| 00:00.07 | Samot | If we went to UPS or generator, someone had to babysit. |
| 00:00.14 | Samot | ? Trunk where? |
| 00:00.19 | Samot | Truck |
| 00:00.29 | drmessano | Im talking about ALL those little generators in the field at the cabinets |
| 00:00.33 | drmessano | Power doesnt come from the CO |
| 00:00.42 | Samot | Ahh, yeah... |
| 00:00.50 | drmessano | All those cabinets are LOCALLY powered |
| 00:00.53 | Samot | Downside, someone still babysits. |
| 00:01.06 | drmessano | They go down OFTEN |
| 00:01.21 | drmessano | I should say, they often do not start when the power fails |
| 00:01.43 | Samot | Well most of those junction boxes need serious overhauls. |
| 00:01.45 | drmessano | So yay, power at the CO.. YAY, power at your house.. but if the cabinet between is down.. Too bad |
| 00:01.47 | Samot | Have you seen some of them? |
| 00:02.14 | drmessano | Even the newer ones |
| 00:02.36 | drmessano | They build out new ones with backup power.. leave them until the power goes out and the genset doesnt fire |
| 00:25.19 | apb1963 | I had my android device/extension working with encryption... and then I managed to mess it up. Not sure what I did... other than encryption I was messing with video and ring strategy.. I think that was it. Here's the call log http://fnpaste.com/zrBK |
| 00:31.59 | apb1963 | Oh I almost forgot... when I call the extension it goes straight to voicemail. |
| 00:48.40 | *** join/#asterisk UncleKiwi (~UncleKiwi@unaffiliated/unclekiwi) |
| 00:51.57 | UncleKiwi | hey people I hope your all doing well - I have a WAN (hub and 2 spokes) asterisk server is at the hub and both spokes have ATA's and hardphones at them one spoke works well and the other has problems on a money and a wednesday. I really dont know how to discover why calls drop during these periods ( the wan is a 5Ghz wireless bridge) |
| 00:53.01 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
| 00:54.05 | UncleKiwi | can I some how get asterisk to show me the reason it drops the audio |
| 00:54.27 | UncleKiwi | sip set debug ? |
| 00:54.54 | UncleKiwi | i dont seem to be getting packet loss over the connection |
| 01:04.17 | drmessano | You say you're not getting packet loss over the connection |
| 01:04.24 | drmessano | What have you done to test? |
| 01:14.53 | *** join/#asterisk Dovid (~dovid@107.19.189.119) |
| 02:03.12 | igcewieling | UncleKiwi: look for people doing automated "cloud backups" on those days. |
| 02:08.20 | *** join/#asterisk MaliutaLap (nikolai@unaffiliated/maliuta) |
| 02:32.12 | UncleKiwi | drmessano: ping plotter, ping |
| 02:33.08 | UncleKiwi | igcewieling: yeah thanks for the tip |
| 02:33.50 | UncleKiwi | i think i need a great tool to test the link - any suggestions ? |
| 02:34.14 | drmessano | iperf |
| 02:35.05 | UncleKiwi | thanks |
| 02:35.13 | UncleKiwi | checking it out now |
| 02:35.17 | UncleKiwi | have not used that b4 |
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| 06:43.49 | *** join/#asterisk maxxer (4f034845@gateway/web/freenode/ip.79.3.72.69) |
| 06:44.49 | maxxer | hi. I've some problems with extensions deregistering, and causing call drop: https://community.asterisk.org/t/clients-often-deregistering-and-registering-again-gigaset-cordless/69555 |
| 06:58.14 | *** join/#asterisk bof22 (~Thunderbi@185.13.183.107) |
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| 08:32.34 | maxxer | hi. I've some problems with extensions deregistering, and causing call drop: https://community.asterisk.org/t/clients-often-deregistering-and-registering-again-gigaset-cordless/69555 |
| 08:33.19 | *** join/#asterisk jkroon (~jkroon@154.73.35.201) |
| 09:03.16 | *** join/#asterisk chendy (~alexc@183.12.64.156) |
| 09:17.06 | dan_j | maxxer: I used gigasets without any problems. |
| 09:17.39 | dan_j | Got about 20 N510IP connected to my asterisk servers without any problems. |
| 09:17.55 | dan_j | I recommend doing a SIP DEBUG and see what is happening. |
| 09:20.54 | *** join/#asterisk kkocaerkek (~kkocaerke@78.186.160.204) |
| 09:30.12 | maxxer | dan_j: thanks. I enabled debug log but I see no reason why they do deregister. I just see deregistration message |
| 09:30.46 | WIMPy | Don't say deregister. That's something that also exists. |
| 09:31.03 | WIMPy | Fix some Router. Or enable keepalives. |
| 09:33.27 | maxxer | WIMPy: what do you mean by "fix router"? keepalives should be enabled on the asterisk server? |
| 09:33.34 | maxxer | extensions are all in LAN |
| 09:34.10 | WIMPy | Keepalives should come from the client. |
| 09:34.29 | WIMPy | But if it's on a LAN, thaen you seem to have some serious network issues. |
| 09:35.13 | maxxer | We even changed the switch, but it still happens. The weird thing is that it happens only on SOME extensions, not on all |
| 09:35.51 | maxxer | [Jan 30 10:35:18] VERBOSE[30186] app_dial.c: -- SIP/102-00000a0c is ringing [Jan 30 10:35:22] VERBOSE[30931] chan_sip.c: -- Unregistered SIP '102' |
| 09:35.56 | maxxer | :( |
| 09:36.44 | WIMPy | And you shouldn't call them extensions, either. |
| 09:37.37 | WIMPy | But that does in deed look like a real unregister, i.e. the phone saying that it doesn't want calls any more / is no longer reacable at that address. |
| 09:37.41 | UncleKiwi | yes they are peers |
| 09:37.51 | UncleKiwi | right ? |
| 09:38.13 | WIMPy | Yes, or devices or whatever. |
| 09:38.37 | WIMPy | Extensions are in extensions.conf. |
| 09:38.46 | maxxer | yeah sorry for bad glossary :( |
| 09:40.56 | maxxer | but that "Unregistered" happens even when there's no action, i.e. no active call |
| 09:42.22 | maxxer | and btw when it happens the ongoing call is dropped, in fact after that "ringing" the call went to another call group |
| 09:43.59 | UncleKiwi | how many phones do you have on the pbx |
| 09:44.04 | UncleKiwi | so far |
| 09:46.50 | UncleKiwi | iptables -F |
| 09:46.52 | maxxer | 8 |
| 09:47.39 | UncleKiwi | have you had a look to see if you have any filtering messing things up |
| 09:47.45 | maxxer | some of the Gigaset has two peers configured, some just one, but this doesn't seem to matter |
| 09:48.00 | maxxer | UncleKiwi: in iptables you mean? there are no active rules |
| 09:48.05 | UncleKiwi | ok |
| 09:48.35 | maxxer | I have a Cisco phone registered and works without deregistrations |
| 09:48.55 | maxxer | and another Gigaset working fine |
| 09:49.09 | UncleKiwi | upgrade the firmware |
| 09:49.18 | maxxer | no updates, I checked |
| 09:49.19 | UncleKiwi | and factory reset it |
| 09:49.21 | UncleKiwi | ahah |
| 09:49.33 | UncleKiwi | configure the faulty one again |
| 09:49.34 | maxxer | but they have different Gigaset devices |
| 09:49.57 | maxxer | how can I debug on the server side, other than debug_log_20170130 => notice,warning,error,debug,verbose,dtmf ? |
| 09:51.19 | UncleKiwi | i see you have nat=yes |
| 09:57.54 | maxxer | I should disable it since they're all on LAN? |
| 09:58.12 | UncleKiwi | i guess you could try it |
| 09:58.20 | UncleKiwi | then sip reload |
| 10:01.40 | UncleKiwi | i dont know if its needed but I would put the asterisk server ip address in the outbound proxy box on the Gigaset |
| 10:05.24 | maxxer | The Asterisk IP is already set into the Proxy and server address. I don't use STUN |
| 10:09.06 | *** join/#asterisk Kaian (~kaian@6.62-99-78.static.clientes.euskaltel.es) |
| 10:14.29 | dan_j | maxxer: I didn't say 'debug log', I said 'sip debug'. |
| 10:15.11 | dan_j | maxxer: Do 'sip set debug on' or 'sip set debug ip 192.168.0.191' and see whats happening. |
| 10:17.17 | maxxer | thanks for the tip |
| 10:24.59 | *** join/#asterisk Rasputin3711 (~Rasputin3@87.255.254.66) |
| 10:28.10 | maxxer | SIP 503 : http://pastebin.ca/3761919 |
| 10:30.20 | maxxer | also, I commented nat=yes but it's still <--- Transmitting (NAT) to 192.168.0.191:5060 ---> |
| 10:32.11 | maxxer | I had global "nat" setting enabled |
| 10:41.04 | *** join/#asterisk thiagoc (~thiagoc@unaffiliated/thiagoc) |
| 10:46.47 | UncleKiwi | try taking the ipaddress out of the 'domain' on the Gigaset |
| 10:49.16 | UncleKiwi | maxxer: check out the last part of this post at the bottom |
| 10:49.20 | UncleKiwi | http://forums.asterisk.org/viewtopic.php?t=12782 |
| 10:49.39 | UncleKiwi | talks about setting the domain = asterisk |
| 10:52.35 | dan_j | maxxer: Can I double check, is the gigaset on the same network as Asterisk? |
| 10:52.57 | maxxer | dan_j: yes, same LAN same subnet |
| 10:53.05 | dan_j | Ok. So you dont need NAT |
| 10:53.29 | maxxer | even if the sip trunk is outside (sorry for the silly question)? |
| 10:54.03 | dan_j | the sip trunk is a separate thing. You arent losing connection to your trunk. You've only shown that you are losing connection to the gigaset. |
| 10:54.39 | dan_j | Gigaset -> Asterisk Asterisk -> SIP Trunk. Two separate things. |
| 10:55.37 | dan_j | What are the SIP Packets immediately before the Unregistered SIP '10X' ? |
| 11:06.42 | UncleKiwi | maxxer: can you try set the outbound proxy to the ipaddress of your asterisk server |
| 11:08.21 | maxxer | UncleKiwi: isn't it already set? https://cdn-enterprise.discourse.org/asterisk/uploads/default/original/2X/0/062788677622c8f224243462ff2205abf914de00.png |
| 11:09.13 | UncleKiwi | second to bottom box in your post |
| 11:10.00 | UncleKiwi | its empty |
| 11:10.13 | maxxer | dan_j: debug excerpt http://pastebin.ca/3761935 . I've set debug on 192.168.0.191, registering sip 101 and 106 |
| 11:10.46 | maxxer | UncleKiwi: ok, sorry, I thought it was the first proxy setting, missed "outbound" |
| 11:11.53 | UncleKiwi | make any difference ? |
| 11:12.17 | maxxer | UncleKiwi: outbound proxy mode auto is fine? |
| 11:12.26 | UncleKiwi | yeah |
| 11:12.55 | maxxer | UncleKiwi: a working Gigaset SIP has indeed set the outbound proxy |
| 11:12.57 | dan_j | maxxer: Pastebin your sip.conf |
| 11:13.27 | UncleKiwi | maxxer: solved ? |
| 11:13.42 | maxxer | but 106 has the outbound proxy, and has the issue (but also had stun enabled) |
| 11:13.59 | maxxer | UncleKiwi: wait a minute, disconnects are random. can be 5 to 30 minutes |
| 11:14.19 | UncleKiwi | disable stun |
| 11:14.40 | maxxer | did that |
| 11:17.09 | maxxer | dan_j: http://pastebin.ca/3761939 |
| 11:19.22 | dan_j | hmm.. Why is your asterisk sending registers to the gigasets? |
| 11:19.51 | dan_j | First, set NAT=no on peer 106, then reload and try again |
| 11:20.39 | dan_j | Also, you have a spelling mistake |
| 11:20.41 | dan_j | defaultexpirey=180 |
| 11:20.51 | dan_j | should be defaultexpiry=180 |
| 11:22.02 | maxxer | fixed, thanks |
| 11:22.21 | UncleKiwi | what fixed it ? |
| 11:22.53 | UncleKiwi | just so we know |
| 11:23.30 | maxxer | fixed the typo |
| 11:23.34 | UncleKiwi | ah ahaha |
| 11:23.36 | UncleKiwi | sorry |
| 11:23.39 | maxxer | :) |
| 11:24.00 | maxxer | apparently neither the STUN or the outbound proxy fixes, but I'm still monitoring |
| 11:24.50 | maxxer | (some gigaset had the outbound proxy set, but they were deregistering. some had stun enabled and i switched it off) |
| 11:26.05 | dan_j | Still not working? |
| 11:26.23 | UncleKiwi | i have never had much fun with stun |
| 11:26.48 | dan_j | I've never needed stun. Got 300+ peers connected in various remote locations. Never used stun. |
| 11:26.57 | UncleKiwi | nice |
| 11:27.27 | dan_j | maxxer: If it's still not working, please pb a new sip debug. |
| 11:28.21 | dan_j | also, increase defaultexpirey= to 3600 on your asterisk, but leave 180 on the gigasets. |
| 11:28.38 | dan_j | defaultexpiry |
| 11:28.40 | dan_j | :) |
| 11:29.40 | maxxer | should I specify defaultexpiry on every sip definition? |
| 11:33.02 | dan_j | No need. Just put it in [general] |
| 11:34.05 | maxxer | dan_j: seems still deregistering http://pastebin.ca/3761941 |
| 11:34.38 | maxxer | i issued a "sip reload" after changing nat=no |
| 11:38.35 | dan_j | Certainly looks like the gigaset is actually unregistering. |
| 11:38.52 | dan_j | <--- SIP read from UDP:192.168.0.191:5060 ---> |
| 11:38.52 | dan_j | REGISTER sip:192.168.0.251 SIP/2.0 |
| 11:38.52 | dan_j | Expires: 0 |
| 11:39.15 | dan_j | Thats the gigaset sending an unregister request. |
| 11:39.40 | *** join/#asterisk pawiecki (~pawiecki@host-89-238-53-32.smgr.pl) |
| 11:40.05 | dan_j | I'm running an identical setup to you at a client's office. |
| 11:40.12 | *** join/#asterisk Oatmeal (~Suzeanne@cpe-65-185-34-151.columbus.res.rr.com) |
| 11:40.17 | dan_j | And you have the same gigaset firmware, so it's not a firmware issue. |
| 11:41.02 | UncleKiwi | dan_j: what distro do you run |
| 11:41.10 | dan_j | Centos + asterisk v11 |
| 11:41.11 | tzafrir | Hi, the Asterisk Sweedish voice package points to http://www.thenordicvoice.com/ . |
| 11:41.17 | UncleKiwi | same with me |
| 11:41.32 | dan_j | Although im moving to v13 |
| 11:41.38 | tzafrir | It doesn't seem to respond. Any idea if whoever operates it is alive? |
| 11:41.41 | UncleKiwi | I never had a lot of fun with ubuntu |
| 11:42.17 | *** join/#asterisk ace_me (~IceChat9@unaffiliated/ace-me/x-814638) |
| 11:42.24 | ace_me | Hi ! I have zoiper and ekiga installed on local laptop and a freepbx running in the virtualbox machine ! When I try to connect ekiga with zoiper which is already logged in as an agent I get I am sorry there is no call parked on that extension ! Please try again... |
| 11:42.26 | dan_j | Yes, but this seems to be an issue with his gigaset rather than his asterisk installation |
| 11:42.36 | ace_me | What could block / cause this ? |
| 11:42.59 | dan_j | ace_me: #freepbx is probably the best place for that question |
| 11:43.39 | UncleKiwi | dan_j: yeah strange that you know this phone and firmware and its doing this strange thing |
| 11:43.50 | maxxer | indeed it could be the client, as I have a C470 which doesn't present the problem |
| 11:44.06 | dan_j | maxxer: same base? |
| 11:44.18 | maxxer | what it's strange is that it has not happening from the beginning, in the past it happened but very very rarely |
| 11:44.37 | dan_j | Factory reset the gigaset base unit and try again |
| 11:44.40 | maxxer | dan_j: no, C470 is a different base. the .191 is an A510 |
| 11:45.00 | dan_j | Ah. I have the N510 but i think they are identical |
| 11:45.12 | dan_j | At least, they run the same firmware. |
| 11:45.21 | dan_j | Note |
| 11:45.30 | dan_j | If you factory reset, you need to pair the cordlesses again. |
| 11:45.44 | maxxer | believed so |
| 11:46.00 | maxxer | but last friday I messed up all the bases, so it's not really a big deal :D |
| 11:46.15 | maxxer | damn, can't really believe |
| 11:47.07 | UncleKiwi | maxxer: how many of the phones are acting like this ? |
| 11:48.09 | maxxer | UncleKiwi: most of them. an A510, A580, C530... |
| 11:49.07 | UncleKiwi | maxxer: how many phones total and how many are acting bad |
| 11:50.25 | maxxer | 8 phones. 1 Cisco working fine, 1 C470 working fine, 1 A580 which seems to be working fine today. Then 2 on the A580 and 2 on the A510, 1 on C530 are deregistering. |
| 11:50.34 | *** join/#asterisk Dovid (~dovid@107.19.189.119) |
| 11:50.52 | maxxer | I have other bases available, I must try with them |
| 11:51.15 | maxxer | and try a factory reset |
| 11:52.53 | *** join/#asterisk GeneralSpongebob (~NameUser@cpc94700-mapp10-2-0-cust122.12-4.cable.virginm.net) |
| 11:53.55 | *** join/#asterisk Oatmeal (~Suzeanne@cpe-65-185-34-151.columbus.res.rr.com) |
| 11:54.12 | dan_j | UncleKiwi: The number of phones is not relevant. It is the number of bases that is relevant since a base can host multiple phones. |
| 11:56.06 | UncleKiwi | dan_j: i was just curious about how many devices he had failing - i understand your point |
| 11:56.08 | maxxer | I have at most 2 SIP phones per base, as Gigasets cannot handle more than 2 simultaneous conversations |
| 11:56.39 | maxxer | you can configure more, but i.e. during a group call only the first two will ring |
| 11:58.30 | Samot | This is for extension 106? |
| 11:58.39 | Samot | Or it's one have an issue? |
| 11:59.01 | Samot | s/have/having/ |
| 11:59.17 | maxxer | Samot: 106 is one of the 5 deregistering. and it's configured on a base along with 101 |
| 11:59.33 | Samot | Yes, but two separate accounts. |
| 11:59.59 | Samot | Expires: 0 <- You send a REGISTER with that as the REGISTER timeout, you're not going to REGISTER. |
| 12:00.13 | Samot | You're not even getting a 401 challenge. |
| 12:01.03 | Samot | If you have one that looks to be REGISTERing just fine and the other account not... |
| 12:01.08 | Samot | One the same device.. |
| 12:01.17 | Samot | Look at your configuration. |
| 12:02.51 | Samot | Swap the accounts. |
| 12:03.03 | Samot | Make the working account 106 and the non working account 101. |
| 12:03.11 | Samot | Just change the user/password fields. |
| 12:03.17 | Samot | See what happens. |
| 12:03.41 | Samot | I bet 106 will register and 101 will exhibit the same behaviour that 106 was just displaying. |
| 12:05.03 | maxxer | Samot: 101 and 106 bot does NOT work. other phones work, but on different base |
| 12:06.06 | *** join/#asterisk maxxer (~quaqua@unaffiliated/maxxer) |
| 12:06.26 | Samot | OK, I see more things now. |
| 12:06.40 | Samot | A verbose output really isn't needed for this. |
| 12:06.49 | Samot | Just adds extra crap. |
| 12:06.57 | Samot | K, so now I can confirm.. |
| 12:07.06 | Samot | I don't see 101 or 106 being challenged. |
| 12:07.20 | *** join/#asterisk evilman_work (~evilman@87.244.6.228) |
| 12:08.04 | Samot | Asterisk is not challenging either acccount. |
| 12:08.08 | Samot | Asterisk is not challenging either account. |
| 12:08.54 | maxxer | so resuming it's a client issue. and right now the only things I can do is try factory default the base and configure it again |
| 12:09.24 | Samot | Yeah, I would try that. |
| 12:10.55 | Samot | Actually, no. |
| 12:10.58 | Samot | Don't do that. |
| 12:11.04 | Samot | Just reboot it. |
| 12:11.10 | Samot | I want to see if this changes: |
| 12:11.16 | Samot | CSeq: 336 REGISTER <-- That IS bad. |
| 12:11.42 | maxxer | Samot: I rebooted all the clients last friday because I had to unplug them, so they've all been restarted |
| 12:11.49 | Samot | I mean right now |
| 12:11.52 | Samot | At this moment. |
| 12:11.57 | Samot | Reboot the device with 101/106 |
| 12:12.14 | Samot | I don't even care if it was rebooted an hour ago. |
| 12:12.20 | Samot | Your CSeq is waaaay off. |
| 12:12.25 | maxxer | ok |
| 12:12.27 | Samot | Reboot to see if that starts at 1 like it should. |
| 12:15.58 | *** join/#asterisk Oatmeal (~Suzeanne@cpe-65-185-34-151.columbus.res.rr.com) |
| 12:16.44 | maxxer | reaching out the office... |
| 12:21.00 | maxxer | Samot: they're not answering, in case are you still around in 45/60 minutes? :) |
| 12:21.20 | Samot | I actually have to leave soon and will be back in about that time. |
| 12:22.01 | maxxer | ok thanks, I'll post here anyway |
| 12:26.43 | dan_j | maxxer: If you have access to the gigaset web interface, you should be able to restart it from there. |
| 12:33.02 | maxxer | I thought a real power off was needed |
| 12:42.48 | maxxer | CSeq: 375 REGISTER |
| 12:42.51 | maxxer | http://pastebin.ca/3761954 |
| 12:42.59 | maxxer | Samot: ^ |
| 12:43.11 | maxxer | dan_j: here's the log after reboot |
| 12:43.53 | *** join/#asterisk salz212 (67ff0552@gateway/web/freenode/ip.103.255.5.82) |
| 12:49.46 | dan_j | I dont see it unregistering apart from at the top. |
| 12:50.32 | *** join/#asterisk mub (~jub@static-173-53-12-18.rcmdva.fios.verizon.net) |
| 12:51.04 | dan_j | 101 still has NAT=yes |
| 12:51.07 | *** join/#asterisk bof22 (~Thunderbi@185.13.183.107) |
| 12:51.27 | dan_j | But i think your next step is factory reset. |
| 13:05.46 | maxxer | maybe I trimmed the shutdown |
| 13:05.57 | maxxer | yes, I deactivated nat on 106 only |
| 13:18.24 | *** join/#asterisk [TK]D-Fender (~joe@216.191.106.165) |
| 13:48.10 | *** join/#asterisk maxxer (~quaqua@unaffiliated/maxxer) |
| 13:48.10 | *** join/#asterisk dmasiero (~doug@lantern.masiero.us) |
| 13:57.23 | maxxer | dan_j: can I just set "nat=no" into the globals section or should I set per sip config? Will it affect upstream trunk? |
| 13:58.43 | [TK]D-Fender | You should be configuring each peer to know what its situation is |
| 13:58.54 | [TK]D-Fender | And properly defining your local subnets, ettc |
| 14:05.58 | *** join/#asterisk DivideBy0 (~DivideBy0@unaffiliated/divideby0x0) |
| 14:05.59 | *** mode/#asterisk [+o DivideBy0] by ChanServ |
| 14:17.52 | Samot | maxxer: The device shouldn't be sending a new REGISTER request to Asterisk with a CSeq other than 1. |
| 14:18.08 | Samot | Especially if you just booted the device and it's the first attempt. |
| 14:18.49 | Samot | I'd factory reset and try again. |
| 14:20.20 | maxxer | I'm trying a factory reset on a different base, let's see if it changes anything |
| 14:20.40 | Samot | One that doesn't work? |
| 14:20.50 | maxxer | yeah, another one not working |
| 14:21.02 | maxxer | I've a base with a single phone, I'll try on that |
| 14:21.03 | Samot | Why aren't you doing the one we've been working on? |
| 14:21.20 | maxxer | because I was waiting for your reply, while doing the reset on the other ;) |
| 14:22.20 | Samot | I warned I was going to be gone. |
| 14:22.28 | *** join/#asterisk chendy (~alexc@183.12.64.156) |
| 14:22.42 | maxxer | sure, so while I was waiting I was doing the reset test on another base |
| 14:22.52 | Samot | And did it work? |
| 14:23.03 | maxxer | in progres... |
| 14:34.49 | *** join/#asterisk jlewis-highwinds (~jlewis@soloth.lewis.org) |
| 14:36.12 | *** join/#asterisk newtonr (RustyNewto@nat/digium/x-mmvfyfusplbgyyoj) |
| 14:36.12 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 14:36.45 | jlewis-highwinds | if anyone from Digium hangs here who can talk about issues with incorrect bridging of calls on your SIP trunks, I'd like to talk. Support has been completely unhelpful, and this is affecting multiple customers. |
| 14:38.31 | jlewis-highwinds | i.e. totally unrelated customers (other than both relying on Digium SIP trunks) are having unrelated calls bridged together by Digium |
| 14:38.53 | jlewis-highwinds | Makes for very interesting conference bridges |
| 14:39.09 | jlewis-highwinds | which turn into party-lines |
| 14:41.58 | Samot | So they get bridged to random calls? |
| 14:42.13 | jlewis-highwinds | bingo |
| 14:42.14 | Samot | Like anyone's random calls? |
| 14:42.22 | Samot | Or just your users random calls? |
| 14:43.07 | jlewis-highwinds | calls to Digium customer A's number get bridged onto an existing call to Digium customer B's number |
| 14:43.34 | Samot | So these are all direct Digium customers? |
| 14:43.44 | Samot | I.e. You're not reselling the plans? |
| 14:44.25 | [TK]D-Fender | So those calls are never hitting your server? |
| 14:44.32 | jlewis-highwinds | so, the most common way we experience this is with conference bridges on our server, into which remote staff used a 10-digit number to reach us...and during the conf call, random calls to other Digium customers will land on our conf bridge, with nothing logged by our server |
| 14:45.06 | Samot | Who's random calls? |
| 14:45.10 | Samot | Your customers? |
| 14:45.21 | Samot | Or a random call from some guy in Texas?! |
| 14:45.29 | jlewis-highwinds | we're not reselling service...just using our own asterisk server and Digium SIP trunking for origination/termination |
| 14:45.53 | Samot | OK. |
| 14:46.05 | *** join/#asterisk Kalavera (~Kalavera@aquiles.novelix.com.pe) |
| 14:46.13 | Samot | So you're saying that other Digium customers are having their calls sent to your PBX... |
| 14:46.21 | Kalavera | hey guys, how do I make asterisk to interact with the FXO card |
| 14:46.30 | jlewis-highwinds | by asking the most recent of these random people who showed up on our bridge what number they'd dialed, I got in touch with that company and verified that they too are a Digium SIP trunk customer |
| 14:46.33 | Samot | And they end up in a conference bridge without Asterisk logging any of it? |
| 14:46.47 | jlewis-highwinds | yep |
| 14:47.01 | Samot | So how did _your_ Asterisk server put these calls in the conference bridge? |
| 14:47.22 | jlewis-highwinds | so it's not so much that the calls are being sent to our server...they've got to be getting bridged at Digium onto an existing call into our server |
| 14:47.39 | Samot | Huh? |
| 14:47.54 | Samot | How can the call be on your server, bridged into your conference room but be bridge at Digium? |
| 14:48.24 | jlewis-highwinds | if the call was hitting us in any normal way, we'd have asterisk logs of the incoming call (and they'd have to enter a passwd to get into the conf bridge) |
| 14:48.27 | jlewis-highwinds | none of that happens |
| 14:48.34 | Samot | "normal way" |
| 14:48.35 | Samot | ? |
| 14:48.41 | Samot | Normal way is to send the request to your PBX. |
| 14:48.51 | Samot | It's up to your PBX to accept it. |
| 14:49.08 | Samot | The calls cant be on your server... |
| 14:49.23 | Samot | AND be bridged at your channel at the provider level. |
| 14:49.37 | Samot | Either your PBX is accepting and bridging calls.. |
| 14:49.39 | jlewis-highwinds | not if its bridged onto an existing call elsewhere |
| 14:49.50 | Samot | Or Digium is bridging calls at a higher level. |
| 14:50.09 | jlewis-highwinds | right...I'm suggesting Digium is bridging calls incorrectly |
| 14:50.19 | Samot | So they are not bridging into your conference room. |
| 14:50.21 | Samot | Or your PBX |
| 14:50.28 | Samot | They are bridging at their network. |
| 14:50.43 | Samot | You can't have both. |
| 14:51.21 | jlewis-highwinds | right...but we experience this as random calls intruding into our conference bridges |
| 14:51.42 | Samot | Just on conf bridge calls? |
| 14:52.28 | maxxer | The resetted base is still unregistering. :( it was fine for ~10 minutes after factory reset and 5m after cordless association it disconnected |
| 14:52.47 | jlewis-highwinds | that's the only instance of this I've been made aware of...but it kind of makes sense since those many/frequent, and often have remote staff dialing in |
| 14:53.00 | Samot | And the conf bridge is on your PBX? |
| 14:53.18 | jlewis-highwinds | meant to say, the conf calls are many/frequent, and tend to last a while |
| 14:53.40 | Samot | maxxer: So it registered and then lost registration? |
| 14:53.45 | jlewis-highwinds | so they present a large target onto which incorrectly bridged calls can lang |
| 14:53.48 | jlewis-highwinds | er...land |
| 14:53.52 | Samot | And the conf bridge is on your PBX? |
| 14:54.02 | *** join/#asterisk cresl1n (Adium@asterisk/libpri-and-libss7-expert/Cresl1n) |
| 14:54.02 | *** mode/#asterisk [+o cresl1n] by ChanServ |
| 14:54.05 | jlewis-highwinds | yep...local conf bridge on our server |
| 14:54.12 | maxxer | Samot: yes |
| 14:54.19 | Samot | OK, so how do Digium even know it's a conference call? |
| 14:54.27 | jlewis-highwinds | they wouldn't |
| 14:54.27 | maxxer | Do you want debug? |
| 14:54.30 | Samot | maxxer: Then start looking at your network. |
| 14:54.34 | Samot | Right. |
| 14:54.37 | jlewis-highwinds | to them, it's just a DID they deliver calls to |
| 14:54.40 | Samot | So how could they only bridge conf calls. |
| 14:55.00 | Samot | It's just a channel to them, that's bridged to the PSTN. |
| 14:55.11 | Samot | It's two channels. |
| 14:55.20 | Samot | Your PBX and the PSTN with them in the middle. |
| 14:55.20 | jlewis-highwinds | I don't know that it only happens to our conf calls...I'm just not aware of any reports of it happening on non-conf calls |
| 14:55.30 | Samot | Well then you need to figure that out. |
| 14:55.47 | Samot | Because so far it's unclear as to where this problem really is. |
| 14:55.55 | jlewis-highwinds | how is that? |
| 14:56.14 | Samot | Because the basics dont add up. |
| 14:56.19 | Samot | I call your DID.. |
| 14:56.27 | Samot | PSTN to Digium. |
| 14:56.34 | Samot | Channel A |
| 14:56.44 | Samot | Digium sends the call to your PBX.. |
| 14:56.46 | Samot | Channel B |
| 14:56.56 | Samot | Your PBX answers, channels are bridged. |
| 14:57.03 | Samot | Two channels. |
| 14:57.05 | Samot | That's it. |
| 14:57.14 | Samot | In order for them to have multiple channels bridged.. |
| 14:57.24 | jlewis-highwinds | theoretically, I suppose it could be some remote staff person's telco provider...but the fact that I know at least the most recent instance was two Digium customer DIDs and this started happening after we switched our SIP trunking to Digium makes me suspect Digium |
| 14:57.26 | Samot | They would have to actively move your channels into a conf bridge. |
| 14:58.03 | Samot | I would be more apt to think they are routing calls to your PBX incorrectly. |
| 14:58.12 | Samot | But that your PBX is dealing with that call.. |
| 14:58.15 | Samot | And bridging it. |
| 14:58.19 | jlewis-highwinds | or have a bug that incorrectly bridges calls connecting channel C to A or B |
| 14:58.34 | Samot | That's now how it works. |
| 14:58.40 | WIMPy | Samot: Have you never experienced two RTP streams on the same port? |
| 14:58.43 | Samot | The channels have to be moved to a conf bridge. |
| 14:58.48 | jlewis-highwinds | if they were routing calls to our PBX, we'd have logs of those incoming calls |
| 14:59.05 | Samot | Then you need to figure out if it's all the calls. |
| 14:59.11 | Samot | Because this should happen on all calls.. |
| 14:59.14 | Samot | Not just conf calls. |
| 14:59.19 | GeneralSpongebob | So, trying to set up a fresh Asterisk box from source I'm stuck at make because "reipe for target chan_iax2.so failed" What have I missed? |
| 14:59.32 | Samot | WIMPy: Not that I recall. |
| 15:00.33 | WIMPy | Asterisk has done that a few times to me. |
| 15:00.42 | jlewis-highwinds | trouble is, their support basically just says "that can't happen. Show us logs or go away." |
| 15:00.51 | Samot | Because it can't. |
| 15:00.59 | jlewis-highwinds | and due to what's happening, there are no logs for it on our end |
| 15:01.11 | WIMPy | I just told you how it can. |
| 15:01.25 | Samot | The bridging? |
| 15:01.35 | WIMPy | That's not bridging. |
| 15:01.44 | Samot | What he's talking about? |
| 15:01.52 | [TK]D-Fender | jlewis-highwinds, Show the call for your end |
| 15:01.57 | WIMPy | It just sounds like one if your end receives two streams. |
| 15:01.58 | jlewis-highwinds | I suspect two RTP streams being merged would give the same effect as what we're seeing |
| 15:02.15 | Samot | OK |
| 15:02.17 | Samot | That's not bridging. |
| 15:02.26 | Samot | When you tell a support person (like me) that it's bridging.. |
| 15:02.28 | WIMPy | no |
| 15:02.30 | Samot | And we say it's not.. |
| 15:02.39 | Samot | It can't happen like that. |
| 15:02.40 | jlewis-highwinds | what else do you call it when unrelated calls are connected to each other? |
| 15:03.04 | [TK]D-Fender | jlewis-highwinds, Show the call for your end |
| 15:03.09 | WIMPy | I guess cross-talk is back. |
| 15:03.27 | *** join/#asterisk cmendes0101 (~cmendes01@47-144-223-7.lsan.ca.frontiernet.net) |
| 15:03.38 | WIMPy | Maybe you should do some RTP debugging. |
| 15:03.59 | WIMPy | Wireshark |
| 15:04.04 | jlewis-highwinds | come to think of it...the times I've experienced this, I think we're only getting half of the other call |
| 15:05.04 | Samot | OK, I can accept a cross-talk theory. |
| 15:05.13 | jlewis-highwinds | i.e. we'll either have a person just "show up" on our conf call, or we'll hear ringing, then hear a doctor's office answering system pick up and play a recording |
| 15:05.57 | jlewis-highwinds | but I don't think we've ever gotten both ends of the "other call" |
| 15:06.12 | Samot | Did you tell Digium that? |
| 15:06.33 | GeneralSpongebob | Is anyone here to give me a hint at why I'm getting "recipe for target chan_iax2.so failed" when running make? |
| 15:07.15 | Samot | I can't. I haven't experienced that error myself. |
| 15:08.19 | jlewis-highwinds | GeneralSpongebob: https://www.dialogic.com/den/developer_forums/f/8/t/13416.aspx ? |
| 15:08.31 | WIMPy | GeneralSpongebob: make V=1 (iirc) |
| 15:10.18 | *** join/#asterisk kharwell (kharwell@nat/digium/x-rztdzjwdqqzhkqas) |
| 15:10.18 | *** mode/#asterisk [+o kharwell] by ChanServ |
| 15:11.09 | GeneralSpongebob | Thanks. That forum post looks like it's talking about specific hardware though, I'm just installing on a Ubuntu 16 VM and have followed the Digium installation guide. What is "V=1" & is this just something missing from the guide? |
| 15:11.17 | jlewis-highwinds | Samot: was just checking with IT (they handle Digium)...they did report that the random calls intruding in our bridges have just been one party of the unrelated call |
| 15:11.36 | WIMPy | GeneralSpongebob: Verbose |
| 15:12.07 | Samot | jlewis-highwinds: OK, well I can't speak for Digium.. |
| 15:12.36 | Samot | But generally in cases like this, the provider (Digium in this case) generally doesn't handle the audio on the calls. |
| 15:12.59 | Samot | They proxy it straight through to the provider. |
| 15:13.14 | GeneralSpongebob | ah, I see. It seems to give the same output. "Makefile.rules:138: recipe for target 'chan_iax2.so' failed" |
| 15:14.49 | Samot | jlewis-highwinds: Here's what I suggest. First, you're going to need times/dates a mystery audio stream shows up. Second, you're going to need logs of this call. Third, you're probably going to have to have a recording. |
| 15:14.58 | Samot | To send to Digium. |
| 15:16.02 | *** join/#asterisk rmudgett (rmudgett@nat/digium/x-kinrgglvagdghqtz) |
| 15:16.02 | *** mode/#asterisk [+o rmudgett] by ChanServ |
| 15:21.11 | Samot | It sucks but that's probably what you have to do. |
| 15:22.36 | jlewis-highwinds | we can't have logs since the call is not being delivered here. Recording all our conf bridges for a while might be an option...and would give us better time-stamping for when this happens |
| 15:22.37 | GeneralSpongebob | also "make NOISY_BUILD=yes" gives the same output about chan_iax2.so failing |
| 15:23.13 | Samot | jlewis-highwinds: You have logs of the conf call. |
| 15:23.33 | Samot | The fact that the "mystery" channel isn't there, it's in your favor. |
| 15:23.39 | jlewis-highwinds | yes...but only of our people who actually dialed in...not the "intruders" |
| 15:23.45 | Samot | RIGHT |
| 15:23.54 | Samot | Which means that call ISN'T at your PBX |
| 15:24.04 | Samot | Proof that it's NOT you. |
| 15:24.22 | Samot | They are asking for proof in the form of data. |
| 15:24.44 | Samot | If you're records are absent of that data, then it's in your favor. |
| 15:25.01 | jlewis-highwinds | but the trouble is, finding someone at Digium beyond front-line support who will believe what's happening is actually something worth investigating |
| 15:25.16 | Samot | There's an escalation process for that reason. |
| 15:25.26 | Samot | You escalate and bypass them. |
| 15:25.45 | Samot | But that isn't going to change the fact that data isn't going to be asked for. |
| 15:25.48 | Samot | I know. |
| 15:26.19 | Samot | It may not be Digium. |
| 15:26.35 | Samot | It may be the carrier that Digium is using and how they are sending calls to/from Digium. |
| 15:26.47 | Samot | I don't know. |
| 15:27.03 | Samot | But these are all questions that are going to be asked up to the carrier level. |
| 15:27.53 | GeneralSpongebob | WIMPy, when building should there be hundreds of lines saying "multiple definition of #######" for a lot of things? |
| 15:28.38 | WIMPy | GeneralSpongebob: Nope |
| 15:29.02 | WIMPy | Are you sure your sources are ok? |
| 15:29.04 | GeneralSpongebob | Good. That means I've, what, extracted something twice? |
| 15:29.19 | GeneralSpongebob | freshly downloaded a few hours ago, no errors on untarring |
| 15:29.53 | WIMPy | It might be somethign else on your machine like bad includes somewhere. |
| 15:31.09 | Samot | Like I said, it sucks. It's a PITA. |
| 15:33.43 | GeneralSpongebob | I deleted the directory and then untarred, configure, make.. same error. What other includes should I be checking? |
| 15:35.21 | maxxer | Samot: quite generic :) I noticed I have a lot of RX error in ifconfig... |
| 15:35.35 | Samot | That's a problem. |
| 15:35.35 | maxxer | but anyway why some clients work perfectly and others won't :( |
| 15:35.42 | Samot | It's traffic. |
| 15:35.48 | *** join/#asterisk Oatmeal (~Suzeanne@cpe-65-185-34-151.columbus.res.rr.com) |
| 15:36.15 | Samot | maxxer: Your device boots up. It sees that it has an active SIP account and it should register for it. |
| 15:36.20 | Samot | So it sends a REGISTER |
| 15:36.28 | Samot | The CSeq is 1 |
| 15:36.32 | Samot | It's the 1st one. |
| 15:37.08 | Samot | Asterisk should challenge that. |
| 15:37.42 | Samot | It's going to send a reply with that challenge, it's CSeq looks like CSeq: 1 REGISTER |
| 15:37.59 | Samot | Your device replies back, with the answer to the challenge.. |
| 15:38.13 | Samot | Asterisk replies back with the result, generally a 200 OK. |
| 15:38.32 | Samot | The entire time, all those messages are CSeq: 1 |
| 15:39.23 | Samot | When you have a high CSeq number for a 1st time REGISTER..that's not good. |
| 15:40.25 | Samot | That means the device is sending REGISTERs and timing out. |
| 15:40.35 | Samot | So it's sending again.. |
| 15:40.37 | Samot | and again.. |
| 15:41.17 | GeneralSpongebob | should asterisk-13.2.0.tar.gz be makeable or is it a possible broken development version? |
| 15:41.44 | Samot | Why are you starting with 13.2? |
| 15:41.57 | GeneralSpongebob | 13.xx is LTS |
| 15:42.13 | Samot | 13.13.1 is the current release of 13 LTS |
| 15:43.01 | Samot | I think you about a baker's dozen or so behind updates of the 13.x branch. |
| 15:43.11 | apb1963 | I had my android device/extension working with encryption... and then I managed to mess it up. Not sure what I did but now when I call the extension it goes straight to voicemail... in addition to encryption I was messing with video and ring strategy.. I think that was it. At this point I've attempted to start over from scratch by deleting the extension and reinstalling zoiper on the android - and it's still not working. Log: |
| 15:43.11 | apb1963 | http://fnpaste.com/641y |
| 15:44.25 | apb1963 | Well it doesn't actually go to vm now that I wiped it.. it just gives me a fast busy. |
| 15:44.36 | apb1963 | Although the android rings... and won't stop. |
| 15:44.46 | Samot | INVITE sip:2774@107.146.220.94:42577;transport=UDP;rinstance=824493d1aeecf876 SIP/2.0 |
| 15:44.51 | GeneralSpongebob | I don't know how I've ended up with an old version. I'll go back and try to get the latest |
| 15:45.10 | Samot | SIP/2.0 415 Unsupported Media Type |
| 15:45.25 | Samot | So you're trying to do encryption over UDP? |
| 15:45.29 | apb1963 | Samot, Yeah... I don't know what media type it's referencing |
| 15:46.21 | file | You are trying to use DTLS encryption which they don't support, you should use SDES |
| 15:46.21 | apb1963 | No, I tried to eliminate the encryption and start over. However, my end goal is encryption... presumably over TCP I guess? TLS & SRTP. |
| 15:46.41 | apb1963 | "they" ? |
| 15:46.51 | file | Zoiper |
| 15:47.11 | file | Your SDP contains a DTLS media type |
| 15:47.27 | apb1963 | where are you seeing that? |
| 15:48.37 | file | The m line in the SDP |
| 15:48.54 | file | It should be RTP/SAVP for SDES |
| 15:49.09 | apb1963 | This? m=audio 14464 UDP/TLS/RTP/SAVP 0 8 3 111 101 |
| 15:49.21 | file | UDP/TLS/RTP/SAVP is DTLS |
| 15:49.30 | apb1963 | ah |
| 15:49.37 | file | And that is the media type which it is referring to |
| 15:49.42 | file | In its response |
| 15:50.09 | apb1963 | Yeah I saw the line.. I didn't know what it was telling me. |
| 15:50.27 | apb1963 | ok let me go find that |
| 15:50.37 | Samot | Yeah, so Zoiper is sending that |
| 15:50.53 | Samot | You have to turn it off in Zoiper too. |
| 15:51.14 | file | The Invite is from Asterisk |
| 15:51.38 | file | Zoiper shouldn't need to be touched |
| 15:53.41 | apb1963 | ok so I already had DTLS disabled... |
| 15:53.43 | Samot | Yeah, my bad. |
| 15:54.18 | apb1963 | I guess it's not taking for some reason. |
| 15:54.26 | Samot | "sip show settings" |
| 15:59.53 | apb1963 | http://fnpaste.com/jYm6 |
| 16:01.19 | igcewieling | apb1963: are you using the cellular data or are you using wifi? I've had Verizon wireless messing up SIP traffic, some sort of proxyish thing. |
| 16:06.12 | *** join/#asterisk libardi (~libardi@187.64.235.241) |
| 16:06.53 | Samot | Huh. Well sorry, that was pointless. I guess because I was looking for it, the DTLS settings are not in there. |
| 16:07.00 | Samot | er never looking for it.. |
| 16:07.02 | apb1963 | igcewieling, wireless |
| 16:07.15 | Samot | apb1963: Do a sip reload.. |
| 16:07.23 | Samot | and then make another call and show that. |
| 16:07.31 | igcewieling | apb1963: oddly both cellular and wifi are wireless. that does not help. |
| 16:10.20 | GeneralSpongebob | Thanks for your help, WIMPy & Samot. I've now managed to make asterisk using version 13.13.1 |
| 16:13.36 | apb1963 | igcewieling, wifi |
| 16:13.46 | apb1963 | Samot, http://fnpaste.com/ngqo |
| 16:16.08 | Samot | m=audio 11898 UDP/TLS/RTP/SAVP 0 8 3 111 101 <-- Same thing. |
| 16:16.32 | Samot | Show the settings for this peer, masking the secret= |
| 16:22.26 | apb1963 | Samot, http://fnpaste.com/PvKa |
| 16:23.45 | [TK]D-Fender | <PROTECTED> |
| 16:23.45 | [TK]D-Fender | <PROTECTED> |
| 16:23.52 | [TK]D-Fender | Doesn't LOOK like TLS to me |
| 16:24.08 | [TK]D-Fender | <PROTECTED> |
| 16:24.12 | [TK]D-Fender | really not at all.... |
| 16:24.29 | Samot | I wanted to see the actual peer settings.. |
| 16:24.32 | Samot | from sip.conf |
| 16:26.54 | *** join/#asterisk miralin (~Thunderbi@194.8.128.48) |
| 16:27.19 | drmessano | Yeah that's literally never going to work |
| 16:27.51 | drmessano | Encryption: No |
| 16:28.21 | drmessano | There's 2 settings that enable TLS+SRTP and both of them are wrong |
| 16:53.38 | *** join/#asterisk thiagoc (~thiagoc@unaffiliated/thiagoc) |
| 17:04.59 | *** join/#asterisk skywayskase (~skywayska@204.148.29.50) |
| 17:31.07 | igcewieling | why bother with TLS? the NSA will decrpypt the calls anyway. 8-| |
| 17:36.45 | *** join/#asterisk genpaku (~genpaku@107.191.100.185) |
| 17:37.06 | *** join/#asterisk yokel (~yokel@unaffiliated/contempt) |
| 17:52.25 | drmessano | igcewieling: Perfectly good waste of a chance to Trump Troll |
| 17:53.30 | drmessano | Like "Why bother encrypting, Trump is only going to make illegal" |
| 17:53.41 | drmessano | Like "Why bother encrypting, Trump is only going to make it illegal" |
| 17:55.42 | igcewieling | drmessano: Trump will be gone some day. The NSA won't. |
| 17:56.08 | drmessano | I dont think you've gotten the hang on this yet |
| 17:56.19 | drmessano | "Trump is going to be impeached, next week" |
| 17:57.22 | igcewieling | I don't want to hear about Trump. That's why I stopped watching the news, the daily show, and cobert's show. |
| 17:58.37 | drmessano | Nothing wrong with some good paranoia |
| 17:58.47 | drmessano | It keeps Reynolds in business |
| 17:59.13 | drmessano | Aluminum foil sales are up massively |
| 18:01.21 | igcewieling | drmessano: I'm plenty paranoid. I don't have a cell phone and always use cash, when I can't use cash (like an online purchase) , I use a pre-paid debit card |
| 18:01.44 | drmessano | I'm genuinely paranoid about the beer industry |
| 18:02.03 | *** join/#asterisk rwb (~Thunderbi@204.13.43.166) |
| 18:02.05 | igcewieling | I use an actual one-way text pager for when work needs to contact me. |
| 18:02.24 | drmessano | Because locally sourced craft beer is made by hipsters. Many hipsters haven't come out of hiding or stopped crying in months. It's putting a damper on production |
| 18:02.39 | drmessano | I had to buy.. Budweiser the other day |
| 18:02.42 | igcewieling | Only privacy zealots and drug dealers still use them. |
| 18:03.40 | drmessano | Well, the medical industry too |
| 18:03.51 | igcewieling | ah, I'd forgotten about that. |
| 18:04.33 | drmessano | Yeah, probably 80% of the industry |
| 18:06.56 | Samot | For a paranoid guy, you spend a lot of time online where your conversations are logged. |
| 18:06.59 | Samot | Just saying. |
| 18:08.17 | drmessano | X-No-Archive |
| 18:11.21 | drmessano | Samot |
| 18:11.28 | apb1963 | Samot, drmessano [TK]D-Fender file igcewieling thank you for the help |
| 18:11.34 | drmessano | Slow-Witted Conspiracy Theorist Convinced Government Behind NASA <-- The Onion |
| 18:12.13 | drmessano | âFollow the money and youâll find out who pulls NASAâs puppet strings: Washington, D.C.,â |
| 18:12.31 | igcewieling | "White House Staff Reminded To Place Lids Firmly On Trash Cans After Steve Bannon Gets Into Garbage Again" <- also the onion |
| 18:12.50 | drmessano | HAH |
| 18:13.38 | apb1963 | file, As a user, it would be helpful to me if it printed out the content type so that I know what media/content type it's referring to. Thank you |
| 18:13.58 | Samot | Hey, live how you want to live. |
| 18:14.33 | Samot | I really don't care until it impedes on other people, especially me. |
| 18:14.46 | drmessano | apb1963: It's all there in the SDP |
| 18:14.56 | apb1963 | <PROTECTED> |
| 18:14.57 | apb1963 | <PROTECTED> |
| 18:15.38 | Samot | I still haven't seen the actually sip.conf peer settings. |
| 18:16.01 | apb1963 | drmessano, Yes I suppose for the trained adminstrator it's obvious... but I probably could have avoided a support visit if it printed it out so I knew what it was talking about. |
| 18:16.05 | Samot | Asterisk thinks this peer should be sending audio encrypted.. |
| 18:16.15 | Samot | Your device doesn't like that. |
| 18:16.20 | apb1963 | Samot, I fixed it. Thank you |
| 18:16.23 | Samot | OK. |
| 18:16.33 | Samot | How? |
| 18:16.35 | drmessano | The peer wasn't configured for TLS or SRTP |
| 18:16.37 | file | Zoiper responded with the Unsupported Media Type, not Asterisk |
| 18:16.43 | file | we don't control Zoiper |
| 18:16.45 | Samot | Right. |
| 18:16.50 | Samot | I said Zoiper didn't like it. |
| 18:16.57 | apb1963 | file, I see. OK |
| 18:17.12 | drmessano | 13:16:05 <Samot> Asterisk thinks this peer should be sending audio encrypted.. <-- Not according to the peer config |
| 18:17.19 | Samot | DTLS |
| 18:17.25 | Samot | It's not shown in that. |
| 18:17.55 | drmessano | Literally had the peer set for UDP only and no encryption |
| 18:17.56 | Samot | sip show settings and a sip show peer <peer> was shown. |
| 18:18.23 | Samot | 11:16:10 AMÂ <Samot>Â m=audio 11898 UDP/TLS/RTP/SAVP 0 8 3 111 101 <-- Same thing. |
| 18:18.31 | Samot | ^ That's Asterisk sending the INVITE. |
| 18:18.34 | Samot | To the device. |
| 18:18.36 | apb1963 | Right.. and I had no idea that DTLS is "UDP/TLS/RTP/SAVP" |
| 18:20.22 | apb1963 | drmessano, I was starting over from scratch so I took out encryption just to get a basic config working and that's what I still couldn't get working right the second time around. Then after the various clues dropped here, I turned encryption back on, etc. and it all seems to be working. |
| 18:20.31 | Samot | RTP doesn't get encrypted. |
| 18:20.37 | Samot | It gets encapsulated. |
| 18:21.30 | drmessano | " I turned encryption back on, etc. and it all seems to be working." |
| 18:21.37 | drmessano | So whats the argument here? |
| 18:21.46 | drmessano | His peer config had it OFF |
| 18:21.49 | drmessano | He turned it oN |
| 18:21.51 | drmessano | It's working |
| 18:21.56 | Samot | Right |
| 18:22.02 | Samot | But oddly enough, not how he wanted it. |
| 18:22.14 | Samot | He doesn't want any encryption. |
| 18:22.16 | drmessano | He didn't? |
| 18:22.30 | Samot | So his signaling wasn't encrypted. |
| 18:22.33 | Samot | But his RTP was. |
| 18:22.38 | Samot | We said DTLS. |
| 18:22.46 | Samot | Asterisk was configured to encapsulate the audio |
| 18:23.00 | Samot | Zoiper was configured not to accept that. |
| 18:23.20 | Samot | He was expecting a standard UDP based call on Zoiper. |
| 18:23.50 | drmessano | 10:46:22 <apb1963> No, I tried to eliminate the encryption and start over. However, my end goal is encryption... presumably over TCP I guess? TLS & SRTP. |
| 18:24.03 | Samot | OK you got me. |
| 18:24.06 | Samot | I missed that one. |
| 18:25.29 | Samot | We saw calls going with unencrypted signaling and encapsulated audio. |
| 18:25.35 | Samot | We said "Hey DTLS is on" |
| 18:25.36 | drmessano | He was trying to get back to where he was and mistakingly used DTLS instead of just enabling TLS and toggling Encryption on |
| 18:25.40 | Samot | Right. |
| 18:25.55 | Samot | I was just trying to see the actual DTLS settings. |
| 18:26.04 | Samot | Which I asked for like an hour ago or so. |
| 18:26.23 | drmessano | In his peer settings he still had UDP and No Encryption. Plain as day |
| 18:26.33 | drmessano | Just had to toggle them on |
| 18:27.08 | Samot | Right. Again, I had it backwards. I thought it was in the elimination process and couldn't get back to start. |
| 18:27.19 | Samot | My confusion. |
| 18:27.41 | Samot | I went to the Blue Oyster by mistake. |
| 18:27.44 | Samot | Seriously. |
| 18:27.47 | Samot | Mistake. |
| 18:28.34 | drmessano | I never understood how someone didn't see the sign |
| 18:28.48 | drmessano | All 23 times they accidentally went there |
| 18:29.32 | drmessano | Someone needs to do an analysis of those movies.. Like where they break down Star Wars as centered entirely around the Jawas |
| 18:29.41 | *** join/#asterisk newtonr (RustyNewto@nat/digium/x-uzxkukxdqcbvmxeg) |
| 18:29.41 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 18:29.51 | drmessano | I bet that whole series centered around that bar |
| 18:33.05 | Samot | haha |
| 18:33.17 | Samot | Well some say it's about the droids really...but who knows. |
| 18:33.38 | Samot | When you get into a franchise like that, you have to find some hidden gags. |
| 18:34.00 | Samot | Look at Psych. |
| 18:34.18 | [TK]D-Fender | Just picked up 2 Cisco 7941 for $10 CAD each.... |
| 18:34.27 | [TK]D-Fender | Finally getting something new(old) to play witth.... |
| 18:34.31 | Samot | The pineapple gag. Every episode has a pineapple in it. |
| 18:34.34 | Samot | hahah. |
| 18:34.36 | Samot | OK. |
| 18:34.53 | Samot | So $20 to have something to smash when angry. |
| 18:34.56 | Samot | Not a bad deal. |
| 18:35.04 | [TK]D-Fender | I'm sure they'll work. |
| 18:35.14 | Samot | Says everyone that buys them for SIP |
| 18:35.34 | [TK]D-Fender | Not as fully configurable as my Poly's, but still decent if you know what you're doing and don't raise your expectations too high ;) |
| 18:35.45 | Samot | You also said it would take an hour for Jack17. |
| 18:35.54 | Samot | I have lost some faith in what you say :) |
| 18:36.08 | [TK]D-Fender | That's why I'm not announcing a deadline for functionality ;) |
| 18:36.21 | [TK]D-Fender | You can't fail if you don't set a goal! |
| 18:36.32 | Samot | Trump's motto. |
| 18:36.37 | Samot | It's how he wins so much. |
| 18:37.03 | apb1963 | Any idea what "Subscribe for Register" is used for in zoiper, and do I need it? |
| 18:37.08 | [TK]D-Fender | How many billions of dollars do you have to lose to "win"? |
| 18:37.45 | maxxer | I even rebooted the server as I saw a lot of RX errors on the interface, still nothing :( |
| 18:37.55 | apb1963 | [TK]D-Fender, If it's not your money? Just one. |
| 18:38.03 | [TK]D-Fender | I figured for $7.50 USD a piece I could at the very least hold down paper in a stiff breeze. Or resell them for 3X what I payed..... |
| 18:38.12 | [TK]D-Fender | but I'm bettting I'll get them running as good as they can |
| 18:38.24 | apb1963 | maxxer, You might have a bad NIC or bad cable if not wireless? |
| 18:38.36 | [TK]D-Fender | in the process of flashing for SIP now (1st attempt ever) |
| 18:38.37 | pjensen00 | 7.50 for a paper weight? Kinda pricey but if it makes beepy noises I can see the (probably short lived) novel appeal |
| 18:38.54 | maxxer | apb1963: I will try switching Asterisk on another nic tomorrow :( |
| 18:39.01 | maxxer | that was the next step |
| 18:39.08 | Samot | Well it's like the Go-Bots of SIP phones.. |
| 18:39.18 | Samot | Even when you transform it to SIP, it's shitty. |
| 18:39.27 | apb1963 | maxxer, should be some testing software for NICs |
| 18:39.27 | [TK]D-Fender | pjensen00, There's still that resale potential where I can plead ignorance on setting them up. #caveatemptor |
| 18:39.51 | apb1963 | maxxer, give a holler in #networking and see if anyone has any ideas for you, for your card. |
| 18:40.43 | pjensen00 | Oooohh crafty. I like it. |
| 18:41.08 | Samot | Go-Bots. |
| 18:41.17 | Samot | Of SIP phones. |
| 18:41.26 | [TK]D-Fender | No, that's Grandstream ;) |
| 18:41.41 | Samot | Not really. |
| 18:41.45 | [TK]D-Fender | Or the ancient PA1688 chipset ones from 15 years ago :p |
| 18:41.47 | [TK]D-Fender | ^ |
| 18:41.49 | Samot | It doesn't transform into SIP shit. |
| 18:41.52 | Samot | It comes out SIP shit. |
| 18:41.54 | [TK]D-Fender | THAT is some Go-Bot shit |
| 18:42.19 | Samot | I had Go-Bots *sigh* |
| 18:42.36 | Samot | Not by choice. Also those relatives that can't tell things apart. |
| 18:42.41 | Samot | I collected Transformers. |
| 18:42.57 | Samot | They tried to be nice at Christmas, I get Go-Bots. |
| 18:44.05 | [TK]D-Fender | Well I've got it flashed to SIP. So far so good... now to get it registered.... |
| 18:44.37 | Samot | Then you can claim you got features like hold and call waiting to work. |
| 18:45.23 | [TK]D-Fender | \o/ |
| 18:45.28 | [TK]D-Fender | FOR GREAT VICTORY |
| 18:49.49 | pjensen00 | If I submit a bug report for Asterisk 14 that also affects Asterisk 13, will the bug be addressed in both? Or is there some other complicated routine involved |
| 18:50.43 | file | it is addressed in all applicable branches that are currently supported |
| 18:50.51 | pjensen00 | ok thanks |
| 19:20.36 | *** join/#asterisk eric_hill (~eric_hill@wsip-184-180-163-60.ks.ks.cox.net) |
| 19:20.41 | *** join/#asterisk UncleKiwi (~UncleKiwi@unaffiliated/unclekiwi) |
| 19:22.17 | *** join/#asterisk Demon_VoIP (~demon@109.60.222.253) |
| 19:23.02 | *** join/#asterisk stac (~stac@april-fools/2013/runnerup/stac) |
| 19:25.28 | *** join/#asterisk tonsofpcs (~mythbuntu@rivendell/member/tonsofpcs) |
| 19:25.30 | Demon_VoIP | asterisk 13. PJSIP. Are there any AMI events tell me IP address of successful registration? |
| 19:25.33 | tonsofpcs | I have a custom dialplan that is dialing extensions on my old phone system to link legacy users. I think the digits may be being 'dialed' too fast - is there a way to slow it down? |
| 19:25.37 | tonsofpcs | exten => _3XX,1,Dial(SIP/provider/16075555555,,D(w${exten})) |
| 19:27.34 | *** join/#asterisk andresmujica (~andresmmu@ubuntu/member/andresmujica) |
| 19:28.31 | Demon_VoIP | tonsofpcs, https://wiki.asterisk.org/wiki/display/AST/Application_SendDTMF |
| 19:29.30 | igcewieling | Demon_VoIP: that is not very useful for what he wants |
| 19:30.01 | igcewieling | tonsofpcs: you can put a w between each digit. |
| 19:30.40 | tonsofpcs | igcewieling: how do I do that? |
| 19:30.53 | Demon_VoIP | replace option D to option, for example M(send-dtmf^${exten}) |
| 19:31.11 | Demon_VoIP | two way to do... |
| 19:32.04 | igcewieling | 4 digit extensions something like : ${EXTEN:0:1}w${EXTEN:1:1}${EXTEN:2:1}${EXTEN:3:1} |
| 19:32.08 | igcewieling | ..e.r.. |
| 19:32.26 | igcewieling | <PROTECTED> |
| 19:34.58 | Demon_VoIP | asterisk 13. PJSIP. Are there any AMI events tell me IP address of successful registration? Any event? Any solution :( |
| 19:35.03 | igcewieling | if the number of digits dialed is not always the same, then it would take a bit of dialplan code to accomplish the same thing. |
| 19:35.56 | [TK]D-Fender | Demon_VoIP, You know the WIKI has a very complete list of what's available.... |
| 19:36.52 | Demon_VoIP | [TK]D-Fender, yes. I can't find any event. I don't know how to search? |
| 19:37.04 | [TK]D-Fender | one by one. |
| 19:37.10 | [TK]D-Fender | The names will probably be VERY obvious |
| 19:37.23 | [TK]D-Fender | And if you can'tt fine one... then clearly it does not exist |
| 19:37.27 | Demon_VoIP | [TK]D-Fender, i've tried a lot of... No one. |
| 19:37.37 | [TK]D-Fender | Then there's your answer |
| 19:37.47 | Demon_VoIP | once again |
| 19:37.56 | igcewieling | that is why I still don't use pjsip. |
| 19:38.22 | igcewieling | too much stuff not up to the level of support in chan_sip |
| 19:38.52 | Demon_VoIP | no UA, no IP.. ok. It is fact. |
| 19:41.31 | [TK]D-Fender | https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+ManagerEvent_PeerStatus |
| 19:41.39 | [TK]D-Fender | What does THIS say? |
| 19:42.20 | Demon_VoIP | [TK]D-Fender, no any events for PJSIP registrations. Only SIP, IAX2, PJSIP endpoints (devices). |
| 19:42.42 | [TK]D-Fender | as in? |
| 19:43.11 | [TK]D-Fender | https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+ManagerEvent_Registry |
| 19:44.05 | Demon_VoIP | [TK]D-Fender, i'll try now again. |
| 19:44.59 | Demon_VoIP | [TK]D-Fender, not IP addres |
| 19:45.14 | [TK]D-Fender | ? |
| 19:45.43 | Demon_VoIP | [TK]D-Fender, event doesn't contain IP address fied |
| 19:45.46 | Demon_VoIP | field |
| 19:45.57 | *** join/#asterisk lvlinux (~ruel@unaffiliated/lvlinux) |
| 19:46.10 | [TK]D-Fender | go parse it out from another dump |
| 19:46.20 | [TK]D-Fender | You have the user part to match by |
| 19:46.29 | [TK]D-Fender | etc |
| 19:47.38 | Demon_VoIP | Ok. Do you tell to convert sip domain to IP? With SRV records? With not one SRV records and to be sure I guess one? |
| 19:48.22 | Demon_VoIP | Yes, I did it. I'm trying to guess the IP address. Other way I don't have |
| 19:49.14 | [TK]D-Fender | [TK]D-Fender> go parse it out from another dump <------ |
| 19:49.25 | [TK]D-Fender | * very clearly shows you this in another command |
| 19:50.01 | Demon_VoIP | sorry, i don't understend. What is another command i should parse? |
| 19:50.44 | [TK]D-Fender | what command already tells you this? |
| 19:50.46 | [TK]D-Fender | in CLI |
| 19:51.38 | Demon_VoIP | i see. just a moment |
| 19:55.13 | Demon_VoIP | PJSIPShowRegistrationsOutbound ? Only all of them. not one. There is not i find |
| 19:55.26 | [TK]D-Fender | ALL is fine |
| 19:55.35 | [TK]D-Fender | you have most of the name to search for |
| 19:56.36 | Demon_VoIP | >5000 AMI events after each register events? |
| 19:58.00 | Demon_VoIP | I do that when script stars to sync states... Only when start. |
| 19:58.22 | [TK]D-Fender | huh? |
| 20:00.41 | Demon_VoIP | [TK]D-Fender, thanks for the desire to help |
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| 21:06.30 | igcewieling | Demon_VoIP: It is more of a compulsion rather than a desire. |
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| 23:35.14 | Samot | Well just had to dump a contractor. |
| 23:35.17 | Samot | Idiot. |
| 23:36.26 | Samot | Alas the issues of having non-local clients. |
| 23:46.43 | pjensen00 | We have had 5:1 ratios of bad:good |
| 23:46.58 | pjensen00 | The one we found that's good is REALLY good. The rest were....... |
| 23:47.08 | pjensen00 | I wish I had more dots ...... |
