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| 09:17.39 | Encrypt | HEllo o/ |
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| 09:54.26 | Encrypt | I still have this question... |
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| 09:55.00 | Encrypt | Is it possible to connect to my remote asterisk server which is behind a NAT as well as I am behin a NAT? |
| 09:55.16 | Encrypt | With a kind of VPN technlogy |
| 09:55.34 | Encrypt | I thought about doing an SSH tunnel |
| 09:55.49 | Encrypt | But since SSH is based on TCP, it is not possibe |
| 09:55.52 | Encrypt | possible* |
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| 10:42.17 | catphish | i seem to be experiencing a nasty crash, can anyone have a quick look and point me in the direction of what might be causing this? http://paste.codebasehq.com/pastes/hys0ngkntdml2lx1j0 |
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| 10:49.16 | catphish | i've unregistered statis, i hope this helps |
| 10:52.22 | catphish | i think using module autoload was a mistake |
| 10:52.55 | catphish | my realtime problems yesterday that i thought were caused by "sip reload", were actually caused by the whole app crashing :( |
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| 13:11.16 | twanny796 | how can i determine a .bin sound file format? |
| 13:11.41 | WIMPy | Try file. Maybe you're lucky. |
| 13:12.11 | twanny796 | hello WIMPy ; file just gave me data /> |
| 13:12.24 | WIMPy | Bad luck then. |
| 13:12.50 | WIMPy | If you have no other clues, you will have to experiment. |
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| 13:13.46 | twanny796 | WIMPy: experiment with what? |
| 13:14.12 | WIMPy | verious formats until it sounds like something. |
| 13:14.34 | WIMPy | That requires some luck, off course. |
| 13:16.03 | twanny796 | WIMPy: with which software? |
| 13:16.29 | WIMPy | I'd try sox or maybe audacity. |
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| 13:20.27 | cw1972 | open it in a hex editor, there may be a header to identify the format |
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| 13:21.02 | twanny796 | cw1972: trying |
| 13:21.33 | Reapster | heya, if I delete endpoint, aor and auth via pjsip dynamic config (/ari/asterisk/config/dynamic/res_pjsip/{endpoint,aor,auth}), the endpoint still shows up in /ari/endpoints, is there some other way to really delete an endpoint? |
| 13:22.20 | file | where are they stored? |
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| 13:23.05 | Reapster | whatever the default is, I haven't set up the sorcery stuff yet, so in memory? |
| 13:23.22 | file | the only default is to pull from config file |
| 13:23.27 | file | which is read-only |
| 13:23.34 | twanny796 | cw1972: excuse my ignorance which hex editor in linux? |
| 13:25.17 | Reapster | I'll configure sorcery for it and check again then, restarting asterisk got rid of everything |
| 13:25.20 | Reapster | thanks file |
| 13:26.21 | file | if you have stuff configured properly so it has a real place to go and deleting it doesn't completely make it go again, then I'd file a bug - there may be a lifecycle problem with the internal Asterisk endpoint and the PJSIP endpoint where one disappears and the other doesn't |
| 13:26.28 | file | it wasn't written with push in mind originally |
| 13:26.44 | file | the end result being it may still appear in /ari/endpoints but the PJSIP endpoint would be gone |
| 13:27.23 | twanny796 | cw1972: HARM_52 ??? |
| 13:29.12 | Reapster | oh ok, thanks for the heads up |
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| 13:34.46 | Zogot | is it right that you get 2 PeerStatus events in the AMI? Asterisk 13.3 |
| 13:35.36 | WIMPy | You probably get a lot. For what exactely? |
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| 13:37.26 | gonzapata | hello everybody |
| 13:38.05 | Zogot | WIMPy: for the same peer, i should of clarified further |
| 13:38.16 | gonzapata | I need some help with libss7 I have tried several asterisk version dahdi etc and asterisk crash |
| 13:38.50 | WIMPy | Zogot: The question remains: For what exact occasion? |
| 13:39.46 | Zogot | when a phone unregisters/registers, i get the exact same event for peerStatus twice |
| 13:39.58 | Zogot | https://gist.github.com/zogot/9aee2e2ecfacfe8a3375 |
| 13:41.19 | WIMPy | I can't look at that, but if it's only one account registering, I'd expect only one event. |
| 13:41.38 | Zogot | WIMPy: same :p lol |
| 13:41.45 | Zogot | i get only 1 for the others |
| 13:41.48 | Zogot | DeviceStateChange and |
| 13:41.52 | Zogot | ChallengeSent |
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| 13:44.10 | catphish | i seem to be experiencing a nasty crash, can anyone have a quick look and point me in the direction of what might be causing this? http://paste.codebasehq.com/pastes/hys0ngkntdml2lx1j0 |
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| 13:44.29 | WIMPy | Oh, is chan_pjsip that verbose? Anyway, file may be able to tell you more if he has the chance to spot it. |
| 13:44.45 | catphish | i've taken the step of removing my statis module, i hope this will prevent the problem for now |
| 13:45.01 | WIMPy | catphish: Strings with embennded newlines? |
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| 13:45.41 | catphish | WIMPy: i have no idea where the invalid strings are coming from, or what's trying to json encode them :( |
| 13:46.19 | WIMPy | And it's killing the whole process? |
| 13:46.21 | file | Zogot, https://issues.asterisk.org/jira/browse/ASTERISK-25116 |
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| 13:46.28 | catphish | WIMPy: yes :( |
| 13:46.43 | WIMPy | That's very evil. |
| 13:47.11 | catphish | my verbosity was not very high, that paste is all the info i have |
| 13:47.24 | Zogot | file: hmm thanks for the link. I am currently on chan_sip, so perhaps someone should comment it has issues in that also |
| 13:47.46 | Zogot | I could also make that comment :p |
| 13:47.52 | file | you could. |
| 13:48.02 | catphish | i've decided to stay on chan_sip and realtime for now, i realised the reason i was losing my peers was purely because asterisk was crashing :( |
| 13:48.04 | seik0 | some says Hangup() in h is not very good (undefined, if strictly), but what is the best way just to stop working in h-extension ? |
| 13:48.58 | seik0 | because i have "_." exten in same context and must define special extens to not be overwritten by "_." |
| 13:49.11 | catphish | the ... in my paste isn't actually actually where lines have been removed, it literally restarts immediately after that JSON error :( |
| 13:49.29 | WIMPy | Oh, chan_sip generates a ChallengeSent event? Never seen that. |
| 13:50.39 | WIMPy | seik0: The best way to use a _. is if you are able to Goto() another context where you don't have that pattern so the special extensions work. |
| 13:51.28 | seik0 | WIMPy: even so first step with undefined "h" exten will make the same goto |
| 13:51.58 | WIMPy | Yes, but you won't have anythign happening at that point, yet. |
| 13:52.48 | seik0 | doesn't make any sense to me |
| 13:52.58 | WIMPy | You won't get to the h extension for a call that hasn't been answered. |
| 13:55.46 | seik0 | but if I really need "_." exten and then make Goto() to Some-Context, that Some-Context also should contain "_." exten and I will face the same problem |
| 13:56.06 | [TK]D-Fender | it should NOT contain that pattern |
| 13:56.17 | WIMPy | That depends on what you want to do. |
| 13:56.18 | [TK]D-Fender | jump OUT of the context that has it |
| 13:57.02 | seik0 | but it can only be achieved by excluding "_." from base context |
| 13:57.41 | WIMPy | If you do a databse lookup for ${EXTEN} ot such, you could still Goto(SomeOtherContext,AA${EXTEN},1) and lookup ${EXTEN:2} in '_AA.' and would be safe, |
| 13:57.46 | seik0 | so there should be one place where need for "_." should be resolved |
| 13:58.02 | seik0 | and in that place I may problably face h-problem |
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| 13:58.43 | WIMPy | And y ou can still have a h extension in the same context, off course. |
| 13:58.48 | seik0 | WIMPy: AA${EXTEN} looks a good workaround |
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| 13:59.14 | seik0 | thanks |
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| 14:31.16 | seik0 | surprisingly magical, but when Originating (say, with call-file) to Local channel , variables in call-file set with Set exist in both legs of Originate |
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| 14:44.30 | [red] | so in asterisk 13, are there any different settings for getting voicemail odbc storage working from 11? i'm pretty confident i have everything right, but asterisk keeps saving voicemails to /var/spool instead of the DB |
| 14:44.39 | [red] | no error messages generated by asterisk |
| 14:45.11 | [TK]D-Fender | * will only load one module to manager voicemail. Apparently you re letting it load the statick file storage one. |
| 14:45.43 | [red] | hmmm, good point |
| 14:45.48 | [red] | i'll give that a try |
| 14:45.54 | [TK]D-Fender | modules.conf <- |
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| 14:51.17 | [red] | any idea of what the module name is? i'm not finding it with CLI> module show |
| 14:52.37 | dajaz27 | What does chan_sip.c:15307 -- Registration for sip provider timeout mean |
| 14:52.40 | [TK]D-Fender | look in your modules folder |
| 14:52.54 | file | Asterisk doesn't have separate modules for voicemail by default, it's chosen at compile time in menuselect under the "Voicemail Build Options" |
| 14:53.04 | [TK]D-Fender | dajaz27: Means * didn't get an answer. You should be looking at what it requested and where it sent that request |
| 14:53.25 | [TK]D-Fender | file: Used to have multiple .so's , when did that change? |
| 14:53.34 | file | never? it has always been like this |
| 14:53.46 | file | there's outside patches to build separate ones, but afaik noone has ever submitted them |
| 14:53.47 | [TK]D-Fender | recalls seeing a piel before... |
| 14:53.53 | [TK]D-Fender | pile* |
| 14:54.56 | file | or they just do it as part of the build process |
| 14:55.00 | dajaz27 | Thank you. |
| 14:55.57 | [red] | thanks guys, it was the compiler flag |
| 14:56.05 | [red] | feels like he should have known this |
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| 15:01.28 | [red] | a warning if you set the odbcstorage= config in voicemail.conf and have directory storage compiled would be super nice :) |
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| 15:58.01 | SpeakerToMeat | I'll add an extra question, what's a good brand and/or model of sip phone that is not too expensive (like big brands), but has good build quality and software in general? |
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| 15:58.21 | [TK]D-Fender | Polycom > All |
| 15:58.43 | [TK]D-Fender | Aastra runs 2nd place, 3rd starts tying up with Yealink, Digium, Snom, etc |
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| 15:59.37 | SpeakerToMeat | Normally I wouldn't consider Polycom on the "not too expensive" realm. But maybe I'm mistaken? |
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| 16:02.02 | [TK]D-Fender | Depends where you are |
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| 16:02.33 | [TK]D-Fender | If you're in North America then there is no reason to cheap out for anything else |
| 16:02.45 | SpeakerToMeat | Not in north america |
| 16:03.05 | [TK]D-Fender | It then comes down to features as to whether Aastra has a partuculr model that is slightly better suited to your intended usage |
| 16:03.25 | [TK]D-Fender | Cisco's SPA series are "good enough" as well. |
| 16:03.45 | [TK]D-Fender | And are generally a top value/$ elsewhere |
| 16:06.35 | igcewieling1 | Has anyone ever solved an issue where Polycom phones seemingly randomly lose the contact directory or revert the contact directory to a previous version. Seems to happen most often on firmware upgrades, but not always. |
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| 16:17.20 | SpeakerToMeat | I guess you guys deal with all sorts of users.... have you found good methods, or low resistance methods of allowing users to manage their own speed dial and/or manage the directory? |
| 16:18.06 | WIMPy | always thought the phones do that. |
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| 16:25.08 | [TK]D-Fender | They do. |
| 16:25.20 | [TK]D-Fender | by default all phone directories are controlled by the phone itself |
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| 16:28.26 | igcewieling1 | by default the phone uploads its contacts to MAC-directory.xml |
| 16:34.04 | SpeakerToMeat | Ok, thanks |
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| 17:23.21 | [red] | igcewieling1: yes. you need to generate a directory listing for each polycom phone. the polycom phones only load the default diretory once if a phone specific directory doesn't exist |
| 17:23.29 | [red] | i made a script to help with this issue |
| 17:23.35 | [red] | at least for freepbx |
| 17:24.09 | [red] | https://git.talkroute.com/red/freepbx-polycom-directory/blob/master/updateContactDirectory.sh |
| 17:27.21 | igcewieling1 | [red]: I am familiar with creating phone specific contact directories. I'm not sure how that script would help with the contacts the person put in their contact directory. |
| 17:27.33 | [red] | oh gotcha |
| 17:28.05 | [red] | yeah, my experience with polycom is that either you can have contacts managed by a central server or manged by the end user |
| 17:28.09 | [red] | both doesn't work out well |
| 17:28.22 | igcewieling1 | [red]: contacts are managed by the local user. |
| 17:28.40 | [red] | and it's still being overwritten/lost? |
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| 17:29.23 | igcewieling1 | [red]: in VERY recent Polycom firmwares you can have a real global directories, see the release notes for the 5.x or 5.1.x firmwares. |
| 17:29.47 | igcewieling1 | [red]: yes. seemingly randomly, but might happen most often during firmware upgrades. |
| 17:29.59 | [red] | oh gotcha, sorry i haven't worked with the newer firmwares yet, don't think i may be of much help |
| 17:30.30 | igcewieling1 | Had a customer complaining today that the 90 contacts they had put into their phone were all gone this morning. Happens once a month or so. |
| 17:31.17 | igcewieling1 | Once a month or so across the 1,000+ phones I manage. |
| 17:31.23 | [red] | ouch |
| 17:32.23 | igcewieling1 | not all 1,000 phones, just one or two out of the 1,000. |
| 17:33.13 | igcewieling1 | I suspect the "correct" solution is to set up an ldap server at each client. |
| 17:33.53 | [red] | that, or, maybe even a quick web script where folks can manage their contacts through a web interface |
| 17:34.03 | [red] | and then that generates the xml files for the phones |
| 17:44.12 | igcewieling1 | [red]: heh, we did that about 5 years ago. Doesn't work nearly as well as it sounds. |
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| 18:31.57 | zopsi | Is there any reason that explains upon reboot asterisk isn't recognizing my G.729 and DPMA licenses in /var/lib/asterisk/licenses? It is owned by asterisk:asterisk. |
| 18:32.53 | TazzNZ | zopsi, what do you have to do to get it to work ? |
| 18:33.41 | zopsi | TazzNZ: what do you mean? They were being recognized fine before. |
| 18:33.58 | zopsi | TazzNZ: currently it does not work |
| 18:34.16 | TazzNZ | right - so it was working, then your rebooted, and now it doesn't work at all ? |
| 18:34.34 | TazzNZ | (well DPMA and g729 :) ) |
| 18:34.35 | zopsi | well I can start asterisk and everything is fine except the licensed modules |
| 18:34.45 | TazzNZ | right |
| 18:34.47 | zopsi | I'm on 13.5 |
| 18:35.03 | TazzNZ | if you open the asterisk cli |
| 18:35.41 | zopsi | it is open |
| 18:36.13 | TazzNZ | and type module show like 729 |
| 18:36.17 | TazzNZ | what does it show ? |
| 18:36.42 | zopsi | codec_g729a.so Digium G.729 Annex A Codec (optimized fo 0 Running core |
| 18:37.23 | TazzNZ | and repeat that with "digi" instead of 729 |
| 18:38.46 | zopsi | https://pastebin.com/uhGwEsYA |
| 18:39.14 | TazzNZ | so why do you think it's not working ? |
| 18:39.26 | zopsi | it doesn't detect the license and thus doesn't function |
| 18:40.06 | zopsi | https://pastebin.com/40nNSrqT |
| 18:40.26 | zopsi | I've confirmed my two licenses are in /var/lib/asterisk/modules and accessible by asterisk user |
| 18:40.28 | TazzNZ | unload and reload the module |
| 18:40.30 | TazzNZ | module unload res_digium_phone.so |
| 18:40.31 | TazzNZ | and |
| 18:40.35 | TazzNZ | module load res_digium_phone.so |
| 18:40.48 | TazzNZ | pastebin the load please |
| 18:41.12 | zopsi | hm |
| 18:41.37 | zopsi | https://pastebin.com/kVet9VgG |
| 18:42.45 | TazzNZ | module reload res_digium_phone.so |
| 18:42.50 | TazzNZ | that might do it too |
| 18:43.25 | zopsi | https://pastebin.com/nttNYM6g |
| 18:44.38 | TazzNZ | what does "/var/lib/asterisk/licenses" show? |
| 18:45.18 | zopsi | a ls or in cli? |
| 18:45.28 | TazzNZ | whoops |
| 18:45.30 | TazzNZ | yes - ls |
| 18:45.30 | TazzNZ | :D |
| 18:46.17 | zopsi | https://pastebin.com/DKDnG6q3 |
| 18:49.24 | TazzNZ | the only digg's on mine is that I don't have group write |
| 18:49.35 | TazzNZ | I am not sure that will change anything |
| 18:49.41 | TazzNZ | but let's try it |
| 18:50.14 | zopsi | sure |
| 18:50.37 | TazzNZ | chmod 644 -v /var/lib/asterisk/licenses/ |
| 18:50.51 | TazzNZ | chmod 644 -v /var/lib/asterisk/licenses/* |
| 18:50.54 | TazzNZ | forgot the * |
| 18:51.41 | zopsi | still problem |
| 18:51.50 | zopsi | how can I tell if my hostid changed? |
| 18:52.00 | TazzNZ | when you reload it still can't find it ? |
| 18:52.16 | TazzNZ | why do you suspect your hostid has changed ? |
| 18:52.17 | zopsi | yeah still nothing |
| 18:53.09 | zopsi | no reason I didn't change hardware or anything but I'm just trying to think of things |
| 18:53.55 | TazzNZ | it won't "just change" |
| 18:55.32 | zopsi | well apparently it did... |
| 18:55.38 | zopsi | how is the host-id calculated |
| 18:56.00 | zopsi | my g729 license that was working yesterday has a different host ID than the one that shows when reloading the module. |
| 18:56.31 | TazzNZ | In general, the host ID is not allowed to change on a given host. Since the host ID is based on hardware fingerprinting, Digium acknowledges that sometimes hardware and systems fail, so it is necessary to move the keys to new hardware |
| 18:56.36 | igcewieling1 | zopsi: based on the MAC addresses on the system. |
| 18:56.42 | igcewieling1 | I don't know any more than that. |
| 18:56.58 | zopsi | literally have no changed anything hardware on my server. |
| 18:57.02 | zopsi | not* |
| 18:57.16 | TazzNZ | http://kb.digium.com/articles/FAQ/Introduction-to-Digium-Asterisk-Module-Licensing |
| 19:01.53 | zopsi | I have no idea how that happned, but I used my last registration on my g729 and DPMA and it fixed it |
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| 19:32.16 | c|oneman | wireshark playback of calls crash for anyone else? |
| 19:32.31 | c|oneman | click on the waveform and *poof* |
| 19:44.26 | SpeakerToMeat | Is there a set out there of CC licensed basic default system recordings? |
| 19:45.07 | SpeakerToMeat | Well... actually, freepbx seems to come with a set, only, I can't hear them :/ I'll keep investigating |
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| 23:42.52 | zopsi | on inbound calls I get a bunch of "res_pjsip/pjsip_distributor.c:221 find_dialog: Could not find matching INVITE transaction for CANCEL request" as well as 9 inbound calls for the same instance |
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