| 00:25.20 | *** join/#asterisk NightMonkey (~NightMonk@pdpc/supporter/professional/nightmonkey) |
| 01:03.37 | *** join/#asterisk TandyUK (~admin@2a02:13a0:a006:1:bc94:11f9:a606:1cb4) |
| 01:04.09 | *** join/#asterisk andresmujica (~andresmmu@ubuntu/member/andresmujica) |
| 01:07.33 | *** join/#asterisk mmlj4 (~mmlj4@47-44-49-2.static.unas.mo.charter.com) |
| 01:27.39 | *** join/#asterisk kraylo (~kraylo@unaffiliated/kraylo) |
| 02:47.27 | *** join/#asterisk fonefreak (~root@c-76-105-87-143.hsd1.ga.comcast.net) |
| 04:22.28 | *** join/#asterisk randolfcri (7a35b6ca@gateway/web/freenode/ip.122.53.182.202) |
| 05:09.10 | *** join/#asterisk lankanmon (~LKNnet@CPE64777dd7e053-CM64777dd7e050.cpe.net.cable.rogers.com) |
| 05:25.01 | *** join/#asterisk gerhard7 (~gerhard7@ip5657ee30.direct-adsl.nl) |
| 05:51.32 | *** join/#asterisk Penguin (~xwQ5kwYl6@our.systems.are.full.of.penguins.at.penguinsystems.net) |
| 06:14.21 | *** join/#asterisk u0m3_ (~u0m3@188.25.18.156) |
| 06:23.05 | *** join/#asterisk pchero (~pchero@109.70.54.56) |
| 07:07.27 | tuxd00d | Is it expected behavior for âremove_existing=yesâ to limit AORs to a single Contact, despite âmax_contactsâ being set to something greater than 1? |
| 07:08.34 | *** join/#asterisk pchero_work (~pchero@109.70.54.56) |
| 07:15.14 | tuxd00d | Oh, somewhat addressed in ASTERISK-27192 |
| 07:35.36 | *** join/#asterisk gerhard7 (~gerhard7@ip5657ee30.direct-adsl.nl) |
| 07:39.43 | *** join/#asterisk sekil (~sekil@cable-89-216-222-166.dynamic.sbb.rs) |
| 07:57.39 | *** join/#asterisk DanB (~DanB@clt-195.192.204.82.ip-anschluss.net) |
| 08:17.08 | *** join/#asterisk hehol (~hehol@gatekeeper.loca.net) |
| 08:20.11 | *** join/#asterisk [d__d] (~d__d]@ec2-54-85-45-223.compute-1.amazonaws.com) |
| 08:46.27 | *** join/#asterisk MacroMan (~MacroMan@host213-123-31-77.in-addr.btopenworld.com) |
| 09:11.46 | *** join/#asterisk boris_t (~boris_t@82.112.41.207) |
| 09:18.08 | *** join/#asterisk somepoortech (~somepoort@72-0-128-228.static.firstlight.net) |
| 09:19.08 | *** join/#asterisk defsdoor (~andy@cpc120600-sutt6-2-0-cust177.19-1.cable.virginm.net) |
| 09:33.44 | *** join/#asterisk Mr_Pleb_Mgoo (~jakeb@103.46.213.148) |
| 09:36.45 | *** join/#asterisk DanB_ (~DanB@fw1.nextpertise.nl) |
| 09:42.50 | *** join/#asterisk DanB (~DanB@clt-195.192.204.82.ip-anschluss.net) |
| 09:44.08 | *** join/#asterisk DanB (~DanB@clt-195.192.204.82.ip-anschluss.net) |
| 11:03.50 | *** join/#asterisk fonefreak (~root@c-76-105-87-143.hsd1.ga.comcast.net) |
| 11:24.13 | *** join/#asterisk DanB_ (~DanB@fw1.nextpertise.nl) |
| 11:28.11 | *** join/#asterisk sh_smith (foobar@cpe-76-174-26-91.socal.res.rr.com) |
| 11:36.17 | DanQuinney | I've got an issue with a number of * servers not replying to packets every so often, tcpdump showing this, but nothing in *.log |
| 11:36.33 | DanQuinney | running latest 13.17.0 |
| 12:01.34 | *** join/#asterisk sekil (~sekil@cable-89-216-221-254.dynamic.sbb.rs) |
| 12:33.51 | *** join/#asterisk pawiecki (~pawiecki@router.dir.pl) |
| 12:45.20 | *** join/#asterisk [TK]D-Fender (~joe@216.191.106.165) |
| 12:51.30 | *** join/#asterisk cervajs2 (~cervajs2@gatekeeper.bm.ipex.cz) |
| 12:52.46 | Samot | DanQuinney: That is vague. |
| 12:53.09 | cervajs2 | any ideas what to try in https://issues.asterisk.org/jira/browse/ASTERISK-15421 ? |
| 12:53.32 | DanQuinney | I'm afraid I currently don't have much to go on Samot, I've turned on debug logging to see if that sheads any further light |
| 12:54.02 | DanQuinney | but as it stands, for ~ 60s every so often * will seemingly not respond to any packets |
| 12:54.24 | DanQuinney | then continue as if nothing had happened |
| 12:56.57 | file | what kind of packets? |
| 12:57.38 | DanQuinney | all SIP packets, INVITES, OPTIONS etc |
| 12:57.51 | DanQuinney | *INVITE's, OPTION's |
| 12:58.52 | DanQuinney | almost as if * is sat processing a packet and until it has finished it won't do anything else |
| 12:58.59 | file | if you are using chan_sip and it is UDP traffic then it is single threaded |
| 12:59.10 | DanQuinney | using chan_pjsip |
| 12:59.15 | DanQuinney | *pjsip |
| 12:59.28 | file | chan_pjsip isn't single threaded in that regard, it dispatches things |
| 12:59.34 | DanQuinney | and yes, it's all UDP |
| 12:59.42 | file | are you using realtime? |
| 13:00.07 | DanQuinney | no, got rid a year ago as we were having issues with it - it's all static config nowadays |
| 13:00.36 | file | then I'd suggest using https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace#GettingaBacktrace-GettingInformationForADeadlock |
| 13:00.43 | file | when it is in that state, as it will show where it is hanging |
| 13:00.45 | file | and filing an issue |
| 13:00.50 | file | also ensure you are using bundled PJSIP |
| 13:01.23 | DanQuinney | thanks file |
| 13:02.06 | file | I could have compressed that into two sentences I suppose, re-reading it is makes it seem I'm slightly crazy |
| 13:02.21 | file | and I just used is instead of it, gasp |
| 13:03.28 | DanQuinney | ban-able offence is it not? |
| 13:03.50 | *** kick/#asterisk [file!uid178970@asterisk/developer-and-muffin-lover/file] by file (Use proper English.) |
| 13:03.53 | *** join/#asterisk file (uid178970@asterisk/developer-and-muffin-lover/file) |
| 13:03.53 | *** mode/#asterisk [+o file] by ChanServ |
| 13:19.04 | *** join/#asterisk newtonr (newtonr@nat/digium/x-pqofyncdnbsrpqbv) |
| 13:19.04 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 13:29.55 | dan_j | Has anyone elses asterisk v13 started to randomly stop responding to packets every so often? Ive got two seperate servers and both started doing it at the same time. Ping tests have no packet loss so I dont think its a network issue. |
| 13:30.13 | DanQuinney | YES |
| 13:30.21 | DanQuinney | I have dan_j |
| 13:30.34 | Samot | We need to see some debugs |
| 13:30.35 | dan_j | They stop responding for a few seconds, not at the same time. |
| 13:30.38 | DanQuinney | currently recompiling * as per file's suggestion |
| 13:30.41 | DanQuinney | https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace#GettingaBacktrace-GettingInformationForADeadlock |
| 13:30.54 | Samot | First, is it a deadlock? |
| 13:31.03 | Samot | We haven't seen any debugs of these failures. |
| 13:31.18 | DanQuinney | that's what I'm getting now Samot |
| 13:31.35 | dan_j | Its not deadlocking. After a few seconds it continues running, but then the logs show many endpoints needing to reconnect. |
| 13:32.19 | Samot | Do the endpoints continue to respond to Asterisk OPTIONS messages? |
| 13:32.24 | DanQuinney | how long are you seeing the packets being ignored for dan_j? We're seeing upwards of 60 seconds on some occasions |
| 13:33.34 | Samot | Do the endpoints continue to respond to Asterisk OPTIONS messages? |
| 13:33.36 | dan_j | So far, within 60 seconds. Ive got an external company monitoring every 60 seconds and receive a DOWN notification followed by an UP notification 60 seconds later. |
| 13:33.40 | Samot | ^^ Both of you |
| 13:34.16 | Samot | dn_j: How is the external company monitoring? What are they doing for this? |
| 13:34.21 | Samot | Are they just pinging an IP? |
| 13:34.31 | Samot | Or are they actually sending OPTIONS? |
| 13:34.45 | dan_j | Ping and SIP monitoring. Its the sip that is being reported as down. |
| 13:34.50 | DanQuinney | Samot: for me we're sending options and * isn't responding to them |
| 13:35.00 | Samot | That wasn't my question |
| 13:35.03 | Samot | Do the endpoints continue to respond to Asterisk OPTIONS messages? |
| 13:35.05 | dan_j | They send sip options and any reply means UP |
| 13:35.29 | Samot | Do things that Asterisk is sending OPTIONS to get replied back during this time? |
| 13:35.34 | DanQuinney | we don't have * send OPTIONS requests |
| 13:35.46 | Samot | You don't have Asterisk qualify your endpoint peers? |
| 13:35.52 | DanQuinney | no |
| 13:35.56 | Samot | You let everything be unmonitored? |
| 13:36.20 | Samot | So when you do a "sip show peers" |
| 13:36.35 | Samot | You never see an OK or UNAVAILABLE or UNREACHABLE for an endpoint? |
| 13:36.56 | DanQuinney | we us pjsip, so obviosuly sip show peers doesn't return anything |
| 13:37.04 | Samot | No need to split hairs |
| 13:37.11 | Samot | When you do a pjsip show endpoints then |
| 13:37.14 | Samot | You know what I mean. |
| 13:37.20 | dan_j | Samot: i have qualify on so i do send options. ive had to step out of the office so I cannot confirm for a couple of hours. Just thought Id ask if anyone had an issue while im waiting for the kids. |
| 13:37.32 | Samot | Well |
| 13:37.40 | Samot | I'm trying to see if this is a two way issue |
| 13:37.56 | DanQuinney | all contacts show "Unknown"; |
| 13:38.07 | Samot | So you're not even monitoring your own endpoints. |
| 13:38.46 | DanQuinney | no, Kamailio sits between the endpoints and * boxes |
| 13:38.46 | dan_j | Samot: there is a major issue with qualify at the moment. So i understand why DanQuinney wouldnt want to use it |
| 13:38.51 | Samot | Are these PBXes public or behind NAT? |
| 13:38.58 | DanQuinney | public Samot |
| 13:39.02 | dan_j | Public |
| 13:39.05 | Samot | I don't have any issues with qualify |
| 13:39.16 | Samot | I have 13.17 boxes running |
| 13:39.33 | Samot | And if there were issues, I would a) see them and b) my phone would be ringing off the hook. |
| 13:39.51 | dan_j | Samot: it affects startup times and causes asterisk to freeze when doing pjsip reload. There are already open issues relating to it. |
| 13:39.59 | Samot | Ahh. |
| 13:40.02 | Samot | So this is PJSIP related. |
| 13:40.06 | Samot | That would be why. |
| 13:40.37 | dan_j | DanQuinney: you are using pjsip too? |
| 13:40.48 | Samot | I don't use PJSIP |
| 13:40.49 | DanQuinney | I am |
| 13:40.55 | Samot | It doesn't offer me anything special. |
| 13:41.25 | sekil | does outboundproxy setting work correctly with pjsip |
| 13:41.43 | Samot | DanQuinney: I mean you have Kamailio, so what is PJSIP giving you that Chan_SIP doesn't? |
| 13:41.45 | file | works more correctly than chan_sip if you configure it correctly |
| 13:41.48 | dan_j | I switched to make use of resourcelists. |
| 13:42.23 | sekil | file: was that for me? |
| 13:42.31 | file | yes |
| 13:42.46 | sekil | file: ah..thanks..wasn't working at all well with chan_sip.. |
| 13:42.59 | dan_j | DanQuinney: when version of asterisk are you currently on? |
| 13:43.10 | DanQuinney | 13.17 |
| 13:43.16 | DanQuinney | upgraded on Saturday |
| 13:43.17 | Samot | Guys.. |
| 13:43.20 | sekil | should try RLS |
| 13:43.25 | Samot | And file don't take this the wrong way |
| 13:43.28 | Samot | But your providers. |
| 13:43.35 | Samot | Stop using PJSIP |
| 13:44.06 | sekil | dan_j: which phones do you use with rls? |
| 13:44.10 | dan_j | DanQuinney: ok. So no point me updating then :) |
| 13:44.52 | Samot | Again, guys.. |
| 13:44.52 | file | Samot: people can use what they want in my book, 'tis up to them |
| 13:44.54 | dan_j | sekil: panasonic kx-ut133 at the mo. |
| 13:45.09 | sekil | dan_j: ah good..I have those...works fine? |
| 13:45.11 | DanQuinney | dan_j: what are you on? |
| 13:45.18 | Samot | file: I know. |
| 13:45.43 | Samot | file: But as a VoIP provider, PJSIP is still wonky. |
| 13:45.50 | dan_j | 13.5 at the moment. Ive been waiting for a couple of patches to come through before updating. |
| 13:46.49 | dan_j | Samot: beginning to think you are right. Was thinking of leaving asterisk and developing my own pbx but maybe switching back to chansip might be better. The lack of resourcelists in chansip is an issue though. |
| 13:47.30 | *** join/#asterisk brad_mssw (~brad@66.129.88.50) |
| 13:47.41 | *** join/#asterisk cresl1n (Adium@asterisk/libpri-and-libss7-expert/Cresl1n) |
| 13:47.42 | *** mode/#asterisk [+o cresl1n] by ChanServ |
| 13:47.42 | Samot | dan_j: That's why Asterisk is never the only thing in a VoIP providers network. |
| 13:48.35 | dan_j | Samot: ive considered other software but at the moment, ive only got asterisk handling my sip traffic. |
| 13:48.37 | DanQuinney | Samot: tbf, I've not run into any other issues with pjsip since we moved to it almost a year ago |
| 13:48.45 | Samot | OK. |
| 13:49.08 | Samot | But again.. |
| 13:49.12 | Samot | What is it offering you? |
| 13:49.23 | DanQuinney | I think saying the pjsip stack is "wonky" is a little unfair |
| 13:49.43 | Samot | By wonky I mean that it is a source of a lot of problems still. |
| 13:49.47 | dan_j | I agree with Samot due to the issues with qualify. |
| 13:49.56 | Samot | And it still doesn't have all the features of Chan_SIP |
| 13:50.32 | Samot | I'm not saying that PJSIP is bad, but in the grand scheme you have to look at it all at a 5K view. |
| 13:51.39 | DanQuinney | https://www.irccloud.com/pastebin/Yh7B8TNO/ |
| 13:51.42 | DanQuinney | ah |
| 13:52.40 | dan_j | How did you get that? |
| 13:53.51 | *** join/#asterisk DanB (~DanB@clt-195.192.204.82.ip-anschluss.net) |
| 13:55.47 | DanQuinney | started appearing after recompiling with DEBUG_THREADS, MALLOC_DEBUG, DONT_OPTIMIZE and BETTER_BACKTRACES enabled |
| 13:56.30 | dan_j | Thanks. |
| 13:56.41 | file | DEBUG_THREADS and MALLOC_DEBUG were probably a bit heavy, for your specific issue compiling without is probably best |
| 13:57.26 | [TK]D-Fender | <Samot> And it still doesn't have all the features of Chan_SIP <- yup, missing 1 big one for me... |
| 13:59.10 | DanQuinney | I'm going to recompile shortly file, I was just following the link you sent me |
| 13:59.52 | dan_j | Not liking the thought of switching back to chansip |
| 13:59.54 | file | [TK]D-Fender: dynamic DNS supporty thingy was merged in and goes out in RCs, which should be tomorrow or early next week due to le holiday |
| 14:01.04 | [TK]D-Fender | :D |
| 14:01.19 | [TK]D-Fender | file, Next time you're here, muffins on me! |
| 14:01.24 | file | noted. |
| 14:01.45 | Samot | dan_j: I guess the question is, will it solve most of your issues.. |
| 14:02.13 | file | there's actually a way to make it cooler, but I'm not sure if I like it... in that it could do a STUN request periodically |
| 14:02.35 | file | so you just tell it to use stun and provided a STUN server is configured, it'd automatically find out your public IP and keep it up to date regularly |
| 14:03.00 | [TK]D-Fender | Just another lookup method. MOAR = GOODER! |
| 14:03.06 | Samot | I don't like it either. |
| 14:03.19 | Samot | Requires another piece of a equipment |
| 14:03.26 | file | it doesn't |
| 14:03.30 | file | there are public STUN servers |
| 14:03.36 | Samot | Right. |
| 14:03.36 | file | Google runs a cluster, for example |
| 14:03.46 | Samot | I didn't say it had to be mine |
| 14:03.53 | Samot | It's another outside point |
| 14:03.56 | file | the same applies for dynamic DNS |
| 14:04.03 | *** join/#asterisk woddy (~woddy@unaffiliated/woddy) |
| 14:04.09 | dan_j | Samot: at the moment, i dont know if pjsip is causing the freezing but switching to chansip will help with reloads and qualify. |
| 14:04.29 | Samot | True. |
| 14:04.36 | file | not required to be used regardless, and still bouncing around my head amongst the other stuff I'm doing |
| 14:04.53 | Samot | I can see how you're on the fence with it |
| 14:05.24 | Samot | dan_j: And if you have a customer base, those are important. |
| 14:07.19 | Samot | Plus I use FOP2 heavily. |
| 14:07.26 | Samot | So PJSIP isn't an option anyways. |
| 14:07.42 | dan_j | Samot: I'm torn. I was hopeful but it looks like its going to be quite a while before pjsip is production quality. |
| 14:07.56 | Samot | dan_j: That's kind of incorrect. |
| 14:08.05 | Samot | PJSIP is production.. |
| 14:08.16 | *** join/#asterisk bford (d8cff501@gateway/web/freenode/ip.216.207.245.1) |
| 14:08.17 | *** mode/#asterisk [+o bford] by ChanServ |
| 14:08.18 | Samot | It works fine for people with average deployments |
| 14:08.28 | Samot | You, on the other hand, are a VoIP provider |
| 14:08.38 | Samot | You have bigger user base |
| 14:08.39 | file | PJSIP hasn't been exposed to as many deployment types/environments as chan_sip, but for a lot of people it does work fine |
| 14:08.48 | Samot | ^^^^ |
| 14:09.04 | Samot | As a VoIP provider you are in a spot where the it hasn't been exposed enough |
| 14:09.11 | Samot | -the |
| 14:09.14 | file | in your usage it may not, thus running into such things |
| 14:09.17 | dan_j | Samot: one server only has 100 endpoints and it still has issues. 100 endpoints is a normal amount of some inhouse deployments. |
| 14:09.39 | Samot | My point is this. |
| 14:09.49 | Samot | You're not an admin for _a_ company PBX |
| 14:10.13 | Samot | So if PJSIP has a problem for a company PBX admin, that's an issue they have to resolve and figure out |
| 14:10.17 | dan_j | I agree that pjsip would work fine for a company with 5 endpoints. But larger companies wouldnt be able to use pjsip and qualify at the same time in its current state. |
| 14:10.28 | Samot | If PJSIP has a problem, you have A LOT more to answer for. |
| 14:10.51 | file | configuration also plays a part |
| 14:11.07 | Samot | ^^ That is 100% true |
| 14:13.47 | dan_j | Hmm. Really torn now but i think you are correct. I cant stay on pjsip at the moment. |
| 14:14.45 | Samot | I have a few on-premises PBX's running PJSIP for things.. |
| 14:15.02 | Samot | But in regards to my hosted or "cloud" servers for people... |
| 14:15.07 | Samot | I don't use it. |
| 14:15.34 | dan_j | Samot: not needed resourcelists yet? |
| 14:15.50 | Samot | No. |
| 14:15.52 | file | RLS, for all the work we put into them, aren't used by many from what I understand |
| 14:16.07 | Samot | Mainly because by the end of the year I'll have my own Presence server. |
| 14:16.16 | dan_j | Panasonic kx-ut133 goes really slow without rls if you make use of all 24 BLFs |
| 14:17.39 | dan_j | But if thats the only reason to stay with pjsip then i dont think its worth it. |
| 14:17.42 | sekil | dan_j: true dat..it can't even respond to pings properly.. |
| 14:18.15 | Samot | I don't use RLS |
| 14:18.18 | Samot | Don't have a need for it |
| 14:18.31 | dan_j | Can chansip bind to multiple ip addresses like pjsip can? |
| 14:18.41 | Samot | No. |
| 14:18.42 | sekil | dan_j: no |
| 14:19.00 | dan_j | Spent a long time on pjsip and cant remember all the cases that i tested and required. |
| 14:19.13 | Samot | How many IPs do you need for trunks? |
| 14:19.43 | dan_j | Each server has a public ip and a private ip for access over a vpn |
| 14:20.04 | Samot | OK. |
| 14:20.14 | Samot | So, they connect over the VP |
| 14:20.16 | Samot | VPN |
| 14:20.32 | *** join/#asterisk kharwell (kharwell@nat/digium/x-ecdbkojdsjgnbatg) |
| 14:20.32 | *** mode/#asterisk [+o kharwell] by ChanServ |
| 14:20.47 | Samot | Are you talking about endpoints or are you talking about trunks to providers? |
| 14:20.51 | dan_j | Most are public. A couple are vpn due to issues with their isp or local config. |
| 14:22.43 | Samot | Wait, are you having PJSIP listen on multiple IPs for different customers? |
| 14:23.00 | Samot | On the same PBX? |
| 14:23.22 | sekil | having the ability not to bind to 0.0.0.0 is really great |
| 14:23.35 | Samot | You don't have to bind to 0.0.0.0 |
| 14:23.39 | Samot | You can bind to the actual IP |
| 14:24.12 | sekil | yeah but one only |
| 14:24.31 | Samot | Why do you need to listen on more than one? |
| 14:24.34 | dan_j | PJSIP listens on 212.somethingpublic and also on 10.somethingprivate. Almost all clients connect to 212 but a handful connect to a vpn server which gives them access to 10 |
| 14:25.40 | [TK]D-Fender | so far that's 2. What's the 3rd you don't want it listening on? |
| 14:25.52 | dan_j | I cannot connect to 0.0.0.0 because the 212.somethingpublic is a floating ip in a cluster. I dont want it to bind on the non-floating ip. |
| 14:26.09 | sekil | Samot: I need to have the ability to different IPs....like when having many trunks and different routing tables |
| 14:26.14 | dan_j | Sorry, connect=bind |
| 14:26.14 | sekil | Samot: and IPs that overlap |
| 14:26.31 | Samot | Routing tables? |
| 14:26.34 | sekil | Samot: that's why I use FS as a SIP trunking gw forc client trunks |
| 14:26.41 | Samot | Oh, I don't use Asterisk as a switch. |
| 14:26.46 | sekil | Samot: yes |
| 14:26.55 | Samot | That's not what Asterisk is meant for. |
| 14:27.39 | sekil | Samot: i.e. you have sip trunk1 for customer1 ...sip trunk 2 for customer 2..usually they're delivered via same optic cable and having IPs in same network... |
| 14:27.53 | sekil | Samot: so you need to differentiate the profiles and binding IPs |
| 14:28.04 | Samot | I don't use different IPs for different trunks. |
| 14:28.06 | sekil | Samot: as well as routes to providers outbound proxy |
| 14:28.17 | Samot | There's no need to have customers on their own IPs for this. |
| 14:28.22 | sekil | Samot: so having to bind to 0.0.0.0 won't do |
| 14:29.03 | Samot | I don't see the need to give each customer their own IP to listen on for a SIP trunk. |
| 14:29.13 | Samot | They can all connect to the same IP without issue. |
| 14:29.36 | sekil | Samot: you either misunderstood or I didn't explained well |
| 14:29.53 | Samot | I'd go with the latter. |
| 14:30.02 | sekil | Samot: so c1 gets a SIP trunk from a provider..it's delivered in a closed network to my servers... |
| 14:30.09 | sekil | Samot: so does c2 from same provider.. |
| 14:30.27 | sekil | Samot: it's all done via different vlans...but provider is usually using same IP network |
| 14:30.32 | Samot | You're not the provider? |
| 14:30.35 | sekil | Samot: to use for connectivity |
| 14:31.02 | sekil | Samot: I am..but there are cases when I terminate a trunk for a client ..other than having to provide one.. |
| 14:31.32 | Samot | Oh you use places like VoIP Innovations and such |
| 14:31.34 | Samot | OK |
| 14:31.58 | sekil | Samot: voip innovations? |
| 14:32.11 | Samot | I cannot trunk my customers directly to my providers/carriers |
| 14:32.15 | Samot | They don't allow it |
| 14:32.30 | Samot | They only want a small number of IPs to send to. |
| 14:32.35 | Samot | Or block |
| 14:32.39 | sekil | Samot: I'm not in US..over here..there's no SIP trunks via Internet mostly |
| 14:32.57 | sekil | Samot: only via provider's transport |
| 14:33.18 | Samot | OK |
| 14:33.25 | Samot | So I have 100 customers connecting to my PBX |
| 14:33.28 | Samot | All with their own ISPs |
| 14:33.36 | Samot | I don't need 100 IPs listening for them |
| 14:33.39 | Samot | I have 1 ip |
| 14:33.43 | Samot | That's the IP they connect to |
| 14:33.44 | sekil | Samot: of course not |
| 14:33.56 | sekil | Samot: but that's not my point nor case.. |
| 14:34.23 | Samot | How many IPs do you need Chan_SIP to listen on? |
| 14:34.27 | Samot | That was my question. |
| 14:34.59 | sekil | Samot: I don't need it..I don't use it... |
| 14:35.11 | Samot | But my question was for dan_j. |
| 14:35.24 | sekil | Samot: I was saying why it's good to have the ability to specify binding ip.. |
| 14:35.27 | sekil | Samot: ah ok.. |
| 14:36.19 | Samot | You can specifically bind the IP. |
| 14:36.30 | Samot | I can have three IPs on my Asterisk server for other things. |
| 14:36.37 | Samot | I can tell Chan_SIP to bind to X IP. |
| 14:38.04 | dan_j | Samot: two IPs out of four |
| 14:38.34 | Samot | So you need Chan_SIP to listen on two IPs? |
| 14:38.40 | sekil | Samot: so if you have overlapping IPs.. |
| 14:38.52 | dan_j | Yes but not the other two |
| 14:40.51 | *** join/#asterisk catphish (~charlie@unaffiliated/catphish) |
| 14:42.01 | catphish | potentially obvious observation of the day: running asterisk 13 with DONT_OPTIMIZE makes it dramatically slower, running it with DEBUG_THREADS makes it unusably slow |
| 14:42.06 | *** join/#asterisk hehol (~hehol@gatekeeper.loca.net) |
| 14:42.27 | catphish | old versions of asterisk i ran with DONT_OPTIMIZE as a matter of course, but with asterisk 13 is actually causes quite serious performance issues in my environment |
| 14:47.21 | *** join/#asterisk RoyK (~roy@gateway/shell/layerbnc/x-lelxmznuwxxkentl) |
| 14:47.31 | *** part/#asterisk RoyK (~roy@gateway/shell/layerbnc/x-lelxmznuwxxkentl) |
| 14:52.03 | *** join/#asterisk andresmujica (~andresmmu@ubuntu/member/andresmujica) |
| 14:58.33 | *** join/#asterisk rmudgett (rmudgett@nat/digium/x-kvnyvpfdwigqvlzu) |
| 14:58.34 | *** mode/#asterisk [+o rmudgett] by ChanServ |
| 15:06.31 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 15:15.33 | *** join/#asterisk woddy1 (~woddy@unaffiliated/woddy) |
| 15:20.11 | qakhan | hi all, what is the reason on caller ID showing like sip:1234567892@x.x.x.x |
| 15:20.31 | Samot | Where is it showing that? |
| 15:21.15 | qakhan | on the Polycom IP phone |
| 15:21.23 | qakhan | Polycom 670 |
| 15:21.37 | Samot | Need to see how it is being sent to the phone. |
| 15:23.37 | qakhan | only calerid is being sent to phone |
| 15:24.02 | *** join/#asterisk u0m3 (~u0m3@188.26.47.10) |
| 15:30.07 | *** join/#asterisk mmlj4 (~mmlj4@47-44-49-2.static.unas.mo.charter.com) |
| 15:32.24 | catphish | hmm, seems i was wrong about DONT_OPTIMIZE, but DEBUG_THREADS definitely kills it :( |
| 15:33.03 | Samot | qakhan: Yes. we need to see how that CallerId is actually being sent to the phone. |
| 15:33.43 | Samot | qakhan: So you need to do a "sip set debug on" or a "pjsip set logger on" and call that device |
| 15:33.45 | Samot | ~pb |
| 15:33.45 | infobot | rumour has it, pastebin is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 15:33.49 | Samot | ^^ Paste the output there |
| 15:35.20 | *** join/#asterisk billxx (49958984@gateway/web/freenode/ip.73.149.137.132) |
| 15:45.41 | *** join/#asterisk Typhon (~Typhon@ipservice-092-218-202-083.092.218.pools.vodafone-ip.de) |
| 15:47.35 | *** part/#asterisk billxx (49958984@gateway/web/freenode/ip.73.149.137.132) |
| 15:47.46 | *** part/#asterisk cervajs2 (~cervajs2@gatekeeper.bm.ipex.cz) |
| 16:17.42 | *** join/#asterisk shanth_ (~shanth@wsip-98-182-126-226.ph.ph.cox.net) |
| 16:21.21 | *** join/#asterisk woddy (~woddy@unaffiliated/woddy) |
| 16:22.20 | *** join/#asterisk overyander (~jeff@12.49.160.131) |
| 16:23.21 | hdon | thanks again DivideBy0 |
| 16:23.59 | dan_j | Samot: just reviewing my traffic capture. it looks like when it happens, asterisk stops responding for about 6 seconds at a time, but does respond eventually. |
| 16:24.12 | dan_j | turning on debug to see if that sheds some light on things |
| 16:36.53 | *** join/#asterisk dakudos (~dakudos@c-73-243-248-133.hsd1.co.comcast.net) |
| 16:38.48 | dan_j | Looking at my pacemaker logs now. The AMI responded with the number of channels, but at the same time, pjsip was not responding. pjsip is being tested by pacemaker via 127.0.0.1 using sipsak. So now I know it's not a network issue but something pjsip related. |
| 16:39.17 | dan_j | nor is it an asterisk core issue as AMI was responding to queries. |
| 16:42.53 | *** join/#asterisk DanB_ (~DanB@fw1.nextpertise.nl) |
| 16:44.52 | dan_j | Please can someone explain this: res_rtp_asterisk.c: No remote address on RTP instance '0x7f339402a2d0' so dropping frame |
| 16:45.09 | dan_j | It repeats a number of times, at the same time that PJSIP stops responding. |
| 16:45.13 | file | it means that we haven't been told an address to send media to |
| 16:45.20 | file | so we can't send media |
| 16:46.04 | dan_j | https://pastebin.com/vmww5tcr |
| 16:46.12 | dan_j | Sorry, 1sec |
| 16:46.13 | Samot | private |
| 16:46.44 | dan_j | https://pastebin.com/vmww5tcr |
| 16:47.10 | dan_j | Could this be some form of DoS attack exploiting something? |
| 16:48.10 | *** join/#asterisk shanth__ (~shanth@wsip-98-182-126-226.ph.ph.cox.net) |
| 16:48.25 | Samot | [Aug 31 17:27:52] DEBUG[18142][C-0000069c] chan_pjsip.c: AST hangup cause 16 (no match found in PJSIP) |
| 16:48.29 | dan_j | It lasts for about 10 seconds with not much else going on. And ask you can see from the start and end of my pastebin, there is always something going on. |
| 16:48.40 | *** join/#asterisk DanB (~DanB@clt-195.192.204.82.ip-anschluss.net) |
| 16:50.32 | dan_j | Samot: Not sure why that would be. That customer is known and trusted. And this issue is happening on two servers I have running. I'll check the other next time it happens to see if it's similar. |
| 16:50.55 | Samot | You're relating this wrong. |
| 16:51.35 | dan_j | Please explain |
| 16:51.38 | Samot | You just said this happens during the "outage" period |
| 16:51.55 | Samot | When nothing is responding on PJSIP |
| 16:52.16 | dan_j | Yes. That collection of 'No remote address on RTP instance' happens when PJSIP isn't responding to SIP packets |
| 16:52.23 | Samot | I don't think the device being known/trusted has bearing. |
| 16:52.49 | dan_j | 'the device being known/trusted' = 'isn't a DoS attack' |
| 16:53.18 | file | DNS? |
| 16:53.34 | Samot | Good question |
| 16:53.40 | file | Where are the identify sections stored? |
| 16:53.41 | Samot | Are you using DNS for anything of this? |
| 16:54.44 | dan_j | Pacemaker is not using DNS to monitor asterisk. It's using 127.0.0.1 |
| 16:55.03 | dan_j | i'm using static files for the endpoints |
| 16:55.37 | *** join/#asterisk shanth_ (~shanth@wsip-98-182-126-226.ph.ph.cox.net) |
| 16:55.40 | file | Everything is in static files? |
| 16:56.02 | dan_j | Yep. I left realtime so that asterisk doesn't stall when the DB is unavailable |
| 16:56.11 | file | Is your system hostname in /etc/hosts? |
| 16:57.22 | dan_j | not fully. ie. not the FQDN, just the start of it. IE the hostname is voip1.example.com and /etc/hosts contains voip1 but not the end part. |
| 16:57.58 | dan_j | I've added it now. |
| 16:58.55 | dan_j | I'm using opendns for my dns servers |
| 16:59.09 | dan_j | opendns.com |
| 17:01.05 | dan_j | Out of interest, why could the hostname is /etc/hosts be relevant? |
| 17:06.18 | file | Asterisk uses that to determine local IP address in some places |
| 17:08.52 | dan_j | Ok. Ill see what happens now. Thanks. |
| 17:16.58 | *** join/#asterisk shanth_ (~shanth@wsip-98-182-126-226.ph.ph.cox.net) |
| 17:19.10 | DanQuinney | It won't make a difference dan_j, well, I have the fqdn set and it hasn't stopped the issue |
| 17:21.12 | Samot | Honestly, and don't take this the wrong way, dan_j probably needs to sit down and review all this. |
| 17:21.47 | Samot | Because as far as I can tell, for the last couple of months dan_j has been "regressing" with his configurations. |
| 17:22.38 | Samot | And the changes that are being made are not small. |
| 17:24.20 | Samot | Nor are the issues prompting the changes. |
| 17:29.42 | *** join/#asterisk pchero (~pchero@109.70.54.56) |
| 17:36.25 | *** join/#asterisk NightMonkey (~NightMonk@pdpc/supporter/professional/nightmonkey) |
| 17:52.49 | *** join/#asterisk sarlalian (~sarlalian@107.170.239.102) |
| 17:54.08 | *** join/#asterisk sarlalian (~sarlalian@107.170.239.102) |
| 18:13.42 | *** join/#asterisk fonefreak (~root@c-76-105-87-143.hsd1.ga.comcast.net) |
| 18:19.10 | *** join/#asterisk miralin (~Thunderbi@91.237.94.4) |
| 18:25.03 | *** join/#asterisk joshelson (~joshelson@64.55.128.243) |
| 18:36.27 | *** join/#asterisk kraylo (~kraylo@unaffiliated/kraylo) |
| 18:41.09 | *** join/#asterisk shanth_ (~shanth@wsip-98-182-126-226.ph.ph.cox.net) |
| 18:51.48 | *** join/#asterisk tzafrir (~tzafrir@bzq-179-40-170.cust.bezeqint.net) |
| 19:17.29 | dan_j | file: /etc/hosts didnt help |
| 19:17.54 | Samot | 1:19:06 PM <DanQuinney> It won't make a difference dan_j, well, I have the fqdn set and it hasn't stopped the issue |
| 19:18.21 | dan_j | Samot: its mostly stable which is why i havent moved back to chansip |
| 19:18.36 | Samot | OK |
| 19:18.53 | Samot | I guess, for me personally, as a provider "mostly stable" isn't good enough |
| 19:18.57 | dan_j | Only just saw dans comment |
| 19:19.30 | dan_j | True. Just annoynng that this issue has suddenly appeared |
| 19:20.59 | DanQuinney | We went from 13.15.rc1 to 13.17 and it appeared, going to look at the internal network tomorrow to see if anything shows up |
| 19:21.37 | DanQuinney | Also, the issue is exasperated with high cpu, so might be worth checking that dan_j |
| 19:23.00 | DanQuinney | Went from 13.15 due to a memory leak |
| 19:23.24 | Samot | See this is the thing.. |
| 19:23.32 | Samot | Chan_SIP/PJSIP = SIP Stacks |
| 19:23.40 | Samot | Customers don't care about the SIP Stack |
| 19:23.50 | Samot | If they try another provider and there are no issues.. |
| 19:23.54 | DanQuinney | The memory leak was in the pub_sub stuff |
| 19:24.03 | Samot | They don't care if the issue is PJSIP or not |
| 19:24.12 | Samot | They just know SIP on A has issues but SIP on B does not |
| 19:24.45 | Samot | And I'm the guy you need to worry about. |
| 19:24.55 | Samot | Or guys like me. |
| 19:25.08 | Samot | That can come in drop them a phone or two for demos with burn numbers.. |
| 19:25.31 | Samot | And side by side with the other provider (who has issues) those won't be present. |
| 19:25.58 | Samot | we both use Asterisk |
| 19:26.05 | Samot | But you're on PJSIP and I'm not. |
| 19:26.57 | Samot | Chronic issue customers are the easiest to get. |
| 19:27.02 | *** topic/#asterisk by kharwell -> #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.17.1 (2017/08/31), 11.25.2 (2017/08/31), Standard: 14.6.1 (2017/08/31); 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 |
| 19:27.35 | DanQuinney | as I've said Samot, this is the first issue we've seen with pjsip, we might be lucky, or simply only use basic features |
| 19:27.48 | DanQuinney | excluding the memory issue, which customers didn't see |
| 19:27.59 | Samot | Well memory issues lead to performance issues |
| 19:28.09 | Samot | So yeah, they can "see" those or potentially see them. |
| 19:28.12 | DanQuinney | I appreciate that 1 issue is stull an issue |
| 19:28.15 | DanQuinney | *still |
| 19:28.31 | DanQuinney | not if you manage it correctly, and have monitoring in place |
| 19:28.43 | DanQuinney | our pbx servers are overspec'd |
| 19:29.06 | DanQuinney | so when I say memory leak, we're talking less than 50% memory usage |
| 19:29.15 | DanQuinney | but noticable internally with monitoring |
| 19:30.20 | Samot | Well, like I said it's probably 5K view time. |
| 19:30.32 | Samot | At least for dan_j.. |
| 19:30.42 | DanQuinney | <PROTECTED> |
| 19:30.46 | Samot | I'm not sure how much his RT issues impacted customers but I remember a few issues |
| 19:30.48 | Samot | Yes. |
| 19:30.53 | DanQuinney | not a saying I've heard of before |
| 19:31.02 | Samot | Instead of looking at the problem for a narrow view. |
| 19:31.07 | DanQuinney | ah |
| 19:31.11 | Samot | Take a step back and take a wider view |
| 19:31.15 | Samot | Like.. |
| 19:31.19 | DanQuinney | yes indeed |
| 19:31.51 | Samot | "Hey we have 50 tickets that were treated as individual issues but when you look at them collectively they shine a light on a bigger underlying issue" |
| 19:32.14 | Samot | Or |
| 19:32.38 | Samot | "In the last 6 months how many times as the RealTime + PJSIP followed by Static + PJSIP caused outages?" |
| 19:33.03 | DanQuinney | we had no end of issues with RT + PJSIP |
| 19:33.20 | Samot | I'm just giving examples based on dan_j's issues |
| 19:33.30 | Samot | He was using RealTime, it has issues. |
| 19:33.32 | Samot | Not he's not. |
| 19:33.39 | Samot | He's having issues with PJSIP.. |
| 19:33.49 | DanQuinney | I've not been following his issues, other than todays |
| 19:33.51 | Samot | not = now |
| 19:34.22 | Samot | When you spend months chasing down little issues.. |
| 19:34.33 | Samot | It's time to see if there are bigger issues mating and making those little issues.. |
| 19:34.49 | DanQuinney | spoken like a pro :) |
| 19:35.01 | Samot | I hate bleeding resources |
| 19:35.36 | Samot | And a good size of small/medium VoIP providers spend more time bleeding cost and resources on support than ever needed. |
| 19:39.30 | Samot | Sadly, too, they are all general Asterisk or Asterisk/FreePBX shops. |
| 19:39.43 | DanQuinney | euch, freepbx |
| 19:39.49 | DanQuinney | *eugh |
| 19:39.52 | Samot | I use it |
| 19:40.15 | Samot | I mean the way I use it is way outside the scope of how it should be used. |
| 19:40.32 | Samot | But it does a lot of the trivial/tedious stuff for me. |
| 19:43.22 | Samot | That's only for my hosted voice environment |
| 19:43.44 | Samot | On-premises systems are usually straight FreePBX installs. |
| 19:44.13 | DanQuinney | i guess it has its place, but I've never needed it |
| 19:44.27 | DanQuinney | we don't do any on-premises stuff |
| 19:45.10 | Samot | Well like I said, even in the hosted environment it's just there to do tedious stuff. |
| 19:45.18 | Samot | Ring Group needed? Done. |
| 19:45.35 | Samot | I don't have to spend my time writing the automation for it. |
| 19:48.46 | Samot | Since a lot of the end users just need generic, cookie-cutter style stuff for things like Ring Groups, etc. |
| 19:50.12 | DanQuinney | we have an interface that sits and generates config, so most things are easy |
| 19:50.22 | Samot | Right. |
| 19:50.25 | Samot | I could do that too |
| 19:50.29 | Samot | But it's just me. |
| 19:50.32 | DanQuinney | anyone could |
| 19:50.40 | Samot | So. |
| 19:50.57 | DanQuinney | so I understand why someone would use freepbx |
| 19:51.16 | Samot | Plus I have customers that need to manage certain things and I don't want them in the CLI |
| 19:51.35 | Samot | It was more of a case of "Why re-invent the wheel" |
| 19:52.12 | DanQuinney | indeed |
| 20:03.27 | *** join/#asterisk reveal (~reveal@stolemybike.org) |
| 20:11.30 | *** join/#asterisk andresmujica (~andresmmu@ubuntu/member/andresmujica) |
| 20:26.29 | reveal | Question: If I have an asterisk server and a Televantage Server with SIP licenses, can I connect the two together so they can talk to each other and dial extensions of each other? |
| 20:27.17 | Samot | That would depend on the Televantage server. |
| 20:27.32 | Samot | But yes, you can peer the Asterisk PBX to it and send calls to it over that trunk. |
| 20:27.35 | reveal | I have added the astrisk server as a sip server in televantage |
| 20:27.48 | Samot | I don't know what that means. |
| 20:27.53 | Samot | I've never touched those boxes before. |
| 20:27.57 | reveal | and i setup a [televantage]in sip.conf and can see sip show peers shows my televantage server |
| 20:28.07 | reveal | type=friend and etc |
| 20:28.09 | Samot | OK. |
| 20:28.12 | reveal | and the servers IP |
| 20:28.13 | Samot | So they are connected. |
| 20:28.25 | [TK]D-Fender | that isn't a "connection" |
| 20:28.29 | reveal | but when i dial an example televantage ext from ym xlite |
| 20:28.36 | reveal | oh |
| 20:28.37 | [TK]D-Fender | that's a profil you can try to match calls from and send calls based on |
| 20:28.41 | [TK]D-Fender | doesn't mean it WORKS |
| 20:28.42 | Samot | reveal: You need to setup routing first... |
| 20:28.46 | reveal | ahhhh |
| 20:28.49 | reveal | routing in both right |
| 20:28.51 | Samot | Both outgoing and incoming |
| 20:29.06 | [TK]D-Fender | Doesn't mean it will auth. or the other side is configured to accept those calls. Is licensed (if required), etc |
| 20:29.07 | Samot | well, you need to tell Asterisk what calls should go to the other PBX |
| 20:29.32 | Samot | Then how to handle calls from the other PBX |
| 20:30.31 | reveal | i added dialing routes in televantage, and setup 8988 to get passed |
| 20:30.45 | reveal | and i setup a context in asterisk to accept 22 |
| 20:30.59 | Samot | Is the trunk pointing to that context? |
| 20:31.05 | reveal | so if i dial 898822 from my phone to contact the asterisk server |
| 20:31.37 | reveal | the trunk from televantage or from asterisk Samot |
| 20:31.47 | Samot | Asterisk |
| 20:31.52 | Samot | The trunk on Asterisk. |
| 20:32.07 | Samot | You have to set the context= setting to a context to handle incoming calls. |
| 20:32.08 | reveal | [televantage] |
| 20:32.08 | reveal | type=friend |
| 20:32.08 | reveal | trustrpid=yes |
| 20:32.08 | reveal | sendrpid=yes |
| 20:32.08 | reveal | context=default |
| 20:32.10 | reveal | canreinvite=no |
| 20:32.13 | reveal | host=192.168.1.252 |
| 20:32.15 | reveal | disallow=all |
| 20:32.17 | Samot | STOP |
| 20:32.18 | reveal | allow=ulaw |
| 20:32.19 | Samot | ~pb |
| 20:32.19 | infobot | it has been said that pastebin is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 20:32.21 | reveal | allow=gsm |
| 20:32.23 | reveal | thats what i have |
| 20:32.31 | Samot | What is in the default context? |
| 20:32.33 | Samot | Do you have one? |
| 20:32.36 | reveal | sorry to many lines |
| 20:32.45 | reveal | default is just public |
| 20:32.52 | Samot | What is in the context? |
| 20:32.58 | Samot | That's a context with dialplan in it |
| 20:33.00 | Samot | What is there? |
| 20:33.23 | Samot | You have to configure all this in Asterisk |
| 20:33.33 | reveal | ahhh |
| 20:33.44 | Samot | Asterisk isn't going to magically route calls to a phone |
| 20:33.52 | Samot | Nor will it magically route calls from your phone. |
| 20:34.05 | Samot | Your phone peer will also have a context= setting |
| 20:34.16 | Samot | That's where calls from the phone should be processed. |
| 20:34.26 | Samot | That's where the pattern matching for digits and what to do with them is done |
| 20:35.04 | Samot | You have to write the dialplan to handle these calls. |
| 20:35.37 | reveal | in extensions.conf right |
| 20:35.43 | Samot | Yes. |
| 20:36.12 | reveal | so would it be better for me to create this context in sip.conf called televantage |
| 20:36.23 | reveal | then in my extensions dialplan create a televantage context |
| 20:36.25 | reveal | with the dialplan |
| 20:37.08 | reveal | where it would look like [televantage] exten => 8988,1,Answer |
| 20:37.37 | reveal | or extend => _1XXX,1,Answer and etc |
| 20:38.26 | reveal | so what i am reading is sure i can add them as a peer but that doesn't do me any good if asterisk doesn't know what televantage extensions are, or how to handle calls to asterisk extensions from televantage |
| 20:38.31 | reveal | am i understanding that correctly |
| 20:49.27 | Samot | Yes. |
| 20:49.35 | Samot | [televantage] |
| 20:49.46 | Samot | exten => XXXX,1,Something |
| 20:50.24 | Samot | Just like for the phones you would do something like [from-phone] |
| 20:50.27 | Samot | And do the same thing. |
| 20:57.54 | hdon | hi all :) i'm reading through node-ari-client examples. in the bridge example, when your channel enters StasisStart, it joins your channel to a "holding" bridge, and then plays moh on your channel -- what is the point of doing this in practice? why put them in a "holding" bridge? |
| 20:58.50 | DivideBy0 | hdon: you could play ringing to it so the caller hears it ring |
| 20:59.17 | hdon | hmm... |
| 20:59.51 | hdon | does the channel have to be on a bridge to play something to it? i kind of expected that maybe the channel could be connected to a bridge, and maybe another channel would play moh to the bridge so that anyone joined to the bridge would hear the music |
| 21:00.09 | hdon | i think i need to re-read the fundamentals of ARI |
| 21:03.35 | DivideBy0 | yes, you can play ringing to a channel instead of a bridge |
| 21:05.05 | hdon | if i join multiple channels to a bridge, is all media received on all such channels automatically mixed and sent to every channel joined to the bridge? |
| 21:16.31 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 21:23.21 | *** join/#asterisk clarjon1 (~clarjon1@unaffiliated/clarjon1) |
| 21:28.03 | *** join/#asterisk TheGallopingFox (~TheGallop@gateway/vpn/privateinternetaccess/thegallopingfox) |
| 21:28.35 | TheGallopingFox | i have a polycom soundpoint 321, when i boot it 3 red lights are on with splash screen, but nothing happens, is it damaged? |
| 21:29.51 | *** join/#asterisk mmlj4 (~mmlj4@47-44-49-2.static.unas.mo.charter.com) |
| 21:31.44 | hdon | TheGallopingFox, i would double check the manual. sounds like it might be bricked. i've got pcom sip 331s and 335s but i haven't seen anything quite like that before |
| 21:32.18 | Samot | It just sits there? |
| 21:32.22 | Samot | Never does anything? |
| 21:32.32 | TheGallopingFox | when i turn the power on, red light is on, and both the line lights are orange, the splash screen shows but nothing happens |
| 21:32.49 | Samot | You mean the MWI light? |
| 21:32.50 | TheGallopingFox | none of the buttons respond |
| 21:32.59 | TheGallopingFox | Line 1 and Line 2 lights |
| 21:33.09 | Samot | You just said they were orange |
| 21:33.15 | Samot | Where is the red light? |
| 21:33.20 | TheGallopingFox | yeah the line lights are orange, the power light is red |
| 21:33.34 | Samot | What power light? |
| 21:33.34 | TheGallopingFox | not flashing just on, and splash screen is showing |
| 21:33.46 | TheGallopingFox | red light right at the top right |
| 21:33.51 | Samot | That's not power |
| 21:33.53 | Samot | That's MWI |
| 21:33.54 | TheGallopingFox | oh |
| 21:33.57 | TheGallopingFox | oh ok |
| 21:33.58 | Samot | Message Waiting Indicator |
| 21:34.01 | Samot | Your phone is bricked |
| 21:34.05 | TheGallopingFox | lol |
| 21:34.11 | Samot | You can try to factory default it |
| 21:34.12 | TheGallopingFox | i could use it as a door stop |
| 21:34.19 | Samot | But it's going to be a PITA |
| 21:34.31 | TheGallopingFox | does that involve opening it up |
| 21:34.35 | Samot | No. |
| 21:34.45 | Samot | But it does involve being able to get to the menus |
| 21:35.09 | TheGallopingFox | menus not working, buttons not responding to anything |
| 21:35.13 | Samot | Right. |
| 21:35.27 | Samot | Was this thing pulling from a provisioning server? |
| 21:35.32 | TheGallopingFox | yes it was |
| 21:35.36 | TheGallopingFox | asterisk |
| 21:35.41 | Samot | No. |
| 21:35.46 | Samot | Asterisk is not a provisioning server. |
| 21:35.49 | TheGallopingFox | oh |
| 21:35.55 | Samot | Did you manually program the phone? |
| 21:36.08 | Samot | Log into the GUI and setup the buttons/accounts? |
| 21:36.12 | TheGallopingFox | it used dhcp |
| 21:36.16 | Samot | OK |
| 21:36.19 | TheGallopingFox | to get the firmare via ftp |
| 21:36.44 | TheGallopingFox | networking not working either so cannot access gui |
| 21:36.46 | Samot | CAn you confirm it's reaching the provisioning server? |
| 21:36.52 | Samot | OK. |
| 21:36.52 | TheGallopingFox | its not |
| 21:36.56 | Samot | So it's completely bricked. |
| 21:36.56 | TheGallopingFox | no message in logs |
| 21:36.59 | TheGallopingFox | lol |
| 21:37.01 | TheGallopingFox | oh well |
| 21:37.09 | TheGallopingFox | it lasted for 10 years |
| 21:37.13 | Samot | There is a method to factory default. |
| 21:37.18 | Samot | Well you got your money's worth |
| 21:37.25 | TheGallopingFox | maybe a leaking cap |
| 21:37.29 | TheGallopingFox | yeah |
| 21:37.49 | TheGallopingFox | is it worth getting another IP321, or are there anything better to get these days on same kind of budget |
| 21:37.57 | Samot | Well those are EOL I believe |
| 21:38.05 | TheGallopingFox | yeah long time ago i imagine |
| 21:38.08 | Samot | I mean you can probably get a 330 or so used pretty decent |
| 21:38.22 | TheGallopingFox | right ok, stick with polycom |
| 21:38.29 | Samot | Well |
| 21:38.37 | drmessano | Sangoma has phones too |
| 21:38.39 | Samot | Your phone is using DHCP to pull a profile. |
| 21:38.45 | TheGallopingFox | yes |
| 21:38.46 | drmessano | But you need HUGE hands to hold the handsets |
| 21:38.51 | TheGallopingFox | lol |
| 21:38.59 | Samot | You're going to need to update that |
| 21:39.06 | Samot | Because it's based on MAC address |
| 21:39.13 | TheGallopingFox | oh ok |
| 21:39.28 | Samot | So a new phone will require new provisioning files. |
| 21:39.50 | TheGallopingFox | i have a mac--phone.cfg |
| 21:39.57 | TheGallopingFox | it uses via ftp |
| 21:40.03 | drmessano | http://i.imgur.com/BIWYvIS.jpg |
| 21:40.05 | TheGallopingFox | yeah sure |
| 21:40.10 | drmessano | You could also use that |
| 21:40.16 | Samot | If you have something replacing the MAC automatically on the server.. |
| 21:40.30 | Samot | But the profiles generally match the MAC |
| 21:40.30 | TheGallopingFox | i could do via pfsense |
| 21:40.35 | TheGallopingFox | or just edit the files manually |
| 21:40.50 | Samot | Well, first get a new phone. |
| 21:41.05 | Samot | If you get another Polycom the templates will be close to the same |
| 21:41.05 | TheGallopingFox | yeah, so there is no way to unbrick it |
| 21:41.12 | Samot | Nope. |
| 21:41.12 | TheGallopingFox | ok that good |
| 21:41.13 | drmessano | You have PFSense? |
| 21:41.16 | TheGallopingFox | yeah |
| 21:41.22 | drmessano | Thats probably what did it |
| 21:41.28 | Samot | lol |
| 21:41.30 | TheGallopingFox | impossible |
| 21:41.39 | TheGallopingFox | what causes a brick? |
| 21:41.42 | drmessano | Nah man |
| 21:41.43 | Samot | http://imgur.com/a/A9vJo |
| 21:41.47 | drmessano | ^ That |
| 21:41.50 | TheGallopingFox | the internal flash not working? |
| 21:41.57 | [TK]D-Fender | <TheGallopingFox> menus not working, buttons not responding to anything <- no menu on BOOT? |
| 21:41.59 | TheGallopingFox | lol |
| 21:42.06 | TheGallopingFox | no menu on boot either |
| 21:42.09 | TheGallopingFox | i will reboot it now |
| 21:42.12 | [TK]D-Fender | Normally there is a countdown |
| 21:42.15 | drmessano | PF probably did butt stuff with your phone |
| 21:42.19 | [TK]D-Fender | You haven't rebooted it YET? |
| 21:42.24 | drmessano | Lots of butt stuff |
| 21:42.29 | TheGallopingFox | yeah just reconnected power |
| 21:42.37 | TheGallopingFox | the splash screen shows instantly with 3 lights on |
| 21:42.39 | TheGallopingFox | and thats it |
| 21:42.42 | Samot | I don't think he checked what you were asking about on the last boot |
| 21:42.45 | Samot | So he needed to reboot |
| 21:42.47 | drmessano | Jesus christ |
| 21:42.48 | drmessano | 17:32:33 <TheGallopingFox> when i turn the power on, red light is on, and both the line lights are orange, the splash screen shows but nothing happens |
| 21:42.55 | TheGallopingFox | yeah thats it |
| 21:42.57 | drmessano | What part of that says "I havent rebooted it yet" |
| 21:43.05 | [TK]D-Fender | "Splash screen" is usually a BOOT MENU |
| 21:43.09 | TheGallopingFox | oh right |
| 21:43.13 | Samot | Guys. |
| 21:43.15 | Samot | It's bricked. |
| 21:43.17 | Samot | Move on. |
| 21:43.18 | TheGallopingFox | menu key doesnt work |
| 21:43.27 | [TK]D-Fender | Didn't say MENU KEY |
| 21:43.29 | drmessano | Its a fucking 10 year old phone that was working before and now isnt |
| 21:43.32 | Samot | He got it 10 years ago |
| 21:43.33 | [TK]D-Fender | the boot menu uses the SOFT KEYS |
| 21:43.37 | Samot | It gave him a good run. |
| 21:43.48 | [TK]D-Fender | And he's giving a SHIT description |
| 21:43.52 | [TK]D-Fender | I want something more solid |
| 21:43.58 | Samot | OK. |
| 21:44.04 | Samot | But having used these for years. |
| 21:44.06 | Samot | It's bricked. |
| 21:44.09 | Samot | But continue. |
| 21:44.11 | drmessano | I'll be back in two hours to see where he is with the berating over the dead phone not being descriptive enough |
| 21:44.19 | drmessano | wanders off |
| 21:44.25 | TheGallopingFox | no key responds to anything, just pressed all of them |
| 21:44.33 | clarjon1 | TheGallopingFox: it's bricked, mate. |
| 21:44.36 | TheGallopingFox | ok |
| 21:44.39 | [TK]D-Fender | I didn't say "just mash" |
| 21:44.52 | [TK]D-Fender | When you start the phone you should get TEXT on the screen with the countdown. |
| 21:44.56 | [TK]D-Fender | Do you, or don't you? |
| 21:45.05 | TheGallopingFox | so is it worth getting another IP321 or are there better ones 10 years later |
| 21:45.17 | Samot | There are a lot a options |
| 21:45.19 | TheGallopingFox | [TK]D-Fender: no i dont get that at all |
| 21:45.20 | Samot | The 330 |
| 21:45.24 | [TK]D-Fender | Obviosuly there are better ones 10 years later |
| 21:45.30 | [TK]D-Fender | that's a ridiculous question |
| 21:45.36 | Samot | The VVX series |
| 21:45.36 | TheGallopingFox | [TK]D-Fender: as soon as it powers on splash shows and nothing else |
| 21:45.39 | drmessano | JFC |
| 21:45.53 | [TK]D-Fender | what is this "splash" exactly? |
| 21:45.59 | TheGallopingFox | Polycom + Logo |
| 21:46.11 | Samot | It's f'ing bricked |
| 21:46.14 | [TK]D-Fender | So never any other text? |
| 21:46.16 | TheGallopingFox | no |
| 21:46.22 | TheGallopingFox | i think its bricked |
| 21:46.25 | [TK]D-Fender | Then I'll concur on "bricked" |
| 21:46.30 | Samot | Thanks. |
| 21:46.35 | drmessano | Glad we have your fucking approval |
| 21:46.38 | drmessano | Thx |
| 21:46.42 | TheGallopingFox | strokes phone and thinks of all the good memories |
| 21:46.54 | clarjon1 | TheGallopingFox: remember that time you made that call? Ahh, good times... |
| 21:46.57 | drmessano | TheGallopingFox: Get a 331 or 335 |
| 21:46.59 | [TK]D-Fender | Glad I got a description that wasn't FUCKING SHIT <- |
| 21:47.00 | TheGallopingFox | yeah |
| 21:47.14 | TheGallopingFox | ok will look out for a 331 or 335 |
| 21:47.17 | TheGallopingFox | thank you |
| 21:47.28 | [TK]D-Fender | 331 is disco'd as well |
| 21:47.35 | TheGallopingFox | what does that mean |
| 21:47.35 | drmessano | But he can get one used |
| 21:47.40 | drmessano | Cheap |
| 21:47.41 | TheGallopingFox | oh i see |
| 21:47.43 | drmessano | and 331s are fine |
| 21:47.47 | drmessano | SO are 335s |
| 21:47.55 | TheGallopingFox | do they have more codecs |
| 21:48.04 | Samot | That's vague |
| 21:48.26 | Samot | But probably not. |
| 21:48.27 | TheGallopingFox | can get them pretty cheap |
| 21:48.32 | drmessano | TheGallopingFox: Newer firmware has more codecs and other options |
| 21:48.42 | TheGallopingFox | looks like they use the same casing |
| 21:48.43 | drmessano | You really need to decide what you want |
| 21:48.49 | drmessano | and go look at the feature matrix |
| 21:49.00 | TheGallopingFox | yeah will do thank you |
| 21:49.09 | drmessano | "More Codecs" isn't really a feature unless you have a need |
| 21:49.13 | drmessano | Like G722 |
| 21:49.14 | TheGallopingFox | 335 has HD voice, interesting |
| 21:49.19 | drmessano | Which a 335 does |
| 21:49.25 | Samot | Right. |
| 21:49.43 | Samot | Anything that supports G722 will be labeled "HD" |
| 21:49.58 | drmessano | But if you're using analog or G711 to your provider, a 331 used will be fine |
| 21:50.00 | drmessano | and cheap |
| 21:50.21 | drmessano | Everything is downsampled to the PSTN anyway |
| 21:50.28 | drmessano | Basically |
| 21:50.40 | Samot | Yup. |
| 21:50.44 | drmessano | So better codecs are really a matter of ext <> ext calls |
| 21:50.47 | Samot | It either ends up G711, G729 or GSM |
| 21:50.49 | Samot | Depending |
| 21:50.50 | TheGallopingFox | ok |
| 21:51.14 | Samot | However.. |
| 21:51.17 | Samot | That being said.. |
| 21:51.20 | Samot | Hardware wise |
| 21:51.28 | Samot | The HD phones do improve quality a bit |
| 21:51.35 | drmessano | True |
| 21:51.43 | TheGallopingFox | my cellphone can take calls what are crystal clear sometimes, not sure how that happens |
| 21:51.50 | Samot | So your G711 codec sounds cleaner |
| 21:51.51 | TheGallopingFox | but sometimes the quality is normal |
| 21:51.57 | drmessano | Newer hardware in general, backlit |
| 21:52.11 | drmessano | Lots of reasons for a new-ish handset |
| 21:52.17 | drmessano | 335 is a good choice |
| 21:52.23 | TheGallopingFox | ok i will get one |
| 21:52.28 | Samot | If you want to go nuts and have Opus |
| 21:52.36 | Samot | You're going to have to go for a higher model |
| 21:53.25 | TheGallopingFox | VVX series are the new ones, i bet they are pricey |
| 21:53.40 | Samot | Naw. |
| 21:53.43 | drmessano | Na |
| 21:53.54 | TheGallopingFox | oh not too bad |
| 21:53.57 | Samot | The 401/411's are not bad |
| 21:54.06 | Samot | Less that $150 if you get them at the right place, new |
| 21:54.19 | Samot | The 301/311 even cheaper |
| 21:54.19 | TheGallopingFox | ok will sit down and do some research on specs |
| 21:54.26 | drmessano | VVX 401 is $100 |
| 21:54.30 | Samot | See. |
| 21:54.39 | Samot | 411 might be a little more but not much |
| 21:54.43 | Samot | Since it offers a bit more |
| 21:55.01 | TheGallopingFox | is the 335 the last of the soundpoint range |
| 21:55.07 | Samot | Nope. |
| 21:55.13 | Samot | 550's are still out there, right? |
| 21:55.29 | drmessano | I dunno |
| 21:55.30 | TheGallopingFox | wow they are big beasts |
| 21:55.50 | Samot | Yeah. |
| 21:55.57 | Samot | It's pretty easy for Polycom |
| 21:55.59 | drmessano | I have a 550 on my desk |
| 21:56.06 | Samot | SP or VXX |
| 21:56.11 | Samot | er VVX |
| 21:56.17 | Samot | The numbering is the same |
| 21:56.23 | TheGallopingFox | do the VVX support opus |
| 21:56.24 | Samot | 3xx models = entry level user |
| 21:56.32 | Samot | 4xxx = mid level user |
| 21:56.41 | Samot | 5xx = high level /exec |
| 21:56.43 | TheGallopingFox | right ok thanks |
| 21:57.10 | drmessano | I think the 401 does OPUS |
| 21:57.14 | Samot | 6xx are like the 5xx with more lines a few more options |
| 21:57.16 | Samot | Nope. |
| 21:57.21 | drmessano | No? |
| 21:57.23 | Samot | Nothing below 600 |
| 21:57.30 | Samot | It's listed |
| 21:57.39 | Samot | But grayed out |
| 21:58.08 | TheGallopingFox | or a Polycom Spectralink 8002 |
| 21:58.13 | TheGallopingFox | something different |
| 22:00.22 | Samot | Yeah, only VVX500 and VVX600 support Opus in that series |
| 22:00.28 | TheGallopingFox | oh right ok |
| 22:00.42 | Samot | It was introduced in UC 5.4 and as of UC 5.6 hasn't changed. |
| 22:00.46 | TheGallopingFox | its not that important to have opus |
| 22:01.01 | Samot | Oh no, I was just giving drmessano some more details |
| 22:01.10 | TheGallopingFox | are cisco phones any good |
| 22:01.20 | Samot | SPA series only |
| 22:02.15 | TheGallopingFox | ok thanks |
| 22:05.52 | Samot | If you don't need anything fancy |
| 22:06.04 | Samot | The Cisco SPA514G is a nice little 4 line phone. |
| 22:18.04 | tuxd00d | Yealinks work as well, easy to provision. |
| 22:23.05 | *** join/#asterisk newtonr (~newtonr@216.186.174.249) |
| 22:23.05 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 22:43.43 | TheGallopingFox | i have a idea, i got remove the guts of my ip321 and stick a raspberry pi inside it and get it working again |
| 22:56.47 | [TK]D-Fender | ~savemoney |
| 22:56.47 | infobot | <Gremlin> It's okay, but I was hoping to save some money with a soldering iron and a bunch of sound cards. |
| 23:00.34 | *** join/#asterisk twitchnln (~twitch@107.130.213.25) |
| 23:01.01 | twitchnln | greetings everyone |
| 23:09.11 | *** part/#asterisk kharwell (kharwell@nat/digium/x-ecdbkojdsjgnbatg) |
| 23:15.06 | *** join/#asterisk cemotyz09 (~cemotyz09@cpe-70-121-157-202.satx.res.rr.com) |
| 23:54.05 | TheGallopingFox | i wonder if my ip321 handset will work on a ip335, incase the 2nd hand one i buy has scratches |
| 23:54.47 | tuxd00d | You donât have kids, do you? |
| 23:54.57 | TheGallopingFox | no |
| 23:55.41 | TheGallopingFox | the only difference between 321 and 335 is hd voice, but that shouldnt make a difference to the handset |
| 23:55.50 | tuxd00d | Iâm just happen when the phones arenât covered in mystery stickiness |
| 23:56.02 | tuxd00d | happy* |
| 23:56.18 | TheGallopingFox | im concerned the one i buy 2nd hand has nasty scratches on it |
| 23:57.01 | tuxd00d | Itâs pretty rare to have scratches. They use a hard plastic. |
| 23:57.08 | TheGallopingFox | oh ok |
| 23:57.19 | TheGallopingFox | i will keep the handset on the bricked phone as a spare |
| 23:57.25 | TheGallopingFox | and the psu |
| 23:59.00 | tuxd00d | Eventually youâll have a whole box of extra parts |
| 23:59.19 | TheGallopingFox | what will never get used |
| 23:59.35 | tuxd00d | If youâre lucky |
| 23:59.51 | TheGallopingFox | actually i could remove the keys too |
| 23:59.53 | TheGallopingFox | just incase |
