[asterisk-users] Debugging issues with setup
Hello, I set up a new server for Asterisk with 11 cert 6 on it. I am migrating from a previous server. I have replicated all the configurations, modules and setup that I know of. However, when I tested an outbound call, it didn’t work. Checking the asterisk message log yielded nothing. Any ideas on how I can isolate and trace the issue? Thank you! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ConfBridge / internal_sample_rate=auto / warning
Hi there, I am running Asterisk 11.9.0 WANPIPE Release: 7.0.10 DAHDI Version: 2.9.0 Echo Canceller: HWEC libpri version: 1.4.12 When I start the ConfBridge application I get the following warning: [2014-10-24 14:36:21] WARNING[29177][C-6934]: config_options.c:790 uint_handler_fn: Attempted to set internal_sample_rate=auto, but set it to 0 instead due to default) I do not specify a specific user- or bridge-profile so the default profiles are used in confbridge.conf (and there the profiles are empty). But before calling the ConfBridge-App I set the channel var internal_sample_rate=auto. Am I making a mistake or is it an indicator for a wrong configuration? Best regards, -Thorsten- -- _ -- 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] Ast 13 beta 3 - Segfault when calling on pjsip trunk with directmedia=yes
On 10/23/2014 05:00 PM, Matthew Jordan wrote:
On Thu, Oct 23, 2014 at 3:32 PM, Dave Fullerton
dfullertaster...@shorelinecontainer.com
mailto:dfullertaster...@shorelinecontainer.com wrote:
Hello all,
I'm setting up a couple of test boxes and I'm running into a
problem. What I need help with is determining whether I'm going
something wrong or if I need to post a bug report. I have two
asterisk 13.0-beta 3 machines set up with extensions connected to
each as such:
3700 AST-A -- AST-B 3800 3801
When I place a call from 3800 to 3700 or the other way around ,
asterisk seg faults on both machines at roughly the same time. All
connections are done using PJSIP. The crash occurs when the ringing
extension is answered.
If I set (directmedia=no) OR (directmedia=yes t38_udptl=yes) on
the trunk then the call completes fine. All phones and servers are
on the same LAN with no firewalls active.
The trunk between AST-A and AST-B is configured like this in
pjsip.conf and is identical on both machines:
[transport-lan]
type=transport
protocol=udp
bind=0.0.0.0
tos=af31
[pbxbeta]
type=endpoint
disallow=all
allow=g722
allow=ulaw
transport=transport-lan
context=phone-level3
aors=pbxbeta
send_rpid=no
send_pai=yes
trust_id_inbound=yes
trust_id_outbound=yes
direct_media=yes
direct_media_glare_mitigation=__outgoing
;direct_media_method=update
tos_audio=46
tos_video=34
t38_udptl=no
t38_udptl_nat=no
[pbxbeta]
type=aor
contact=sip:{remote IP address}:5060
[pbxbeta]
type=identify
endpoint=pbxbeta
match={remote IP address}
The phones have the following set in pjsip.conf (snippet):
type=endpoint
disallow=all
allow=g722
allow=ulaw
transport=transport-lan
send_rpid=no
send_pai=yes
direct_media=yes
tos_audio=46
tos_video=34
Is there something I'm doing wrong here?
Thanks
Asterisk shouldn't crash.
Please file a bug report ASAP at issues.asterisk.org
http://issues.asterisk.org, with a properly generated backtrace:
https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com http://asterisk.org
Created: https://issues.asterisk.org/jira/browse/ASTERISK-24448
Let me know if you need any more information.
Thanks
-Dave
--
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Re: [asterisk-users] AstriDevCon 2014:Agenda item Deprecate AMI/AGI(Ben Klang)
On Oct 23, 2014, at 1:58 PM, Kevin Larsen kevin.lar...@pioneerballoon.com wrote: From: Paul Albrecht palbre...@glccom.com Seems like now is as good a time as any to raise these issues, in fact, sooner is better than later because once developers start down a path it’s very difficult to get them change their minds no matter how much sense it makes. The fact that developers are even considering taking away user functionality like the dial plan is in of itself a very serious problem because it demonstrates they don’t see Asterisk from the user perspective. Don’t object to extending the Asterisk user interface or changing Asterisk internals. Do object to is taking away functionality that users expect, are familiar with, and has made the Asterisk project successful. Then your experience is atypical. Asterisk has been unstable for several years as developers have continually shoveled new features into the code base over several releases. That’s not necessary objectionable, it’s even to be expected; however, at some point developers need to turn their attention to less glamorous less exciting things like stability and performance. I don't think anyone is objecting to you bringing this up, as it has been mentioned at the dev con. Perhaps it is just that the tone doesn't come across properly in an email, but you are coming across as confrontational and alarmist and it seems to be setting people on edge. Matt has already chimed in that he doesn't see how it would be possible to deprecate the dial plan at this time and even if it were possible, the process would take on the order of years, giving you plenty of time to enact any contingency plans you might need. Scott G. from Digium even posited that if it were to be removed from the core, it would likely end up as a loadable module so that it wouldn't burden those who don't need it and could be loaded for those who do. 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. As for Scott’s suggestion, I don’t see how you can have it both ways: on the one hand ARI can’t work in an environment supporting AGI/AMI and the dial plan, and on the other you can support AGI/AMI in an optional module. It’s just not believable. If you’re not convinced, run menuconfig and look through the applications and functions sections. All of this stuff would have to change and you think that at the end of that process the dial plan would survive? I don’t think so. These developers do not exist in a vacuum, nor do they have total control over where Asterisk goes. Influence, sure, but there is still a corporate structure out there that finds it necessary to be customer oriented. They would have to be monumentally stupid (something which I haven't seen previous evidence of) to kill off the dial plan without providing a path forward for those who depend on it. Furthermore, even if they did pull a stunt so bad as to alienate half their users, the open source code would be forked so fast as to make your head spin or people would migrate to other similar packages (Freeswitch comes to mind). Digium sells their own PBX hardware that I am sure depends on these technologies that you are afraid will go away. They have direct skin in this game too. Totally get why Ben Klang thinks everyone should use adhearsion and that all the resources of the Asterisk community should be devoted to his project. But of course everyone thinks their project is the most important project in the world. What I don’t get is why one project or framework is deemed to be so important that is trumps everyone else in the Asterisk community. I would be interested to know just how atypical my experience is. I have found that on my 1.6 systems I would have random crashes over time. After upgrading over multiple sites, my 11.x systems have been rock solid for the most part. I did have a case where I did a store and forward of a fax that if I tried to forward the fax and it had no file to forward would cause a crash, but other than that, I haven't seen any problems in normal day to day usage. I always thought that the general consensus was that the 11.x series was quite a bit more stable than the older versions. Wouldn’t use the 1.6 release as a basis of comparison as that release was regarded as DOA by most folks. A better example would be Asterisk 1.4. We use it because it’s stable, we don’t need any of the new features and definitely want to avoid the performance degradation in the later releases. Having said that, Asterisk 1.6 was an interesting release because that’s when “async agi” was introduced and
[asterisk-users] Call forwarding from Phones and getting the referrer IP
Hi I'm using asterisk 1.8 but I'm sure this applies to other versions. If someone puts a call divert on a handset such as a Snom phone I get this type of SIP message on receipt of an inbound call: Got SIP response 302 Moved Temporarily back from xxx.xxx.xxx.xxx:x Which then triggers a local channel to make the call. Is there any way I can access that IP address inside my dialplan? I've done a ChanDump and there's no sign of it. Regards Ish -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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] Debugging issues with setup
On Fri, 24 Oct 2014, Marco Carvalho wrote: I set up a new server for Asterisk with 11 cert 6 on it. I am migrating from a previous server. I have replicated all the configurations, modules and setup that I know of. However, when I tested an outbound call, it didn’t work. Checking the asterisk message log yielded nothing. Any ideas on how I can isolate and trace the issue? I'd fire up the CLI, bump up debug and verbose, enable SIP debugging (if your outbound provider is SIP), and observer the output when making a call. There should be some clues there. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000-- _ -- 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 )
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 Command 'module load chan_sip' failed. SIP channel loading... == Parsing '/etc/asterisk/sip.conf': Found == Parsing '/etc/asterisk/test/droid2.sip.conf': Found [Oct 24 16:47:39] ERROR[2743]: config.c:1549 process_text_line: The file 'exts/droid.sip.conf' was listed as a #include but it does not exist. --
[asterisk-users] Asterisk 12 Dialplan
In https://wiki.asterisk.org/wiki/display/AST/Introduction+to+ARI+and+Chann els it is stated: channel-dump.js in action Here's sample output from channel-dump.js. When it first connects there are no channels in Asterisk - (sad) - but afterwards a PJSIP channel from Alice enters into extension 1000. This prints out all the information about her channels. After hearing silence for a while, she hangs up - and our script notifies us that her channel has left the application. end of quote Is there some way the call can be moved to the next priority or context in the dial plan from the stasis app? It seems the caller is stuck in stasis. Thanks -- _ -- 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 10/24/2014 02:21 PM, Jeffrey Ollie wrote: restorecon -rv /etc/asterisk I'd never have guessed. Thanks. I owe you a beer. At least one. 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
Re: [asterisk-users] Asterisk 12 Dialplan
On Fri, Oct 24, 2014 at 1:19 PM, Murthy Gandikota mgandik...@nts.net
wrote:
In
https://wiki.asterisk.org/wiki/display/AST/Introduction+to+ARI+and+Chann
els
it is stated:
channel-dump.js in action
Here's sample output from channel-dump.js. When it first connects there
are no channels in Asterisk - (sad) - but afterwards a PJSIP channel
from Alice enters into extension 1000. This prints out all the
information about her channels. After hearing silence for a while, she
hangs up - and our script notifies us that her channel has left the
application.
end of quote
Is there some way the call can be moved to the next priority or context
in the dial plan from the stasis app? It seems the caller is stuck in
stasis.
Once a channel hangs up it is controlled by hangup handlers and h extens.
If however you want to kick an active channel out of your stasis application
to run dialplan then you use the
POST /channels/{channelId}/continue
ARI command.
Richard
[1]
https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Channels+REST+API#Asterisk12ChannelsRESTAPI-continueInDialplan
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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] Questions on musiconhold.conf custom mode
Hello, I need to play some musiconhold content starting at a random duration from the start. Thanks to mode=custom option and either madplay or mpg123 programs, I could successfully get what I was after on a Debian Wheezy system. Now I realized sox version on my target system (Debian Squeeze) cannot convert to MP3 format. So I'm looking after workarounds. 0. I've read many mpg123 or madplay examples. All of them are clutered with option converting MP3 input file into an appropriate format that Asterisk requires for music on hold. What is the name of this appropriate format ? sln ? wav ? 1. Is there a player like mpg123, that can repeat content in appropriate format (see above) to stdout but can read from anything different from MP3 ? 2. Is there an option on Squeeze to convert audio files to MP3 (reverse coversion works OK). 3. Which options could I have for such custom MOH, if I was building on system without g729 transaltion capabilites ans with g729-only SIP trunks or phones ? Regards -- _ -- 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
