Re: [asterisk-users] On Fedora, kernel update resets /var/run/asterisk owner to root.root
Are you using the Fedora RPMs or are you compiling it yourself? If you're compiling it yourself, you'll need to create a file called /etc/tmpfiles.d/asterisk.conf with this content: d /run/asterisk 0755 asterisk asterisk On Fedora /var/run is a symlink to /run, and /run is a tmpfs partition, which means it gets wiped out every time you reboot. systemd-tmpfiles uses the files in /etc/tmpfiles.d and /usr/lib/tmpfiles.d to recreate files/directories at boot time. On Tue, Dec 2, 2014 at 1:22 PM, sean darcy seandar...@gmail.com wrote: On Fedora 20, every time the kernel updates, /var/run/asterisk owner is set to root.root. I'm running asterisk under user asterisk. Is there any way to keep /var/run/asterisk as asterisk.asterisk. Or do I find a new place to put asterisk.pid? sean -- _ -- 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 -- Jeff Ollie -- _ -- 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 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. -- Jeff Ollie -- _ -- 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 2, 2014 at 3:00 PM, sean darcy seandar...@gmail.com wrote: Put asterisk.conf in /etc/tmpfiles.d, and all worked. It needs to be included in the rpm. It's already in the RPMs distributed by Fedora. I wouldn't know about the RPMs distributed by Digium. -- Jeff Ollie -- _ -- 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-dev] AstriDevCon 2014:Agenda item Deprecate AMI/AGI (Ben Klang)
On Mon, Oct 27, 2014 at 2:04 PM, Paul Albrecht palbre...@glccom.com wrote: The reason the dial plan can never be deprecated is because Asterisk wouldn’t be Asterisk without the dial plan. Sure, you could re-engineer Asterisk so that it would be “better for a small select group of users at the expense of the majority of community that use the product as designed for the purpose it was originally intended. However, you’re either very naive or delusional if you think the community is going to follow you down that path. Do you really believe the community is going simply chuck their dial plans and walk away from their investment in Asterisk? Not likely, dude. My comment/question wasn't really about dial plans, per se. My question was about you insisting that Digium make such unqualified promises about the future of Asterisk. Even though Digium is a private company, I believe that they are still bound by U.S. laws regarding forward-looking statements[1]. So even if they wanted to (which I doubt), there's no way you're going to get the promise that you're looking for. [1] http://en.wikipedia.org/wiki/Forward-looking_statement -- Jeff Ollie -- _ -- 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-dev] AstriDevCon 2014:Agenda item Deprecate AMI/AGI(Ben Klang)
On Fri, Oct 24, 2014 at 10:09 AM, Paul Albrecht palbre...@glccom.com wrote: When Matt says deprecating the dial plan would be difficult and would take a long time it seems to me he’s being evasive and misleading. He doesn’t say it’s never going to happen and he doesn’t share whatever he thinks the Asterisk vision actually is which he should presumably be aware of since he is the Asterisk engineering manager. Why do you keep insisting that Digium promise to *never* deprecate dial plans? I don't think that's a promise that's really worth anything as there may be really good reasons in the future to do so. I think that you've gotten the best that you will get: they've said that there are no plans within Digium to deprecate the dial plan, and if there were plans, they'd give people a long time prepare before it actually happens. It's probably a good time to refresh your understanding of Digium's support policies: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions Version 13 will be around until at least 2018, so you'll have *at least* that long to prepare for the switch, since version 13 is feature frozen so there's no way the dial plan would be removed from 13. And all of this talk of deprecating the dial plan isn't even coming from Digium. It's something that was suggested by a community member at the developer conference. I wasn't there so I don't know how seriously it was taken there, but it would have been impolite of everyone involved to just ignore it. -- Jeff Ollie -- _ -- 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] 11.13.1: unable to load sip.conf (or iax )
Depending on how the data was copied from one install to the other, you may be running into SELinux issues. Try running: restorecon -rv /etc/asterisk and see if that helps. On Fri, Oct 24, 2014 at 11:56 AM, sean darcy seandar...@gmail.com wrote: On 10/23/2014 01:19 PM, sean darcy wrote: On 10/23/2014 11:26 AM, sean darcy wrote: Running 11.13.1 on Fedora. This is a new install, but a copy of a previous - working -install. module load chan_sip Unable to load module chan_sip Command 'module load chan_sip' failed. SIP channel loading... [Oct 23 14:46:08] NOTICE[669]: chan_sip.c:31438 reload_config: Unable to load config sip.conf I don't think it's permissions: ls -ld /etc/asterisk /etc/asterisk/sip* drwxr-x---. 4 asterisk asterisk 4096 Oct 23 00:34 /etc/asterisk -rw-r-. 1 asterisk asterisk 3588 Oct 22 18:37 /etc/asterisk/sip.conf -rw-r-. 1 asterisk asterisk 91033 Oct 23 00:28 /etc/asterisk/sip.conf.rpmnew -rw-r-. 1 asterisk asterisk 790 Oct 23 00:28 /etc/asterisk/sip_notify.conf ps aux | grep asterisk asterisk 294 0.1 5.5 1076736 33364 ? Ssl 14:36 0:03 /usr/sbin/asterisk -f -C /etc/asterisk/asterisk.conf The sip module itself is loaded: module show like chan_sip Module Description Use Count chan_sip.soSession Initiation Protocol (SIP) 0 1 modules loaded I've tried my old config, and just the sip.conf.sample. Same result. FWIW: ls -l /usr/lib64/asterisk/modules/chan* -rwxr-xr-x. 1 root root 72808 Oct 23 00:29 /usr/lib64/asterisk/modules/chan_agent.so -rwxr-xr-x. 1 root root 16032 Oct 23 00:29 /usr/lib64/asterisk/modules/chan_bridge.so -rwxr-xr-x. 1 root root 347920 Oct 23 00:29 /usr/lib64/asterisk/modules/chan_iax2.so -rwxr-xr-x. 1 root root 41888 Oct 23 00:29 /usr/lib64/asterisk/modules/chan_local.so -rwxr-xr-x. 1 root root 118144 Oct 23 00:29 /usr/lib64/asterisk/modules/chan_mgcp.so -rwxr-xr-x. 1 root root 67424 Oct 23 00:29 /usr/lib64/asterisk/modules/chan_motif.so -rwxr-xr-x. 1 root root 11936 Oct 23 00:29 /usr/lib64/asterisk/modules/chan_multicast_rtp.so -rwxr-xr-x. 1 root root 44392 Oct 23 00:29 /usr/lib64/asterisk/modules/chan_phone.so -rwxr-xr-x. 1 root root 755296 Oct 23 00:29 /usr/lib64/asterisk/modules/chan_sip.so Any help appreciated. sean Weirdness: made iax.conf.simple: [general] autokill=yes [idefisk] type=friend host=dynamic context=phones (extra credit for remembering the source) module unload chan_iax2.so Unable to unload resource chan_iax2.so Command 'module unload chan_iax2.so' failed. [Oct 23 16:53:26] WARNING[669]: loader.c:571 ast_unload_resource: Unload failed, 'chan_iax2.so' is not loaded. module load chan_iax2.so Unable to load module chan_iax2.so Command 'module load chan_iax2.so' failed. [Oct 23 16:53:36] ERROR[669]: chan_iax2.c:13488 set_config: Unable to load config iax.conf But then: cp -a iax.conf.simple iax.conf cp: overwrite ‘iax.conf’? y ls -l iax* -rw-r-. 1 asterisk asterisk 74 Oct 23 16:52 iax.conf -rw-r-. 1 asterisk asterisk 652 Oct 22 18:37 iax.conf.real -rw-r-. 1 asterisk asterisk 74 Oct 23 16:52 iax.conf.simple module load chan_iax2.so Loaded chan_iax2.so cp iax.conf.real iax.conf cp: overwrite ‘iax.conf’? y module unload chan_iax2.so Unloaded chan_iax2.so module load chan_iax2.so Loaded chan_iax2.so So the simple config will load. Then if I unload it, and the real config will load !! This approach also works for sip.conf, but now have another problem : it won't recognize any of the #includes. For instance: module load chan_sip.so Unable to load module chan_sip.so Command 'module load chan_sip.so' failed. SIP channel loading... [Oct 23 17:13:43] ERROR[669]: config.c:1549 process_text_line: The file '/etc/asterisk/exts/droid.sip.conf' was listed as a #include but it does not exist. [Oct 23 17:13:43] ERROR[669]: chan_sip.c:31461 reload_config: Contents of sip.conf are invalid and cannot be parsed grep exts/droid.sip sip.conf #include /etc/asterisk/exts/droid.sip.conf ls -l /etc/asterisk/exts/droid.sip.conf -rw-r--r--. 1 asterisk asterisk 316 Oct 22 18:37 /etc/asterisk/exts/droid.sip.conf I also tried relative addressing, exts/droid.sip.conf , same problem. And, of course, all this works on the 11.10.2 server. sean Weirder yet: ls -ld /etc/asterisk/test /etc/asterisk/exts drwxr-xr-x. 3 644 asterisk 4096 Oct 24 16:41 /etc/asterisk/exts drwxr-xr-x. 2 644 asterisk 4096 Oct 24 16:44 /etc/asterisk/test cp exts/droid.sip.conf test/droid2.sip.conf ls -l /etc/asterisk/exts/droid.sip.conf /etc/asterisk/test/droid2.sip.conf -rw-r--r--. 1 644 asterisk 316 Oct 22 18:37 /etc/asterisk/exts/droid.sip.conf -rw-r--r--. 1 644 asterisk 316 Oct 24 16:44 /etc/asterisk/test/droid2.sip.conf grep droid sip.conf #include test/droid2.sip.conf #include exts/droid.sip.conf module load chan_sip Unable to load module chan_sip
Re: [asterisk-users] 11.13.1: unable to load sip.conf (or iax )
On Fri, Oct 24, 2014 at 1:47 PM, sean darcy seandar...@gmail.com wrote: On 10/24/2014 02:21 PM, Jeffrey Ollie wrote: restorecon -rv /etc/asterisk I'd never have guessed. Yeah, if you mv the data instead of cp the data from one place to the other, the SElinux labels don't get updated. I like SElinux, but it would be nice if there were better error messages... Although, if you're on Fedora 20, this is a pretty good description of what was going wrong and how to diagnose/solve it: http://danwalsh.livejournal.com/65777.html -- Jeff Ollie -- _ -- 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] PJSIP and NAT behind a dynamic IP address
What should the PJSIP configuration be if your external IP address is dynamic, as is common with most home networks, and probably a lot of small business networks as well? The external_media_address and external_signaling_address transport settings are static. It would be possible to write a script that would detect the external IP address and rewrite the pjsip configuration file, but since you can't change transports without a full restart of the server that doesn't seem very friendly. Is the only alternative to rely on your firewall/router to fix up the address in the SDP? -- Jeff Ollie -- _ -- 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] IP Phone recommendation
On Wed, Feb 10, 2010 at 12:23 PM, Brent Torrenga li...@torrenga.com wrote: Coming from someone who uses 7940's and 60's: has Cisco/Linksys embraced SIP compatibility with asterisk more completely with the SPA504G's than they have the 7940 series? Lack of features on the 7940's is frustrating, and makes me hesitant to try other Cisco phones, even if the SPA504G is newer. SIP support in newer generations of the 79XX series is much better. I believe that their goal is to have 100% feature parity between the SIP and the SCCP images, they are probably 90% now. Whether Asterisk supports all of those features is another matter though. -- Jeff Ollie -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
