| 00:00.05 | *** join/#asterisk nix8n82 (~AndChat27@67-130-74-235.dia.static.qwest.net) |
| 00:02.57 | *** join/#asterisk evil_gordita (robert@ip70-188-56-139.rn.hr.cox.net) |
| 00:52.38 | *** join/#asterisk burnbrighter (~kurt@50-79-211-169-static.hfc.comcastbusiness.net) |
| 00:58.23 | *** join/#asterisk burnbrighter (~kurt@50-79-211-169-static.hfc.comcastbusiness.net) |
| 00:58.54 | *** join/#asterisk michael_work_ (~michael_w@bzq-82-168-31-134.red.bezeqint.net) |
| 01:01.39 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 02:02.36 | *** join/#asterisk slav3_kitten (~frankthet@unaffiliated/slav3-kitten/x-0866809) |
| 02:09.45 | *** join/#asterisk slav3_kitten (~frankthet@unaffiliated/slav3-kitten/x-0866809) |
| 03:12.28 | *** join/#asterisk geek-man (~geek-man@59-100-0-250.syd.static-ipl.aapt.com.au) |
| 03:14.35 | geek-man | Hi all, I'm having an issue with Asterisk Realtime and SIP registrations, I was wondering if someone could help? |
| 03:15.00 | geek-man | I'm trying to setup a two-node active/active "cluster", and I'm using RT to store SIP registrations for the endpoints. |
| 03:15.08 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 03:15.21 | *** join/#asterisk jasonwert (~jasonwert@71.89.137.28) |
| 03:15.47 | geek-man | I've found that when using cached RT registrations, with qualify configured, registrations from both servers will appear on each node. |
| 03:16.10 | geek-man | And only once the qualify timeout expires does the registration become unreachable. |
| 03:17.42 | geek-man | This is a problem for me because I've written a "clusterDial" subroutine that uses ChanIsAvail to detect if a SIP account is locally registered or not, and if a SIP account has not already had it's registration cached, it gets loaded into memory and temporarily is seen as reachable from the wrong node. This leads ChanIsAvail to incorrect return an available channel to the device. |
| 03:19.15 | geek-man | I was wondering if there's any way I can either 1) prevent registrations against server B from being loaded into memory on server A and vice versa; or 2) When loading registrations into the cache, can they be set to default as unreachable? |
| 03:37.31 | cyford33 | i clustered using 2 nfs servers for all asterisk files, and a central db with realtime or freepbx- and opensips to ballance both |
| 03:37.57 | cyford33 | opps i mean 2 asterisk 1 nfs nas server |
| 03:39.38 | geek-man | I'm not using OpenSIPs⦠The registrations of the end devices are going straight to Asterisk. Perhaps that coud be changing my experience. |
| 03:43.00 | mjordan | It is. In the SIP Proxy + Asterisk deployment, Asterisk generally has little to no knowledge of the location of the endpoints, as the SIP proxy is the registrar |
| 03:43.29 | mjordan | geek-man: you could use Views in your SQL database on the shared table, one for each server |
| 03:43.43 | mjordan | you'll need to find how you want each server's view to be constructed however. |
| 03:46.59 | geek-man | Ah yes. Views is a good idea. |
| 03:47.49 | geek-man | Basically, separate tables is the only way from preventing these false cached registrations from appearing? Aside from offloading registrations to OpenSIPs/Kamilio/etc? |
| 03:49.07 | geek-man | I've thought about using a separate SIP proxy, but using Realtime is already significantly more complex than our existing setup (just static text files), and I want to try and put a cap on the level of complexity -- not too many different parts to the solution. |
| 03:49.46 | geek-man | Would you guys say that something like Kamilio/OpenSIPs is a better/easier solution overall? Or should I just go with table views? |
| 03:54.48 | cyford33 | in my setup even though its 2 servers, everything was identical and using realtime. so i believe if u was registered on the one, u was registered on the other.. and if u was on server a and dialing a extenstion which landed on server b then routing was enabled.. |
| 03:55.59 | cyford33 | only issue i had needing opensips was to keep it persistant if a phone registers on A then all calls need to go to A server B would shoot Unathorized... |
| 03:56.20 | geek-man | cyford33: That's what I'm aiming for. But it doesn't work right when the device is registered against both system, it tries to Dial locally and fails -- and eventually the "qualify" fails and it becomes unreachable anyway, so I didn't think you COULD have devices registered to both at the same time. |
| 03:56.42 | cyford33 | i also used A2billing, and opensips authenticated using mysql db that was ran through opensips |
| 03:56.45 | geek-man | At the moment, I've configured IAX+DUNDi to route calls between both nodes. |
| 03:57.17 | geek-man | And I've written a subroutine to replace Dial() calls, which will detect if a SIP device is not local, and then route via DUNDi if it has to. |
| 03:57.41 | geek-man | But then, this is where I'm running into the issue with ChanIsAvail and Realtime. |
| 03:58.00 | geek-man | I'm thinking about just removing SIP registrations from Realtime entirely. |
| 03:58.13 | cyford33 | the server used @Hostname for registrations |
| 03:58.32 | cyford33 | 2 servwers to differnt hostnames in the same db |
| 03:59.10 | geek-man | Hrm okay⦠How dod you do that? |
| 03:59.16 | geek-man | it might be simpler? |
| 03:59.57 | geek-man | I don't believe I'm using "@hostname" for my registrations. |
| 04:00.04 | geek-man | I'm not familiar with that. |
| 04:00.24 | cyford33 | it was also freepbx. i forget exactlly, but i think there was actually a gui option to use hostname |
| 04:00.32 | geek-man | Ah OK. |
| 04:00.35 | cyford33 | it was a while back.. |
| 04:00.44 | geek-man | I'm just using plain Asterisk 11, I'll have to investigate. |
| 04:00.55 | geek-man | Thanks for the pointer though! |
| 04:01.36 | cyford33 | if u can use a variable in the db it will always echo on system correctlly |
| 04:01.56 | cyford33 | im thinking , but not sure.. |
| 04:02.47 | cyford33 | i will be at trying to do the same setup very soon |
| 04:04.19 | *** join/#asterisk spditner (~simon@76-10-137-81.dsl.teksavvy.com) |
| 04:05.07 | cyford33 | i think by defualt it uses ip.. and that should work too.. in dialplan if $exten = @serverB ip route through trunk to server b |
| 04:06.26 | geek-man | Hrm. That's a good idea. |
| 04:07.57 | *** join/#asterisk MarkSX (~MarkSX@unaffiliated/marksx) |
| 04:19.09 | *** join/#asterisk bazman (~army@202.161.89.18) |
| 04:41.00 | *** join/#asterisk Vutral (~ss@mirbsd/special/Vutral) |
| 04:57.06 | *** join/#asterisk timahvo1 (~rogue@mail.cickenya.com) |
| 04:59.50 | *** part/#asterisk mjordan (mjordan@nat/digium/x-ddxioqsxykeinhty) |
| 05:01.52 | *** join/#asterisk geek-man (~geek-man@59-100-0-250.syd.static-ipl.aapt.com.au) |
| 05:23.40 | *** join/#asterisk mjordan (~mjordan@75.76.55.191) |
| 05:23.40 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 05:25.43 | *** join/#asterisk rudilee (~rudilee@116.50.28.75) |
| 05:33.11 | *** join/#asterisk rhys-h (~geek-man@1.130.180.59) |
| 05:38.22 | *** join/#asterisk gryphon (~gryphon@82.140.120.164) |
| 05:49.12 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
| 05:58.27 | *** join/#asterisk geek-man (~geek-man@59-100-0-250.syd.static-ipl.aapt.com.au) |
| 06:47.02 | *** join/#asterisk jhlavacek (~jirka@84.19.95.180) |
| 06:49.45 | *** join/#asterisk tparcina (~tomo@212.92.200.41) |
| 06:55.58 | *** join/#asterisk bkruse (~Adium@172.56.28.25) |
| 07:03.09 | *** join/#asterisk ghostlines (~ghostline@i191002.upc-i.chello.nl) |
| 07:05.28 | *** join/#asterisk ghostlines (~ghostline@i191002.upc-i.chello.nl) |
| 07:05.48 | *** join/#asterisk bulkorok (~Benjamin@85.183.61.47) |
| 07:22.23 | *** join/#asterisk riess82 (~riessma@mail.p-riess.at) |
| 07:58.54 | *** join/#asterisk jhlavacek (~jirka@84.19.95.180) |
| 07:59.06 | *** join/#asterisk gerhard7 (~gerhard7@77-172-82-111.ip.telfort.nl) |
| 08:08.34 | *** join/#asterisk Darkerr (~Libor@static-84-42-235-44.net.upcbroadband.cz) |
| 08:18.12 | *** join/#asterisk Vutral (~ss@mirbsd/special/Vutral) |
| 08:21.05 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 08:33.02 | *** join/#asterisk gringo (~gringo@unaffiliated/gringo) |
| 08:41.06 | *** join/#asterisk aurs (~aurs@83.84-48-57.nextgentel.com) |
| 08:43.18 | *** join/#asterisk CustosL1men (~CustosLim@unaffiliated/cust0slim3n) |
| 08:50.45 | Kunsi | hm, i've got a question regarding call pickup |
| 08:53.32 | Kunsi | i have two contexts, one for internal phones, other for connections via POTS. POTS extension dials SIP/76&SIP/1000 - now i want to pick up that call from internal context - which works if i set __PICKUPMARK and create a dedicated internal extension wo pick up that channel, but i can't use a BLF (SIP/76) on my snom phone wo pick up that call |
| 08:54.51 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 08:55.01 | Kunsi | what i want: POTS is ringing, BLF (SIP/76) is flashing, I press button on my other phone and get the call |
| 08:56.40 | Kunsi | currently, i have http://pastebin.com/srN7JjPh - where snoms use context "internal" and POTS uses "incoming" |
| 09:02.37 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 09:24.17 | *** join/#asterisk bibz (~bibz@ip212232024148.rev.nessus.at) |
| 09:25.38 | bibz | hey there. I'm using asterisk 11.6-cert2 and I keep randomly getting this problem: there are quite a few calls per day which can't get established and they terminate with the hangupcause: SIP 180 Ringing or SIP 100 Trying.. |
| 09:25.53 | ChannelZ | Kunsi: Well BLF is just an indication. Presumably the snom has a means to configure what happens when you press the blinking line key (IE dial an extension which in turn performs the pickup) |
| 09:26.37 | bibz | I can provide a trace for one of those calls if its needed.. |
| 09:28.30 | Kunsi | ChannelZ: yes, it has, but if i configure BLF to call *1, i can't pick up SIP/76 via *876 - so i basically want to pick up incoming@PICKUPMARK using *876 |
| 09:29.22 | *** join/#asterisk areski (~areski@80.174.128.104.dyn.user.ono.com) |
| 09:33.29 | ChannelZ | well if you're wanting to pickup a specific extension (76) it seems to me like you want to make it Pickup(${EXTEN:2}@Softphone) |
| 09:34.36 | ChannelZ | or @internal if that's what the peer's context is. I'm not actually sure if Pickup 'sees through' included contexts |
| 09:40.22 | Kunsi | Pickup(${EXTEN:2}@internal) is working, yes |
| 09:42.03 | Kunsi | but i thing we're not understanding each other :p |
| 09:44.04 | Kunsi | but after reading a lot of web pages, i understand that it's not possible to do what i want |
| 09:44.23 | Kunsi | unless i put stuff from incoming into internal |
| 09:44.54 | bibz | I wish there would be more documentation out there on asterisk :( |
| 09:45.09 | ChannelZ | Well you've said two different things.. that you configured the phone to dial *1 for BLF pickup, and that you want to pickup a specific ringing extension with *8xxx - they're two different extensions in your dialplan that do two different things. |
| 09:46.09 | Kunsi | say, 1000@incoming is ringing SIP/76&SIP/1000 - i now want to pick up that call using SIP/91, but i can't use Pickup(76@internal), because it's a different context and a different extension |
| 09:47.24 | ChannelZ | Because you want to pickup 1000 not 76. 1000 is the extension that got dialed causing SIP/76 and SIP/1000 to be ringing. |
| 09:47.36 | ChannelZ | You're confusing extensions with devices which aren't related. |
| 09:50.17 | Kunsi | mh |
| 09:50.21 | Kunsi | misunderstand that |
| 09:50.37 | Kunsi | so Pickup(1000@incoming) should work then? |
| 09:52.09 | ChannelZ | IE SIP/76 is not "extension 76" - it's a SIP device you named 76. "extension 76" is something in your dialplan that you programmed to do something -- possibly dial SIP/76, but extension 76 and SIP/76 don't inherently have anything to do with each other just because they both have 76 in them |
| 09:55.30 | ChannelZ | Make it less confusing. Say you renamed SIP/76 to SIP/Joe. exten => 76 dials SIP/Joe. The Pickup extension would be 76, because that's the thing in your dialplan (the extension) causing SIP/Joe (the device) to ring. |
| 09:58.09 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:4cb0:63dd:fda3:f086) |
| 09:59.55 | *** join/#asterisk michael_work_ (~michael_w@bzq-82-168-31-134.red.bezeqint.net) |
| 10:01.38 | bibz | I've just set progressinband=never and prematuremedia=no, maybe this will help with my SIP 180 Ringing problem.. |
| 10:06.56 | *** join/#asterisk rhys-h (~geek-man@59-100-0-250.syd.static-ipl.aapt.com.au) |
| 10:07.01 | *** join/#asterisk war9407 (war@c-71-62-169-29.hsd1.va.comcast.net) |
| 10:13.17 | Kunsi | ChannelZ: ok, got it now. different question, if i have an extension 1, which uses Goto(2), do i have to pick up 1 or 2? |
| 10:14.50 | ChannelZ | 2 |
| 10:14.55 | Kunsi | ok |
| 10:14.57 | Kunsi | thank you |
| 10:15.30 | ChannelZ | Sure. It's ultimately whatever extension executed the Dial application |
| 10:16.14 | Kunsi | ok |
| 10:16.26 | Kunsi | should i use Goto() or ChannelRedirect()? |
| 10:17.25 | bibz | is this the right place to look for freelancers? and are there any rules for this channel? if so, where do I find them |
| 10:18.07 | Kunsi | hm, i think ChannelRedirect() isn't what i think it is |
| 10:19.27 | ChannelZ | ChannelRedirect is more for.. well, redirecting arbitrary channels. Goto is what you want for just general dialplan logic jumping |
| 10:20.07 | ChannelZ | bibz, yeah you can try to find help for hires here |
| 10:21.43 | ChannelZ | (see the 'code of conduct' link in the channel topic for general rules to live by :) |
| 10:24.20 | ChannelZ | IRC specifics are things like no spamming, no flooding; if you have configs/debugs to share, don't barf 10 lines in here, use pastebins and give the link instead |
| 10:26.03 | ChannelZ | goes to bed |
| 10:26.23 | Kunsi | good night (even if it's 11:26am here in germany) |
| 10:26.34 | ChannelZ | 3am for me :) |
| 10:27.13 | Kunsi | so, PST? |
| 10:27.48 | ChannelZ | MST for me (Colorado USA) |
| 10:28.59 | Kunsi | ok |
| 10:31.55 | *** join/#asterisk CriminalMinds (~hola@unaffiliated/liuyan) |
| 10:32.16 | *** join/#asterisk ghostlines (~ghostline@82.201.93.50) |
| 10:43.01 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:d4b0:ea1c:5375:f316) |
| 10:47.48 | *** join/#asterisk bsdice (~bsdice@meran.embinet.com) |
| 10:50.10 | *** join/#asterisk wasanzy (~wasanzy@41-66-254-58-dedicated.4u.com.gh) |
| 10:50.12 | wasanzy | hello |
| 10:50.31 | bibz | hello |
| 10:51.18 | wasanzy | is it possible to do something like this? dial 500*12 so 12 is extension under 500, so when it is dial like that, u are taken straight to the extenion 12 |
| 10:54.31 | Kunsi | wasanzy: you wnat to call an extension not in your phones context? |
| 10:56.26 | wasanzy | Kunsi: I mean I want to be able to dial an extension straight to the system under my phone context. say the code 500 will take me to the phone context, 500*12 should send me to extension 12 at once |
| 10:56.44 | Kunsi | something like "exten => _XXX*XX,1,Goto(${EXTEN:-3},${EXTEN:5},1)" may work |
| 10:57.07 | bsdice | make sure you catch user errors |
| 10:57.11 | bsdice | boy that is ugly |
| 10:57.27 | wasanzy | bsdice: me? |
| 10:57.47 | *** join/#asterisk FreezingCold (~FreezingC@135.0.41.14) |
| 10:57.49 | bsdice | yes |
| 10:58.03 | wasanzy | my idea is ugly? |
| 10:58.32 | bsdice | idea and solution |
| 10:58.43 | bsdice | okay but maybe it works though :) |
| 10:58.46 | bsdice | whatever... |
| 11:01.30 | bsdice | maybe my idea of "extension" is not quite what you think it is; for me, extension is a number not visible from outside of the dialed up pbx, i.e. you can't dial it directly but have to rely on other party's dialplan to be put through |
| 11:02.32 | *** join/#asterisk realityloop (~realitylo@c114-77-94-34.sunsh1.vic.optusnet.com.au) |
| 11:03.18 | realityloop | I'm having trouble getting inbound calls to work, is anyone able to help me through debugging it please? |
| 11:03.29 | realityloop | outbound calls are working fine |
| 11:03.46 | realityloop | inbound the call just drops pretty much |
| 11:04.00 | bsdice | check on firewall if RTP is getting through |
| 11:04.04 | realityloop | and isn't shown in Call Detail Records.. |
| 11:04.31 | bsdice | or UDP to 5060 for call signalling for that matter |
| 11:06.06 | realityloop | bsdice: ports are forwarded.. |
| 11:06.17 | realityloop | the server is behind nat |
| 11:07.29 | bsdice | use ngrep on internal interface "ngrep -d eth2 udp and port 5060" if it is Linux to see what reaches internal asterisk when call comes in |
| 11:07.59 | bsdice | you have to learn SIP protocol to understand what might go wrong |
| 11:09.28 | realityloop | what does -d do with grep, seems the version I have here doesn't support it.. |
| 11:09.59 | bsdice | device |
| 11:10.07 | bsdice | ngrep not grep! |
| 11:10.11 | bsdice | network grep |
| 11:10.15 | realityloop | box doesn't have ngrep |
| 11:10.29 | realityloop | it's a NAS |
| 11:11.35 | bsdice | does "sip show registry" on Asterisk console show Registered ? |
| 11:12.00 | realityloop | bsdice: yes |
| 11:18.11 | bsdice | you have to debug your network, maybe your NAS messes with it too much |
| 11:18.29 | bsdice | you have to discern if 5060 reliably reaches your Asterisk from outside |
| 11:18.38 | bsdice | even after long waits (NAT timeout) |
| 11:19.05 | bsdice | if that works, you have to debug RTP ports and examine, if RTP is flowing as it should |
| 11:19.11 | bsdice | from outside over your NAS to inside |
| 11:21.55 | realityloop | bsdice: wouldn't the fact that I can hear audio properly in both directions on an outgoing call mean that is working? |
| 11:22.26 | Kunsi | realityloop: if you call out, port 5060 is not needed, since you are connecting |
| 11:22.36 | realityloop | Kunsi: ah.. |
| 11:22.44 | Kunsi | otherwise, firewall or something blocks 5060 incoming |
| 11:23.23 | realityloop | Kunsi: I just updated the target IP for my port forward from my working FreePBX server.. so unless it is the NAS specifically stopping it.. |
| 11:31.28 | *** join/#asterisk michael_work (~michael_w@bzq-82-168-31-134.red.bezeqint.net) |
| 11:32.20 | *** join/#asterisk michael_work_ (~michael_w@192.117.186.204) |
| 11:37.42 | *** join/#asterisk michael_work (~michael_w@192.117.186.204) |
| 11:38.59 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 11:39.53 | *** join/#asterisk andresmujica (~andresmmu@ubuntu/member/andresmujica) |
| 11:51.58 | *** join/#asterisk michael_work_ (~michael_w@192.117.186.204) |
| 11:54.59 | realityloop | sorted.. it was some config settings on the trunk.. |
| 11:55.21 | realityloop | thanks for trying to help bsdice & Kunsi |
| 11:58.09 | *** join/#asterisk CeBe (~CeBe@port-92-200-15-215.dynamic.qsc.de) |
| 12:06.41 | *** part/#asterisk realityloop (~realitylo@c114-77-94-34.sunsh1.vic.optusnet.com.au) |
| 12:07.27 | *** join/#asterisk areski (~areski@81.Red-83-53-30.dynamicIP.rima-tde.net) |
| 12:10.37 | *** join/#asterisk TriJetScud (~TriJetScu@van-app-svr.ad.v10networks.ca) |
| 12:11.07 | *** join/#asterisk nanoha-sama (~nanoha-sa@van-app-svr.ad.v10networks.ca) |
| 12:13.46 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 12:20.47 | *** join/#asterisk geek-man (~geek-man@59-100-0-250.syd.static-ipl.aapt.com.au) |
| 12:21.05 | *** join/#asterisk qdel (~Rholk@bny92-1-82-67-178-101.fbx.proxad.net) |
| 12:34.48 | *** join/#asterisk michael_work_ (~michael_w@bzq-82-168-31-134.red.bezeqint.net) |
| 12:36.45 | cyford33 | after a start pjsips crashes with /usr/sbin/safe_asterisk: line 164: 4482 Segmentation fault (core dumped) nice -n $PRIORITY "${ASTSBINDIR}/asterisk" -f ${CLIARGS} ${ASTARGS} > /dev/${TTY} 2>&1 < /dev/${TTY} |
| 12:37.16 | cyford33 | when a pjsip exten is dialed |
| 12:37.18 | Kunsi | in MixMonitor(name, options, command), i suppose command is an asterisk command!? |
| 12:37.23 | *** join/#asterisk riess82 (~riessma@mail.p-riess.at) |
| 12:39.22 | Kunsi | cyford33: please provide a backtrace of asterisk crash (using gdb and core file asterisk created) |
| 12:39.41 | wasanzy | apart from using read() what other way can I make make asterisk accept more than one digit terminated by # |
| 12:40.26 | *** join/#asterisk wonderworld (~ww@ip-62-143-156-254.hsi01.unitymediagroup.de) |
| 12:40.59 | WIMPy | wasanzy: Make lots of extensions. |
| 12:43.51 | wasanzy | WIMPy: I don't understand, so when I call, I should be able to press 123# |
| 12:45.32 | *** join/#asterisk brad_mssw (~brad@66.129.88.50) |
| 12:45.47 | WIMPy | Make an extension for every possible length. |
| 12:48.43 | cyford33 | Kunsi where is the core file located |
| 12:48.57 | Kunsi | usually in ~ |
| 12:51.50 | wasanzy | is actually a channel number the system will generate automatically, so I will need # as terminator to indicate end of channel number |
| 12:53.14 | WIMPy | When/where do you need it? Read() would wait for # by default. |
| 12:53.35 | *** join/#asterisk MadHatter42 (~MadHatter@unaffiliated/madhatter42) |
| 12:54.06 | wasanzy | oh if read() will wait for # by default, then I think it is the right thing |
| 12:54.45 | wasanzy | when u dial in, the system will ask u to enter channel number follow by # then u are taken to that channel |
| 12:57.53 | Kunsi | hm, i'm getting http://pastebin.com/z32btjE2 when trying to compile asterisk 13.1.0 |
| 13:00.52 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 13:02.04 | cyford33 | Kunsi , http://pastebin.com/UJcrLXPz |
| 13:07.43 | Kunsi | hm, don't really know much about pjsip, but maybe one of the others is able to help you |
| 13:10.55 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 13:12.35 | wasanzy | when u dial in, the system will ask u to enter channel number follow by # then u are taken to that channel |
| 13:16.02 | Kunsi | wasanzy: you already said that |
| 13:20.48 | *** join/#asterisk FreezingCold (~FreezingC@135.0.41.14) |
| 13:23.52 | WIMPy | wasanzy: Make it happen. |
| 13:24.40 | krapper | good morning... how would one prevent asterisk from accepting calls on the default context for clients who aren't registered but are attempting to use as a SIP proxy? We use explicit firewall rules today but I want to create a solution where this box could be open to registrations from any IP. Essentially want to ensure the box is secure from clients that are NOT registered. :-) |
| 13:25.52 | WIMPy | Not possible. |
| 13:26.09 | WIMPy | Use a proxy that supports that. |
| 13:26.45 | bsdice | krapper put up a stateful firewall that keeps track of outgoing registrations (via IP) |
| 13:27.08 | bsdice | use qualify to keep those states open in the firewall |
| 13:27.13 | bsdice | everyone else will be blocked |
| 13:27.28 | bsdice | do you have clients from external sources registering as well? |
| 13:27.39 | krapper | bsdice: all registrations are external sources |
| 13:28.10 | bsdice | put all abusers in default context, play them a nice message |
| 13:28.13 | bsdice | then hang up, done |
| 13:28.27 | bsdice | there will always be abuse attempts on 5060 |
| 13:29.22 | bsdice | Playback(feature-not-avail-line) Playback(goodbye) Wait(1) Hangup(54) |
| 13:30.42 | bsdice | requires sip.conf context=incoming_guests and allowguest=yes or similar |
| 13:31.12 | bsdice | you split guests (default context) and users (not default context) and done |
| 13:31.13 | krapper | we don't use the default context for anything today.. am i correct in that all abusers attempts will always be directed to the default context... so this is where they are greeted with funny allison? |
| 13:31.40 | [TK]D-Fender | krapper: allowguest=no" <- Done. |
| 13:32.03 | bsdice | what do you have as context= in [general]? |
| 13:32.18 | bsdice | usually it is called "default" |
| 13:32.20 | [TK]D-Fender | krapperwe don't use the default context for anything today.. am i correct in that all abusers attempts will always be directed to the default context... so this is where they are greeted with funny allison? <- being greeting = dialplan. THat's for you to create. |
| 13:32.43 | [TK]D-Fender | krapper: But you said you just wanted to block them. So "allowguest=no" . |
| 13:33.03 | krapper | [TK]D-Fender: allowguest=no perfect! |
| 13:33.21 | krapper | correct.. we don't want to waste anytime with abusers... hehe |
| 13:34.22 | bsdice | I recommend to test this well |
| 13:34.43 | bsdice | i.e. simulate an abuser |
| 13:34.57 | bsdice | just to make sure everything works as expected |
| 13:35.16 | *** join/#asterisk qdel (~Rholk@bny92-1-82-67-178-101.fbx.proxad.net) |
| 13:35.52 | krapper | yup, just did... NOTICE[2212][C-00000058]: chan_sip.c:25357 handle_request_invite: Failed to authenticate device <sip:adsf@meow.meow.org;transport=UDP>;tag=ea308fds |
| 13:35.54 | bsdice | I once ran a fake answer and recorded everything the other side said |
| 13:36.54 | bsdice | my text for default (abuser context) went like "Hello? (Pause) Hello who is this? (Pause) Are you Zaid? From Tel-Aviv, my long lost brother?" |
| 13:37.14 | krapper | bahaha |
| 13:37.27 | krapper | then monitor those channels, funny! |
| 13:37.40 | bsdice | I scripted it to send me an MP3 of it by email |
| 13:37.55 | bsdice | inserted long pauses to get a good voice print |
| 13:38.04 | bsdice | now I am waiting for Google to invent reverse audio search ;-) |
| 13:38.11 | bsdice | like with pictures only audio |
| 13:38.23 | bibz | still experiencing the "SIP 183 Session Progress" hangup-cause on random calls, even after setting progressinband to never... any ideas? |
| 13:38.31 | krapper | should have it attempt a SIP device and really speak with them... other then forward on to that for MP3 delivery. hehe |
| 13:39.31 | bsdice | krapper I thought of that but given those were Chinese, Russian, Spanish and Latino time zones we are talking about here (basically round the clock) I valued my sleep somewhat more |
| 13:39.47 | *** join/#asterisk creativx (~ntttt@226.62-97-205.bkkb.no) |
| 13:39.49 | krapper | bsdice, right right :-) |
| 13:40.33 | bsdice | "why are you on amphetamins??" "need to stay up all night to talk to asshats trying to abuse my voip server" uhhh - no :) |
| 13:45.55 | *** join/#asterisk MadHatter42 (~MadHatter@unaffiliated/madhatter42) |
| 13:46.58 | krapper | any pointers on T.38? I'm done a bit of research and still struggling with what I believe would be the reinvite on outbound faxes from simple machines via ATA adapter. (Grandstream HT701) It does appear that the signaling is detected on inbound fax calls and udptl is invoked. |
| 13:47.05 | *** join/#asterisk CeBe1 (~CeBe@port-92-200-15-215.dynamic.qsc.de) |
| 13:48.38 | bsdice | no clue, I am using iaxmodem with hylafax for that - works at 14.400bps at times when provider calls in using G.722 wideband |
| 14:09.55 | Kunsi | hm, i'm getting http://pastebin.com/z32btjE2 when trying to compile asterisk 13.1.0 |
| 14:11.27 | file | your PJSIP does not appear to have shared libraries which are required for the functionality in Asterisk |
| 14:11.54 | *** join/#asterisk coppice (~chatzilla@123203240102.ctinets.com) |
| 14:12.09 | file | Kunsi, how did you install pjsip? |
| 14:23.15 | Eric-K | I have this strange error popping up on the Asterisk 11 console: Codec mismatch on channel SIP/1234-00001463 setting write format to slin from alaw native formats (alaw) |
| 14:23.31 | Eric-K | Strange because there is no audio problem and alaw works like it should. |
| 14:23.49 | Eric-K | Why does Asterisk give me this error? |
| 14:24.14 | *** part/#asterisk rue_bed (~rue@d205-250-205-216.bchsia.telus.net) |
| 14:28.42 | wasanzy | WIMPy: I am lost of ideas that is why am asking |
| 14:29.11 | Kunsi | file: i didn't |
| 14:29.23 | Kunsi | (or did I? let me check) |
| 14:29.23 | file | Kunsi, it's on your system... |
| 14:30.34 | Kunsi | hm |
| 14:30.42 | Kunsi | seems like "compile from source" |
| 14:32.33 | Kunsi | yeah |
| 14:32.43 | Kunsi | file: compiled from source, default options |
| 14:32.54 | file | then that would be why |
| 14:32.59 | file | PJSIP by default does not build shared libraries |
| 14:33.05 | *** join/#asterisk newtonr (newtonr@nat/digium/x-slmwydxluelfftif) |
| 14:33.05 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 14:33.23 | Kunsi | so i'd have to --enable-shared in pjsip? |
| 14:33.27 | file | https://wiki.asterisk.org/wiki/display/AST/Building+and+Installing+pjproject |
| 14:34.36 | Eric-K | Kunsi, I configured PJSIP on Debian 7 x64 as follows (might be of help): ./configure --prefix=/usr --enable-shared --disable-sound --disable-resample --disable-video --disable-opencore-amr CFLAGS='-O2 -DNDEBUG' |
| 14:34.55 | Eric-K | make dep, make, make install, ldconfig |
| 14:35.53 | file | you need to remove your old pjproject or install in the same place, otherwise you'll have two installs on the system and it'll go wonky |
| 14:36.08 | Kunsi | yeah, i know that ⺠|
| 14:36.31 | *** join/#asterisk sruffell (sruffell@asterisk/the-kernel-guy/sruffell) |
| 14:36.31 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 14:37.07 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:37.08 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:37.15 | *** join/#asterisk ks3 (~ks3@205.142.197.104) |
| 14:38.20 | *** join/#asterisk D-Boy (~D-Boy@unaffiliated/cain) |
| 14:39.03 | hfor | :D AlarmReceiver() is awesome. :) |
| 14:59.08 | *** join/#asterisk aross42 (~aross@192-0-133-151.cpe.teksavvy.com) |
| 14:59.14 | *** join/#asterisk slackology_ (~slackolog@unaffiliated/quintux) |
| 15:03.44 | *** join/#asterisk bkruse (~Adium@12.22.93.195) |
| 15:06.21 | *** join/#asterisk creativx (~ntttt@226.62-97-205.bkkb.no) |
| 15:06.43 | *** join/#asterisk jeffspeff (~Jeff@12.49.160.131) |
| 15:07.52 | *** join/#asterisk michael-i (~michael-i@157.Red-83-58-34.dynamicIP.rima-tde.net) |
| 15:09.20 | michael-i | Hi all. Just curious if anyone has a working sip peer+user for a Ring Central manually provisioned phone. I'm having trouble mapping their SIP Domain, Outbound Proxy, Username, Auth ID and password into a valid setup. Currently registration works but inbound/outbound calls fail. |
| 15:14.18 | *** part/#asterisk LiuYan (~hola@unaffiliated/liuyan) |
| 15:15.06 | Kunsi | okay, pjsip works now, but i get another error: http://pastebin.com/dTTENrzD |
| 15:16.07 | *** join/#asterisk wasanzy (~wasanzy@41-66-254-58-dedicated.4u.com.gh) |
| 15:16.46 | *** join/#asterisk andresmujica (~andresmmu@ubuntu/member/andresmujica) |
| 15:17.19 | *** join/#asterisk Milenco (~Milenco@home.milenco.net) |
| 15:18.24 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 15:20.05 | *** join/#asterisk mjordan (mjordan@nat/digium/x-rzuzfxjpewxjdkvc) |
| 15:20.05 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 15:21.59 | newtonr | Kunsi, I have no idea what is going on there, but did you try the install_prereq script to get all the prerequisites installed? |
| 15:23.25 | Katty | guise. |
| 15:23.29 | Katty | i has query. |
| 15:23.38 | Katty | so if something was monitor your linux box for process asterisk. |
| 15:23.44 | Katty | at say, a client |
| 15:23.53 | Katty | and for whatever reason, it stopped running. how quickly would you want to know? |
| 15:29.38 | Kunsi | if it's a production system, as fast as i can get it, if "something" is sure there's a problem |
| 15:31.12 | *** join/#asterisk rmudgett (rmudgett@nat/digium/x-wqmplzqjbldxcovt) |
| 15:41.50 | *** join/#asterisk sekil (~sekil@business-89-135-166-219.business.broadband.hu) |
| 15:43.03 | *** part/#asterisk bulkorok (~Benjamin@85.183.61.47) |
| 15:49.42 | *** join/#asterisk Micc (~Mic@208.52.162.35) |
| 15:50.00 | michael-i | victory! (*0.5) inbound calls work, still working on outbound |
| 15:50.03 | Kobaz | was there anything special to enable SIP_CAUSE on 1.8 |
| 15:50.22 | Kobaz | i totally forgot about how that works |
| 15:55.28 | Kobaz | there it is |
| 15:55.30 | Kobaz | storesipcause |
| 15:56.00 | *** join/#asterisk jiuweigui (~jiuweigui@unaffiliated/jiuweigui) |
| 16:02.30 | clinet2 | I have a freepbx 12 w/ asterisk 11 and digium PRI card. Using the fax pro module for configuration. The pri is running super clean and the volume levels seem good. I can send and recieve faxing the UCP at full speed with up to 50 pages to all of my test sites. However when I send to about 40% of the destination numbers we have tried, I can't get even 1 page to go. I get the error, no response after sending a page. I have trie |
| 16:06.53 | *** join/#asterisk andresmujica (~andresmmu@ubuntu/member/andresmujica) |
| 16:09.20 | *** join/#asterisk aness (~aness@cm-84.215.80.229.getinternet.no) |
| 16:10.03 | Kobaz | ~freepbx |
| 16:10.03 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 16:16.03 | clinet2 | I'm in there too. This certainly feels like a spandsp related issue. |
| 16:16.49 | clinet2 | At this point I'm looking for any feedback on testing and debugging this issue in asterisk, and in freepbx. |
| 16:19.43 | *** join/#asterisk jmetro (490924dd@gateway/web/freenode/ip.73.9.36.221) |
| 16:19.45 | Kobaz | not too many low-level people here |
| 16:19.54 | jmetro | Anyone have a good replacement for the Aastra 6757i? |
| 16:19.58 | Kobaz | spandsp would be a good mailing list question |
| 16:20.27 | *** join/#asterisk sgriepentrog (sgriepentr@nat/digium/x-ngbhikbpnoxgtmix) |
| 16:20.41 | Kobaz | Digium D70 |
| 16:21.58 | jmetro | are those easily provisionable? |
| 16:22.05 | Kobaz | yeap |
| 16:22.17 | *** join/#asterisk sparetire_ (~sparetire@unaffiliated/sparetire) |
| 16:22.17 | Kobaz | or you can do a polycom 550 |
| 16:22.25 | Kobaz | but from what you pasted, it sounds like you want a screen with lots of buttos |
| 16:22.28 | Kobaz | buttons |
| 16:22.50 | jmetro | I just liked the Aastras huge screen, easy provision, great sound =/ |
| 16:23.17 | jmetro | going to miss all that room for text. |
| 16:23.19 | *** join/#asterisk jasonwert (~jasonwert@75-134-81-98.static.aldl.mi.charter.com) |
| 16:24.01 | Kobaz | i don't like aastra buttons |
| 16:24.08 | Kobaz | feels like a tv remote |
| 16:24.34 | jmetro | True, but very simple provisioning and formatting the screen |
| 16:24.49 | Kobaz | hah yeah, the cfg is like 5 lines |
| 16:24.53 | Kobaz | versus polycom |
| 16:25.01 | jmetro | polycoms i dont want to touch provisioning that |
| 16:25.08 | Kobaz | but once you get the hang of polycom, you'll be happy you have like 500 things to configure, when someone asks for something custom |
| 16:25.24 | Kobaz | i've never been disappointed |
| 16:26.12 | jmetro | E4 suggested some color screen Aastra as a replacement and my experience with the color screen phones is "dont do it" |
| 16:26.51 | Kobaz | the polycom 650 is nice |
| 16:27.02 | Kobaz | stay away from the vvx |
| 16:27.10 | Kobaz | they look pretty, but they are crazy slow |
| 16:27.17 | Kobaz | i would never deploy them in a real office |
| 16:27.20 | coppice | do you have a mix of audio and T.38 FAXes? |
| 16:27.21 | Kobaz | people would get pissed at me |
| 16:27.38 | jmetro | Kobaz : that was my exact experience XD |
| 16:27.52 | jmetro | Beautiful display, can type about 1 digit per minute. |
| 16:27.55 | Kobaz | haha |
| 16:27.56 | Kobaz | yeah |
| 16:28.46 | *** join/#asterisk gusto (~gusto@2a02:810d:8640:248:8219:34ff:fecf:17f0) |
| 16:29.09 | coppice | clinet2: do you have a mix of audio and T.38 FAXes? |
| 16:29.45 | jmetro | maybe i can get a few 6757i's that havent been sold yet.. |
| 16:32.12 | *** join/#asterisk slav3_kitten (~frankthet@unaffiliated/slav3-kitten/x-0866809) |
| 16:38.05 | clinet2 | No we just have the PRI faxing. All in and out are running through our PRI. |
| 16:39.18 | coppice | strange results over PRI is usually cause by the PRI being incorrectly sync'ed |
| 16:45.40 | *** join/#asterisk w9sh (~sph@50-79-224-193-static.hfc.comcastbusiness.net) |
| 16:46.43 | *** part/#asterisk w9sh (~sph@50-79-224-193-static.hfc.comcastbusiness.net) |
| 16:47.03 | jmetro | Does Digium offer any phones with a wifi connector rather than eth? |
| 16:48.46 | mjordan | nope |
| 16:52.41 | *** join/#asterisk gusto (~gusto@2a02:810d:8640:248:8219:34ff:fecf:17f0) |
| 16:52.53 | jmetro | mjordan danke |
| 16:54.11 | WIMPy | Can someone explain what "Exceptionally long voice queue length" really means? |
| 16:54.49 | mjordan | WIMPy: every channel has a queue of frames that are supposed to be serviced by some thread. That message occurs when the queue has gotten very long - which typically implies that nothing is reading frames off that channel |
| 16:55.00 | *** join/#asterisk gusto (~gusto@2a02:810d:8640:248:8219:34ff:fecf:17f0) |
| 16:55.28 | WIMPy | I sometimes see it on IAX channels. |
| 16:55.47 | WIMPy | And it's always the last gasp before locking up completely. |
| 16:55.48 | mjordan | WIMPy: that would imply that something (the IAX driver) is pushing frames onto the channel, but nothing is trying to read the frames off |
| 16:56.18 | mjordan | WIMPy: The bad thing about that message is that it is generic - it can happen to any channel that gets itself into a bad state. The trick is finding the cause. |
| 16:56.59 | WIMPy | There was a time when I could reproduce it by transferring a call. |
| 16:57.40 | WIMPy | And I'm positive it must have something to do with timestamps. |
| 16:58.16 | WIMPy | Does it happen at some stage where unexpected timestamps could cause a queue to build up? |
| 16:58.53 | mjordan | I'm not sure how that would occur, but it could be a weird IAX thing |
| 16:59.15 | *** join/#asterisk kj22594 (32f52289@gateway/web/freenode/ip.50.245.34.137) |
| 16:59.25 | WIMPy | Any idea how to dog deeper if I encounter such a thing again? |
| 16:59.30 | WIMPy | dig |
| 16:59.38 | mjordan | get the iax debug enabled, and a debug log |
| 16:59.47 | mjordan | that's usually about the only thing you can do |
| 16:59.54 | mjordan | as the rest is trying to analyze why a thread stopped servicing it |
| 17:00.45 | WIMPy | Bad unless you have a way to reproduce. But the one I knew doesn't seem to exist any more. |
| 17:02.13 | kj22594 | Hey everyone, wondering if anyone has used WebRTC in unison with Asterisk, and if so do you have any suggestions as to how to implement it? |
| 17:02.15 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 17:02.25 | WIMPy | It must be some weird evil bug anyway. It started with 11.7.1, but there was no change from 11.7.0 that could be related in any imaginable way. |
| 17:02.41 | *** join/#asterisk gusto (~gusto@2a02:810d:8640:248:8219:34ff:fecf:17f0) |
| 17:11.40 | *** join/#asterisk bmurt (~brendan@8.39.115.8) |
| 17:12.15 | bmurt | hi guys & gals... are older versions of asterisk (1.4) bugs available to be searched/browsed online? |
| 17:12.34 | bmurt | jira goes to 1.8, but we have an older instance that im trying to research |
| 17:13.08 | wasanzy | this will accept 3 digits terminated by #? Read(chanel,,3) |
| 17:13.35 | *** join/#asterisk BakaKuna (~BakaKuna@82-169-251-128.ip.telfort.nl) |
| 17:14.21 | WIMPy | 3 digits OR teminated by #. |
| 17:18.04 | *** join/#asterisk jhlavacek (~jirka@84.19.95.180) |
| 17:18.04 | mjordan | bmurt: issues.asterisk.org contains the old bug reports from Mantis |
| 17:18.12 | mjordan | bmurt: so any issue from the 1.4-ish era is there as well |
| 17:18.25 | mjordan | bmurt: now, anything new reported against unsupported versions is simply closed, as... well. Unsupported. |
| 17:18.30 | bmurt | ahh, i guess i just can't filter off of 'affected version' |
| 17:19.02 | mjordan | a lot of times in the olden, dark days, people didn't fill in the version field |
| 17:19.07 | mjordan | shameful |
| 17:19.44 | bmurt | lol |
| 17:19.48 | bmurt | burn them at the stake |
| 17:20.35 | *** join/#asterisk retentiveboy (~retentive@2001:558:6011:78:157:3142:bcbc:6bc9) |
| 17:22.30 | bmurt | ty mjordan |
| 17:40.16 | wasanzy | WIMPy:3 digits but I want # should be the terminator |
| 17:40.45 | [TK]D-Fender | wasanzy: Then you'll have to do it yourself digit by digit |
| 17:41.21 | WIMPy | Then you you want to have to enter #, you have to allow more than 3 digits. |
| 17:44.25 | wasanzy | I also want to validate the input, so I want something like Gotoif($[ "${LEN(${digit})}" == "3"]?success:nosuccess) is that correct? |
| 17:45.03 | wasanzy | [TK]D-Fender and WIMPy, am a bit lost |
| 17:45.35 | wasanzy | how do I do it digit by digit? |
| 17:45.40 | [TK]D-Fender | wasanzy: What is there to be lost over? |
| 17:45.54 | [TK]D-Fender | wasanzy: in the DIALPLAN. |
| 17:46.22 | [TK]D-Fender | wasanzy: Read 1 digit. Are you done? Read MORE digits. Got a "#"? Is it at the end of the 3rd digit? deal with it either way |
| 17:46.28 | WIMPy | you need to start. |
| 17:46.44 | wasanzy | is this example here what you are mean I should do? http://www.voip-info.org/wiki/view/Asterisk+cmd+Read |
| 17:47.23 | WIMPy | Yes. |
| 17:47.48 | WIMPy | We have already been that far yesterday. |
| 17:47.59 | WIMPy | Start using it. |
| 17:48.13 | wasanzy | WIMPy: ok |
| 17:48.17 | *** join/#asterisk jmetro (490924dd@gateway/web/freenode/ip.73.9.36.221) |
| 17:51.01 | *** join/#asterisk gusto (~gusto@2a02:810d:8640:248:8219:34ff:fecf:17f0) |
| 17:51.28 | wasanzy | is this line necessary? exten => s,1,Set(wait=2) can I change it to exten => 1,1,Set(wait=2) ? |
| 17:52.05 | [TK]D-Fender | wasanzy: Do you understand extensions at all? |
| 17:52.09 | WIMPy | No. |
| 17:52.21 | WIMPy | Maybe you should start with the dialplan basics in the |
| 17:52.23 | [TK]D-Fender | wasanzy: It's "s" because it's "s". You can do whatever you want so long as you calls land on the right place |
| 17:52.26 | WIMPy | ~book |
| 17:52.26 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 17:52.29 | [TK]D-Fender | ^^ |
| 17:53.42 | wasanzy | ok |
| 17:54.33 | wasanzy | I know dialpan. I was confusing myself |
| 17:56.52 | [TK]D-Fender | nOTHING TO GET CONFUSED OVER |
| 17:57.27 | [TK]D-Fender | Read the digits one at a time yourself to get the effect you want |
| 17:57.49 | wasanzy | ok |
| 17:58.04 | WIMPy | Well, i guess, I'd get confused as well, when looking at that detail for a whole day. |
| 17:59.37 | bsdice | wasanzy Debugging a dialplan is twice as hard as writing one. That is, if you are writing your dialplan as good as you can, you are, by definition, not smart enough to debug it. |
| 18:00.01 | bsdice | wasanzy So don't be nutty about perfection, leave some flaws and bugs in there. ;-) |
| 18:00.08 | jmetro | Debugging dialplan is easy UNLESS youre trying to look into the scattered mess that is freepbx |
| 18:00.45 | wasanzy | ok |
| 18:00.48 | WIMPy | No need to debug. Just do it right from the start :-) |
| 18:00.51 | jmetro | ^ |
| 18:02.03 | wasanzy | sure |
| 18:03.08 | bsdice | to fully appreciate Asterisk, I recommend starting with 0 config files. That's what I did, then see what it wants and continue writing every config file, by hand, one by one. :-) |
| 18:03.40 | bsdice | freepbx appears to lead people the other way round into its configuration pile |
| 18:04.46 | [TK]D-Fender | FreePBX isn't really "bad". FreePBX USERS usually don't bother really learning * regardless, or just enough to poitn at every single line of debug and ask "Is that line the problem?!" |
| 18:07.57 | *** join/#asterisk benasse (~benasse@nowhere.cicogna.fr) |
| 18:09.08 | *** join/#asterisk riess82 (~riessma@188.22.32.192) |
| 18:12.38 | *** join/#asterisk andresmujica (~andresmmu@ubuntu/member/andresmujica) |
| 18:13.57 | *** join/#asterisk andresmujica (~andresmmu@ubuntu/member/andresmujica) |
| 18:15.35 | drmessano | [TK]D-Fender, is that like resolving kernel issues by removing the kernel? |
| 18:16.03 | drmessano | FreePBX has its own distro, so why do I need Linux on here? |
| 18:16.22 | [TK]D-Fender | drmessano: No. Removing the kernel takes actual effort. FreePBX users would have none of it! ;) |
| 18:16.33 | drmessano | Excellent point |
| 18:16.51 | drmessano | Just make sure you update too often and dont test |
| 18:17.38 | drmessano | "I upgraded my box to 1.2.3 RC1, and everything was fine, so I waited a week and rolled out 1.2.3.1 to all my customers" |
| 18:17.49 | drmessano | depr |
| 18:17.54 | [TK]D-Fender | No, not RC.... ALPHA |
| 18:17.55 | drmessano | derp |
| 18:18.21 | drmessano | The ALPHA worked FINE, so why would anything PAST that not work??? |
| 18:18.28 | drmessano | U PEOPL JUST DUNNO |
| 18:19.45 | WIMPy | Reminds me of famous error messages "Deleting the utility module is foolish". |
| 18:19.59 | drmessano | hahah |
| 18:20.09 | drmessano | Thats good stuff |
| 18:20.28 | WIMPy | Utilitymodule being the name of the RISC OS kernel. |
| 18:20.34 | *** join/#asterisk AlHafoudh (~AlHafoudh@echo.freevision.sk) |
| 18:20.45 | drmessano | Oh.. badass |
| 18:22.32 | drmessano | I dont know what Mozilla and Google are trying to prove, but who the hell came up with the whole versioning scheme for each |
| 18:22.47 | drmessano | "Chrome 39" is totally useless to me |
| 18:23.16 | WIMPy | Just use a date. |
| 18:24.20 | drmessano | Exactly |
| 18:25.14 | drmessano | Say what you want, but 14.10 and 12.04 not only identify the Ubuntu release, but give me an idea of age |
| 18:25.30 | drmessano | So why not go with something that easy |
| 18:27.39 | [TK]D-Fender | Because they've already reached 39. Putting out 15.01 would feel like it's an OLD version :) |
| 18:27.58 | drmessano | Chrome NG? |
| 18:28.03 | drmessano | ChromeR |
| 18:28.08 | file | I would argue every major update is NG |
| 18:28.15 | file | #troll |
| 18:28.33 | drmessano | hands file a bridge |
| 18:28.47 | file | is it an Asterisk bridge? can I connect channels together using it? |
| 18:29.26 | Qwell | file: yes, but it's through a non-optimizing local channel bridge. |
| 18:29.35 | file | Qwell, ouch |
| 18:29.54 | drmessano | app_troll sits under the bridge and interrupts channels randomly |
| 18:30.03 | *** join/#asterisk andresmujica (~andresmmu@ubuntu/member/andresmujica) |
| 18:32.51 | drmessano | Wouldn't surprise me if in Asterisk 69 there was an app_troll which attaches to a bridge, which means for the last 15 years Asterisk has been the build up to one gigantic punchline |
| 18:33.19 | *** join/#asterisk mjordan (~mjordan@75.76.55.191) |
| 18:33.19 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 18:34.17 | drmessano | Much like my theory that Windows 13 or so will be the 3D version of Microsoft Bob, which proves that coders never stop trying to make people use their code, even if we didnt like it the first time |
| 18:34.51 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
| 18:35.19 | *** join/#asterisk bmurt (~brendan@8.39.115.8) |
| 18:35.43 | drmessano | You'll turn on the machine, enter your "home" and Rover will be sitting there.. a little more grey and the years having caught up with him, but he will be waiting for you like you never left |
| 18:37.01 | kj22594 | ';l;l;;';l;p54359;' |
| 18:37.04 | *** join/#asterisk andresmujica (~andresmmu@ubuntu/member/andresmujica) |
| 18:42.16 | *** join/#asterisk kj22594 (32f52289@gateway/web/freenode/ip.50.245.34.137) |
| 18:44.23 | kj22594 | Has anyone used WebRTC in unison with Asterisk, if so any suggestions on how to implement this? |
| 18:50.14 | *** join/#asterisk cyford33 (~allen@76.122.73.37) |
| 18:52.17 | cyford33 | how do i fix presence issues : res_pjsip_pubsub.c:608 subscription_get_handler_from_rdata: No registered subscribe handler for event presence.winfo |
| 18:52.44 | file | that event type has no implementation, there is nothing you can do about that |
| 18:52.56 | wasanzy | I have written a simple dialplan to accept a max 4 digits and take the user to another context. This is not working, find dialplan and log here: http://pastebin.com/jK1AanWE |
| 18:53.55 | *** join/#asterisk robl^ (~robl@pdpc/supporter/active/robl) |
| 18:55.10 | [TK]D-Fender | wasanzy: -- User entered nothing. |
| 18:55.12 | cyford33 | <PROTECTED> |
| 18:55.24 | cyford33 | (( ) |
| 18:56.21 | [TK]D-Fender | and same => n,Gotoif(${chanel}?exist:noexist) <- this gotoIF makes no sense That is not a proper test |
| 18:57.45 | WIMPy | And a Goto to a Goto doesn't make too much sense, either. |
| 18:58.51 | cyford33 | same => n,GotoIf(${ISNULL(${SIPPEER(${EXTEN},codecs)})}?:100) |
| 18:59.21 | cyford33 | something like that? |
| 18:59.46 | [TK]D-Fender | cyford33: No. |
| 19:00.21 | [TK]D-Fender | cyford33: Your test is nothing like what he'd be testing for. Why would you think yours would serve as a basis? |
| 19:01.02 | cyford33 | it looks like if checkin if channel exsist then go new context |
| 19:01.07 | WIMPy | And we have no idea, what input would be valid. |
| 19:01.18 | [TK]D-Fender | cyford33: He isn't. |
| 19:01.37 | [TK]D-Fender | cyford33: No idea how you'd have come up with that thought. |
| 19:01.39 | wasanzy | I wanted to check and make sure the digit entered is 4 digit. |
| 19:02.13 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 19:02.15 | [TK]D-Fender | wasanzy: well just chossing the variable before the "?" is not a TEST |
| 19:02.57 | WIMPy | And certainly not for a length of 4. |
| 19:05.22 | wasanzy | Gotoif($[ "${LEN(${chanel})}" == "4"]?4:100) is this correct? |
| 19:07.33 | wasanzy | same => n,Read(chanel,,4) also doesn't seem to be getting the input |
| 19:07.36 | litn | hey guys, I have two network interfaces, with different public IPs. I would like to have some redundancy here where if one goes out then we still have phone connection, however, I must change my sip_nat to the other IP address whenever this occurs. |
| 19:07.43 | litn | how can I set this up so that it will work automatically? |
| 19:09.21 | drmessano | litn, script it |
| 19:09.31 | cyford33 | WIMPy exten => s,n,Read(get,"silence/1",,,,,${wait}) |
| 19:09.39 | litn | that's the only way? I was going to but I wanted to make sure tehre wasn't something already built into asterisk |
| 19:09.43 | litn | a setting I overlooked or something :) |
| 19:09.47 | drmessano | Nope |
| 19:09.52 | litn | so in this case, will I drop calls no matter what? |
| 19:09.58 | drmessano | Yep |
| 19:10.02 | litn | I was thinking about how it may be able to use both network connections |
| 19:10.29 | litn | the phones are registered through a third nic that I am not worried about, and the SIP provider will take connections from either public IP on sip registration |
| 19:10.47 | litn | however I can also set it to route by ip and have a failover or secdonary route instead of sip registration |
| 19:10.50 | WIMPy | cyford33@ Are you trying random examples now? |
| 19:11.18 | [TK]D-Fender | Looking like... |
| 19:11.42 | cyford33 | why u say that |
| 19:11.53 | cyford33 | u ask to capture 4 digits right |
| 19:12.13 | cyford33 | with asterisk 1.8+ that should work |
| 19:12.33 | cyford33 | http://www.voip-info.org/wiki/view/Asterisk+cmd+Read |
| 19:12.54 | WIMPy | I don't see where version matters. |
| 19:13.15 | cyford33 | it wouldnt work in 1.2 |
| 19:13.31 | cyford33 | <PROTECTED> |
| 19:13.37 | WIMPy | That's prehistoric. |
| 19:13.49 | cyford33 | wouldnt work on 1.6 ether |
| 19:13.51 | *** join/#asterisk Greek-Boy (~Greek-Boy@41.188.150.227) |
| 19:14.06 | [TK]D-Fender | [14:09]cyford33WIMPy exten => s,n,Read(get,"silence/1",,,,,${wait}) <- where does this limit to 4? Why are you passing QUOTES with a filename to play? Why are you changing the VARIABLE NAME he's reading into? Where did you come up with this ${wait} variable you're including at the end? |
| 19:14.40 | [TK]D-Fender | Pretty much every piece of that is wrong... |
| 19:14.45 | WIMPy | Random examples? |
| 19:15.15 | [TK]D-Fender | And didn't bother following at least the syntax of the code he's working with in using "same" |
| 19:15.16 | Greek-Boy | I'm trying to setup asterisk behind pfsense. I opened all necessary ports. (SIP 5060 and RTP 10000 to 20000). When an external user calls the pbx everything works fine (Echo test, etc) but when users try to call each others extensions no audio. |
| 19:15.26 | cyford33 | ,,,,, |
| 19:15.28 | drmessano | It was copy pasted from the Wiki article.. which makes no sense to me, because the only thing its helping the OP with is that we use Read() |
| 19:15.39 | cyford33 | captures 4 digits |
| 19:16.00 | [TK]D-Fender | [14:09]cyford33WIMPy exten => s,n,Read(get,"silence/1",,,,,${wait}) <- what part of THIS says "4 digits"? |
| 19:16.34 | cyford33 | WIMPy exten => s,n,Read(get,"silence/1",X,X,X,X,${wait}) |
| 19:16.38 | [TK]D-Fender | and still... quotes = BAD, and that "wait" variable... he doesn't HAVE one named that. |
| 19:16.45 | WIMPy | The imaginary one :-) |
| 19:16.58 | cyford33 | lol |
| 19:17.10 | cyford33 | it came from wiki i didnt make it |
| 19:17.12 | [TK]D-Fender | And you're reding into a DIFFERENT variable name than he was. |
| 19:17.25 | WIMPy | s/he/you/ ? |
| 19:17.39 | [TK]D-Fender | cyford33: You're proposing it as a sample. It's crap and no piece of that looks like anything he should be doing |
| 19:18.25 | cyford33 | i am asking him would that work |
| 19:18.26 | WIMPy | Maybe it's like that game where two people say random words until they say the same. |
| 19:18.30 | drmessano | same => Read(derp,,,,,,,,) would have been more accurate |
| 19:18.31 | wasanzy | I don't know why the Read is not getting the input digits |
| 19:18.34 | cyford33 | i didnt propose anything |
| 19:18.42 | WIMPy | We give random examples until two of them match. |
| 19:18.46 | [TK]D-Fender | cyford33: He doesn't have a clue, and no, that's horribly broken |
| 19:18.53 | cyford33 | if it dont work a sipple no is fine lol |
| 19:18.58 | cyford33 | simple |
| 19:19.29 | cyford33 | its on an wiki site, |
| 19:19.39 | WIMPy | wasanzy@ Well, if your DTMF is broken that's a completely different matter. |
| 19:19.43 | cyford33 | ok ill freaking make one and test it |
| 19:19.50 | cyford33 | or you can just correct it |
| 19:20.46 | drmessano | cyford33, you pasted one line from a complete example of a specific task from an outdated wiki site, and the snippet you posted does little more than illustrate Read() with the rest being useless and requiring discard. I think that's the problem |
| 19:21.17 | drmessano | Continue to apply blue to face while holding breath, but it wont become less wrong over time |
| 19:21.39 | wasanzy | http://pastebin.com/UFzDnR9Q I have modified the diaplan but still getting " -- User entered nothing." |
| 19:21.55 | [TK]D-Fender | wasanzy: Stop using BACKGROUND first |
| 19:22.16 | [TK]D-Fender | wasanzy: this is not something you can background. Read() has an option for a prompt to play first |
| 19:22.24 | *** join/#asterisk bmurt (~brendan@8.39.115.8) |
| 19:22.38 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-28-58.cablep.bezeqint.net) |
| 19:22.45 | [TK]D-Fender | wasanzy: Fix this then show us a new call. BTW... WAIT until the prompt is finished before entering anything |
| 19:22.55 | cyford33 | exten => _X.,n,WaitExten(4) |
| 19:23.00 | [TK]D-Fender | wasanzy: If you still have nothing... your DMTF mode is wrong |
| 19:23.08 | wasanzy | ok |
| 19:23.11 | [TK]D-Fender | [14:22]cyford33exten => _X.,n,WaitExten(4) <- NO. |
| 19:23.52 | [TK]D-Fender | cyford33: So much wrong with that I'd rather not start... |
| 19:24.00 | cyford33 | waits for 4 digit dtmf |
| 19:24.09 | WIMPy | no |
| 19:24.11 | drmessano | Waits for an extension to be entered; gives the caller the opportunity to push a new extension onto the stack |
| 19:24.25 | drmessano | From the famous outdated Wiki |
| 19:24.39 | drmessano | 4 digits of DTMF != new extension |
| 19:25.01 | WIMPy | And the Parameter is not a number of digits. |
| 19:25.19 | drmessano | That too |
| 19:25.24 | drmessano | WaitExten(seconds) |
| 19:25.26 | cyford33 | aww |
| 19:25.34 | drmessano | Youre just making shit up now |
| 19:25.43 | WIMPy | waits for someone to suggest DISA next. |
| 19:25.46 | drmessano | HAHAH |
| 19:25.52 | drmessano | Wait... |
| 19:25.56 | drmessano | YES, DISA |
| 19:26.07 | cyford33 | WaitExten(seconds) |
| 19:26.13 | drmessano | or it's big brother WaitGotoDISA |
| 19:26.28 | [TK]D-Fender | cyford33: taht assumes you already are ON an exten matching a 2+ digit extn (WTF?) and then waits 4 .. SECONDS for input. THat has NO implication on how many didigts to read |
| 19:26.44 | [TK]D-Fender | cyford33: And that depends on other EXTENSIONS existing in that context. Horribly wrong. |
| 19:26.49 | drmessano | WibblyWobbly() |
| 19:26.54 | cyford33 | i see now lol |
| 19:26.56 | *** join/#asterisk [44] (~slackolog@unaffiliated/quintux) |
| 19:27.12 | drmessano | WibblyWobbly(4|spacetime) |
| 19:27.16 | [TK]D-Fender | ? book |
| 19:27.18 | drmessano | That should work |
| 19:27.22 | [TK]D-Fender | ~book |
| 19:27.22 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 19:27.23 | WIMPy | Goto(${RAND}) |
| 19:27.23 | [TK]D-Fender | READ IT ^^^ |
| 19:27.34 | drmessano | WIMPy, LOL |
| 19:28.39 | cyford33 | who? |
| 19:28.56 | drmessano | Read(spacetime,tardis,${RAND},,,) |
| 19:29.01 | WIMPy | callroulette.com |
| 19:29.40 | cyford33 | <PROTECTED> |
| 19:29.40 | [TK]D-Fender | X ~MAYBE 10 <- ILLOGICAL operators |
| 19:29.49 | [TK]D-Fender | cyford33: Just stop.... |
| 19:29.49 | WIMPy | Yes. Let's make app_tardis. |
| 19:30.08 | [TK]D-Fender | cyford33: that has nothing to dow ith limiting the size of input, or validating it... |
| 19:30.11 | WIMPy | And seconds again. |
| 19:30.26 | cyford33 | can i see an example |
| 19:31.03 | cyford33 | WIMPy this is digit |
| 19:31.03 | drmessano | WIMPy, introduces WibblyWobbly() which does rand ..... things.... to the dialplan. Also SonicScrewdriver() which fixes everything in any prior lines of dialplan |
| 19:31.07 | [TK]D-Fender | cyford33: You don't understand any of the dialplan basics.... what's the point? |
| 19:31.18 | WIMPy | cyford33: no |
| 19:31.24 | cyford33 | why u say that |
| 19:31.29 | [TK]D-Fender | cyford33: If you don't understand the individual commands you're even looking at, what's the point of pasing a TON of them? |
| 19:31.34 | cyford33 | i understand alot after seing it |
| 19:31.48 | wasanzy | http://pastebin.com/ssDxA1mv still didn't work |
| 19:31.54 | drmessano | The book is full of words |
| 19:32.04 | wasanzy | I took out the background |
| 19:32.20 | cyford33 | well im not the one that really needs its it |
| 19:32.20 | [TK]D-Fender | [14:29]cyford33last guess lol ; exten => s,4,Set(TIMEOUT(digit)=4) <- Where does your guess show how it is that you "understand" this to being a way to limit # of digits entered "somewhere"? |
| 19:32.31 | drmessano | cyford33, actually, you do need it.. a lot |
| 19:32.35 | WIMPy | wasanzy@ Not in that PB. |
| 19:32.57 | cyford33 | <PROTECTED> |
| 19:32.57 | [TK]D-Fender | wasanzy: I just told you to get rid of that Background and do the playback in your Read() |
| 19:33.25 | cyford33 | i am asking out of criousity, WIMPy needs ity |
| 19:33.46 | [TK]D-Fender | cyford33: No, WIMPy is more than qualified to do this 5 different ways. |
| 19:33.48 | wasanzy | [TK]D-Fender: does the Read need playback to work? |
| 19:33.55 | [TK]D-Fender | wasanzy: No. |
| 19:34.01 | wasanzy | ok |
| 19:34.10 | [TK]D-Fender | wasanzy: Use Read() to play that prompt back, not Background() |
| 19:34.35 | wasanzy | ok but I took out the background, so nothing is play anymore |
| 19:34.40 | cyford33 | [TK]D-Fender why read a book on a function app i dont need |
| 19:34.46 | cyford33 | i finished my projects |
| 19:35.11 | cyford33 | i may not be on another asterisk projkject |
| 19:35.11 | WIMPy | Did that project involve some sort of spam? |
| 19:35.14 | [TK]D-Fender | cyford33: You're proposing things to others to "help" them and you don't seem to have a clue. |
| 19:35.16 | *** join/#asterisk slackology__ (~slackolog@unaffiliated/quintux) |
| 19:35.37 | cyford33 | <PROTECTED> |
| 19:35.50 | [TK]D-Fender | cyford33: Random copy/paste is not "help" |
| 19:35.54 | *** join/#asterisk MadHatter42 (~MadHatter@unaffiliated/madhatter42) |
| 19:36.01 | cyford33 | i may be wrong at times and u are too |
| 19:36.03 | [TK]D-Fender | cyford33: And you seemed to have "belief" that it was right |
| 19:36.11 | WIMPy | Giving wrong suggestions is usually not considered helping. |
| 19:36.27 | [TK]D-Fender | cyford33: You don't know what any of those commands even do from what you gave him. |
| 19:37.02 | cyford33 | yeah i didnt know they where wrong lol and im sorry for wasnt yall freakin time lol |
| 19:37.24 | drmessano | This has to be some kind of trolling.. because this level of serious is just dangerous |
| 19:37.37 | cyford33 | i remember i created and ivr ike 5 years ago , was trying to look it up for him |
| 19:38.04 | cyford33 | that dialplan i am sure i can fix in 30 mins though if needed |
| 19:42.27 | drmessano | I'm henry the 8th I am, henry the 8th I am, I am |
| 19:42.46 | wasanzy | same => n,Read(chanel,${GETHEARD_WLC},4) the audio was played, but still after entering 4 digits, it tells me "User entered nothing." |
| 19:43.37 | WIMPy | Then you need to fix your DTMF. |
| 19:44.10 | [TK]D-Fender | [14:19]WIMPywasanzy@ Well, if your DTMF is broken that's a completely different matter. |
| 19:44.16 | [TK]D-Fender | [14:22][TK]D-Fenderwasanzy: If you still have nothing... your DMTF mode is wrong |
| 19:45.24 | wasanzy | [TK]D-Fender: how can I deal with the DTMF |
| 19:45.33 | [TK]D-Fender | FIX THE MODE |
| 19:45.43 | [TK]D-Fender | It isn't matching whatever the other side is using |
| 19:45.59 | kj22594 | join #xlive5 |
| 19:46.11 | *** part/#asterisk kj22594 (32f52289@gateway/web/freenode/ip.50.245.34.137) |
| 19:47.32 | wasanzy | I have no idea about DTMF mode |
| 19:48.13 | WIMPy | Let's start at the beginning. You are using a SIP phone? |
| 19:48.38 | wasanzy | am using Hand Set |
| 19:48.43 | [TK]D-Fender | ON WHAT? |
| 19:48.52 | WIMPy | Connected to what? |
| 19:49.42 | wasanzy | Am using Sangoma card, which I insert a GSM SIM in it and attached it to the Asterisk server |
| 19:50.02 | [TK]D-Fender | Show us the call |
| 19:50.11 | wasanzy | ok |
| 19:50.11 | WIMPy | Ugh |
| 19:50.34 | [TK]D-Fender | http://pastebin.com/jK1AanWE |
| 19:50.41 | [TK]D-Fender | It's DAHDI |
| 19:51.04 | wasanzy | yea am using DAHDI |
| 19:51.11 | [TK]D-Fender | There is no "mode" |
| 19:51.29 | wasanzy | ok |
| 19:51.29 | [TK]D-Fender | I'm not sure how that card presents it. |
| 19:51.44 | *** join/#asterisk areski (~areski@80.174.128.104.dyn.user.ono.com) |
| 19:51.45 | [TK]D-Fender | wasanzy: Check your sangoma configs for hardware DTMF decode |
| 19:51.50 | WIMPy | Or if the card does. |
| 19:51.58 | [TK]D-Fender | wasanzy: Read their docs to find out wherer that is an make sure to enable it |
| 19:52.04 | WIMPy | Or its driver or dahdi... |
| 19:52.14 | wasanzy | ok |
| 19:52.42 | *** join/#asterisk linuxgeek (~linuxgeek@2a03:b0c0:2:d0::5f:2001) |
| 19:53.28 | WIMPy | But one thing is for sure: GSM is not the best precondition for DTMF. |
| 19:53.48 | Chainsaw | WIMPy: I bet it fares like the violins on your average hold music track. |
| 19:54.45 | WIMPy | Well, it's most probably not the GSM CODEC you know from the VOIP world. That's been dead for like 15 years and for a good reason. |
| 19:54.57 | WIMPy | But still... |
| 19:55.20 | [TK]D-Fender | It's have to be OOB |
| 19:55.41 | wasanzy | I think I should use SIP phone and see if that will work first |
| 19:55.52 | WIMPy | I don't think that exists. |
| 19:56.13 | WIMPy | But it was possible even with the original GSM FR CODEC. |
| 19:56.40 | WIMPy | My first GSM phone had built-in voicemail. That worked. |
| 19:56.48 | wasanzy | WIMPy: me? |
| 19:57.06 | WIMPy | no. |
| 19:57.11 | WIMPy | It's a good idea to use another phone. |
| 20:10.35 | *** join/#asterisk mjordan (~mjordan@75.76.55.191) |
| 20:10.35 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 20:10.59 | *** join/#asterisk andresmujica (~andresmmu@ubuntu/member/andresmujica) |
| 20:12.11 | drmessano | Is users.conf still a real thing? |
| 20:12.27 | [TK]D-Fender | it technically exists. |
| 20:12.34 | [TK]D-Fender | And the stupid thing still gets parsed. |
| 20:12.38 | [TK]D-Fender | Let it DIE man.... |
| 20:12.45 | wasanzy | Hmm, using a sip phone, Read is able to read the digits entered now |
| 20:12.59 | drmessano | I was actually thing it would be perfect for a project |
| 20:13.03 | drmessano | thinking |
| 20:13.21 | drmessano | I kinda need its stupidity |
| 20:13.22 | wasanzy | so I think the problem is from DTMF as you said earlier |
| 20:17.14 | drmessano | I want to build a basic switch.. Add/del users, everyone can call everyone else. No PSTN. |
| 20:17.26 | drmessano | Simple and easy dialplan |
| 20:18.30 | drmessano | Looking at users.conf though, I can't decide if it's less work or more work than using sip.conf and extensions.conf and templates |
| 20:20.33 | drmessano | basic GUI for extension and secret |
| 20:20.59 | drmessano | Might be a fun weekend project |
| 20:23.39 | *** join/#asterisk genpaku (~genpaku@107.191.100.185) |
| 20:29.20 | [TK]D-Fender | drmessano: remember the key is the extra crap itt creates including dialplan.... you'd certainly not want to get stuck in that paradigm |
| 20:37.10 | *** join/#asterisk mjordan (mjordan@nat/digium/x-rnnpogdsocxxnfsd) |
| 20:37.10 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 20:47.14 | drmessano | yes true |
| 20:47.57 | drmessano | I guess I was thinking from the config standpoint, but from the actual created diaplan standpoint, yeah that's a mess |
| 20:53.14 | [TK]D-Fender | that;s what it does... using single parms to set up things like voicemail, etc. |
| 20:53.38 | [TK]D-Fender | So much easier to start from scratch the FreePBX way of generating complete config segments you can "include" in |
| 20:53.49 | [TK]D-Fender | Rather that try a "parsed" approach |
| 20:53.53 | [TK]D-Fender | than |
| 20:54.00 | *** join/#asterisk CeBe1 (~CeBe@port-92-200-15-215.dynamic.qsc.de) |
| 20:54.15 | *** join/#asterisk mjordan (mjordan@nat/digium/x-fzzmpecehmxhmuob) |
| 20:54.16 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 21:00.00 | cyford33 | how can i find out the status on making it possible to get the pjsips registry in the dialplan ? |
| 21:05.06 | rrittgarn | having an issue where callers get dropped mid conversation. Looking at debugs i see called (obj->txf = (nil)) right before the hangup (hangup events at: http://pastebin.com/uTMsz3ng) Any thoughts as to if this is a bug, or a misconfiguration / resource issue? Or how i can continue troubleshooting? I dont even know what caused it... |
| 21:17.21 | [TK]D-Fender | cyford33: Has a ticket actually been opened to add that functionality? |
| 21:17.44 | *** join/#asterisk Nugget (nugget@rennsport.macnugget.org) |
| 21:19.20 | wasanzy | http://wiki.sangoma.com/troubleshooter-boards-pri-dtmf |
| 21:19.26 | wasanzy | am seeing something there |
| 21:21.26 | cyford33 | [TK]D-Fender a cupple weeks ago when i was looking for a way, i was told it didnt exist but should be available in a cupple weeks.. i am guessing there was a tkt on it |
| 21:21.30 | *** join/#asterisk willo (~quassel@fedora/willo) |
| 21:21.41 | [TK]D-Fender | Then go look for it |
| 21:21.53 | cyford33 | thats what i am doing |
| 21:22.21 | cyford33 | i cant find it |
| 21:22.38 | cyford33 | but maybe im lookin wrong, where should i be lookin |
| 21:26.24 | *** join/#asterisk Nugget (nugget@rennsport.macnugget.org) |
| 21:26.31 | [TK]D-Fender | Where ARE you looking? |
| 21:28.29 | cyford33 | google and here https://issues.asterisk.org/jira/browse/ASTERISK-23173?jql=text%20~%20%22pjsip%22 |
| 21:31.44 | mjordan | well, that isn't talking about the registry |
| 21:31.45 | [TK]D-Fender | https://issues.asterisk.org/jira/browse/ASTERISK-24654 |
| 21:31.50 | [TK]D-Fender | That loks like your submission |
| 21:32.16 | [TK]D-Fender | mjordan: partly |
| 21:32.28 | [TK]D-Fender | this is the other part of what's he's looking to do |
| 21:32.37 | [TK]D-Fender | first is the fact it fails outright |
| 21:33.16 | [TK]D-Fender | Then someone (you?), or file, etc seemed like they were going to take up the issue of not being able to query the reg string from the dialplan like you can for chan_sip via ASTDB |
| 21:33.29 | [TK]D-Fender | Right now he's loking for status on that part |
| 21:33.34 | [TK]D-Fender | I don't see the ticket yet... |
| 21:33.49 | [TK]D-Fender | Mind you his issue PREVENTING him from registering makes that a moot point. |
| 21:33.55 | [TK]D-Fender | And on that note... checkout time... |
| 21:33.57 | [TK]D-Fender | BBL |
| 21:34.23 | mjordan | well, I can go comment on that issue again, but that log file doesn't appear to show an error |
| 21:35.03 | *** join/#asterisk retentiveboy (~retentive@2001:558:6011:78:157:3142:bcbc:6bc9) |
| 21:37.55 | mjordan | and I don't think it's a bug anyway |
| 21:39.47 | cyford33 | no error anymore |
| 21:40.15 | cyford33 | i can register, it just doesnt send the extra header to aor |
| 21:40.31 | cyford33 | it strips some of the uri |
| 21:41.02 | cyford33 | so when there is a way to quiry it , ill only get a little bit of it |
| 21:42.14 | cyford33 | cuts it off here X-PUSH-URI= |
| 21:43.50 | *** join/#asterisk PaybackTony (~Adium@2601:7:700:7fa:9864:1ed:93ea:5db4) |
| 21:48.09 | PaybackTony | Hey guys, having a little issue perhaps someone can point me in the right direction. Got 11.15 compiled and trying to get WebRTC working (SIP.js). ICE was compiled with it. The issue I'm having is no audio on WebRTC calls. The odd part is that when turning on RTP debug, asterisk is attempting to send RTP (via ICE) packets to the same IP that SIP.js says it was using (ICE / Stun). SIP messages flow fine (hangup, etc), but the RTP pack |
| 21:49.03 | PaybackTony | For non-webrtc calls everything works fine. |
| 21:52.47 | *** join/#asterisk MadHatter42 (~MadHatter@unaffiliated/madhatter42) |
| 21:52.52 | *** join/#asterisk yokel (~yokel@unaffiliated/contempt) |
| 21:57.30 | cyford33 | WOW yall closed it |
| 21:58.38 | drmessano | Why wouldnt they? |
| 21:59.46 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 22:03.11 | cyford33 | becuase it is not working as it suppose to |
| 22:03.19 | drmessano | Its not a bug |
| 22:03.33 | drmessano | As mjordan clearly stated, it's something not implemented |
| 22:03.38 | drmessano | A feature request, at best |
| 22:03.43 | cyford33 | no |
| 22:03.49 | cyford33 | it doesnt cut of the head |
| 22:04.04 | cyford33 | it seems more like a character limitation |
| 22:04.13 | cyford33 | and it works with sip_chan |
| 22:04.39 | cyford33 | not allowing this will not let push notifications work with pjsip |
| 22:05.19 | cyford33 | and pjsip is the new defualt |
| 22:10.51 | [TK]D-Fender | No, it isn't |
| 22:11.01 | [TK]D-Fender | Do not call it "default". That is not at all the case |
| 22:11.06 | [TK]D-Fender | It is still new & growing |
| 22:11.07 | *** join/#asterisk yokel (~yokel@unaffiliated/contempt) |
| 22:12.14 | cyford33 | ohh on freepbx it is sorry |
| 22:12.47 | cyford33 | if u do defualt freepbx 5060 is pjsip |
| 22:13.07 | cyford33 | hmm |
| 22:13.25 | [TK]D-Fender | Means nothing here |
| 22:13.47 | cyford33 | before i upgraded pjsip to 2.3 i couldnt even register, |
| 22:14.04 | cyford33 | now i can register and case closed |
| 22:17.59 | rrittgarn | does [2015-01-15 15:19:18] DEBUG[22720] res_odbc.c: odbc_release_obj2(0x1e762e8) called (obj->txf = (nil)) imply there is an issue that needs further investigation, or is it just a red herring in my troubleshooting process? |
| 22:19.05 | file | it's a debug message, it doesn't mean there's a problem |
| 22:19.49 | rrittgarn | I should clarify that i am trying to track down a problem. Specifically right after that message a call unexpectedly ends |
| 22:20.19 | rrittgarn | the part specifically that makes me think it is something to chase is that it is referencing an object that is (Nil) |
| 22:21.47 | file | that's a transaction, it means whatever ODBC thing occurred did not involve a transaction |
| 22:25.54 | rrittgarn | ok that makes sense |
| 22:28.26 | wasanzy | hmm am still struggling with this DTMF thing |
| 22:28.46 | *** join/#asterisk spditner (~simon@206-248-160-115.dsl.teksavvy.com) |
| 22:29.22 | *** join/#asterisk yokel (~yokel@unaffiliated/contempt) |
| 22:30.10 | rrittgarn | what does audiohook.c do? |
| 22:32.29 | rrittgarn | scenario is as follows: Digium TE133 PRI card to Sip endpoints. Getting reports of a lot of dropped calls. Went through debugs for a reported dropped call, found that in the SIP capture, the PBX initiates the hangup. The PRI doesn't. The phone doesn't either. Trying to find out why it ended the call |
| 22:33.45 | WIMPy | So Asterisk terminates both calls? |
| 22:34.53 | *** join/#asterisk CeBe (~CeBe@port-92-200-121-66.dynamic.qsc.de) |
| 22:35.02 | rrittgarn | yes |
| 22:35.11 | cyford33 | plus yall the ones advised me to open a my first bug on this |
| 22:35.18 | *** join/#asterisk CeBe1 (~CeBe@port-92-200-121-66.dynamic.qsc.de) |
| 22:35.41 | rrittgarn | i am using mixmon to record the call and do hear the conversation just drop out in the middle |
| 23:22.34 | *** join/#asterisk infobot (ibot@rikers.org) |
| 23:22.34 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.1.0 (2014/12/15), 11.15.0 (2014/12/15), 1.8.32.1 (2014/11/20); Standard: 12.8.0 (2014/12/15); DAHDI: DAHDI-linux 2.10.0 (2014/08/13), DAHDI-tools 2.10.0 (2014/08/13); libpri 1.4.15 (2014/06/16) -=- Asterisk wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
| 23:25.15 | WIMPy | What do you guys use to detect Asterisk being locked-up? |
| 23:26.09 | pawiecki | Well, all I need for now is to make sure, that after caller hear all the introduction and call would proceed to another person - the recording is set and ready to catch the call. |
| 23:26.21 | pawiecki | locked up? What do you mean? |
| 23:26.35 | WIMPy | Go for MixMonitor then. |
| 23:26.39 | *** join/#asterisk MissionCritical (~MissionCr@unaffiliated/missioncritical) |
| 23:27.00 | WIMPy | If Asterisk just sits there as if everything is ok, but it won't process calls any more. |
| 23:28.06 | pawiecki | WIMPy: it's easy - clients will call with lot's of warm and friendly words to tell you ;) |
| 23:28.24 | *** join/#asterisk foulou (~foulou@tests.lagoon.nc) |
| 23:28.30 | WIMPy | Yes. I'd like to optimize that a little. |
| 23:28.37 | *** join/#asterisk moy (sid47040@gateway/web/irccloud.com/x-xozifzvhdabbmdxw) |
| 23:28.50 | *** join/#asterisk D-Boy (~D-Boy@unaffiliated/cain) |
| 23:28.53 | *** join/#asterisk newtonr (newtonr@nat/digium/x-fudyslffphuknzxr) |
| 23:28.53 | *** mode/#asterisk [+o newtonr] by barjavel.freenode.net |
| 23:29.05 | *** join/#asterisk GameGamer43 (sid5533@gateway/web/irccloud.com/x-czmjynahxgchktwt) |
| 23:29.34 | *** join/#asterisk sgriepentrog (sgriepentr@nat/digium/x-uiczggpbhwwlypth) |
| 23:30.36 | pawiecki | test accounts, and some kind of custom script to run every few minutes or so would be good |
| 23:30.41 | *** join/#asterisk jrose_atDigium (jrose_atDi@nat/digium/x-slrbpufnfbsibhwh) |
| 23:30.57 | *** join/#asterisk mjordan (mjordan@nat/digium/x-fquetveypjuxoedx) |
| 23:31.15 | *** join/#asterisk tm1000 (sid6728@gateway/web/irccloud.com/x-vubmkpvpnloucdco) |
| 23:32.12 | pawiecki | for example if test account foo can't contact bar - send warning, sms. Or even better - send some kind of a messages when all is ok, but whene there's no messages for 15 minutes - raise alarms |
| 23:32.23 | *** join/#asterisk spicyramen_ (~Adium@2601:9:83c0:10:3d01:a4e2:4a16:69cc) |
| 23:33.06 | pawiecki | i'm not sure how to implement it practically but the basic idea is there |
| 23:38.31 | WIMPy | The touble is that it's likely not dead enough to not reply at all. |
| 23:38.49 | *** join/#asterisk bmurt (~brendan@64-121-3-32.c3-0.upd-ubr2.trpr-upd.pa.cable.rcn.com) |
| 23:47.21 | *** part/#asterisk mjordan (mjordan@nat/digium/x-fquetveypjuxoedx) |
| 23:50.00 | *** join/#asterisk hfor (~hforjehik@gateway/tor-sasl/hforjehikiasq6n2) |
| 23:52.11 | *** join/#asterisk CeBe1 (~CeBe@port-92-200-26-195.dynamic.qsc.de) |
| 23:53.25 | *** join/#asterisk yokel (~yokel@unaffiliated/contempt) |
| 23:57.40 | *** join/#asterisk KNERD (~KNERD@64.31.22.134) |