats. IIRC the license of the FPM ones prevented Digium from distributing modified copies. I am currently using asterisk 1.8 with the mpg123 processes (mode=mp3 or mode=quietmp3 in the conf file). If you use that mode, you're probably doing something wrong following an ancient guide. This means that there is one single shared stream of moh for all channels that are using the same class of moh. If I were to start using wav files (mode=files), is there a way to have the same kind of shared stream of moh to reduce the load on the machine in the case where a lot calls are on hold? Is it even worth it to try reducing the load (maybe asterisk handles playing wav files very efficiently and the extra load generated by it is negligible)? I am looking to upgrade to asterisk 11 in the future. Is any of this different for that version? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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] Installing Asterisk on Virtual Machine
On Mon, Apr 22, 2013 at 03:44:45PM +0530, Sandeep Raju wrote: Hi, I'm trying to install Asterisk 11.2 on a virtual machine in my private opestack cloud.. When I compile Asterisk 11.2 from source (./configure, make, make install) as specified in the Asterisk book and run it, it gives me the error: Illegal instruction (core dumped). Any ideas how I can solve this? What operating system do you have installed there? What CPU? What is the output of: uname -a Illegal instruction means that you tried running an instruction that the CPU cann't run. Maybe an incorrect choice of optimization flags? Maybe this is due to libraries not matching your architecture? Next thing to do: get a trace from the core file that was dumped using gdb. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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] TigerJet 320G Chip / TDM400 Chipset / DAHDI Support
On Tue, Apr 02, 2013 at 03:27:55PM -0500, Marshall Henderson wrote: Hi, I'm curious what chip Digium is using in the latest TDM400 cards. Specifically, to my recollection, they used to use the TigerJet 320G, however somewhat recently, Tigerjet was bought out, and now the 320G is no longer produced. Maybe a better question is: is there a way I can take the latest DAHDI source and get a list of supported chipsets from it? Latest tarballs: http://downloads.asterisk.org/pub/telephony/dahdi-linux/ But the latest source is in the git repository: git clone git://git.asterisk.org/dahdi/linux.git It lacks the full history of DAHDI and Zaptel. You can find those in the Subversion repositories in svn.asterisk.org or . -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 about zapata.conf
On Mon, Mar 25, 2013 at 03:15:24PM +, Salaheddine Elharit wrote: thank you so much fo the upgrade from zptel to dahdi, if there is any possibility to upgrade to dahdi without impacting my installation of asterisk and other application already installed in my server. if you can tell how to upgrade using dahdi drivers Asterisk 1.4 is at build time set to use either DAHDI or Zaptel (but not both). (try: 'strings /usr/sbin/asterisk | grep /dev'). So you'll have to at least rebuild Asterisk vs. DAHDI. Asterisk of older versions does not support DAHDI at all. You should also note that even the branch 1.4.x is no longer actively supported, and this would be a good time to upgrade. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 about zapata.conf
On Mon, Mar 25, 2013 at 10:44:47AM +, Salaheddine Elharit wrote: hello list, i have a question related to zapata.conf,if i do any change in zapata.conf i must restart asterisk or just i restart zapata ,and how to do . “service zaptel restart” or there is any other command /etc/asterisk/zapata.conf is a configuration ifle of Asterisk's chan_zap.so alone. So changes to it would generally require no more than restart of Asterisk. The simpler of them would be applied with a simple reload (or 'reload chan_zap.so' as you mention). -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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] Optimizing Asterisk Environment
On Sat, Mar 23, 2013 at 09:33:38AM -0400, Nick Khamis wrote: Hello Everyone, We are getting some rather poor results (relative) with our Asterisk setup. Not sure if we are using the sipp correctly etc.. but nevertheless, is there any documentation that describes how we can get the most our of our Asterisk box. For example when we hit the too many file error, and fixing it using ulimit. Also, is there any way we can allocate sufficient memory to our Asterisk instance when starting the PBX. On any decent OS the OS should do that automatically for you. An up to date and in-depth tutorial that covers this would be great. A quick search yielded pretty motivating success stories, but no little to no description on how to achieve them. What do you want to optimize? What is the current bottleneck? Run the system in full capacity and provide us some data. For starters: free -m uptime vmstat -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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] red alarm on span - do channels in the group automatically get skipped over?
On Tue, Mar 05, 2013 at 12:31:09AM -0600, Hose wrote: Hello, If I put two spans' worth of channels, say 1-23 from span 1 and 25-47 in span 2, in one group, but only span 2 was showing OK and the other was down / showing a RED alarm, would asterisk automatically skip over trying to use channels 1-23 when doing outbound calls? e.g., dial(dahdi/g1/(number) would just jump to channel 25? Yes. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 and Oslec
On Tue, Feb 26, 2013 at 11:38:15AM -0500, Doug Lytle wrote: I'm hoping someone can help me here. I've purchased replacement systems for 3 aging 1.4.x installs. I'm hoping to setup Asterisk 11, dahdi 2.6.1 and Oslec. I'm also moving those installs from Mandriva 10.0 to Debian 6.06 (Squeeze). Grab a recent dahdi-source from somewhere: http://packages.debian.org/search?keywords=dahdi-source If that does not work, try: http://updates.xorcom.com/pkg-voip/ With that installed, run: m-a a-i dahdi It already includes OSLEC. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 AMI - Create a daemon (background process)
On Tue, Feb 26, 2013 at 11:29:44AM -0500, Eric Wieling wrote: PHP has had memory leak issues in the past, though modern versions are apparently much better. The thing is, when you write a daemon you must EXPECT it to exit at some point, maybe the socket went away or the system restarted or you are out of memory, whatever. You need to make sure you handle these situations regardless of which language you write in. Using inittab is one way, daemon-tools is another. Some Linux distributions nowadays have upstart or systemd. That makes writing an extra daemon simpler. Regardless of the language. For me, PHP with its C-like syntax, decent string handling, Perl Regex, and POSIX process control (including fork), it is good enough for most stuff that I do. Gah. Just write it in Perl. Or, if it's a glorified script, maybe it just needs to be a shell script. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 and Oslec
On Tue, Feb 26, 2013 at 12:31:59PM -0500, Doug Lytle wrote: Grab a recent dahdi-source from somewhere: Did just that and ran your listed command. The compile fails at: make[3]: Entering directory `/usr/src/linux-3.6.9' CC [M] /usr/src/modules/dahdi/drivers/dahdi/dahdi-base.o /usr/src/modules/dahdi/drivers/dahdi/dahdi-base.c:52:28: error: linux/smp_lock.h: No such file or directory What version did you use? 2.6.1 or the older 2.3.0 one? If the latter, try 2.6.1 . -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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] Quick start configuration sample for chan_dahdi.conf
Hi, On Mon, Feb 11, 2013 at 03:38:09PM +0100, Shitian Long wrote: I am really a beginner of PRI ISDN board, I am wondering if there is a quick start chan_dahdi.conf configuration I could use. For starters, there's the example / reference chan_dahdi.conf: http://svn.asterisk.org/svn/asterisk/trunk/configs/chan_dahdi.conf.sample Replace 'trunk' with 'tags/version' for the a specific version. I tried to install two FreePBX boxes follow the instructions from http://www.cadvision.com/blanchas/Asterisk/DahdiT1trunk.html; Minor nit: generally there's no need to quote FreePBX and URLs. I believe that for URLs it actually does harm as it increases the chance of people accidentally copying the quote marks as part of the URL. Do you have FreePBX installed or not? You can consider Asterisk to be a PBX toolkit and FreePBX to be a PBX created with that toolkit. FreePBX has its own versions for configuration files. Specifically the version it has for chan_dahdi.conf does not look like the one you have. connected them between PRIs, It worked. And now if I refer the FreePBX chan_dahdi.conf it looks like http://pastebin.com/kfWWL6dm; and it That's just wrong. The section name [global] is not special in chan_dahdi.conf. It will be parsed as a specific channel section (parsed after [channels]) The section names [general] (that you happen to use) and [globals] will not be parsed: they are reserved for future use. Ignoring the section [trunkgroups], which you should not care about in your setup, your configuration should look either as: [channels] ; global settings ; Per-channel-settings channel = chan-spec1 ; Per-channel-settings channel = chan-spec2 or: [channels] ; global settings [sec1] dahdichan = chan-spec1 ; Per-channel-settings [sec2] dahdichan = chan-spec2 ; Per-channel-settings Alternatively: [channels] ; global settings #include dahdi-channels.conf and generate /etc/asterisk/dahdi-channels.conf (as well as /etc/dahdi/system.conf) by running dahdi_genconf. That should get you started. seems there is no specific configuration in FreePBX chan_dahdi.conf. And now I tried to add [global] [3:33pm] #include dahdi-channels.conf into chan_dahdi.conf. and do a static-host*CLI dahdi restart still seems no progress… Note that Asterisk will barf if it encounters an #include of a file that does not exist. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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-channels.conf parameters
On Tue, Feb 05, 2013 at 02:09:12PM -0600, Hose wrote:
Hi,
I've always used dahdi-genconf to just create the dahdi-channels.conf
and since our PRI is fairly simple (just dump all the channels into one
group) it works with dialing with dahdi/g1/(number). I'm trying to
understand the file though for my own reference.
It seems the file looks like this:
group=0,11
context=from-pstn
switchtype = national
signalling = pri_cpe
channel = 1-23
context = default
group = 63
So what I don't get is why group is specified with 0,11. Is that groups
0 and groups 11? And then it has a random group = 63 at the end. And I
dial with group 1 (dahdi/g1), but it seems to work? :) It's completely
confused me as to why this actually works.
'group = 63' is a work around an old (and long-ago fixed, I believe) in
chan_dahdi (or maybe chan_zap - and never made it to chan_dahdi?). You
can just write 'group = ' instead to reset the groups.
It would be even nicer to write the above as:
[dahdi-span-1]
group=0,11
context=from-pstn
switchtype = national
signalling = pri_cpe
dahdichan = 1-23
('dahdi-span-1' is an arbitrary title). This only works for Asterisk
= 1.6.0, and thus I never got to switching dahdi_genconf to use it.
--
Tzafrir Cohen
icq#16849755 jabber:tzafrir.co...@xorcom.com
+972-50-7952406 mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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 updates
On Mon, Jan 28, 2013 at 01:55:09PM +, Steven Howes wrote: Hi All, Who do I need to poke to get the yum repository / RPM files updated? The dahdi RPMs are not up to date with the CentOS kernel versions any more, it's making doing an installation a bit tricky due to dependancies, I'd rather not roll back / remove new kernels if I don't have to.. See also: http://git.tzafrir.org.il/?p=rpm/dahdi-linux.git;a=summary http://git.tzafrir.org.il/?p=rpm/dahdi-tools.git;a=summary Sadly I don't yet have the packages built from them available. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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] User busy issue in A400P 4 FXO card
On Thu, Jan 03, 2013 at 09:44:43AM +, A J Stiles wrote:
On Thursday 03 January 2013, Selva M wrote:
Hi,
I setup PBX with A400P 4 x FXo board. There are one analog line plugged
into port 1.
Internal extension cane make calls to PSTN without any issue.
When I make inbound call, caller get busy tone user busy' message right
away.
Asterisk log shows following log and internal extension (200) rings for
that call and hangup (log below).
I tested the system with some other service provider and it worked fine
for IB and OB calls.
i would like to get your feedback to resolve the issue and will
appreciate your feedback.
Thanks
Selva
Don't try to run before you can walk. First of all, simplify your dialplan
right down to the minimum. Have just this context for calls coming in from
the card:
[from-pstn]
s,1,NoOp(Incoming call from ${CALLERID(num)})
s,2,Dial(200)
Huh?
s,3,Hangup()
; end of from-pstn context
The example extensions.conf provides a simple IVR context called 'demo'.
Either use that file or copy the part starting with '[demo]' up until
the next '[section'] to your extensions.conf and in your
chan_dahdi.conf (or dahdi-channels.conf) set 'context=demo'. Reload, and
try again.
In the Asterisk CLI you should see output for 'dialplan show demo'. You
should see the context your dahdi channels go to in the output of 'dahdi
show channels'.
--
Tzafrir Cohen
icq#16849755 jabber:tzafrir.co...@xorcom.com
+972-50-7952406 mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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] Build asterisk for VIA C3
On Thu, Jan 03, 2013 at 10:38:40AM -0500, neo haux wrote: Is it difficult to publish a build asterisk.deb compiled for VIA C3 architecture ? Instead of using the binary just for me. So any one trying to install it on C3 CPU will need just to do: aptitude install asterisk The one that is installed by default doesn't work for such a CPU Should I contact debian dev team for that? Any problem with the standard Debian one? Could you please be more specific regarding the versions? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 for Razberry Pi
On Wed, Jan 02, 2013 at 09:55:44AM -0500, Robert Rawlinson wrote: Has anyone ported Asterisk to the Razzberry Pi? If so could you point me to info on doing so? apt-get install asterisk -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 11 and no RTP
On Thu, Dec 20, 2012 at 05:21:24PM -0500, Jerry Geis wrote: Error loading module 'res_rtp_asterisk.so': /usr/lib64/libavformat.so.52: undefined symbol: av_tree_node_size This is the error I get when trying to start Asterisk 11 on centos 5. Asterisk 11 works fine on my centos 6 box - I also verified that on centos 6 I do not have the above mentioend file. libav is from ffmpeg (nowadays there's even a fork of ffmpeg called libav). -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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
[asterisk-users] wcb4xxp extra hardware IDs
Hi Can anybody verify (confirm / deny) https://issues.asterisk.org/jira/browse/DAHLIN-220, which is actually a simple patch from http://bugs.debian.org/597760 ? It basically asks to add a few hardware IDs to the driver. But the report is a bit inconclusive. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 roll-over / move / rotate an Asterisk Master.csv call detail record (CDR) file every 15 minutes
On Tue, Dec 04, 2012 at 09:17:39AM -0600, Danny Nicholas wrote: Elaborate please. When does this feature come into effect? man logrotate.conf #? Just list the files you want to rotate in /etc/logrotate.d/asterisk (or whereever). -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 roll-over / move / rotate an Asterisk Master.csv call detail record (CDR) file every 15 minutes
On Tue, Dec 04, 2012 at 10:05:47AM -0800, Earl Ruby wrote: Paul: Four reasons not to use logrotate: 1. logrotate does not provide log rotation every 15 minutes. apt-get install logtail logtail2 - print log file lines that have not been read Every 15 minutes use it to get the new lines from the CSV file. Rotate it when needed with logrotate. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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] leading ghost 0
Un-top-posting, On Tue, Nov 20, 2012 at 03:49:49PM +0100, Leandro Dardini wrote: 2012/11/20 Frederic Van Espen frederic...@gmail.com On Tue, 2012-11-20 at 15:03 +0100, gincantalupo wrote: I'm sure nobody has added something... tried prilocaldialplan and pridialplan but nothing changed. Question: if pridialplan or prilocaldialplan would work, should I see the 0 inside PRI frame with intense debug or it is hidden? Somebody correct me if I'm wrong but I think you have to restart asterisk when you change these settings on dahdi. Keep that in mind. Not only, you have to restart dahdi/zaptel as well. No need for that. Some of the settings in chan_dahdi.conf (most of the per-span settings) are not applied at configuration reload. There are some slightly less brutal ways than fully restarting Asterisk to apply them: In the Asterisk CLI: dahdi restart Or, again in the Asterisk CLI: module unload chan_dahdi.so module load chan_dahdi.so No need to load / unload any kernel modules and such. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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] Detected alarm on channel 5: Red Alarm
On Thu, Nov 15, 2012 at 10:11:20AM +0530, Harish Mandowara wrote: Dear, i using this scenario. jitsi--- asteriskEPABX-- Local Telephone What DAHDI device is used for the connection 'asterisk - EPABX'? A digital one? (PRI?) or an analog one? (FXO?) when i am calling from jitsi to no 88 its giving this message and getting busy tone. == Using SIP RTP CoS mark 5 -- Executing [88@myphones:1] Dial(SIP/sandeep-0004, DAHDI/g0/88,20,rt) in new stack -- Called g0/88 [Nov 15 09:53:54] WARNING[3169]: chan_dahdi.c:7536 handle_alarms: Detected alarm on channel 5: Red Alarm -- Hanging up on 'DAHDI/5-1' -- Hungup 'DAHDI/5-1' == Everyone is busy/congested at this time (1:0/0/1) -- Executing [88@myphones:2] Hangup(SIP/sandeep-0004, ) in new stack == Spawn extension (myphones, 88, 2) exited non-zero on 'SIP/sandeep-0004' -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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] Dial plan order of operations
On Mon, Sep 24, 2012 at 02:17:29PM -0700, Steve Edwards wrote:
On Mon, 24 Sep 2012, Asterisk Newb wrote:
Thanks, situated the problem with the following:
exten = _212555.,1,Authenticate(/etc/asterisk/pins||3,j)
exten = _212555.,2,Dial(SIP/${EXTEN:3}@level3,90,tr)
Two suggestions:
1) Using the 'n' priority will make your dialplans more maintainable.
Asterisk 1.2 does not have it, IIRC.
1.1) Upgrade to a newer version of Asterisk :-(
--
Tzafrir Cohen
icq#16849755 jabber:tzafrir.co...@xorcom.com
+972-50-7952406 mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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 hangs while starting in OpenSuse 12.2
On Thu, Aug 30, 2012 at 01:42:06PM +0200, Patrick Lists wrote: On 08/30/2012 09:45 AM, Gopalakrishnan N wrote: Hi, I have installed libpri, Dahdi 2.6 and Asterisk 1.8.15, in normal host, I am not using any virtualbox, still i struck in loading the modules. Please do not top post. Install strace and then start asterisk with the command: # strace asterisk Asterisk will fork into the background and the process you trace will exit. strace -f asterisk #? strace asterisk -f #? Just in case you wonder, 'asterisk -f strace' will not work as you might have expected from the above examples. Nither will '-f strace asterisk'. '-U asterisk ' may also come in handy. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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] libpri error
On Mon, Jul 30, 2012 at 10:50:20AM +, Kamlesh Kumar wrote: when I issue 'make' command, below output comes. [root@localhost libpri-1.4.11.3]# make gcc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -fPIC -O2 -MD -MT copy_string.o -MF .copy_string.o.d -MP -c -o copy_string.o copy_string.c [snip] Looks OK. Make is quite noisy when it encounters an error: $ echo something invalid q921.c $ LANG=C make gcc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -fPIC -O2 -MD -MT copy_string.o -MF .copy_string.o.d -MP -c -o copy_string.o copy_string.c gcc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -fPIC -O2 -MD -MT pri.o -MF .pri.o.d -MP -c -o pri.o pri.c gcc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -fPIC -O2 -MD -MT q921.o -MF .q921.o.d -MP -c -o q921.o q921.c q921.c:3113:1: error: unknown type name 'something' q921.c:3113:1: error: expected '=', ',', ';', 'asm' or '__attribute__' at end of input make: *** [q921.o] Error 1 $ echo $? 2 Had I not sabotaged the code: $ make gcc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -fPIC -O2 -MD -MT copy_string.o -MF .copy_string.o.d -MP -c -o copy_string.o copy_string.c gcc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -fPIC -O2 -MD -MT pri.o -MF .pri.o.d -MP -c -o pri.o pri.c [snip] gcc -shared -Wl,-hlibpri.so.1.4 -o libpri.so.1.4 copy_string.lo pri.lo q921.lo prisched.lo q931.lo pri_aoc.lo pri_cc.lo pri_facility.lo asn1_primitive.lo rose.lo rose_address.lo rose_etsi_aoc.lo rose_etsi_cc.lo rose_etsi_diversion.lo rose_etsi_ect.lo rose_etsi_mwi.lo rose_other.lo rose_q931.lo rose_qsig_aoc.lo rose_qsig_cc.lo rose_qsig_ct.lo rose_qsig_diversion.lo rose_qsig_mwi.lo rose_qsig_name.lo version.lo /sbin/ldconfig -n . ln -sf libpri.so.1.4 libpri.so $ echo $? 0 '$?' (the value of the shell variable named '?') holds the return status of the last command. If it's not 0, it means that this command returned an error. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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] problem with 8 port card
On Mon, Jul 30, 2012 at 09:43:56AM -0500, Russ Meyerriecks wrote: On Mon, Jul 30, 2012 at 04:32:16PM +0530, akhilesh chand wrote: Hi, After upgrading the dahdi-Linux now I'm get failed probing type for channel 1 at /usr/lib/perl5/site_perl/5.8.8/dahdi/config/gen/system.pm This could be an incorrect dahdi-tools version problem. These are pretty general support questions. The fastest way for you to resolve these sort of install issues would be to contact your card's manufacturer. They should be able to walk you through any specific installation issues you may be having, very quickly. The quickest wy to check that is to download the latest version (latest tarball from asterisk.org and extract it, checkout from svn, whatever), without installing it (not evet ./configure), and run: ./xpp/dahdi_hardware or: ./xpp/lsdahdi If the copy in the system gives you the error and the copy you downloaded doesn't, and if it bothers you, upgrade. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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] file and on SayNumber() app
On Mon, Jul 23, 2012 at 02:05:50PM +0300, נפתלי מאיר wrote: Thank you guys. I found say.conf example file with and. No need for that. Hebrew is already supported. Just set the language properly. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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] file and on SayNumber() app
On Mon, Jul 23, 2012 at 10:55:54AM +0300, נפתלי מאיר wrote: Hello, I use the SayNumber() with variable. for example the number 1234 - asterisk play the number without and. -- Executing [888@from-internal:1] Set(SIP/103-035d, LANGUAGE=en) in new stack -- Executing [888@from-internal:2] SayNumber(SIP/103-035d, 1234) in new stack -- SIP/103-035d Playing 'digits/1.ulaw' (language 'en') -- SIP/103-035d Playing 'digits/thousand.ulaw' (language 'en') -- SIP/103-035d Playing 'digits/2.ulaw' (language 'en') -- SIP/103-035d Playing 'digits/hundred.ulaw' (language 'en') -- SIP/103-035d Playing 'digits/30.ulaw' (language 'en') -- SIP/103-035d Playing 'digits/4.ulaw' (language 'en') It`s not will to be: ; one - thousand - two - hundred - *and* - thirty - four ?? Asterisk says numbers slightly differently depending on the language. You would expect that if the language is, say, Hebrew. If the LANGUAGE is set to 'he' or something that begins with 'he_', you'll get the 'and'. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 10/1.6.1 and Dahdi/Libpri compatilities in BRI /PtmP
On Fri, Jun 22, 2012 at 08:07:54PM +1200, Alec Davis wrote: Have a look at the latest blacklist sample in dahdi trunk http://svnview.digium.com/svn/dahdi/tools/trunk/blacklist.sample?view=log file: blacklist.sample ... # Some mISDN drivers may try to attach to cards supported by DAHDI. If you # have a card which is *not* supported by DAHDI but supported by one of the # below drivers you should feel free to remove it from the blacklist below. blacklist hfcmulti May collide with wcb4xxp blacklist netjet May collide with wctdm and some other older drivers. blacklist hfcpci May collide with zaphfc. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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] Setting span orders with Astribank and Sangoma A101
On Sun, Jun 10, 2012 at 10:10:29PM +1200, Duncan Turnbull wrote: Hi All Just a quick check on the best way to ensure multiple cards/devices load in the correct order. Asterisk 1.8 with Sangoma A101 had no problems until we introduced an Astribank. root@pabx377:/etc/asterisk# dahdi_hardware -v usb:001/004 xpp_usb+ e4e4:1162 Astribank-modular FPGA-firmware LABEL=[usb:X1060395] CONNECTOR=@usb-:00:1d.7-3 XBUS-00/XPD-00: FXS (14) Span 2 XBUS-00/XPD-10: FXS (8) Span 3 XBUS-00/XPD-20: FXS (8) Span 4 XBUS-00/XPD-30: FXS (8) Span 5 pci::05:00.0 wanpipe- 1923:0300 Sangoma Technologies Corp. A101 single-port T1/E1 But now they have changed order after reboot to 1-4 and I am guessing that makes the A101 span 5 but I want to make these setting permanent I see mention of /etc/dahdi/xpp_order and dahdi_genconf xpporder but will that help me set the spans within Dahdi so they always appear on the same number The word load is not the correct one here. The Astribank's span will only appear once you run 'dahdi_registration on' . Normally this happens in the dahdi init script, but you can tweak it to happen elsewhere. Another option: if Dahdi 2.6 is an option (no idea if Sangoma supports it): that version has an option to reserve span numbers explicitly for specific hardware devices. With this we can do away with relying on the load order. This is considered experimental (and I wonder how its scriptary interacts with the Sangoma scriptary). http://docs.tzafrir.org.il/dahdi-linux/#_span_assignments , but there are better ways. Let me know what you want to do. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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] chan_dahdi with asterisk 1.4 and new Linux versions
On Mon, May 07, 2012 at 02:59:17AM -0700, bilal ghayyad wrote: Dear Tzafrir; For sure I re run ./configure Actually, I formatted the machine and reinstalled .. also I removed the asterisk 1.4 and extracted again and I ran ./configure, make and make install. All of these I tried ! As long the Linux OS is new, then it is not possible to get chan_dahdi. And if the OS is new, then we can not compile old DAHDI (versions before 2.4 and maybe 2.4 it self can not be compiled if the OS is new). I tried Fedora and Ubuntu. But with asterisk 1.8, things are running well. What is chan_dahdi missing? ./menuselect/contrib/dummy-select -c ./menuselect/contrib/dummy-select -m chan_dahdi -v -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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] chan_dahdi with asterisk 1.4 and new Linux versions
On Fri, May 04, 2012 at 09:24:56AM -0700, bilal ghayyad wrote: What is happening with me that when I used fedora core 16, I compiled and installed dahdi 2.6 and then compiled and installed asterisk 1.4 and it did not create chan_dahdi. I tried to select it by running make menuselect and I discover that it is not possible !! By the way: this problem is not existed with old linux versions .. Have you re-run ./configure ? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 1.6.2.22 backtrace
On Fri, May 04, 2012 at 08:34:49PM +0200, Jonas Kellens wrote: Hello, what does it mean when you read in the backtrace file : Reading symbols from /lib64/libgcc_s.so.1...(no debugging symbols found)...done. No debugging symbols are avaialble for libgcc_s . Libgcc is an external library, which is part of gcc and includes code generated by it. Its debug information would probably be included in a package such as libgcc-debuginfo . Try installing it. That said, debug information for external libraries is often not that important for Asterisk crashes. Loaded symbols for /lib64/libgcc_s.so.1 -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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
