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| 00:48.58 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: Asterisk 11.11.0 (2014/07/10), 1.8.29.0 (2014/07/10); Standard: Asterisk 12.4.0 (2014/07/10); DAHDI: DAHDI-linux 2.9.0 (2014/01/30), DAHDI-tools 2.9.0.1 (2014/01/31); libpri 1.4.15 (2014/06/16) -=- Visit the official Asterisk wiki: wiki.asterisk.org -=- Read the Code of Conduct bit.ly/1hH6P22 |
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| 09:01.11 | XATRIX | Hi guys, i have a problems with a new setup. I'm using elastic 3.0 RC1, and i can't connect my X-Lite software to... tcpdump shows some packets incoming from my IP, but X-lite can't get connected. I've created an Organization and a user in. Is there any log file i can check for ? |
| 09:01.30 | XATRIX | I mean, where can i found some asterisk auth-attempts or connections logs ? |
| 09:05.26 | r00f | try /var/log/asterisk/messages |
| 09:05.40 | r00f | however, i am not sure if elastics uses that path |
| 09:07.24 | XATRIX | Yea, i have no such |
| 09:08.21 | XATRIX | I have only '/var/log/asterisk/full' there , and some directories |
| 09:08.30 | r00f | check the full |
| 09:08.47 | XATRIX | r00f: also, i have another software that can succsessfully connect to, but can't make calls anywhere: |
| 09:08.51 | XATRIX | [Jul 16 12:07:10] VERBOSE[3495] chan_sip.c: -- Unregistered SIP 'admin_parkovka.ua' |
| 09:08.51 | XATRIX | [Jul 16 12:07:11] VERBOSE[3495] chan_sip.c: -- Registered SIP 'admin_parkovka.ua' at 127.0.0.1:5060 |
| 09:08.51 | XATRIX | [Jul 16 12:07:15] NOTICE[3495] chan_sip.c: Peer 'admin_parkovka.ua' is now UNREACHABLE! Last qualify: 3 |
| 09:09.17 | XATRIX | I suppose there should be my external office IP , but i can't make it appear there :) |
| 09:09.42 | r00f | mysteries and secrets... |
| 09:10.37 | XATRIX | ( |
| 09:12.19 | r00f | this log does not show any call attempts |
| 09:12.49 | XATRIX | No, there's no call attempts. Only connection one |
| 09:13.05 | XATRIX | I dial a number - nothing happening |
| 09:13.47 | r00f | could be nat issue, if you say that that ip is the external one |
| 09:14.40 | XATRIX | Yes, but i have another commercial service with the same pbx, and it works ! |
| 09:14.58 | XATRIX | I'd like to build my own one |
| 09:16.25 | r00f | enable sip debug and see why it goes unreachable right after registering. tbh, i don't know why does it time out in 4 seconds |
| 09:17.53 | XATRIX | Sorry where can i enable debug ? |
| 09:18.43 | XATRIX | sip set debug ? |
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| 09:20.49 | cw1972 | sip set debug on in the asterisk cli |
| 09:23.27 | XATRIX | Yes, and when i restart my X-lite - log shows me completely nothing in |
| 09:26.05 | XATRIX | That's what i have http://imgur.com/PXNYZWL |
| 09:26.22 | XATRIX | I also tried Domain: pbx.parkovka.ua |
| 09:26.37 | XATRIX | Nothing happens as well |
| 09:27.09 | XATRIX | But my second softphone MicroSIP can connect to, and successfully auth |
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| 10:21.08 | XATRIX | I tried to change the server's IP address to domain name - and now my MicroSIP says 483 Too Many Hops |
| 10:21.40 | XATRIX | What does it mean ? I don't actually do any routings... I simply want to connect it to my Asterisk |
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| 10:33.07 | XATRIX | Any ideas what can be there ? |
| 10:39.14 | *** join/#asterisk bibz (~bibz@router.ais.at) |
| 10:39.16 | bibz | hey guys |
| 10:39.46 | bibz | exten => _+.,n,Set(CALLERID(number)=${CALLERID(number):1}) |
| 10:40.03 | bibz | this won't shorten the number in the "From:" header.. why is that? |
| 10:45.18 | bibz | any clues? |
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| 11:06.27 | bibz | nobody? :( |
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| 11:26.13 | XATRIX | Is there any good SIP client for Windows Linux which can give me verbose logging ? |
| 11:26.22 | bibz | is it even possible to modify the From: header? |
| 11:28.20 | hindi | XATRIX, Phoner |
| 11:30.28 | hindi | bibz, sip.conf fromuser |
| 11:31.09 | hindi | bibz, or want to do this in dialplan? |
| 11:31.59 | bibz | i'm doing it in dialplan |
| 11:32.58 | hindi | I think asterisk only give you access to headers of the initial INVITE |
| 11:33.00 | bibz | I'm having "fromuser=myvoipuser" in my voip.conf... |
| 11:33.12 | bibz | sip.conf* |
| 11:33.18 | bibz | what do you mean by that? |
| 11:33.34 | hindi | you only can acces the from in the invite and only read it |
| 11:33.41 | hindi | no write access |
| 11:34.12 | bibz | my provider wants me to send the callerID without a 0 in front of it.. |
| 11:34.17 | bibz | they want international prefix |
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| 11:34.28 | hindi | ok thats not in the from :) |
| 11:34.29 | bibz | like 436643024243 instead of 0043664302492 |
| 11:35.04 | bibz | well, they seem pretty incompetent.. they sent me an email with my wrong request and marked out the errors |
| 11:35.07 | hindi | CALLERID(num)=136643024243 |
| 11:35.36 | hindi | can you pastebin the INVITE = |
| 11:35.46 | bibz | yes, hold on. |
| 11:38.10 | bibz | http://pastebin.com/jzkxMvd0 |
| 11:38.33 | bibz | i changed the numbers and logins to something random |
| 11:38.43 | hindi | np |
| 11:38.43 | bibz | i hope it doesn't confuse you. |
| 11:39.06 | hindi | what did they mark as wrong? |
| 11:39.07 | bibz | the call takes place, but since the "From:" doesn't meet the providers requirements, I'm calling out as "unknown" |
| 11:39.20 | hindi | ok :) |
| 11:39.34 | hindi | use Set(CALLERID(num)=<yournumber> |
| 11:39.36 | bibz | From: ">>>>06912186925<<<<<" <sip:4321321125086925@node1.myprovider.at>;tag=asdddd6476 |
| 11:39.38 | hindi | try it |
| 11:39.41 | bibz | the >>> <<< is what they marked |
| 11:39.51 | hindi | ok |
| 11:40.03 | bibz | it should be like this: |
| 11:40.04 | hindi | its the CallerID(name) |
| 11:40.06 | bibz | From: "436912186925" <sip:4321321125086925@node1.myprovider.at>;tag=asdddd6476 |
| 11:40.22 | bibz | no zero (prefix), but instead the country (international) prefix |
| 11:40.40 | bibz | this is my set: exten => _+.,n,Set(CALLERID(num)=${CALLERID(num):1}) |
| 11:40.55 | hindi | yes include Set(CALLERID(name)=436912186925) and Set(CALLERID(number)=436912186925) and try it again |
| 11:41.15 | bibz | isn't it CALLERID(num) ? |
| 11:41.40 | bibz | or are those 2 different functions.. |
| 11:42.04 | hindi | sry its num |
| 11:42.10 | bibz | oh ok |
| 11:42.37 | hindi | but they marked the name |
| 11:42.56 | hindi | so please try to set CALLERID(name) |
| 11:43.29 | hindi | "06912186925"<--- NAME <sip:4321321125086925 <---NUMBER @node1.myprovider.at> |
| 11:43.58 | bibz | they marked only the From: ">>>Number<<<" |
| 11:44.06 | bibz | thats the weird thing about it.. |
| 11:44.22 | hindi | so you are from austria? |
| 11:44.46 | hindi | maybe join asterisk-de so we can discuss in german :) |
| 11:45.05 | bibz | oh, great. |
| 11:45.10 | bibz | bis gleich :) |
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| 12:29.26 | Ibrahim22 | Hi, everyone! I have a question related to the Queue app. In my dialplan callers get into a queue. Callers can then be transferred (blind transfer). If this transfer fails, they rejoin the queue. I use AMI events to keep track of state. Every time a transfer is done, I get a AgentComplete event. The weird thing is that when a caller gets transferre |
| 12:29.26 | Ibrahim22 | d and that transfer fails and he/she rejoins the queue, all the subsequent AgentComplete events still sends out the original Uniqueid (when they first got in the queue), even though other events related to Queue do not. |
| 12:30.32 | WIMPy | Smells like local channels. |
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| 12:35.41 | Ibrahim22 | Jep, I use local channels for the agent interfaces + transfers + rejoining queues |
| 12:36.41 | WIMPy | So you're seeing the IDs of the local channels until they disappear or are optimized away. |
| 12:38.24 | Ibrahim22 | Is there a better way to uniquely identify callers then? |
| 12:39.29 | WIMPy | If you use local channels, you have to track their creation and link them to the original channel yourself. |
| 12:41.08 | Ibrahim22 | Okay, so I should not be using AgentComplete |
| 12:43.05 | WIMPy | I'd just read what's all being sent and see if the wanted information is directly available somewhere. But if you create new chennels you might have to catch their creation and the channel they belon |
| 12:44.39 | Ibrahim22 | Okay, I'll check out my events. Thank you for the advice |
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| 13:19.44 | XATRIX | Ok, guys, i've made it with my clients, but now i have a different kind of a problem. I have 2 extensions. 110,120. I call 110 from my 120, and it says "your call cannot be completed at the moment, please check your number and dial again" |
| 13:20.08 | [TK]D-Fender | ? pb |
| 13:20.10 | [TK]D-Fender | ^^^ |
| 13:20.18 | XATRIX | http://fpaste.org/118428/14055167/ - there's a log file of the calls |
| 13:20.22 | [TK]D-Fender | show us the call |
| 13:20.42 | [TK]D-Fender | There is no verbose in there |
| 13:20.53 | [TK]D-Fender | "core set verbose 10" |
| 13:21.06 | XATRIX | Wait a bit. I'll do it |
| 13:21.53 | XATRIX | http://fpaste.org/118429/51690514/ |
| 13:22.01 | XATRIX | I hope it's what you asked about |
| 13:23.44 | XATRIX | It says i have no such recipient ? |
| 13:23.50 | XATRIX | Or i have to reload configs |
| 13:23.58 | XATRIX | Am i right ? |
| 13:25.50 | XATRIX | Ok, i've reloaded and now it says, "110 is unavailable, please leave the message after the beep" And that's a call log: http://fpaste.org/118431/05517095/ |
| 13:26.46 | [TK]D-Fender | that is FreePBX dialplan... |
| 13:27.02 | [TK]D-Fender | And looks like 110 is not defined. |
| 13:27.43 | [TK]D-Fender | "dialplan show pbx.parkovka.ua-from-internal" |
| 13:27.51 | [TK]D-Fender | "dialplan show ext-local" |
| 13:27.54 | [TK]D-Fender | pastebin those together |
| 13:29.27 | XATRIX | Ok, It's Elastix server |
| 13:29.50 | [TK]D-Fender | guess you didn't apply your changes... |
| 13:30.24 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 13:31.33 | [TK]D-Fender | <PROTECTED> |
| 13:31.43 | XATRIX | http://fpaste.org/118435/17493140/ |
| 13:31.49 | XATRIX | Thats' the dial plans |
| 13:32.03 | [TK]D-Fender | So now it tries to dial an is supposedly "busy". Actual SIP debug from CLI would confirm exactly what responded and how. |
| 13:32.18 | [TK]D-Fender | Dial plans are fine... now taht you applied your changes |
| 13:32.33 | XATRIX | I clicked reload :) |
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| 13:35.05 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 13:38.10 | XATRIX | [TK]D-Fender: ok, look. When 110 calls 120 - he can reach me. But if i call 120 -> 110 - the person or number is unavailable |
| 13:38.31 | XATRIX | I didn't set any DND status or whatever else |
| 13:38.56 | [TK]D-Fender | "sip set debug on" <--- look at the call in proper detail |
| 13:42.20 | XATRIX | http://fpaste.org/118437/51813214/ |
| 13:43.14 | [TK]D-Fender | Clearly not enabled... |
| 13:43.35 | XATRIX | Sorry ? What is not enabled ? Debug or extension ? |
| 13:43.40 | [TK]D-Fender | SIP DEBUG |
| 13:43.44 | [TK]D-Fender | I jsut gave you the command..... |
| 13:43.59 | XATRIX | But the last lines : free-189*CLI> sip set debug off SIP Debugging Disabled |
| 13:44.12 | XATRIX | So it was enabled before and verbose is 10 |
| 13:44.24 | [TK]D-Fender | that is AFTER the call |
| 13:44.38 | XATRIX | Ok, let me try again |
| 13:45.43 | XATRIX | http://fpaste.org/118441/51829314/ |
| 13:45.47 | XATRIX | Is it ok ? |
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| 14:00.25 | XATRIX | Ok, seems like i've made it. I don't know how, but it works now |
| 14:00.29 | XATRIX | thanks a lot! |
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| 14:06.11 | [TK]D-Fender | Still no debug for the call. |
| 14:06.17 | [TK]D-Fender | But if it's working.... |
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| 14:13.35 | *** mode/#asterisk [+o sruffell] by ChanServ |
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| 14:33.31 | RadJackson | Hi , A call to B , and C is spying on the call, when the A-B conversation is over how to catch C in a context ? |
| 14:33.34 | RadJackson | i use ChanSpy |
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| 14:44.24 | [TK]D-Fender | You don't catch in a context. If anything it'll continue on the next priority where it left off. |
| 14:44.54 | RadJackson | Thats wy i told myself too . but no , when A or B hangs up , C remains Chanspying , btw , I'm doing ChanSpy(C) |
| 14:45.42 | RadJackson | its logic , you can keep spying an agent , even if he isnt on a bridged call. Any idea how to spy the call so when A or B hangs up , C leaves the ChanSpy ? |
| 14:45.51 | [TK]D-Fender | "E: Exit when the spied-on channel hangs up." <-------------- |
| 14:46.32 | RadJackson | i will check that |
| 14:46.40 | *** join/#asterisk sgriepentrog (~sgriepent@nat/digium/x-eingzaavqpfjaqpc) |
| 14:48.40 | RadJackson | it just doesnt do it ... http://pastebin.com/1y7CR0BM , is there a way to exit chanspy when the spied on channel isnt on a bridged call ? |
| 14:49.52 | [TK]D-Fender | "b: Only spy on channels involved in a bridged call." |
| 14:50.53 | RadJackson | any documentation , I cant see that nowhere |
| 14:52.42 | [TK]D-Fender | "core show application chanspy" <- you don't seem to be looking in the right place |
| 14:53.08 | file | and https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_ChanSpy?src=search |
| 14:57.59 | RadJackson | Its not working , indeed it spies on a agent connected to a bridged call, but when the bridged call is over , i am still connected to the chanspy , it just doesnt move to the next step |
| 14:58.00 | RadJackson | http://pastebin.com/7wfaXPY7 |
| 15:00.37 | [TK]D-Fender | We don't see the call.... |
| 15:01.09 | [TK]D-Fender | And you didn't tell it to exit |
| 15:01.22 | [TK]D-Fender | exten => s,n,ChanSpy(${agent},bqX(9)) <-- no exit |
| 15:01.23 | RadJackson | the variable ${agent} contents SIP/606 , which is an agent connected to a call |
| 15:01.36 | [TK]D-Fender | [10:45][TK]D-Fender"E: Exit when the spied-on channel hangs up." <-------------- |
| 15:01.37 | RadJackson | i tested with the E too :( |
| 15:01.37 | RadJackson | 1s |
| 15:01.46 | [TK]D-Fender | You say "tested", but we don't see it |
| 15:01.57 | [TK]D-Fender | Show the actual call with actual code |
| 15:02.54 | RadJackson | http://pastebin.com/5YDEY5B7 |
| 15:04.08 | RadJackson | with this code i can spy on ${agent} being on a call , but when ${agent} hangs up or his recipient , in all cases i am still spying.. |
| 15:04.09 | [TK]D-Fender | this time you left off the BRIDGE ONLY option |
| 15:04.19 | [TK]D-Fender | exten => s,n,ChanSpy(${agent},EqX(9)) <--------- no BRIDGE-ONLY |
| 15:07.12 | RadJackson | http://pastebin.com/afJJSXnd still not working :( sorry |
| 15:08.20 | [TK]D-Fender | exten => s,n,ChanSpy(${agent},EqX(9)) <-------------- STILL NOT THERE |
| 15:10.56 | RadJackson | now should i put the b Option or not ? , i tried'em all ... bEqX , EqX , bqX , you mean this => http://pastebin.com/5YDEY5B7 ? tried it aswell :\ |
| 15:11.12 | [TK]D-Fender | BOTH |
| 15:11.34 | [TK]D-Fender | You keep saying, we aren't seeing. |
| 15:11.42 | [TK]D-Fender | Show us an actual call. |
| 15:13.00 | RadJackson | http://pastebin.com/dPt6XAE1 |
| 15:13.01 | RadJackson | there is it |
| 15:13.12 | RadJackson | The Call code is on the voyeur_boot context. |
| 15:13.23 | RadJackson | The spier code is on the voyeur_thing context |
| 15:14.39 | [TK]D-Fender | hits capacity and aborts |
| 15:14.43 | [TK]D-Fender | Perhaps someone else will pick this up from here |
| 15:15.20 | RadJackson | its ok , i would be happy to help |
| 15:15.23 | RadJackson | ^^ |
| 15:15.57 | RadJackson | so any idea why the chanspy is stuck even when the bridged call is over ? |
| 15:21.06 | *** join/#asterisk rmudgett (~rmudgett@nat/digium/x-mbimkpftxyzyzxqm) |
| 15:34.28 | *** join/#asterisk hurdman (~ygcheny@cylon.r0b0t.fr) |
| 15:38.12 | hurdman | hello , i have two asterisk server, same sip conf ( of course different IP ) on the same networks. On this asterisk server, i have got two linphone, one per asterisk, with same friend sip conf. When i make a call (allowguest=yes ) from one linphone to for exemple , the default extension "1" that call the second linphone (the own that's register on this asterisk ), it seems that asterisk change the IP of my account (sip show peers) from the IP of the linp |
| 15:40.37 | *** part/#asterisk LiuYan (~LiuYan@unaffiliated/liuyan) |
| 15:54.16 | BakaKuna | Hi, I am trying to match an extension by maximum amount of characters. We now have _1[48]!, but need to match something like _1[48]XX. Where the last X can be there, but does not have to be. So that both 144 and 1444 would be matched, but not 14444. I must be missing something but I can't find it. |
| 15:55.53 | BakaKuna | So i have . which matches one or more and ! which matches zero or more, but i thing I'm looking for something that matches zero or one. |
| 15:55.54 | [TK]D-Fender | BakaKuna: You need multiple patterns |
| 15:56.47 | [TK]D-Fender | BakaKuna: there is no "maximum" option. You're looking for something that doesn't exist which is why you weren't finding it. |
| 15:57.00 | *** join/#asterisk raub (~raub@ip70-171-13-167.ga.at.cox.net) |
| 15:57.31 | BakaKuna | [TK]D-Fender: Thnxs, I wanted to know that before trying other approaches. |
| 15:58.01 | [TK]D-Fender | BakaKuna: Read the book. Patterns are well documented. If you don't see it there ... it doesn't exist/. |
| 15:59.04 | BakaKuna | [TK]D-Fender: I have it in front of me :), never hurts to check if my reading skills are good enough to comprehend what I'm actually reading. |
| 16:06.24 | *** join/#asterisk lbazan (~LoKoMurdo@fedora/LoKoMurdoK) |
| 16:06.34 | lbazan | morning! |
| 16:07.08 | lbazan | I have issue with the flash operator panel |
| 16:07.40 | lbazan | I'm in the right place to ask about this? |
| 16:11.29 | [TK]D-Fender | We don't directly support it here, but someone night have the answer you're looking for if you're lucky. |
| 16:12.35 | *** join/#asterisk protocoldoug (~quassel@unaffiliated/protocoldoug) |
| 16:15.53 | *** join/#asterisk MarcoZink (~marcozink@187-177-157-192.dynamic.axtel.net) |
| 16:18.52 | *** join/#asterisk raub (~raub@ip70-171-13-167.ga.at.cox.net) |
| 16:28.50 | *** join/#asterisk timahvo1 (~rogue@197.237.134.227) |
| 16:33.31 | *** join/#asterisk cmendes0101| (~cmendes01@pool-96-251-59-96.lsanca.fios.verizon.net) |
| 16:36.51 | *** join/#asterisk igcewieling1 (~ewieling@ip98-183-26-100.pn.at.cox.net) |
| 16:54.32 | *** join/#asterisk wonderworld (~ww@ip-62-143-157-238.unitymediagroup.de) |
| 16:55.01 | *** join/#asterisk bkruse (~Adium@24.42.207.11) |
| 16:57.02 | *** join/#asterisk FreezingCold (~FreezingC@135.0.41.14) |
| 17:06.18 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 17:27.03 | *** join/#asterisk ChannelZ-Wk (~bobm@spark.idolum.com) |
| 17:35.58 | c|oneman | I wonder if people from digium switchvox hide in this channel |
| 17:39.05 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 17:44.19 | mjordan | I wouldn't say I hide |
| 17:45.00 | mjordan | This may be a surprise, but if you work at Digium, and you work on Asterisk, there is a *good* chance that at some point in time, on some day, in some fashion, you may be asked to at least look at SwitchVox. Shocking, I know. |
| 17:46.38 | file | is mildly electrocuted |
| 17:48.39 | mjordan | file: the shocks will continue until morale improves |
| 17:50.29 | file | or until the potato runs out of juice |
| 17:50.47 | ChannelZ-Wk | <PROTECTED> |
| 17:51.11 | mjordan | now with more electricity! |
| 18:08.49 | *** join/#asterisk mgob (~mgob@ip70-161-141-168.hr.hr.cox.net) |
| 18:12.16 | mgob | Has anyone else run into this 'bug'[unconfirmed]? We seem to be hitting it pretty hard https://issues.asterisk.org/jira/browse/ASTERISK-23998 - Asterisk 11.11.0 / FreePBX 2.11 - Any queue agent that transfers to another agent or voicemail results in the long queue issue. |
| 18:15.32 | *** join/#asterisk coppice (~chatzilla@123203240102.ctinets.com) |
| 18:18.38 | *** join/#asterisk lnb (~lnb@CPE4c5e0c417c51-CM602ad06bec2f.cpe.net.cable.rogers.com) |
| 18:19.50 | lnb | new 11.4 server, minimal configuration, sip debug on, don't see anything, tcpdump shows register from softphone but no answer back from asterisk server. |
| 18:20.13 | lnb | why would nothing show in cli ? |
| 18:20.18 | [TK]D-Fender | Porve * is listening |
| 18:20.20 | [TK]D-Fender | prove |
| 18:20.29 | lnb | you want to see the tcpdump |
| 18:20.33 | [TK]D-Fender | no. |
| 18:20.36 | lnb | then what? |
| 18:20.39 | [TK]D-Fender | I want proof that * is listening |
| 18:20.50 | lnb | how to show that? |
| 18:22.45 | [TK]D-Fender | netstat -an|grep udp |
| 18:23.17 | lnb | udp 0 0 0.0.0.0:5060 0.0.0.0:* |
| 18:24.28 | lnb | why is it not logging or showing logging in cli ? |
| 18:24.39 | lnb | i see logging only in tcpdump |
| 18:25.19 | [TK]D-Fender | netstat -anp|grep udp |
| 18:25.24 | [TK]D-Fender | verify that it is * |
| 18:25.39 | lnb | the only logging in /var/log/asterisk/messages is [Jul 15 16:40:03] Asterisk 11.11.0 built by root @ sip2.internal on a x86_64 running Linux on 2014-07-15 16:27:41 UTC |
| 18:25.48 | [TK]D-Fender | looging = trash |
| 18:25.50 | [TK]D-Fender | live CLI only |
| 18:26.07 | lnb | sip set debug on |
| 18:26.11 | lnb | core set verbose 10 |
| 18:26.13 | lnb | and nothing |
| 18:26.25 | [TK]D-Fender | show us. |
| 18:26.32 | [TK]D-Fender | along that chan_sip is loading |
| 18:26.34 | lnb | udp 0 0 0.0.0.0:5060 0.0.0.0:* 3414/asterisk |
| 18:26.49 | [TK]D-Fender | that you have any peers configured, etc |
| 18:27.10 | lnb | sip2*CLI> core set verbose 10 |
| 18:27.10 | lnb | Console verbose was 5 and is now 10. |
| 18:27.10 | lnb | sip2*CLI> |
| 18:27.14 | lnb | thats it |
| 18:27.16 | lnb | nothing |
| 18:27.21 | [TK]D-Fender | PB it all.... |
| 18:27.51 | lbazan | [TK]D-Fender: tks! |
| 18:27.58 | lnb | sip2*CLI> module show like sip |
| 18:27.58 | lnb | Module Description Use Count |
| 18:27.58 | lnb | app_adsiprog.so Asterisk ADSI Programming Application 0 |
| 18:27.58 | lnb | chan_sip.so Session Initiation Protocol (SIP) 0 |
| 18:27.58 | lnb | 2 modules loaded |
| 18:28.07 | pabelanger | ~pb |
| 18:28.07 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 18:28.11 | pabelanger | lnb, ^ |
| 18:28.14 | lnb | ok |
| 18:28.25 | lbazan | When you enter in the flash operator panel the DAHDI Trunks section no have activity |
| 18:28.31 | pabelanger | get everything into one document |
| 18:36.05 | lnb | http://www.pastebin.ca/2821867 |
| 18:36.23 | lnb | its a very basic new asterisk 11.11 from source |
| 18:37.38 | lnb | i was following along asterisk site to setup and test out hello-world |
| 18:37.46 | lnb | but i dont see anything in cli> |
| 18:37.51 | lnb | only in tcpdump output |
| 18:42.39 | [TK]D-Fender | Show us your looging to CLI and issuing the debug commands, and dumping your peers like I asked |
| 18:42.53 | lnb | but there is nothing in cli |
| 18:44.23 | lnb | refresh pb post |
| 18:44.29 | lnb | thats what you want? |
| 18:46.46 | lnb | enabled core debug |
| 18:47.08 | lnb | on windows box, started zoiper again ... nothing in cli> |
| 18:47.14 | lnb | refresh pb post |
| 18:49.57 | lnb | http://imagebin.ca/v/1TXqyr2BbrVe |
| 18:53.09 | *** join/#asterisk sgriepentrog (~sgriepent@nat/digium/x-lapbfhlfqetwycrj) |
| 18:56.11 | igcewieling1 | LNB: How many years have you been trying to set up Asterisk? |
| 18:58.43 | Nugget | Like most of us, as soon as LNB got tt-monkeys playing he lost a lot of his momentum. |
| 19:02.00 | *** join/#asterisk nuken (~nuken@open.integrada.coop.br) |
| 19:02.04 | nuken | hi all |
| 19:02.26 | nuken | i'm trying to configure a Planet VIP-480FO with my freepbx |
| 19:03.01 | nuken | i'm able toreceive calls in my asterisk |
| 19:03.12 | igcewieling1 | Nugget: *nod* |
| 19:03.18 | nuken | but when i try to make a call, circuits-are-busy |
| 19:03.42 | nuken | i'm trying to register the Planet gateway in the asterisk, but it not works |
| 19:03.53 | pabelanger | nuken, #freepbx might be a better place for help |
| 19:03.54 | igcewieling1 | Nugget: I miss the old days when we just hit them with a copy of ATFoT and then ignore them. |
| 19:04.27 | nuken | pabelanger, ok i will try there, thank you |
| 19:05.53 | [TK]D-Fender | [14:42][TK]D-FenderShow us your looging to CLI and issuing the debug commands, and dumping your peers like I asked |
| 19:06.33 | lnb | <PROTECTED> |
| 19:06.33 | lnb | Name/username Host Dyn Forcerport Comedia ACL Port Status Description |
| 19:06.33 | lnb | 6001 (Unspecified) D Auto (No) No 0 Unmonitored |
| 19:06.33 | lnb | 1 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 1 offline] |
| 19:06.51 | igcewieling1 | Best of luck [TK]D-Fender, I'm outta here. |
| 19:06.55 | *** part/#asterisk igcewieling1 (~ewieling@ip98-183-26-100.pn.at.cox.net) |
| 19:08.59 | lnb | http://pastebin.com/Ab55PHK1 |
| 19:09.55 | lnb | zoiper trying to register but nothing in cli, therefore nothing to pb |
| 19:10.05 | [TK]D-Fender | "sip show settings" |
| 19:11.02 | lnb | http://pastebin.com/BS9zRicm |
| 19:12.14 | [TK]D-Fender | Dump your firewall |
| 19:13.19 | lnb | http://pastebin.com/Usi9zRL8 |
| 19:14.40 | lnb | those rules are default centos 7 |
| 19:15.05 | [TK]D-Fender | "default" means nothing to me |
| 19:15.10 | [TK]D-Fender | trash it |
| 19:15.31 | lnb | i am going to put rules i use in all other live freepbx servers we have |
| 19:17.23 | lnb | default in this case means whatever centos 7 x64 puts there during install |
| 19:17.45 | lnb | like the nic prior to 7 was eth0 now its enp0s3 |
| 19:19.04 | [TK]D-Fender | trash it |
| 19:20.59 | *** join/#asterisk phyrexianslug (~phyrexian@S0106000c29543ac5.wp.shawcable.net) |
| 19:21.08 | *** join/#asterisk BakaKuna (~Thunderbi@82-169-251-128.ip.telfort.nl) |
| 19:22.47 | lnb | great no such thing as service iptables |
| 19:22.56 | lnb | ps auxw |grep iptables shows the grep |
| 19:25.52 | phyrexianslug | ... iptables isn't a service..? |
| 19:26.08 | phyrexianslug | It's an application that makes changes to the iptables configuration |
| 19:26.14 | *** join/#asterisk bkruse (~Adium@24.42.207.11) |
| 19:26.22 | phyrexianslug | what are you trying to do? (sorry, I may have missed something?) |
| 19:26.42 | lnb | # service iptables stop |
| 19:26.42 | lnb | Redirecting to /bin/systemctl stop iptables.service |
| 19:26.42 | lnb | Failed to issue method call: Unit iptables.service not loaded. |
| 19:26.56 | lnb | this is centos 7 |
| 19:27.08 | lnb | i want to stop iptables period |
| 19:27.36 | phyrexianslug | http://www.cyberciti.biz/tips/linux-iptables-how-to-flush-all-rules.html |
| 19:28.02 | *** join/#asterisk navaismo (~navaismo@189.241.81.92) |
| 19:28.03 | phyrexianslug | You can't "stop" ip tables, it's just a set of rules that determine what the kernel does with packets. |
| 19:28.41 | lnb | -F |
| 19:28.48 | lnb | now i cannot get in either :) |
| 19:29.05 | phyrexianslug | iptables -P INPUT ACCEPT first. :P |
| 19:29.28 | phyrexianslug | if the default rule is deny, than flushing sets an inherit deny all. |
| 19:36.29 | lnb | this is a freeking nightmare |
| 19:36.42 | lnb | why did they make this changes |
| 19:36.44 | lnb | wtf |
| 19:39.33 | lnb | forget this nonesense install ubuntu |
| 19:40.06 | [TK]D-Fender | "they"? |
| 19:40.08 | [TK]D-Fender | "changes"? |
| 19:40.41 | lnb | garbage destroyed poof |
| 19:40.55 | lnb | they whoemevr wrote centos 7 |
| 19:41.08 | lnb | changes: there is no more 'iptables' commands |
| 19:41.12 | lnb | it has changed |
| 19:41.24 | lnb | install centos 7 and see for yourself |
| 19:41.34 | lnb | the vm is deleted poof garbage gone |
| 19:41.36 | [TK]D-Fender | Your description is garbage. |
| 19:41.42 | lnb | no its not |
| 19:41.47 | lnb | i am just really pissed off |
| 19:41.52 | [TK]D-Fender | it is |
| 19:41.56 | [TK]D-Fender | And this is consistent |
| 19:42.01 | lnb | ]you should see me in a car when i am like this |
| 19:43.21 | lnb | by the way, i started the asterisk install using https://wiki.asterisk.org/wiki/display/AST/Getting+Started |
| 19:46.29 | phyrexianslug | Wow, |
| 19:46.58 | phyrexianslug | Yep, Cent 7 totally revamped how IPtables worked? |
| 19:47.00 | phyrexianslug | http://stackoverflow.com/questions/24729024/centos-7-open-firewall-port |
| 19:48.04 | lnb | i dont have days on end to get this to work, and I followed the instructions closely |
| 19:48.28 | phyrexianslug | Eh, IMHO you're not missing anything with "anything besides centos". :P |
| 19:48.34 | lnb | instructions on the asterisk wiki i mean |
| 19:48.43 | lnb | huh>? |
| 19:49.02 | phyrexianslug | <- doesn't like centos. |
| 19:49.05 | lnb | oh you mean use something like ubuntu server |
| 19:49.13 | lnb | i see your point CLEARLY |
| 19:49.19 | lnb | especially with 7 |
| 19:49.21 | phyrexianslug | :P |
| 19:49.59 | lnb | i should perhaps just build a FreeBSD 10 vm |
| 19:50.23 | *** join/#asterisk Sean-Der (~Sean-Der@siobud.com) |
| 19:50.26 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 19:50.54 | lnb | before wiping centos 7, installed system-config-firewall-tui, ran it, it would not let me |
| 19:50.56 | Sean-Der | Has anyone ever seen a SIP over UDP DOS for the username 'Fady' before? |
| 19:50.59 | lnb | pile of rubbish |
| 19:51.02 | phyrexianslug | Hmmn, never done Asterisk on BSD |
| 19:51.13 | lnb | let me see if asterisk is in ports |
| 19:51.16 | lnb | one sec |
| 19:51.23 | phyrexianslug | Build from source on Debian, always works fine for me! |
| 19:51.31 | Sean-Der | It just seems so strange it was only one username, and it brought everything to a halt. From the IP 5.152.208.58 |
| 19:52.26 | lnb | Port:asterisk11-11.8.1_1 |
| 19:52.26 | lnb | Path:/usr/ports/net/asterisk11 |
| 19:52.26 | lnb | Info:An Open Source PBX and telephony toolkit |
| 19:52.26 | lnb | Maint:flo@FreeBSD.org |
| 19:53.18 | *** join/#asterisk D-Boy (~D-Boy@unaffiliated/cain) |
| 19:54.06 | lnb | dahdi and libpri are ports too |
| 19:54.53 | lnb | but installing those ports will not teach me anything like installing from src |
| 19:59.55 | phyrexianslug | At some point you'll want to upgrade or make a low-level change anyhow. |
| 20:00.07 | phyrexianslug | Installing from source for * is highly recommended! :D |
| 20:00.49 | lnb | well i was rereading the book, and it seems to imply one can have multi tenant with asterisk |
| 20:01.04 | lnb | the definitive guide book |
| 20:01.30 | lnb | i am only used to FreePBX and it does not support multi-tenant |
| 20:01.38 | phyrexianslug | Huh, Never got into multi-tennant on asterisk |
| 20:01.48 | phyrexianslug | Just fired up VM's for each customer. :P |
| 20:01.59 | lnb | the other i do use for it is fusionpbx which is on top of freeswitch |
| 20:02.24 | phyrexianslug | you doing PBX's or Service Provider + PBX? |
| 20:02.29 | lnb | i like asterisk and want to better know how to use it |
| 20:02.35 | lnb | pbx's |
| 20:02.45 | phyrexianslug | a "multi tennant" Private Branch eXchange is kinda silly, IMHO. :P |
| 20:02.46 | lnb | host them for clients |
| 20:03.17 | lnb | multi-tennant is good for places that have 2 or 3 people |
| 20:03.24 | phyrexianslug | That's not very private anymore then. :P |
| 20:03.31 | lnb | sure it is |
| 20:03.40 | lnb | everyone can have ext 100 |
| 20:03.54 | phyrexianslug | No, I mean the server isn't very private. :P |
| 20:03.55 | lnb | and one cannot call the other by ext # |
| 20:03.59 | lnb | why? |
| 20:04.03 | lnb | they cannot get into it |
| 20:04.07 | phyrexianslug | Yeah, |
| 20:04.11 | phyrexianslug | there's the problem |
| 20:04.14 | lnb | why? |
| 20:04.21 | lnb | they do NOT want to deal with it |
| 20:04.24 | phyrexianslug | PBX serves one business, |
| 20:04.36 | phyrexianslug | Asterisk does Exchanges fine, |
| 20:04.39 | lnb | have you ever installed and used fusionpbx ? |
| 20:04.50 | [TK]D-Fender | [16:03]lnbmulti-tennant is good for places that have 2 or 3 people <- No. |
| 20:04.56 | lnb | why |
| 20:04.58 | [TK]D-Fender | It for more than one TENNENT |
| 20:05.05 | lnb | no no |
| 20:05.06 | phyrexianslug | 90+% of the gui's don't deal with service provider stuff. |
| 20:05.08 | [TK]D-Fender | I can have ONE company with 5 PEOPLE working at it |
| 20:05.10 | lnb | what i mean is this |
| 20:05.20 | [TK]D-Fender | Again terrible descriptions |
| 20:05.37 | lnb | its not economically viable to build a VM for a client and install/setup freepbx for 1 or 2 users |
| 20:05.52 | lnb | its just not unless they want to pay for it |
| 20:05.55 | lnb | and they dont |
| 20:06.34 | lnb | if it was a user like some of the people here, they would want a VPS and do it all themselves |
| 20:06.37 | lnb | thats another story |
| 20:07.14 | lnb | but the clients I have want to press the email button, word button and excel button, do their work and go home |
| 20:07.16 | lnb | thats it |
| 20:07.18 | phyrexianslug | "its not economically viable to build a VM for a client and install/setup freepbx for 1 or 2 users" Funny, I do that ~5 times a week? |
| 20:07.46 | lnb | <lnb> if it was a user like some of the people here, they would want a VPS and do it all themselves |
| 20:07.48 | phyrexianslug | Are you saying "renting a VM for each"? |
| 20:07.54 | lnb | yes |
| 20:07.56 | phyrexianslug | Oh |
| 20:08.09 | phyrexianslug | there's the problem, we have racks of virtual servers. :P |
| 20:08.22 | lnb | i dont put more than 1 business in a FreePBX or asterisk vm server |
| 20:08.23 | phyrexianslug | cost per server is trivial. |
| 20:08.26 | lnb | really? |
| 20:08.33 | lnb | calculate bandwidth |
| 20:08.38 | lnb | and other charges |
| 20:08.45 | phyrexianslug | Bandwidth is free at an imbalance. |
| 20:08.48 | tm1000 | phyrexianslug: notice the whole âunless they want to pay for itâ âand they dontâ bit |
| 20:08.50 | lnb | i have cage at peer1 and its NOT cheap |
| 20:08.58 | tm1000 | pricing yourself out of the biz |
| 20:09.04 | lnb | hi tm1000 |
| 20:09.18 | lnb | well i am all ears if you have better solution |
| 20:09.37 | lnb | s/eyes/ears |
| 20:09.45 | lnb | err backwards |
| 20:09.47 | phyrexianslug | If you don't host anything yourself, it's likely to be expensive to you? |
| 20:09.49 | tm1000 | lnb: we do the same thing as phyrexianslug, we have many clients that do the same thing as phyrexianslug |
| 20:09.56 | tm1000 | it works, people make money, lots of money |
| 20:10.07 | lnb | phyrexianslug: i own all the servers... they are not rented |
| 20:10.23 | lnb | phyrexianslug: i pay for the cage and bandwidth and electricity |
| 20:10.25 | phyrexianslug | You need better ISP agreements then! :P |
| 20:10.33 | lnb | no kidding |
| 20:10.36 | file | naturally depends on location too |
| 20:10.38 | lnb | they are very expensive |
| 20:10.39 | tm1000 | we do everything through solus, we probably have 20ish asterisk servers all as vm clients |
| 20:10.50 | lnb | i use solusvm too |
| 20:10.57 | lnb | but its the cost of PEER1 |
| 20:11.09 | lnb | the cage to put 20 physical servers |
| 20:11.36 | lnb | i cant get 1gb up/down here at my home office |
| 20:11.43 | lnb | if i could i would |
| 20:12.39 | phyrexianslug | So get a dark line to an internet exchange, and chat with some telco's! |
| 20:12.39 | lnb | phyrexianslug: others have told me to get out of peer1 but i cannot find a place that will give me any better deals |
| 20:12.51 | lnb | at least here in Toronto canada (crackland) |
| 20:13.20 | tm1000 | phyrexianslug: a "multi tennant" Private Branch eXchange is kinda silly, IMHO. <â yet people complain about how freepbx doesnt have this ability all day long |
| 20:13.24 | phyrexianslug | lol, <- Winnipeg |
| 20:13.35 | tm1000 | but here is a vanilla user of asterisk saying even HE/SHE doesnt do MT |
| 20:13.40 | lnb | phyrexianslug: you know les.net? |
| 20:13.46 | phyrexianslug | Yep! |
| 20:13.54 | lnb | i got his name from [TK]D-Fender |
| 20:14.04 | lnb | they have been good to me |
| 20:14.06 | phyrexianslug | Les is awesome. Great dude. |
| 20:14.14 | lnb | indeed he is |
| 20:14.23 | phyrexianslug | I'm litterally ~100 ft from his building. >.> |
| 20:14.29 | lnb | really.. wow |
| 20:14.35 | phyrexianslug | Well, that's a lie, more like ~500ft |
| 20:14.36 | lnb | maybe you are les? |
| 20:14.43 | phyrexianslug | lol, Naw |
| 20:14.51 | lnb | never know |
| 20:14.52 | phyrexianslug | he's on here from time to time though! |
| 20:14.58 | lnb | he is a good man |
| 20:15.05 | lnb | i thank [TK]D-Fender for sending me there |
| 20:15.37 | phyrexianslug | How's Toronto? :D |
| 20:15.58 | lnb | compared to? |
| 20:16.12 | phyrexianslug | In general? |
| 20:16.16 | lnb | its cool for this time of year |
| 20:16.37 | phyrexianslug | Yeah, we just went through the coldest july week on record here. :P |
| 20:16.42 | lnb | which is good for al gore and his world overheating project |
| 20:16.47 | phyrexianslug | haha |
| 20:17.53 | lnb | tell me something, when you install do you tell it (i forgot right now what 'it' is) to use /usr/lib or /usr/lib64 ? |
| 20:17.58 | [TK]D-Fender | Canada exports Global Warming... |
| 20:18.05 | lnb | really |
| 20:18.54 | lnb | i checked temperatures in the extreme north of some countries, and like ~ -50 is not exactly skinny dipping weather |
| 20:19.06 | lnb | not today but in winter |
| 20:20.31 | phyrexianslug | Asterisk? |
| 20:20.41 | phyrexianslug | I generally do the build defaults these days. |
| 20:20.50 | phyrexianslug | Works fine 99+% of the time |
| 20:21.42 | lnb | tm1000: what is private branch exchange? |
| 20:21.52 | [TK]D-Fender | heads home... |
| 20:32.06 | marceloamorim | guys |
| 20:32.43 | marceloamorim | I didn't find the documentation about which number I need to set on the chan_dahdi.conf about tonezone for brazil |
| 20:33.19 | beardy | w/wg 69 |
| 20:43.31 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 20:46.36 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 20:50.21 | *** join/#asterisk wonderworld (~ww@ip-62-143-157-238.unitymediagroup.de) |
| 20:57.53 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 21:03.36 | marceloamorim | guys |
| 21:04.05 | marceloamorim | I think this tonezone substitute the loadzone, so I don't need to use |
| 21:04.28 | marceloamorim | after to figure this out, I move on to the next problem |
| 21:04.55 | marceloamorim | I need to change my dtmfcidlevel on the chan_dahdi, but I have some issues trying to do this |
| 21:06.16 | marceloamorim | I'm wondering how I suppose to get this energy trigger to stabilize to one fix number |
| 21:18.49 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 21:22.12 | bibz_away | how can I change the CallerID in the first invite i'm sending out? |
| 21:22.53 | phyrexianslug | How are you setting it in the first place? |
| 21:24.42 | bibz | i'm setting it somewhere in the dialplan |
| 21:24.51 | bibz | but before a Dial |
| 21:25.17 | bibz | I don't know why the first invite is without the modified CallerID.. |
| 21:25.28 | bibz | when I look into the debug,after a few functions it changes |
| 21:26.19 | phyrexianslug | pastebin of: config (likely just need extensions.conf / full log with verbose >10 / sip debug? |
| 21:27.30 | bibz | hold on just a sec |
| 21:27.45 | phyrexianslug | No rush, I'm here another hour or so! |
| 21:31.39 | bibz | but I warn you.. its a quite big diaplan |
| 21:32.16 | bibz | DEBUG > http://pastebin.com/BcMRAAbb |
| 21:32.49 | bibz | DIALPLAN > http://pastebin.com/gQR4BJSw |
| 21:35.22 | bibz | I've tried setting the callerID at the very top of the diaplan, but the first invite is still unmodified |
| 21:35.51 | phyrexianslug | And what's the call flow like? |
| 21:36.06 | phyrexianslug | Peer has what as the initial outgoing context? |
| 21:36.40 | phyrexianslug | "mw1-gw-outgoing-call |
| 21:36.42 | phyrexianslug | " |
| 21:36.42 | bibz | pbx3-out |
| 21:36.47 | bibz | is the context from the sip.conf |
| 21:38.02 | phyrexianslug | Don't see that context in the dump anywhere |
| 21:38.16 | bibz | I should mention, that the call gets initiated |
| 21:38.24 | [TK]D-Fender | turn off the core debug. that is excessive |
| 21:38.39 | phyrexianslug | LOOKS like it's dialing an account code first maybe? |
| 21:38.48 | bibz | but my provider requires me to send the CallerID in E164 format (436643029838 instead of +436643029838).. |
| 21:39.05 | bibz | thats why I'm getting so much debug infos.. :D |
| 21:39.08 | bibz | I'll do so |
| 21:40.14 | bibz | I just don't get it.. even when setting the CallerID in the first extension, it won't be modified in the INVITE |
| 21:41.11 | bibz | the problem is, that my calls are forwarded without a number, since the provider has a policy where the number must be in E164 format - as I already mentioned.. |
| 21:41.11 | bibz | :( |
| 21:41.41 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-yjsuzvtcyfxcbrlx) |
| 21:41.42 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 21:44.06 | phyrexianslug | bibz: need the context for the peer you're dialing from from sip.conf |
| 21:44.43 | *** join/#asterisk bkruse (~Adium@64.89.97.127) |
| 21:44.43 | phyrexianslug | outgoing looks strange in extensions, need to see where the entry point is |
| 21:45.43 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-28-58.cablep.bezeqint.net) |
| 21:45.57 | bibz | ok, hold on just a second. |
| 21:47.00 | bibz | http://pastebin.com/M91yB0jT |
| 21:48.25 | lnb | isnt the syntax supposed to be exten => 100,1,Answer() and not exten = 100,1,Answer() |
| 21:48.42 | [TK]D-Fender | both work |
| 21:48.50 | lnb | ok thanks |
| 21:48.58 | [TK]D-Fender | bitpastebin another call and kill the core debug |
| 21:49.05 | [TK]D-Fender | bibz: pastebin another call and kill the core debug |
| 21:49.31 | bibz | how is it defined in the logger.conf? |
| 21:49.37 | bibz | the core debug |
| 21:49.41 | [TK]D-Fender | forget logs |
| 21:49.44 | [TK]D-Fender | live * CLI only |
| 21:50.01 | bibz | oh. I was using the log created by logger.conf > full => notice,warning,error,debug,verbose,dtmf,fax |
| 21:50.16 | phyrexianslug | asterisk console |
| 21:50.24 | phyrexianslug | "core set verbose 10" |
| 21:50.30 | phyrexianslug | "core set debug off" |
| 21:51.25 | bibz | got it |
| 21:51.52 | bibz | http://pastebin.com/Bpm0UEJi |
| 21:52.03 | bibz | ok, this isn't the whole call. does it have to be a fully initated call? |
| 21:52.21 | *** join/#asterisk MasterSenpai9654 (9f8cfe6b@gateway/web/freenode/ip.159.140.254.107) |
| 21:52.46 | [TK]D-Fender | you didn't reach a DIAL in there |
| 21:52.58 | [TK]D-Fender | [17:50]phyrexianslug"core set verbose 10" <---- |
| 21:53.04 | [TK]D-Fender | you kill verbose as well |
| 21:53.08 | [TK]D-Fender | killed* |
| 21:53.26 | bibz | when starting up the CLI with -rvvvvvvv, doesn't it increase the verbose level? |
| 21:53.33 | bibz | I'll do the core set verbose, hold on another second |
| 21:54.34 | MasterSenpai9654 | I think you can only increase verbose up to 3 v's |
| 21:54.54 | phyrexianslug | asterisk -rvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvv |
| 21:55.02 | phyrexianslug | Nope, seems to give me 40. :P |
| 21:55.41 | phyrexianslug | Depends on the version I think, some allow more at the console, some don't/ |
| 21:56.19 | bibz | ok, here is the new live debug of a call |
| 21:56.19 | bibz | http://pastebin.com/ri5jX9FA |
| 21:56.40 | MasterSenpai9654 | there should be a asterisk -rvover9000! |
| 21:56.46 | MasterSenpai9654 | lol |
| 21:57.08 | bibz | "Console verbose was OFF and is now >9000" |
| 21:57.14 | bibz | :P |
| 21:58.21 | bibz | should I try setting the callerid before EVERY Dial in the extensions.conf? |
| 21:58.37 | bibz | but I don't think that would solve the problem, since putting it on top won't... |
| 21:59.04 | lnb | on new 11.4 asterisk install, tried to subscribe but cli displays error because peer has no mailbox |
| 21:59.18 | lnb | i did run make samples |
| 21:59.38 | MasterSenpai9654 | it looks like in the pastebin it tries to access to a database, but fails to |
| 21:59.39 | lnb | i presume one of those same conf files says the peer has to have a mailbox |
| 22:00.11 | bibz | it fails on some functions.. but that shouldn't affect the callerID.. |
| 22:00.38 | bibz | I even tried commenting out everything that has CallerID in it.. like some database INSERTs.. |
| 22:00.57 | bibz | thats not even logical, I know.. but I'm already for too many hours on this problem |
| 22:02.08 | bibz | any ideas? |
| 22:02.58 | *** join/#asterisk stasdizzi (~stasdizzi@159.224.69.125) |
| 22:03.51 | [TK]D-Fender | <PROTECTED> |
| 22:03.53 | [TK]D-Fender | <PROTECTED> |
| 22:04.07 | [TK]D-Fender | From: "36644762522" <sip:436644762522@node1.ipaustria.at>;tag=as4aad651b |
| 22:04.10 | [TK]D-Fender | it's set |
| 22:05.31 | phyrexianslug | you're calling from 06644762522 -> to 431235078922 ? |
| 22:05.51 | [TK]D-Fender | <PROTECTED> |
| 22:06.02 | [TK]D-Fender | INVITE sip:+4369919542270@node1.ipaustria.at SIP/2.0 |
| 22:07.03 | lnb | 6001/6001 192.168.10.3 D Auto (No) No 32556 Unmonitored |
| 22:07.16 | lnb | a lot better with centos 6.5 |
| 22:07.31 | [TK]D-Fender | [17:59]lnbi presume one of those same conf files says the peer has to have a mailbox <- no.. that's ASSume. |
| 22:08.07 | lnb | what i did was # rsync -az /etc/asterisk/ /etc/asterisk_orig/ |
| 22:08.15 | lnb | and just cp back the 4 files |
| 22:08.20 | lnb | then it registered |
| 22:08.31 | lnb | well restarted asterisk as well of course |
| 22:09.06 | lnb | this is like starting from scratch as I've never installed/setup asterisk from source |
| 22:09.35 | lnb | centos 7 x64 could not get anything from cli |
| 22:10.12 | [TK]D-Fender | Centos has nothing to do with it |
| 22:10.17 | [TK]D-Fender | configuring your system is your job |
| 22:10.25 | phyrexianslug | bibz: ? |
| 22:11.06 | lnb | umm don't know what to tell you. perhaps you should install centos 7 x64 and install asterisk according to the wiki.asterisk.org and see what happens |
| 22:11.11 | *** join/#asterisk hursjohn (~hursjohn@206.51.86.2) |
| 22:11.24 | lnb | one thing for sure, this is working |
| 22:11.38 | lnb | centos 6.5 x64 + asterisk 11.4 |
| 22:12.07 | MasterSenpai9654 | I am wondering if anyone as connected an Asterisk server to Freeswitch using Freeswitch's dialplan? There was a documentation on Freeswitch's site to set that up, but it doesn't get to the extent of configuring in bound\out bound routes on FreePBX Admin to the FreeSwitch. |
| 22:12.34 | [TK]D-Fender | FreePBX is another question altogether |
| 22:12.57 | bibz | oh, excuse me, I was away |
| 22:13.23 | phyrexianslug | lnb where did you find cent7 on asterisk wiki? |
| 22:13.27 | bibz | wait, it is set? |
| 22:13.36 | lnb | phyrexianslug: i didn't |
| 22:13.39 | phyrexianslug | bibz: you're calling from 06644762522 -> to 431235078922 ? |
| 22:13.40 | bibz | hold on a second.. why is the CALLERID(name) 1 character short |
| 22:13.43 | bibz | right |
| 22:14.01 | lnb | phyrexianslug: i did not know ahead of time what the nightmare(s) might be |
| 22:14.58 | bibz | seems like I'm missing a character, thats why I doesn't get passed right.. |
| 22:15.11 | phyrexianslug | That would make sense? |
| 22:15.44 | [TK]D-Fender | exten => _+.,n,Set(CALLERID(num)=${CALLERID(num):1}) <-- You are chopping one off here. |
| 22:16.10 | [TK]D-Fender | bibz: You should lok at your code and what you're actually doing. |
| 22:16.29 | bibz | what the heck |
| 22:16.50 | [TK]D-Fender | exten => _+.,n,Set(CALLERID(name)=${CALLERID(num):}) <- just broken |
| 22:16.55 | bibz | ok, before chopping that one off, the "From:"number was for example: 069938928323 |
| 22:17.07 | [TK]D-Fender | Go look at your code |
| 22:17.10 | [TK]D-Fender | it is setting what you are telling it to do. |
| 22:17.10 | bibz | so I chopped one off and added "43" infront of it |
| 22:17.23 | bibz | seems like it got lost in the dialplan the "43" |
| 22:17.55 | bibz | but even then, why is there 1 character chopped of the callerid(num) and 2 characters off callerid(name).. |
| 22:18.36 | lnb | now to make this a bit more usable |
| 22:19.32 | bibz | oh, god. I don't need to modify the callerid(num), only the callerid(name).. |
| 22:19.50 | phyrexianslug | *cheer* |
| 22:27.46 | phyrexianslug | bibz: This call, it's someone internally calling out to a number at a provider? Or someone calling in FROM the provider? |
| 22:28.29 | phyrexianslug | I don't understand why the call is going to "public-in", default in sip.conf? |
| 22:29.50 | bibz | its going out to a mobile |
| 22:29.54 | bibz | of a employer |
| 22:30.09 | bibz | so when no-one answers at the office, its gets forwarded (outside) to a mobile number |
| 22:44.50 | *** join/#asterisk AlHafoudh (~AlHafoudh@echo.freevision.sk) |
| 22:51.20 | phyrexianslug | So you're calling INTO the system and trying to fake the system calling out so the mobile station sees the caller ID of the person who called in initially? |
| 22:56.29 | bibz | right |
| 22:57.13 | bibz | btw, thanks a lot for the help to everyone |
| 23:01.15 | *** join/#asterisk Draecos (~Draecos@101.112.20.73) |
| 23:02.09 | *** join/#asterisk lnb (~lnb@CPE4c5e0c417c51-CM602ad06bec2f.cpe.net.cable.rogers.com) |
| 23:21.15 | *** join/#asterisk bmurt (~brendan@208-58-116-232.c3-0.upd-ubr1.trpr-upd.pa.cable.rcn.com) |
| 23:26.01 | *** join/#asterisk elguero (~miguel323@2001:470:1f06:12c4::2) |
| 23:32.08 | *** join/#asterisk ddemarteleire (ad0a66b2@gateway/web/freenode/ip.173.10.102.178) |
| 23:33.07 | ddemarteleire | What is the best way to provide people a way to call out using the main number instead of their DID. A second line? |
| 23:35.37 | *** join/#asterisk MaliutaLap (~nobusines@unaffiliated/maliuta) |
| 23:39.04 | carrar | add digit or two digits in front of the number they are dialing? |
| 23:39.31 | carrar | that seems simple |
| 23:39.49 | WIMPy | Does it? |
| 23:40.04 | carrar | sure |
| 23:40.18 | WIMPy | Have you tried it? |
| 23:40.21 | carrar | yes |
| 23:40.35 | carrar | have a few customers using that actually |
| 23:40.44 | WIMPy | And how do you select your outgoing identiti when uding your phone book? |
| 23:40.51 | WIMPy | y |
| 23:40.58 | carrar | you don't |
| 23:40.59 | carrar | or |
| 23:41.06 | carrar | add that 12 digit number in your phone book |
| 23:41.23 | carrar | or just manually dial it |
| 23:41.32 | WIMPy | Yes. Very simple. |
| 23:41.35 | carrar | sure |
| 23:41.39 | carrar | seems like it |
| 23:41.52 | carrar | WHats simplier? |
| 23:42.23 | WIMPy | Two accounts? |
| 23:42.41 | carrar | What if they have 20 different caller ID numbers they may want to use? |
| 23:43.23 | WIMPy | I guess they have to do it your way then. |
| 23:43.23 | carrar | granted in this case they don't |
| 23:43.53 | carrar | yeah a extra line would be simpler for the user if just one extra caller id |
| 23:44.05 | carrar | more work for the admin |
| 23:44.28 | WIMPy | Well... |
| 23:45.08 | WIMPy | Unless you have a free admin, using a VOIP system might not be a good idea anyway. |
| 23:45.18 | carrar | heh |
| 23:52.09 | ddemarteleire | So set up an outbound route, and for the dial pattern, put a prefix, and override extension CID? |
| 23:52.41 | ddemarteleire | Oh no, wait only 1 route per trunk |
| 23:56.37 | *** join/#asterisk Draecos (~Draecos@101.112.20.73) |