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| 01:18.46 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.19.2 (2018/02/21), Standard: 15.2.2 (2018/02/21); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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| 04:53.59 | klow | Is it is possible to reload the the dialplan through an ARI call ? |
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| 07:57.52 | clarnist | hello |
| 07:58.28 | clarnist | I want to manage a php script for making call and i only make call in local extensions |
| 07:58.52 | clarnist | Can't outbound. Can anybody know how to make outgoing calls? |
| 07:59.09 | clarnist | context = 'from-sip-external' |
| 08:02.59 | Samot | You need to clarify that request a bit more |
| 08:10.20 | clarnist | I have a php script that the idea is to call on phone and retrieve status of this: answer, rejected or not anwered |
| 08:11.03 | Samot | Still not clear. |
| 08:11.08 | Samot | You want to check the actual phone? |
| 08:11.15 | Samot | Or you want to check the device state in Asterisk? |
| 08:11.16 | clarnist | I am using agi and when I put Channel="SIP/$number" where number is a extension number locally it call it |
| 08:13.15 | Samot | So you want to call the phone? |
| 08:13.24 | clarnist | Yes from php |
| 08:13.47 | clarnist | No audio only status of connection |
| 08:13.50 | Samot | Then you need to do an AMI Origination or the Originate function/app via the diplan |
| 08:14.00 | Samot | What do you mean "status of connection" |
| 08:14.08 | Samot | If you don't have audio, you don't have a call. |
| 08:14.14 | Samot | The call still needs to be answered. |
| 08:14.16 | clarnist | Answered, no answered, rejected |
| 08:14.43 | Samot | 3:13:50 AM <Samot> Then you need to do an AMI Origination or the Originate function/app via the diplan |
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| 08:15.16 | clarnist | I have originate |
| 08:15.21 | clarnist | <PROTECTED> |
| 08:15.21 | clarnist | <PROTECTED> |
| 08:15.21 | clarnist | <PROTECTED> |
| 08:15.21 | clarnist | <PROTECTED> |
| 08:15.21 | clarnist | <PROTECTED> |
| 08:15.22 | clarnist | <PROTECTED> |
| 08:15.26 | clarnist | <PROTECTED> |
| 08:15.28 | clarnist | <PROTECTED> |
| 08:15.30 | clarnist | <PROTECTED> |
| 08:15.32 | clarnist | <PROTECTED> |
| 08:15.34 | clarnist | <PROTECTED> |
| 08:15.40 | clarnist | It can call inbounds but can't go outside, f.e. cell phone |
| 08:18.07 | Samot | Because SIP/<external number> isn't a valid endpoint |
| 08:18.22 | Samot | It needs to be SIP/<trunk peername>/<external number> |
| 08:19.41 | clarnist | Thanks it really helps ! |
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| 09:50.22 | sotoz | Hello I'm using asterisk 14.7.6 and when I do MP3Playback() I get the error `Poll timed out/errored out with 0` |
| 09:50.29 | sotoz | The file gets played back correctly |
| 09:50.39 | sotoz | but then I get 10 secs of delay and this error |
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| 10:22.02 | Syco | Hello, I have a quick question.. Can I do Dial(IAX2/012345@ipaddress) ? Can it work without any peer or authentication? Of course the other server needs to accept anonymous calls, but I can't find if that's a valid dial string. |
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| 11:06.47 | sibiria | Syco: at least that's the channel syntax for old IAX, if that might help |
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| 12:06.53 | pawiecki | Hi. Client had a problem calling some extensions, and we restarted asterisk under pressure, which wasn't smart, but it fixed the issue. Now i'm trying to figure out what could be the problem before restart, in logs I see some queued hints right before the restart and there's no "Ringing" after * calls the user. Does that mean that * coundn't send new state to selected users, and couldn't call them? |
| 12:06.55 | pawiecki | https://paste.fedoraproject.org/paste/JYjKFjVUgyJtcJ4v3oxl7A/raw |
| 12:07.15 | pawiecki | couldn't* |
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| 12:26.47 | kippi | hey |
| 12:27.03 | kippi | is there away from /proc/ I can see how many calls asterisk is processing ? |
| 12:31.42 | pawiecki | kippi: what do you mean exactly? |
| 12:32.30 | kippi | pawiecki: I want to view how many calls are active, but I want to do this very often and don't want to be calling rasterisk....... |
| 12:33.54 | pawiecki | kippi: what do you need this for? |
| 12:34.30 | kippi | pawiecki: we would like a "realtime" status from our asterisk boxes |
| 12:35.43 | sotoz | Hello I'm using asterisk 14.7.6 and when I do MP3Playback() I get the error `Poll timed out/errored out with 0`. The file gets played back correctly but then I get 10 secs of delay (till the dialplan progresses) and this error. |
| 12:36.02 | sotoz | anyone had seen that before? |
| 12:36.23 | pawiecki | kippi: why not something simple like 'watch asterisk -rx "core show calls" > calls.txt' ? |
| 12:36.47 | pawiecki | then just refresh that file |
| 12:37.36 | pawiecki | sotoz: show us logs of this thing hasppening |
| 12:37.40 | pawiecki | happening* |
| 12:39.44 | kippi | pawiecki: is it really a good idea to be hitting asterisk like that once a second? |
| 12:42.13 | pawiecki | kippi: once every 2 seconds by default. I'm not an expert, but it's a simple solution, and it's a very simple command for * to parse. There can be more elegant solutions. |
| 12:43.52 | kippi | pawiecki: with our load we have seen issues with using the asterisk console, this is why I was hoping there might be something in proc I can ref |
| 12:56.51 | sibiria | when restarting over the CLI using 'core restart when convenient', will this actually cause asterisk to exit? |
| 12:57.29 | sibiria | (so that safe_asterisk can relaunch it) |
| 12:57.43 | pawiecki | sibiria: "his command waits until Asterisk has no calls in progress, and then it restarts the service. It does not prevent new calls from entering the system." |
| 12:58.10 | pawiecki | https://wiki.asterisk.org/wiki/display/AST/Stopping+and+Restarting+Asterisk+From+The+CLI |
| 12:58.14 | sibiria | yes, the description doesn't entirely explain if it just restarts "core" or quits the process |
| 12:58.39 | pawiecki | "and then it restarts the service" |
| 12:58.55 | pawiecki | what is core? |
| 13:04.43 | sibiria | as i suspected, asterisk doesn't exit at all |
| 13:04.50 | sibiria | it doesn't restart the service in that sense |
| 13:05.04 | pawiecki | sibiria: are there atcive calls in progres? |
| 13:05.24 | sibiria | no, and that doesn't matter |
| 13:05.28 | sibiria | it jsut reloads all modules |
| 13:05.34 | sibiria | that's what "core" means in this context |
| 13:05.51 | sibiria | and that's what a "restart" means |
| 13:06.10 | sibiria | rather than actually exiting asterisk so that safe_asterisk can fire it up again (to clear out f.e. memory leaks) |
| 13:06.55 | pawiecki | sibiria: why are you restarting asterisk if you want safe_asterisk to kick in? Wouldn't stopping asterisk work better in this scenario? |
| 13:07.22 | sibiria | to me, a restart implies asterisk exiting and starting again |
| 13:07.45 | sotoz | @pawiecki: 2018-03-07 12:47:31.000 Playing mp3 file. |
| 13:07.45 | sotoz | 2018-03-07 12:47:49.154 NOTICE[22083][C-0000a176] app_mp3.c: Poll timed out/errored out with 0 |
| 13:07.46 | sotoz | 2018-03-07 12:47:58.000 asterisk[14890]: ChannelID is 1520426851.127082 and Hangupcause is 1. |
| 13:08.17 | sibiria | i'm simply looking for a way to combine the convenient part with an actual restart of the process (because we're seeing some memory leaks, and we can't update to latest 13.x in the nODear future) |
| 13:08.27 | sibiria | near* |
| 13:09.15 | pawiecki | sibiria: why not 'service asterisk restart'? |
| 13:10.18 | sibiria | pawiecki: because that doesn't involve the "convenient" part. it just TERMs the process |
| 13:10.19 | pawiecki | sotoz: have you seen this: https://issues.asterisk.org/jira/browse/ASTERISK-26085 ? |
| 13:10.47 | sibiria | i want the graceful and patient aspect of it so that ongoing calls aren't abruptly ended |
| 13:10.55 | sibiria | i'll find a way around it though |
| 13:12.55 | sibiria | adding a new mode to the sysvinit file should suffice - one that issues "core stop when convenient" and then immediately starts the service again |
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| 13:39.00 | sotoz | pawiecki: this means that this issue is fixed on th 15.x version? |
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| 13:44.23 | pawiecki | sotoz: please read the bug report and release notes for 15.x if you are not sure. |
| 13:45.20 | sotoz | well apparently they say that the target release version is 15.0.0 and the resolution is fixed so yes |
| 13:46.54 | file | the change went into other branches, but because it wasn't automatically tagged not all versions are in fixed |
| 13:47.30 | file | if you go to the review - https://gerrit.asterisk.org/#/c/3738/ you can see in cherry picks the branches it went in, and then if you click the "Included in" drop down it'll tell you every release it is in |
| 13:55.22 | sotoz | ok thanks @file |
| 13:55.50 | sotoz | I see that it is included in 14.7.6 also so I would say that this wasn't my issue after all :( |
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| 19:11.06 | kodomo | Hi all! I'm trying to migrate form chan_sip to pjsip (therefore working with asterisk 13 and pjsip 2.7.1), but are having trouble setting up TLS. Is it possible that pjsip does not support a series of cipher suites previously supported by chan_sip? (most notably: TLS_ECDHE_RSA_WITH_AES_256_CBC_SHA ) |
| 19:11.31 | kodomo | TLS works fine with chan_sip, but there seems to be no cipher suite overlap between my clients and pjsip :( |
| 19:16.07 | kodomo | pjsip seems to support every component of abovementioned suite individually, but apparently not in the needed combination (it's missing in pjsip list ciphers - and I'm wondering whether I'm merely missing a config item or whether support for the suite has indeed been dropped) |
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| 19:35.15 | mh166 | Hi guys. I'm just trying out the pjsip_wizard.conf |
| 19:35.56 | mh166 | Now I'm wondering: is there an option to have a custom replaceable parameter? |
| 19:37.16 | mh166 | For example, I'd like to have a parameter for my phone number as my SIP provider loves to use it throughout different options. |
| 19:37.47 | mh166 | The final outcome could look something like this: |
| 19:37.59 | mh166 | remote_hosts = tel.t-online.de |
| 19:38.22 | mh166 | phone_number = +4900000000 |
| 19:38.57 | mh166 | endpoint/outbound_proxy=sip:{$PHONE_NUMBER}@${REMOTE_HOST}:5060\;lr |
| 19:39.12 | mh166 | client_uri_pattern = sip:{$PHONE_NUMBER}@${REMOTE_HOST}:5060 |
| 19:39.18 | mh166 | --- |
| 19:40.57 | mh166 | As I've got several different phone numbers to register, this would make using templates soo much easier. |
| 19:41.46 | mh166 | I then would only have to specify the phone_number per instance and the rest could come from the template as almost every other option is the same. |
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| 19:43.31 | mh166 | Is there such an option and I just haven't found it yet? Or would this be a nice thing that could possibly be added to the pjsip wizard? |
| 19:58.49 | file | there is no such thing afaik, and it would be something that would go into the common configuration file stuff |
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| 20:09.59 | kodomo | Is there a more appropriate place to discuss pjsip ciphersuite support than this channel? |
| 20:31.56 | kodomo | This is getting frustrating... so according to https://www.openssl.org/docs/manmaster/man1/ciphers.html TLS_ECDHE_RSA_WITH_AES_256_CBC_SHA translates into ECDHE-RSA-AES256-SHA |
| 20:32.14 | kodomo | cipher=ECDHE-RSA-AES256-SHA is supported (listed in pjsip list ciphers) |
| 20:33.13 | kodomo | but whether I specify it in the transport definition or not (transport is created), * claims that there's no shared cipher and fails the handshake |
| 20:33.41 | kodomo | (even though I see it offered in the Client Hello message I have in the pcap trace) |
| 20:34.47 | kodomo | pjsip show transport transport-tls shows me the config just like I expect it... any suggestions? |
| 20:49.42 | *** join/#asterisk Dovid (~dovid@ool-45738ae3.dyn.optonline.net) |
| 20:52.05 | Dovid | Is there any simple way in asterisk set an array? It seems that the function array and the application mset sort of do the same thing. What I want to do is basically set an array and then later just see if it exists. Something like members = $abc = array(1,2,3,10,20) then if some enters 20 I see if $abc[20] is set. |
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| 21:59.06 | jamesaxl | Dovid: you can try something like Set(ARRAY(var1,var2)=1,2) |
| 21:59.53 | Dovid | jamesaxl: I realized it wont work. I need a lot lonfer then 255 characters |
| 22:00.24 | jamesaxl | Dovid: did you try HASH? |
| 22:00.33 | Dovid | I did not. I will look @ it in a sec |
| 22:07.10 | Samot | Dovid: There's no Array() option. |
| 22:07.33 | Samot | In regards to a data array, be it associative or not. |
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