| 00:00.00 | lnb_ | in my head, extensions here connect to local asterisk, local asterisk sends the calls to remote freepbx and remote freepbx sends the calls out based on the NPANXXXXX |
| 00:00.13 | goodtime | i guess treated would mean that iit is going to have some kinda of action taken where seen would just be looked at or just noticed |
| 00:00.32 | goodtime | or not |
| 00:00.37 | goodtime | lol |
| 00:01.12 | lnb_ | the local asterisk will have routes configured, based on the route, it chooses a trunk to send the calls |
| 00:01.27 | lnb_ | at least thats how its done with remote freepbx server |
| 00:01.37 | goodtime | is that correct [TK]D-Fender |
| 00:01.39 | lnb_ | i am just adding antoher asterisk |
| 00:01.44 | lnb_ | in the soup mix |
| 00:02.39 | [TK]D-Fender | goodtime, sort of ... partially |
| 00:03.03 | goodtime | ok |
| 00:03.25 | goodtime | what did you mean by that? |
| 00:03.31 | goodtime | then.. |
| 00:03.36 | [TK]D-Fender | goodtime, "seen as" denotes implied behaviour. that would be as though the term had specific resulat automatically associated to it. |
| 00:04.07 | goodtime | okay |
| 00:04.21 | goodtime | understood |
| 00:04.24 | lnb_ | treated as in this case makes more sense |
| 00:04.28 | [TK]D-Fender | goodtime, "treated as" implies that YOU will process it in a way consistent with a specific objective, possibly identically to the way you'd process other simiar calls. |
| 00:04.39 | lnb_ | since something will be done 'treated' |
| 00:04.59 | lnb_ | seen is i saw, i looked |
| 00:05.13 | lnb_ | she's ugly, i left |
| 00:05.18 | goodtime | ha |
| 00:05.23 | lnb_ | no |
| 00:05.25 | [TK]D-Fender | goodtime, Asterisk has no such "default behavior". Everything is dialplan, and that is entirely your job. Any given call from any given device could be treated entirely different from any other. |
| 00:05.26 | goodtime | understood |
| 00:05.30 | lnb_ | that wouuld be treated as too |
| 00:05.34 | lnb_ | -> i left |
| 00:05.37 | lnb_ | i did something |
| 00:05.46 | WIMPy | See how you treated look |
| 00:06.20 | goodtime | so it is like a verb |
| 00:06.33 | lnb_ | i think by looking at sip_additional.conf there would be a trunk between the local and remote servers |
| 00:06.45 | [TK]D-Fender | lnb_, No. |
| 00:06.54 | lnb_ | y? |
| 00:07.30 | [TK]D-Fender | lnb_, a device entry. |
| 00:07.42 | [TK]D-Fender | lnb_, SIP is SIP. |
| 00:08.21 | [TK]D-Fender | lnb_, Ther is no technical difference in one call VS another in that sense. The concept of "trunk" is LOGICAL ONLY and exists only in that capacity |
| 00:08.28 | lnb_ | [TK]D-Fender: i have say just 1 SIP phone here, it will have its config pointing to the local asterisk box.... |
| 00:09.10 | [TK]D-Fender | lnb_, Then you'll want a device config to match it and tell it how to communicate with it, and where to start processing calls from it in. |
| 00:09.11 | lnb_ | that local asteriskbox will have extensions_additonal.conf with somethin like... |
| 00:09.23 | *** join/#asterisk robl^ (~robl@pdpc/supporter/active/robl) |
| 00:09.43 | lnb_ | exten => _1NXXNXXXXXX,n,Macro(dialout-trunk,2,${EXTEN},,off) |
| 00:09.46 | [TK]D-Fender | lnb_, Next.. you're in raw Asterisk... forget "sip_additional.conf" and "extensions_additional.conf" |
| 00:09.54 | lnb_ | ok |
| 00:10.06 | lnb_ | it all goes in sip.conf ? |
| 00:10.11 | lnb_ | or extensions.conf |
| 00:10.12 | [TK]D-Fender | that's what * knows |
| 00:10.36 | [TK]D-Fender | "sip_additional.conf" and "extensions_additional.conf" <- these are not files * magically looks for. |
| 00:10.56 | [TK]D-Fender | It is TOLD to by the main in FreePBX. |
| 00:10.59 | lnb_ | no those files are includes from perhaps sip.conf |
| 00:11.01 | [TK]D-Fender | But we're not in FreePBX here |
| 00:11.04 | lnb_ | correct |
| 00:11.09 | lnb_ | raw basic asterisk |
| 00:11.25 | lnb_ | not for the weak at heart |
| 00:12.15 | lnb_ | anyway the local asterisk will see the outbound call and see the match for a route and then send it to trunk |
| 00:12.19 | lnb_ | correct? |
| 00:12.28 | [TK]D-Fender | no. |
| 00:12.35 | [TK]D-Fender | * doesn't see an outbound call |
| 00:12.41 | [TK]D-Fender | There is no such thing |
| 00:12.46 | [TK]D-Fender | All calls to * ar3e INBOUND |
| 00:12.46 | WIMPy | hands lnb_ some Fishermans Friends. |
| 00:12.47 | lnb_ | well thats what -> exten => _1NXXNXXXXXX,n,Macro(dialout-trunk,2,${EXTEN},,off) says |
| 00:12.55 | [TK]D-Fender | YOU say where they go once they arrive |
| 00:12.55 | lnb_ | WIMPy: ? |
| 00:13.06 | lnb_ | no |
| 00:13.09 | [TK]D-Fender | *COUGH* |
| 00:13.13 | [TK]D-Fender | That's what he means |
| 00:13.13 | lnb_ | thats where asterisk says they go |
| 00:13.31 | goodtime | this is good |
| 00:14.08 | [TK]D-Fender | lnb_> well thats what -> exten => _1NXXNXXXXXX,n,Macro(dialout-trunk,2,${EXTEN},,off) says <- this doesn't say anything. So far with that one line.. it goes nowhere, does nothing because what it refers to doesn't exist yet |
| 00:14.15 | lnb_ | exten => _1NXXNXXXXXX,n,Macro -> go to trunk 2 |
| 00:14.15 | WIMPy | "If they're too strong, you're too weak" |
| 00:14.21 | goodtime | all as i know is there are no stupid questions |
| 00:14.24 | lnb_ | ahh ok got it |
| 00:14.32 | goodtime | just the one that was never asked |
| 00:14.33 | [TK]D-Fender | lnb_, You are assuming huge pieces of infrastructure exist, have a purpose, and are intrinsic. NONE of these are the case |
| 00:14.45 | lnb_ | ok you're right |
| 00:14.54 | lnb_ | <PROTECTED> |
| 00:14.54 | [TK]D-Fender | lnb_> exten => _1NXXNXXXXXX,n,Macro -> go to trunk 2 <- says what? |
| 00:15.04 | lnb_ | its a macro |
| 00:15.17 | [TK]D-Fender | lnb_, The's FreePBX code. Not yours |
| 00:15.17 | lnb_ | use trunk 2 to make the call |
| 00:15.22 | lnb_ | correct |
| 00:15.25 | lnb_ | not my code |
| 00:15.31 | [TK]D-Fender | And the entire concept of what "2" is.... is their own fabrication |
| 00:15.32 | WIMPy | Macros have been deprecated for ages. |
| 00:15.36 | lnb_ | oh |
| 00:15.37 | lnb_ | ok |
| 00:15.38 | [TK]D-Fender | Stop talking about FreePBX code |
| 00:15.41 | [TK]D-Fender | That's out the window |
| 00:15.41 | lnb_ | ok |
| 00:15.45 | lnb_ | got it |
| 00:15.47 | [TK]D-Fender | You aren't using their constructs. |
| 00:15.55 | lnb_ | correct |
| 00:16.06 | lnb_ | (unless i cheat and install freepbx) |
| 00:16.15 | lnb_ | but then i wont learn sfa |
| 00:16.16 | [TK]D-Fender | So never talk about them until you have a proper grasp of how * works |
| 00:16.46 | lnb_ | so now your telling me i have to invent my own code to make the calls? |
| 00:17.02 | [TK]D-Fender | WELCOME TO ASTERISK |
| 00:17.17 | [TK]D-Fender | All call processing = dialplan. All dialplan = YOUR JOB. |
| 00:17.21 | lnb_ | first of all, how to determine what files i need to populate /etc/asterisk with besides sip.conf |
| 00:17.42 | lnb_ | since i did not run make samples |
| 00:17.50 | [TK]D-Fender | Time to go learn what all those other wonderful files are for. |
| 00:18.02 | lnb_ | i see the samples in /src/asterisk/ll/configs |
| 00:20.55 | goodtime | nice |
| 00:21.08 | goodtime | im so glad i found this # |
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| 00:24.12 | lnb_ | does res_pjsip/chan_pjsip work with asterisk 11 ? |
| 00:24.18 | lnb_ | or leave it alone? |
| 00:24.35 | [TK]D-Fender | Does not exist in 11 |
| 00:24.39 | lnb_ | ok |
| 00:24.41 | lnb_ | thanks |
| 00:24.44 | [TK]D-Fender | 12+ |
| 00:24.55 | [TK]D-Fender | Where 12 = DOA |
| 00:24.57 | lnb_ | 12 seems to have issue with BLF |
| 00:25.09 | lnb_ | not sending the callerid |
| 00:25.13 | [TK]D-Fender | <lnb_> 12 seems to have issue with BLF <- nothing proven |
| 00:25.21 | lnb_ | so you dont know should i pick the call or not |
| 00:25.39 | lnb_ | i proved it alright |
| 00:25.40 | [TK]D-Fender | huh? |
| 00:25.54 | [TK]D-Fender | You never had full configs & debug at the same time across both versions |
| 00:26.01 | [TK]D-Fender | You kept saying you'd get it |
| 00:26.07 | [TK]D-Fender | You never provided them all. |
| 00:26.09 | lnb_ | just like i proved the issue with fail2ban and then they fixed it |
| 00:26.12 | lnb_ | ok nm |
| 00:26.14 | [TK]D-Fender | Not going through this again |
| 00:32.45 | lnb_ | man this is complicated |
| 00:45.41 | goodtime | well dont give up |
| 00:46.05 | goodtime | im sure everything can be figured out |
| 00:46.24 | goodtime | or it become suseless |
| 00:46.34 | WIMPy | Even if you figure out it's not possible :-) |
| 00:46.42 | goodtime | becomes useless* |
| 00:46.59 | goodtime | ahhh |
| 00:47.37 | goodtime | yeah i guess one a come to that conclusion |
| 00:48.19 | goodtime | wow that could be discurageing |
| 00:48.47 | WIMPy | You get used to that. |
| 00:48.49 | lnb_ | i have a similar setup where call flow is: itsp <--> a2billing server w/freepbx <--> clients freepbx server <--> sip phones (endpoints) |
| 00:49.17 | lnb_ | and looking at that a2billing/freepbx server in extensions_custom.conf |
| 00:49.29 | lnb_ | the only exten is _X |
| 00:50.05 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
| 00:50.27 | lnb_ | so it accepts anything by the look of it |
| 00:50.44 | lnb_ | but passes it to AGI(a2billing.php..) |
| 00:50.51 | WIMPy | _X would only accept one-digit numbers. |
| 00:51.04 | lnb_ | ok maybe _X. |
| 00:51.30 | WIMPy | Ok, so it only has to start with a digit then. |
| 00:51.35 | lnb_ | exten => _X.,1,NoOp(A2Billing Start) |
| 00:51.35 | lnb_ | exten => _X.,n,Agi(a2billing.php,1) |
| 00:51.35 | lnb_ | exten => _X.,n,Hangup |
| 00:51.55 | lnb_ | that looks like anything almost |
| 00:52.39 | lnb_ | i am sort of trying to replicate the flow without using a2billing at all |
| 00:52.49 | lnb_ | two pbx's talking to each other |
| 00:53.05 | lnb_ | i am just thinking out the logic |
| 00:53.50 | lnb_ | i don't want the local asterisk to have voicemail... if the phone is not answered, the remote freepbx sends call to vm |
| 00:54.32 | lnb_ | so that seems to imply that this local asterisk will be treated as an extension |
| 00:54.39 | lnb_ | by the remote freepbx |
| 00:55.08 | [TK]D-Fender | <WIMPy> Even if you figure out it's not possible :-) <- ESPECIALLY not then. http://sw-family.org/sites/default/files/fanart/demotivationals/airshaft-duel-denial.jpg |
| 01:00.25 | goodtime | lol cool [TK]D-Fender |
| 01:00.55 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 01:01.21 | lnb_ | did i install wrong version? Asterisk SVN-branch-11-r430589 |
| 01:01.36 | [TK]D-Fender | Wrong for what? |
| 01:01.43 | lnb_ | i wanted asterisk 11 |
| 01:01.49 | [TK]D-Fender | Sure looks like 11. |
| 01:01.55 | lnb_ | ok |
| 01:02.44 | [TK]D-Fender | By asking that pretty much says you didn't look up what that release was from SVN |
| 01:03.05 | [TK]D-Fender | http://lists.digium.com/pipermail/asterisk-commits/2015-January/071713.html |
| 01:04.27 | lnb_ | i followed a page to install it |
| 01:05.10 | [TK]D-Fender | And had no idea what you even grabbed |
| 01:05.34 | [TK]D-Fender | remembers the good old days when people would jsut download from asterisk.org and follow the instructions.... |
| 01:08.15 | lnb_ | the page i used did not say wget blah is was svn |
| 01:09.50 | goodtime | rtfm huh [TK]D-Fender? |
| 01:09.57 | goodtime | lol |
| 01:10.15 | [TK]D-Fender | The page you linked was in RUSSIAN. |
| 01:10.22 | WIMPy | Too many instructions to choose from... |
| 01:10.45 | goodtime | yeah reading up has always been the way to go if im realy serious about my intrests |
| 01:10.55 | [TK]D-Fender | <lnb_> http://asterisk-pbx.ru/wiki/asterisk_install/asterisk-12_freepbx-12_centos-7 |
| 01:10.56 | [TK]D-Fender | <lnb_> i can follow it along because commands are english |
| 01:11.15 | [TK]D-Fender | Yeah... anyone can copy & paste commands. ... and when you don't know what they are and why they are.... |
| 01:11.49 | goodtime | ok so it is redhat or something like that then..... |
| 01:12.24 | goodtime | maybe fedroa |
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| 01:13.19 | goodtime | i mostly am fimilar with apt-get/debian based OS |
| 01:13.47 | goodtime | but i use a kde plasma GUI |
| 01:14.08 | goodtime | guess i like the eyecandy heh |
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| 01:35.14 | lnb_ | had he looked at the russian page, he would have seen there is NO svn ... that came from some other page |
| 01:36.23 | lnb_ | russin page -> wget http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-12-current.tar.gz |
| 01:36.30 | lnb_ | russian i meant |
| 01:37.25 | lnb_ | i will blow this away and start fresh |
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| 02:02.09 | kemmler | What's the proper way to take ${CDR(billsec)} after a call ends? Is there a proper way to do it from the dialplan or should I use an independent script that parses the CDR. |
| 02:04.01 | WIMPy | Where/when do you want it? |
| 02:04.25 | kemmler | I'd like to update mysql with it after the call ends |
| 02:04.40 | kemmler | after each call |
| 02:05.03 | WIMPy | Use te h extension. |
| 02:05.10 | WIMPy | the |
| 02:05.18 | WIMPy | But why don't you just put the cdr to mysql? |
| 02:06.32 | kemmler | I'm doing that as well. I simply have a separate counter I'm using |
| 02:07.18 | kemmler | should i use DeadAGI in conjunction with the h exten? |
| 02:07.48 | kemmler | if i'm going to be using an external script to perform some operations beforehand? |
| 02:08.27 | WIMPy | Do you want to return anything to Asterisk? |
| 02:08.32 | kemmler | negative |
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| 02:08.51 | WIMPy | Then AGI doesn't make sense. |
| 02:09.01 | WIMPy | Use a normal System(). |
| 02:09.14 | kemmler | will do. thank you for the info |
| 02:11.44 | [TK]D-Fender | I would just use FUNC_ODBC and keep it all internal. |
| 02:12.47 | kemmler | I'll entertain that option as well. Just trying to feel out how I can go about it for now |
| 02:14.59 | [TK]D-Fender | Lower overhead, no scripts smaller code, all win, more fibre. |
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| 04:29.25 | hanuman | Hi, what is live recording and stored recording in asterisk ari |
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| 04:38.47 | dijib | hello Penguin and all else |
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| 04:43.12 | hanuman | Hi, what is live recording and stored recording in asterisk ari |
| 04:44.10 | ChannelZ | What do you mean by "what is" |
| 04:46.11 | hanuman | ChannelZ: is there any possible to soter recordings in remote server with live recordings |
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| 04:47.26 | ChannelZ | I suppose with NFS if you need to do it live |
| 04:47.52 | ChannelZ | Otherwise you could build some mechanism to upload or sync them after-the-fact |
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| 04:52.43 | hanuman | ChannelZ: In asterisk ari what is target_uri |
| 05:03.33 | ChannelZ | I am not overly familiar with ARI but I think that's the URI of the channel you're manipulating/issuing commands for |
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| 09:19.45 | bsv | hi |
| 09:20.08 | bsv | is there anyway (in asterisk) to control what rtp port the client will use? |
| 09:20.24 | bsv | like rport in the via header for sip |
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| 10:13.10 | SupaYoshi | Can someone tell me why this dialplan wont put me through to 112 with PSTN? |
| 10:13.30 | SupaYoshi | (112xS0<:@gw0>|0[89]xx.<:@gw0>|<:003130>[1-9]xx.|<0:0031>[1-7]xxxx.|00xx.) |
| 10:14.29 | WIMPy | Definitely not without telling us un what device. |
| 10:14.37 | SupaYoshi | SPA 3000 |
| 10:14.38 | SupaYoshi | sorry. |
| 10:14.46 | SupaYoshi | I got a dialplan from here, tweaked it a little |
| 10:14.54 | SupaYoshi | http://www.callvoip.nl/technotes/voip_sipura3000_dialplan.pdf |
| 10:15.03 | SupaYoshi | But when I call 112, to try it out. |
| 10:15.16 | SupaYoshi | It says, you have dialed an invalid number which means its going over the SIP trunk. |
| 10:16.43 | wdoekes | x is 0-9 right? then 112x would be 112[0-9], no? |
| 10:18.28 | wdoekes | even the pdf says so |
| 10:18.34 | SupaYoshi | no |
| 10:18.38 | wdoekes | yes it does |
| 10:18.42 | wdoekes | check (A) |
| 10:18.42 | SupaYoshi | 112 is emergancy |
| 10:19.09 | wdoekes | the pdf is broken at the bottom, but ok at the top |
| 10:19.14 | SupaYoshi | Ohhh.. |
| 10:19.16 | SupaYoshi | So It needs to be |
| 10:19.22 | SupaYoshi | (11xS0<:@gw0>|0[89]xx.<:@gw0>|<:003130>[1-9]xx.|<0:0031>[1-7]xxxx.|00xx.) |
| 10:19.29 | SupaYoshi | right? |
| 10:19.32 | wdoekes | can't vouch for the rest |
| 10:19.44 | SupaYoshi | kk |
| 10:19.45 | wdoekes | but it's obvious that if it says X here and Y there, one of them must be wrong |
| 10:19.54 | SupaYoshi | I jus need to make sure |
| 10:19.56 | SupaYoshi | why it wont work :3 |
| 10:22.00 | SupaYoshi | k back |
| 10:22.08 | SupaYoshi | changed it, jus getting busy line tone. |
| 10:22.17 | SupaYoshi | so i guess its still not correct :x |
| 10:23.42 | wdoekes | start with: x.<:@gw0> |
| 10:23.51 | wdoekes | it should put everything on the pstn |
| 10:23.58 | wdoekes | if that doesn't work, you need to check why |
| 10:23.58 | SupaYoshi | let me test that :D thanks |
| 10:25.45 | SupaYoshi | yep that wont work, busy tone as well |
| 10:25.58 | SupaYoshi | So i think i have to modify the dialplan on the PSTN tab too. |
| 10:26.09 | SupaYoshi | In contro dictionary with what all these guides say somehow? |
| 10:26.25 | SupaYoshi | x.<:@gw0> |
| 10:26.48 | SupaYoshi | PSTN line tab has dialplans too |
| 10:30.55 | SupaYoshi | Could it be because I connected it through to an old pbx? |
| 10:31.04 | SupaYoshi | I get a busy tone when I call through PSTN |
| 10:31.16 | SupaYoshi | when I unplug internet however, and say PSTN auto fallback |
| 10:31.29 | SupaYoshi | It will route the calls over my old PBX though... so im wondering if the dialplan is still wrong. |
| 10:31.36 | SupaYoshi | or that its working just fine now |
| 10:38.36 | wdoekes | SupaYoshi: appendix A http://www.callvoip.nl/docs/sipura/spa3000_guide.pdf |
| 10:38.46 | wdoekes | - entire dp must be surrounded by () |
| 10:42.28 | SupaYoshi | Yeah i got these. |
| 10:42.37 | SupaYoshi | (11xS0<:@gw0>|0[89]xx.<:@gw0>|<:003130>[1-9]xx.|<0:0031>[1-7]xxxx.|00xx.) |
| 10:42.44 | SupaYoshi | See ( - ) |
| 10:43.08 | SupaYoshi | But I think its working |
| 10:43.15 | SupaYoshi | just not working with my old pbx (1994) behind it |
| 10:43.21 | SupaYoshi | so going to test it on an old line later? |
| 11:00.07 | SupaYoshi | KAy |
| 11:00.08 | SupaYoshi | 112 works |
| 11:00.26 | *** join/#asterisk marceloamorim (~marcelo@189.90.192.72) |
| 11:01.03 | SupaYoshi | Now 0800 - 0900 numbers |
| 11:01.05 | *** part/#asterisk marceloamorim (~marcelo@189.90.192.72) |
| 11:02.19 | SupaYoshi | These wont work still |
| 11:02.28 | SupaYoshi | (11xS0<:@gw0>|0[89]xx.<:@gw0>|<:003130>[1-9]xx.|<0:0031>[1-7]xxxx.|00xx.) |
| 11:02.31 | SupaYoshi | This is my dialplan. |
| 11:02.37 | SupaYoshi | Calling 0800 0402 |
| 11:02.41 | SupaYoshi | Results in a busy line tone. |
| 11:02.55 | SupaYoshi | But I thought 0[89]xx. would cover for that. |
| 11:03.57 | SupaYoshi | wdoekes, ya there? |
| 11:17.17 | SupaYoshi | Okay have it working thanks |
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| 11:23.50 | wdoekes | SupaYoshi: so, what's the final version (and what was the troublesome part)? |
| 11:24.30 | WIMPy | I have never done anything to in-device dialplans other than just delete them. |
| 11:27.11 | SupaYoshi | It works now :) |
| 11:27.16 | SupaYoshi | Oh, |
| 11:27.17 | SupaYoshi | Well. |
| 11:27.31 | SupaYoshi | (11xS0<:@gw0>|0[89]xx.<:@gw0>|<:003130>[1-9]xx.|<0:0031>[1-7]xxxx.|00xx.) final version for LINE 1 (VOIP) |
| 11:27.38 | SupaYoshi | but the catch is, that noone tells or tests? idk? |
| 11:27.41 | SupaYoshi | That with the SPA3000, |
| 11:27.56 | SupaYoshi | You need to put the dialplan in PSTN LINE too |
| 11:28.00 | SupaYoshi | So, in PSTN line I put, |
| 11:28.05 | SupaYoshi | (11xS0|0[89]xx.) |
| 11:28.12 | SupaYoshi | Works like a charm :D |
| 11:28.19 | SupaYoshi | I even called 911 on accident :/ |
| 11:28.25 | SupaYoshi | damnit |
| 11:28.41 | SupaYoshi | But yah was okay said sorry. x.x |
| 11:29.05 | zekoZeko | how else can you know it's working? :) |
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| 11:31.22 | SupaYoshi | lawd ikr |
| 11:31.33 | SupaYoshi | there aint no testing number |
| 11:31.36 | SupaYoshi | well i could have called 113 |
| 11:31.42 | SupaYoshi | which is animal emergancy |
| 11:31.43 | SupaYoshi | :P |
| 11:31.46 | SupaYoshi | x |
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| 11:34.30 | zekoZeko | that must be some US-ian practical joke :) In these parts of the world 113 is the police :) |
| 11:34.54 | WIMPy | Where's that? |
| 11:34.59 | zekoZeko | most of europe |
| 11:35.23 | WIMPy | is in Europe, but that's new to me. |
| 11:35.33 | zekoZeko | maybe not then |
| 11:35.49 | zekoZeko | what do you have for emergency? I thught it's 112 except UK (999) |
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| 11:36.16 | WIMPy | Yes, 112 is standard. |
| 11:36.42 | WIMPy | And if you're using a GSM mobile, that's worldwide. |
| 11:37.17 | zekoZeko | ah i see now, i thought 113 is "standardised" also, but seems not. |
| 11:37.37 | Rac-on | we used to have 114 as animal-emergency in holland, but that got changed to 144 a while back |
| 11:38.12 | eirirs | 112 are emergency call to police here |
| 11:38.29 | zekoZeko | 113 is police in italy, luxemburg and slovenia only it seems :) wonder why we picked it. |
| 11:39.02 | zekoZeko | 112 is general emergency here in .si, they call fire, ambulance and/or police then |
| 12:26.32 | SupaYoshi | We have 113 |
| 12:26.38 | SupaYoshi | Is suicide line |
| 12:26.56 | SupaYoshi | italian guy in the Netherlands calls 113 for Police. |
| 12:26.59 | SupaYoshi | Hello is this police? |
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| 12:58.13 | michael-i | Hi all. I'm having some trouble making outbound calls through RingCentral. I get a "Everyone is busy/congested at this time (1:0/0/1)" message when attempting the Dial but tracing SIP shows now attempt is made to RC. This is on Asterisk 13.1.0. |
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| 13:41.50 | andremar | Hy friends. Anybody out there that can help with an Alsa issue? |
| 13:42.28 | WIMPy | ~ask |
| 13:42.28 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 13:45.06 | andremar | All right. Anybody knows why this: "WARNING[3007][C-00000011]: chan_alsa.c:568 alsa_indicate: Don't know how to display condition 26 on ALSA/plughw:1,0" happens when I dial a SIP channel via console dial. Also and maybe related, only the first call I make using this method has sound. To get audio again one has to module unload chan_alsa.so and then load again. |
| 13:48.55 | michael-i | Update on RingCentral outbound calls: I got past my 407 Proxy auth problems and the destination rings. I pick up and the call cuts immediately after connecting. Feels like codec mis-match? |
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| 13:58.38 | [TK]D-Fender | michael-i: Is that ... a question? Does it give you that as an error? Do we have a call to comment on? |
| 13:59.28 | michael-i | [TK]D-Fender: just giving an update and still fiddling, i'll have a test call soon |
| 14:00.00 | michael-i | [TK]D-Fender: no error readily apparent, looking to get more info with some more config changes |
| 14:00.36 | [TK]D-Fender | Readily apparent requires that we actually see it. Let us know when you've got something to show... |
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| 14:03.46 | Johnny5 | Ring Central should have a list of supported codecs, which are negotiated during the invite, and should reject accordingly if you are only offering off-the-wall choices that they do not. This is all speculation without a debug of PCAP. |
| 14:05.59 | [TK]D-Fender | Lack of debug leads to speculation. Speculation leads to wasting of time. Wasting of time leads to the dark side.... |
| 14:06.05 | [TK]D-Fender | </yoda> |
| 14:08.53 | michael-i | [TK]D-Fender: Looks like my stripped down modules may have something to do with it. It passes SDP negotiation but blows up here: WARNING[1953][C-00000000]: bridge.c:776 in bridge_base_init: Bridge e3eebb82-e145-4e5e-9386-e06e3a30eb7b: Could not create class basic. No technology to support it. |
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| 14:10.55 | file | you need bridge_simple |
| 14:11.02 | michael-i | [TK]D-Fender: aka all kindly ignore, pretty sure I've got it :) Thanks anyway, fine wizards |
| 14:11.15 | michael-i | file: yup, rebuilding now :) |
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| 14:22.22 | kj22594 | Hey all, running into this issue when trying to use sipml5 in unison with Asterisk. It appears every time I hit a button on the built in keypad. The error is "ERROR[23573]: res_pjsip_dtmf_info.c:139 dtmf_info_incoming_request: Invalid DTMF event signal in INFO message." |
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| 16:18.08 | *** join/#asterisk bibz (~bibz@ip212232024148.rev.nessus.at) |
| 16:18.58 | bibz | hey there. I keep getting the hangup-cause "SIP 183 Session Progress", which leads to lost calls.. Does anyone have experience with this error? |
| 16:19.22 | bibz | I'm even willing to pay for helpful advice.. |
| 16:20.56 | wdoekes | 183 is not an error, how do you see that as a hangup-cause? |
| 16:21.18 | bibz | wait, it is not?! |
| 16:21.50 | Johnny5 | Nope |
| 16:22.18 | bibz | guess "180 Ringing" also not? |
| 16:22.19 | wdoekes | it's used for early media: audio before the call is picked up |
| 16:22.30 | wdoekes | are you trolling us? |
| 16:22.51 | bibz | uh, sorry if it looks like it, but no, i'm not trolling. |
| 16:24.10 | bibz | but I don't understand how it is not an error, since most of the calls end with a "SIP 200 OK", and others do with "183 Session Progress".. |
| 16:25.29 | wdoekes | no, a call does not end with 200. a call starts with an INVITE (which gets acknowledged by a 200 and an ACK). a call ends with a BYE (which gets acknowledged by a 200). |
| 16:25.32 | Johnny5 | Neither are cause for hangup in and of themselves |
| 16:27.40 | file | well, or a failure response |
| 16:27.59 | bibz | I was given a 1700 line long dialplan and i'm trying to understand how this whole thing works. It is writing the hangup-cause for every call in an MSSQL database, in the "hangup_cause" table.. and 80% of the calls have "SIP 200 OK" as the hangup-cause. the other 10% are busy, and the other ones are either "SIP 180 Ringing" or "SIP 183 Session Progress".. |
| 16:28.28 | bibz | I tought, that these 10% are lost calls, since they don't have an "SIP 200 OK".. |
| 16:34.50 | Johnny5 | bibz: Response code definitions: http://goo.gl/fTFCo |
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| 16:36.12 | bibz | thanks! |
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| 16:42.27 | michael_work | hmmm. we keep VM records inside mysql as blobs. now i see there are few of them that fail to be inserted, no futher error description :( all i see is app_voicemail.c: SQL Direct Execute failed! and res_odbc.c: SQL Execute error! Verifying connection to asterisk [asterisk-connector]... |
| 16:43.09 | michael_work | is asterisk writting it to hd and only than insert to db or is it straight to db? if 1st why would file be deleted on failed insert |
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| 17:02.21 | *** join/#asterisk RPerre (~rperre@alpes.nortenet.pt) |
| 17:04.18 | RPerre | Hello! Can someone answer me a couple of questions about ARI ? |
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| 17:11.57 | mjordan | ~ask |
| 17:11.57 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 17:13.54 | *** join/#asterisk elguero (~miguel323@2001:470:1f06:12c4::2) |
| 17:14.04 | mjordan | bibz: the HANGUPCAUSE will return the last known technology specific response received for that channel. See https://wiki.asterisk.org/wiki/display/AST/Hangup+Cause |
| 17:14.22 | mjordan | bibz: if you don't have a terminating message, your last message may be a 183, or a 200, etc. |
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| 17:18.53 | *** join/#asterisk igcewieling1 (~ewieling@ip98-170-196-157.pn.at.cox.net) |
| 17:19.43 | RPerre | Ok, first of all I'm new to Asterisk, I'm developing (trying at least) a application using ARI. My question is: Can I 'originate' from A to B and control the call? For example if someone 'hangup', can I know who it was? If someone rejects the call, can I know the 'cause'? |
| 17:20.09 | igcewieling1 | Has anyone here worked with Digium and Adran regarding Asterisk's OPTIONS packets being considered by the Adtran to be invalid when proxy mode is enabled (not SIP ALG, an actual SIP Proxy) |
| 17:20.15 | igcewieling1 | Adtran says the packets are not valid. |
| 17:21.46 | RPerre | *also, I'm using phpari |
| 17:23.55 | igcewieling1 | RPerre: Have you considered using PHPAGI? |
| 17:25.27 | RPerre | igcewieling1, but I'll be using AGI yes? I need to use ARI. |
| 17:27.06 | igcewieling1 | Must be something I'm not familiar with. The only ARI I know about is Asterisk Recording Interface |
| 17:27.39 | RPerre | Its a REST interface for Asterisk. https://wiki.asterisk.org/wiki/display/AST/Getting+Started+with+ARI |
| 17:28.58 | *** part/#asterisk lorenzo (lorenzo@unaffiliated/lorenzo) |
| 17:32.22 | file | RPerre, you can't directly originate from A to B - you can originate calls out and then what happens is up to you |
| 17:32.36 | file | RPerre, you have to bridge them, if one hangs up you need to hang the other up |
| 17:36.11 | RPerre | file, (I'm using phpari). Lets say I have a Stasis Application running: "hello world". I have a dialplan context called 'mycontext', and it delivers the call to this stasis app. Should I originate A, then originate to B and then bridge the two? |
| 17:36.31 | RPerre | Should i do it inside my stasis application? |
| 17:36.43 | file | RPerre, if you have a channel in your application then you already have a channel - it depends on whether you want to connect THAT channel to something else or not... |
| 17:36.45 | RPerre | (sorry, but ARI confuses me alot) |
| 17:37.30 | RPerre | file, did you ever used phpari? maybe i can give you an example |
| 17:37.37 | file | nope |
| 17:37.52 | RPerre | ok, then let me try to describe it |
| 17:37.57 | file | but phpari is just an SDK around ARI itself - it doesn't change how ARI works |
| 17:39.21 | RPerre | Lets say i have a stasis application running |
| 17:39.44 | RPerre | and I use (outside of this application) 'originate' |
| 17:40.09 | RPerre | ...sorry, i'm not making sense |
| 17:40.22 | RPerre | give me a second |
| 17:41.03 | RPerre | this is a basic stasis aplication: https://github.com/greenfieldtech-nirs/phpari/blob/master/examples/BasicStasisApplication.php |
| 17:41.35 | RPerre | if i have a context (on dialplan) that sends the call to this stasis |
| 17:43.18 | RPerre | this is how i originate: https://github.com/greenfieldtech-nirs/phpari/blob/master/examples/ChannelOriginate.php |
| 17:43.36 | RPerre | my REAL question is: |
| 17:43.56 | RPerre | i want to use this originate example, and make it pass tru the stasis application |
| 17:43.59 | RPerre | i have no clue how |
| 17:44.08 | RPerre | (sorry, i know i'm not make alot of sense) |
| 17:45.06 | file | ARI allows you to specify a stasis application (along with arguments) that the originated channel is sent to upon answer, in the above example this is done by setting "app" and "appArgs" and leaving out "extension", "context", "priority" |
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| 17:53.26 | RPerre | file, does this make any sense? http://pastebin.com/F2tiXYU9 |
| 17:53.43 | file | yes |
| 17:53.55 | file | well, without context |
| 17:54.02 | RPerre | so if 'perre' is: |
| 17:54.03 | RPerre | [perre] |
| 17:54.03 | RPerre | exten => _XX.,1,NoOp() |
| 17:54.03 | RPerre | <PROTECTED> |
| 17:54.03 | RPerre | <PROTECTED> |
| 17:54.11 | RPerre | that should be it, right? |
| 17:54.28 | file | I don't understand the question |
| 17:54.36 | RPerre | the context |
| 17:54.41 | RPerre | in the dialplan |
| 17:54.52 | file | that would originate a call into your Stasis application and once answered send that into your Stasis application |
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| 17:55.51 | RPerre | ok, i'll do some tests |
| 17:55.54 | RPerre | thanks file |
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| 18:07.50 | JaniceKitten | hopes she's not meant to be banned |
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| 18:09.17 | JaniceKitten | hello voxter |
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| 18:11.41 | eirirs | JaniceKitten: why |
| 18:11.49 | JaniceKitten | eirirs, wat |
| 18:12.02 | eirirs | meant to be banned |
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| 18:17.02 | mjordan | RPerre: you can originate from Asterisk to some other device using POST /channels. When the far end answers, the created channel will either be put into your Stasis application or will go execute dialplan, depending on what you told it to do. Example of this can be seen in the mixing bridge example on the wiki: https://wiki.asterisk.org/wiki/display/AST/ARI+and+Bridges%3A+Basic+Mixing+Bridges#ARIandBridges:BasicMixingBridges-Example:Impleme |
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| 18:17.23 | mjordan | er, sorry, just saw file answered your question :-P |
| 18:17.48 | file | I'm not quite sure what the real question was in the end, but I'll roll with it |
| 18:17.52 | file | just like butter |
| 18:17.57 | mjordan | file: does that mean you auto-succeed? |
| 18:18.33 | file | mjordan, maybe |
| 18:19.07 | JaniceKitten | hi |
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| 18:43.47 | mjordan | lunches |
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| 20:46.27 | PaybackTony | Could anyone point me in a direction as to what on the asterisk side may be stopping a sip client (SIP.js in this case) from reaching ice state connected (from ice state complete)? |
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| 20:49.32 | PaybackTony | I have two setups, one works, one doesn't. Same asterisk config on both but each server has gone through a few other packages / upgrades. On the one that no longer works, everything seems fine up until the client itself sends the ACK (and should go from ice state complete to ice state connected). Asterisk begins sending RTP to the public IP of the client (via ICE), but since the ice connection never reaches connected state no audio |
| 20:49.39 | PaybackTony | 11.15 |
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| 21:08.04 | pabelanger | PaybackTony, fire up tcpdump and see what is happening |
| 21:12.14 | PaybackTony | pabelanger, doing that at the moment, first started with sniffing rtp traffic and no rtp seems to be going outbound, doing a few more things to triple check |
| 21:27.42 | PaybackTony | It looks like my machine (sip.js client) is receiving the packets as I suspected, but I am not very experienced with tcpdump however I can see what seems to be the ICE negotiation happening with packets flowing back and forth and after that there is a consistent pattern of many packets sent from asterisk to the client and then just two packets sent back. Console log on chrome doesn't show any of this activity after the ice negotiati |
| 21:29.18 | PaybackTony | I should note again that rtp debug on asterisk side doesn't appear to show those 2 packets actually making it back |
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