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| 03:30.53 | britto | Hi |
| 03:34.52 | britto | Some one uses asterisk-java |
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| 06:44.24 | dermont | Hi, can anyone tell me how to dial outbound with PJSIP on extensions.conf? |
| 06:44.30 | dermont | in SIP it was like Dial(SIP/xxx@trunk) |
| 06:44.34 | dermont | can't seem to make it work in PJSIP |
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| 07:10.44 | ChannelZ | pretty much the same... PJSIP/number@place |
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| 07:21.35 | r00f | i am having strange errors sometimes... asterisk decides that peer is congested without showing any acutal error or even attempt to call. how can it decide that peer is congested without trying it? |
| 07:21.47 | *** join/#asterisk cw1972 (~cw1972@host81-136-221-45.in-addr.btopenworld.com) |
| 07:21.52 | r00f | there are no call limits configured on that peer |
| 07:22.27 | r00f | [Jul 9 09:29:31] VERBOSE[4045][C-00001720] app_dial.c: -- Called SIP/voxbeam/0011101XXXXXXXXXXXXX |
| 07:22.27 | r00f | [Jul 9 09:29:31] VERBOSE[4045][C-00001720] app_dial.c: == Everyone is busy/congested at this time (1:0/0/1) |
| 07:24.48 | dermont | I tried that ChannelZ |
| 07:24.49 | dermont | I get |
| 07:25.09 | dermont | ERROR[6894]: chan_pjsip.c:1818 request: Unable to create PJSIP channel - endpoint 'trunk' was not found |
| 07:26.02 | ChannelZ | r00f, I'm sure the remote end is returning that. Maybe they don't like the number dialed |
| 07:26.40 | r00f | ChannelZ, but shouldn't it show something like "Git SIP reply "480 Not Available"" or something like that? |
| 07:26.56 | r00f | asterisk usually shows this |
| 07:27.06 | ChannelZ | Not particularly, no. You can look at the SIP debug if you want to see exactly whats going on. |
| 07:27.37 | r00f | oh. i thought it always shows |
| 07:27.43 | r00f | okay, will debug then |
| 07:27.49 | ChannelZ | Maybe it does at a much higher verbose level or something, I'm not sure |
| 07:27.49 | r00f | thanks |
| 07:27.59 | r00f | Got SIP response 486 "Busy here" back from |
| 07:28.24 | r00f | that's what i mean. i see these messages after failed calls - but not after failed calls i'm asking right now |
| 07:28.29 | ChannelZ | dermont, does [trunk] exist as a type=endpoint in your pjsip config? |
| 07:28.47 | dermont | pjsip.conf? |
| 07:28.53 | dermont | yes |
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| 08:18.03 | r00f | ChannelZ, after some debugging you seem to be correct. asterisk acts like that when it gets 404 reply. for some reason, it prefers not to show it in console, while it shows all other sip replies like 403, 480 or 503 |
| 08:18.15 | r00f | idk why was it decided to design sip messages like that |
| 08:18.18 | r00f | maybe it makes sense |
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| 08:34.14 | Guest9930 | Hi, a general SIP question(in an asterisk system though). How can I see on a RE-INVITE that the User agent is being put in hold? |
| 08:51.25 | marlinc | Would it be possible to tell a agent in a queue information about the phone number being called? |
| 08:51.47 | marlinc | For example through a voice message before the call gets transferred |
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| 09:01.51 | doome__ | maybe you could do in a macro, given to the queue application |
| 09:02.12 | doome__ | (see Queue app documentation) |
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| 10:00.31 | marlinc | Is it possible to return the value of a variable and when that value is empty return another? |
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| 10:30.30 | kritzikratzi | hi! iâm trying to play a sound file, and keep playing it until a valid digit (eg 2 and 5 are allowed) is pressed. |
| 10:30.53 | kritzikratzi | this works nicely with the background command and catching single digit extensions like so: |
| 10:31.03 | kritzikratzi | exten => 1,1,Playback(custom/you-pressed-one) |
| 10:31.30 | kritzikratzi | is it possible to also catch the star in a similar way? i tried |
| 10:31.43 | kritzikratzi | exten => *,1,Playback(custom/you-pressed-asterisk) |
| 10:31.46 | kritzikratzi | but it doesnât work |
| 10:31.47 | kritzikratzi | any ideas? |
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| 10:51.06 | racho | hello. running Asterisk 11.11.0 and using websockets and webrtc during the calling phase i can hear the dialing/ringing audio but as soon as one side picks up no auido is passing through |
| 10:51.38 | racho | the error message is that asterisk can't resolve the contact addr -> 'xxxx.invalid' |
| 10:52.19 | racho | both the clients and server are in a LAN with nat=no |
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| 11:02.11 | ChannelZ | kritzikratzi, it should work, but if it doesn't respond right away you might have some other extension in that context using a pattern that it could possibly match, so it's waiting for more digits |
| 11:03.13 | kritzikratzi | ChannelZ: yea, i realized. iâm now thinking about making my own copy of the read command |
| 11:03.19 | kritzikratzi | if iâm correct i can just go in here: https://github.com/bklang/asterisk/blob/trunk/apps/app_read.c#L221 |
| 11:03.48 | kritzikratzi | iâm thinking about getting rid of the maxdigits arg, and using a string of allowed digits instead |
| 11:04.04 | kritzikratzi | if the char that was read is on the list, then it would cancel, if not it would continue playback |
| 11:04.32 | WIMPy | What's wrong with doing it the background way? |
| 11:04.36 | kritzikratzi | i played around with the background command some more, but the waiting period was too long for me |
| 11:05.18 | WIMPy | If you have to wait, clean up your dialplan. |
| 11:05.58 | kritzikratzi | ah, that speeds things up? kk, then iâll have to make a new context. currently i have a lot of tests in a single context under different extensions. |
| 11:06.04 | kritzikratzi | let me give it a new try :) |
| 11:06.30 | WIMPy | Just als ChannelZ told you. |
| 11:11.31 | kritzikratzi | WIMPy: i didnât quite understand the implications of what he said i guess (pretty new to asterisk still). |
| 11:12.08 | kritzikratzi | anyways, i created a new context and it works beautifully! plus i can catch # too (which gave me a bit of a headache with read). |
| 11:12.16 | kritzikratzi | thx for the help guys! |
| 11:12.49 | WIMPy | Yeh. I did a patch to accept # once. It just didn't pass the style police :-) |
| 11:13.08 | kritzikratzi | ^^ |
| 11:14.25 | kritzikratzi | WIMPy: one more question. is there a way to donate to the asterisk? i canât really find a donate button on the website. (iâd prefer to donate towards documentation, but any donate button would be fine imho :) ) |
| 11:15.08 | WIMPy | No. Either you can by Digium hardware or you can contribute to the code or documentation. |
| 11:16.22 | kritzikratzi | iâm way too noobish at the moment to contribute. but iâm having so much fun with this thing, so my skill should improve :) |
| 11:17.25 | kritzikratzi | (my living room now has a guestbook which you can call. it records you, and then plays back everything anyone said) |
| 11:17.36 | kritzikratzi | sox and asterisk are great fun in combination! |
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| 12:17.57 | elitas | hello, what password sip account strenght could be ? symblos, letters ? |
| 12:20.04 | nbjoerg | the book is using base64 encoded random numbers and drops a few special characters |
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| 13:12.23 | Cyford33 | has anyone comapred rates to * ? |
| 13:13.32 | Cyford33 | in max calls |
| 13:16.03 | [TK]D-Fender | huh? |
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| 13:28.05 | Rico | is "Alembic" the right way to create asterisk 13 realtime database ? |
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| 13:37.10 | alexschomb | hey. i'm trying to setup PJSIP ACLs in combination with ODBC (MariaDB) ARA but can't get it to work. ps_endpoints and others work fine. is there any example documentation? can't find it in the wiki. |
| 13:37.46 | alexschomb | Asterisk v13 |
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| 13:43.49 | Rico | is "Alembic" the right way to create asterisk 13 realtime database ? |
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| 13:51.09 | Rico | realtime seems to be more complex on asterisk 12/13 than on asterisk 1.x . Isn't it ? |
| 13:51.34 | file | alembic is the mechanism used to create the tables and update them as new fields are added |
| 13:51.49 | Rico | file: ok, got it working |
| 13:52.50 | alexschomb | file: can you point me towards correct implementation of ACLs with realtime? |
| 13:52.59 | file | any documentation would be on the wiki |
| 13:53.11 | file | but do realize... that means every packet coming into your Asterisk would trigger a database lookup |
| 13:53.50 | alexschomb | makes sense. what would be the recommended way to restrict access to endpoints defined in realtime? |
| 13:54.21 | file | define it in the config file? |
| 13:55.07 | alexschomb | how can i make sure it gets applied to every (or only a set of) enpoints? wiki says templates won't work with realtime |
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| 13:55.17 | file | as well, currently there is no mechanism to have ACLs only apply to individual endpoints |
| 13:55.50 | alexschomb | oh okay. then i understood something wrong concerning ACLs |
| 13:56.21 | alexschomb | that means that I just define any ACL and it will be applied automatically to all endpoints? |
| 13:56.38 | alexschomb | (because that doesn't seem to work) |
| 13:56.44 | file | it applies to all traffic coming in |
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| 13:57.07 | file | I have not used it with realtime |
| 13:58.05 | alexschomb | before switching to realtime the acls were only applied when adding a type=acl section with the same name as the endpoint to pjsip.conf |
| 13:58.05 | alexschomb | this section pointed towards acl.conf |
| 13:58.10 | alexschomb | when i didn't have this section the acl in acl.conf won't be applied |
| 13:58.18 | file | the name of the section doesn't matter, the type=acl does |
| 13:58.32 | alexschomb | okay got it. |
| 13:58.42 | alexschomb | just create any section and refer to acl :) |
| 13:58.49 | alexschomb | thanks a lot for your help! |
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| 14:00.40 | alexschomb | works, thanks! |
| 14:01.13 | file | you're welcome |
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| 15:07.48 | Rico | I'm trying to initialize realtime voicemail using alembic, but I get an error : http://pastie.org/10282156 |
| 15:09.20 | Rico | any idea ? |
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| 15:15.34 | Cuzner | why are you asking for help with alembic in an asterisk channel? |
| 15:16.02 | Rico | Cuzner: I'm asking for help about asterisk realtime engine, not about alembic |
| 15:16.08 | Cuzner | ERROR [alembic.util] No such revision or branch 'a541e0b5e89' |
| 15:16.47 | Rico | Cuzner: is there a clean way to create / maintain tables without using alembic ? |
| 15:17.19 | Rico | I'm just following asterisk 13 tutorial |
| 15:17.33 | Rico | https://wiki.asterisk.org/wiki/display/AST/Managing+Realtime+Databases+with+Alembic |
| 15:17.42 | Rico | Update script_location to the schema to update. Asterisk currently supports two sets of schemas: |
| 15:17.42 | Rico | config - the set of schemas for Asterisk Realtime databases |
| 15:17.42 | Rico | voicemail - the schema for ODBC VoiceMail |
| 15:17.52 | Rico | config works fine |
| 15:17.55 | Rico | voicemail does not |
| 15:19.05 | Cuzner | Rico: sorry, i must confess i'm not too familliar with 13, but that looks like a git checkout error for alembic :) |
| 15:19.19 | Rico | Cuzner: ok...* |
| 15:19.38 | Rico | but I think alembic use local branches |
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| 18:04.06 | omgwtfbbq | Hey guys, I have a question regarding the asterisk with heartbeat..for an end user would I have him configured with the virtual ip address that is specified in the haresources ? |
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| 18:26.34 | nbjoerg | is there a self-contained way to test basic asterisk functionality? |
| 18:26.49 | nbjoerg | without having to install dozen of extra dependencies for the full test suite |
| 18:33.50 | *** join/#asterisk jasonwert (~jasonwert@75-134-81-98.static.aldl.mi.charter.com) |
| 18:38.16 | [TK]D-Fender | nbjoerg: install centos. Add Digium's repo. install the packages and the dependencies will work themselves out |
| 18:40.35 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 18:40.58 | nbjoerg | [TK]D-Fender: given that I want to ensure that it works correctly on SmartOS, it would defeat the purpose... |
| 18:48.32 | Cyford33 | i have a issue when my peer trunk goes offline, and calls try an go out on it i get an dialstatus = Answered.. how can i correct it |
| 18:48.50 | Cyford33 | this is a asterisk to asterisk senerio |
| 18:49.22 | *** join/#asterisk jasonwert (~jasonwert@75-134-81-98.static.aldl.mi.charter.com) |
| 18:53.05 | *** join/#asterisk robl^ (~robl@pdpc/supporter/active/robl) |
| 18:54.08 | *** join/#asterisk airjump (~Thunderbi@p5B0A3765.dip0.t-ipconnect.de) |
| 18:55.49 | [TK]D-Fender | nbjoerg[TK]D-Fender: given that I want to ensure that it works correctly on SmartOS, it would defeat the purpose... <- then the only way to know if it works on SmartOS... is to install on SmartOS |
| 18:55.58 | [TK]D-Fender | mbYou've answered your own question based on those terms |
| 18:56.46 | [TK]D-Fender | nbjoergwithout having to install dozen of extra dependencies for the full test suite <- You need the dependencies, that's why they're called dependencies. And if you want to see it run on your specific environment.,. then you're just oging to have to do it |
| 19:05.13 | nbjoerg | [TK]D-Fender: there could have been a basic light weight test that I missed |
| 19:05.19 | nbjoerg | make tests doesn't seem to do anythting |
| 19:06.24 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 19:09.10 | *** join/#asterisk joshaidan (~brianj@mr.hamster.vianet.ca) |
| 19:10.14 | [TK]D-Fender | what "make tests"? |
| 19:11.57 | nbjoerg | there is a tests subdirectory, but I can't figure out how to do anything with it |
| 19:12.14 | [TK]D-Fender | ignore it |
| 19:12.57 | [TK]D-Fender | install the sample configs. trash the parts you should be doing 100% by hand (sip, extensions, and a few others) and go learn the rest and use the samples for reference |
| 19:17.21 | *** join/#asterisk kritzikratzi (~kritzikra@cpe90-146-150-86.liwest.at) |
| 19:28.32 | *** join/#asterisk kleszcz (tick@linuxmafia.pl) |
| 20:08.00 | *** join/#asterisk azerus (~badass@unaffiliated/badass) |
| 20:38.11 | *** join/#asterisk tzafrir (~tzafrir@bzq-179-40-172.cust.bezeqint.net) |
| 20:56.34 | *** join/#asterisk kritzikratzi (~kritzikra@cpe90-146-150-86.liwest.at) |
| 20:58.16 | *** join/#asterisk andoh (0c015aa4@gateway/web/freenode/ip.12.1.90.164) |
| 20:58.21 | andoh | Anyone alive? |
| 21:00.43 | andoh | I am having some problems iwth this python agi script here. It parses and whatnot with no errors, but when I am trying to use it for processing live calls, it seems to skip parts of the script and exits with a 0 without doing anything else. http://pastebin.com/NucUSGp3 |
| 21:00.54 | andoh | Anyone have an idea? |
| 21:04.21 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 21:12.56 | newtonr | andoh, Probably want to pastebin an Asterisk log with "agi set debug on" output included, (probably want to turn "verbose" up as well). That will help someone determine where it is failing |
| 21:13.58 | andoh | Gotcha, give me a sec. |
| 21:14.57 | andoh | http://pastebin.com/0xZmna8M |
| 21:15.43 | andoh | That's with agi debug on, debug and verbose set to 30. |
| 21:17.04 | andoh | If any of the dependancies are needed to be pastebinned, I can do that too. |
| 21:18.13 | newtonr | andoh, You don't have debug going to the console, so turning it up isn't doing anything. You may be missing some other log types. Look at "logger show channels" , edit your logger.conf accordingly and "logger reload" |
| 21:19.12 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 21:20.11 | newtonr | https://wiki.asterisk.org/wiki/display/AST/Logging+Configuration , https://wiki.asterisk.org/wiki/display/AST/Basic+Logging+Commands |
| 21:20.21 | andoh | Hmm.. Ok. |
| 21:20.36 | andoh | Let's see then.. I'll kick myself if it's something that simple. lol |
| 21:23.40 | newtonr | If no one responds here then you may want to post the script and log on the forums or mailing list so others will be able to look over them. A very small slice of the community participates on IRC. It probably is something simple. I'm not an AGI guy so I probably won't spot it easily. |
| 21:32.12 | *** join/#asterisk gravspeed (~SWigle@66.242.174.180) |
| 21:33.59 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 21:38.14 | andoh | Yea, no. Logger wasn't the issue. |
| 21:40.20 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 22:11.34 | r00f | i'd put extensive debug logging into python script itself. when i was troubleshooting phpagi issue, i've modified its code to dump all in/out text to file, so then it was easy to find that asterisk was not always replying to request. |
| 22:12.41 | *** join/#asterisk xochilpili (~xochilpil@unaffiliated/xochilpili) |
| 22:12.45 | xochilpili | hi all |
| 22:12.50 | r00f | sup |
| 22:14.18 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 22:14.19 | xochilpili | i have a snob doubt; i have a dedicated voip trunk from isp (i don't know how to name it), they gave me a ip address net mask and gw, and i have my lan, over 10.0.1.251 which is asteriskNOW |
| 22:15.35 | xochilpili | i have 2 eth cards, one to this voip trunk, and the other on my lan, then, i have a clearOS as a router, as a gateway of my lan |
| 22:16.04 | xochilpili | eth0 it's gw is the clearOS, and the eth1 it's gw is provided by isp |
| 22:16.56 | xochilpili | from internet, i can register to an extension, but i loss audio, i tried to add a route, in asterisk over eth0 and eth1 |
| 22:17.23 | xochilpili | but the same, some one tell me to make a bridge between both eth's |
| 22:17.44 | xochilpili | i don't know what's am i supposed to do |
| 22:18.49 | xochilpili | if i make route to the voice i lose internet in asterisk, if i make the route to clearOS and make the route to voice i lose audio |
| 22:19.42 | r00f | so, asterisknow connected to voip and clearos, while clearos is internet router? how do you register from internet? |
| 22:19.50 | WIMPy | If you have multiple IPs you will also have multiple routes. This is basic networking. |
| 22:21.14 | xochilpili | WIMPy, i have a clears with one ip, and it's on internet, then it's supposed to provide internet to asterisk which is behind clearOS, asterisk now on eth0 has clearos's gateway |
| 22:21.39 | xochilpili | in the other eth card of asterisk i have the voip trunk from isp |
| 22:22.14 | xochilpili | i have both routes, route-eth0 0.0.0.0/0 via 10.0.1.252 (clearOS) dev eth0 |
| 22:22.56 | xochilpili | and route-eth0 x.x.x.x/xx via x.x.x.x dev eth1 this params were provided by iso |
| 22:24.09 | xochilpili | in that way i have internet in asterisk, but i lose audio in a call from internet. locally i have no audio issues |
| 22:25.11 | [TK]D-Fender | Your description is too messy |
| 22:25.20 | [TK]D-Fender | You have not been clear on the basics |
| 22:25.30 | [TK]D-Fender | Please confirm the following : |
| 22:26.04 | WIMPy | And 2 of the 3 netorks haven't been described, yet. |
| 22:26.05 | [TK]D-Fender | A: Your ClearOS router has 2 interfaces? 1 WAN (with public IP), and 1 LAN (private IP). |
| 22:26.19 | [TK]D-Fender | B: Your * server ONLY has a private IP. |
| 22:26.26 | xochilpili | A: yes |
| 22:26.50 | WIMPy | And do you also use VOIP on the internet or only on that extra route from yout ISP? |
| 22:27.14 | xochilpili | B: private ip is the voice trunk provided by iso? |
| 22:27.31 | [TK]D-Fender | xochilpili, Your question makes NO snese |
| 22:27.48 | [TK]D-Fender | On your asterisk server what NETWORK is it on? |
| 22:27.50 | [TK]D-Fender | sense* |
| 22:27.52 | WIMPy | What is your asterisk connected to? |
| 22:28.31 | WIMPy | And do you have two internet connections or where does the extra network to your itsp come from? |
| 22:28.54 | xochilpili | asterisk is connected to clearOS lan on eth0 and also connected to a ipaddres of a voice trunk on eth1 |
| 22:29.12 | xochilpili | in the voice trunk there's is no internet, only voice |
| 22:29.18 | [TK]D-Fender | xochilpili, Where is eth1 lead to? |
| 22:29.31 | [TK]D-Fender | You are not clearly confirming where anything leads to |
| 22:29.44 | [TK]D-Fender | "eth1" means nothing to me |
| 22:29.53 | WIMPy | Maybe you should make adrawing of what you have. |
| 22:30.01 | [TK]D-Fender | And you are not connected TO an IP address |
| 22:30.06 | [TK]D-Fender | An interface HAS an address |
| 22:30.25 | [TK]D-Fender | Giive a PROPER description |
| 22:30.39 | [TK]D-Fender | And WIMPy's suggestion of drawing a picture would be a very good idea |
| 22:31.01 | [TK]D-Fender | Because your wording is not leading to anything we can draw in our heads |
| 22:31.13 | xochilpili | sorry by my english |
| 22:32.01 | xochilpili | on eth1 i have a sip trunk |
| 22:32.16 | xochilpili | eth0 lan with clearos |
| 22:32.17 | [TK]D-Fender | NO |
| 22:32.31 | *** join/#asterisk superscrat (asanders@nat/digium/x-wsutliazimvullyz) |
| 22:32.46 | [TK]D-Fender | Provide a CLEAR description of the networking |
| 22:33.08 | [TK]D-Fender | That means giving DETAILS about what is plugged into what. IP ADDRESSES that are assigned. |
| 22:33.26 | [TK]D-Fender | <xochilpili> on eth1 i have a sip trunk <- this is vague garbage. It says NOTHING about the networks your data has to cross |
| 22:33.31 | WIMPy | And Netmasks! |
| 22:33.52 | [TK]D-Fender | Or what the next hops are, any network transformations that happen, or anything else |
| 22:34.32 | *** part/#asterisk mjordan (mjordan@nat/digium/x-lcqhvoxuelenvnik) |
| 22:36.09 | xochilpili | i am asking to the isp |
| 22:36.20 | xochilpili | and i making a draw |
| 22:36.43 | [TK]D-Fender | You shouldn't have to ask your ISP anything. |
| 22:36.47 | [TK]D-Fender | This is YOUR equipment. |
| 22:37.01 | [TK]D-Fender | If you don't know what networks and interfaces you have on them you have a VERY serious issue |
| 22:37.38 | WIMPy | And no realistic chance with Asterisk. |
| 22:38.49 | xochilpili | yes i know |
| 22:39.09 | xochilpili | bu you know me, im always trying to understand and learn |
| 22:39.54 | xochilpili | all of you know my process. Since i install asterisk with no idea of it... remember? |
| 22:40.11 | [TK]D-Fender | You're still there.... |
| 22:40.31 | [TK]D-Fender | you don't know what you HAVE. |
| 22:40.45 | [TK]D-Fender | You're dead in the water right from the start |
| 22:41.24 | xochilpili | yeah, well, still here, kicking to be out |
| 22:41.50 | WIMPy | And the networking is really easy compared to Asterisk. |
| 22:45.01 | r00f | can i ask a question regarding some asterisk design decision, or it is better to address some -devel maillist? |
| 22:45.19 | WIMPy | ~ask |
| 22:45.21 | 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. |
| 22:45.34 | r00f | ) |
| 22:46.03 | r00f | i've noticed that upon receiving 404 sip reply, asterisk doesn't show "Got SIP response ..." on console |
| 22:46.24 | r00f | so i went to source code, added that message, recompiled - and now i get the message, and nothing broke |
| 22:46.35 | r00f | so just wondering why was it decided to hide it from user |
| 22:46.48 | r00f | asterisk 11 |
| 22:47.12 | [TK]D-Fender | it should, and have never heard of any case to the contrary |
| 22:47.33 | [TK]D-Fender | You'd have to be more specific about exactly what branch you applied your patch to, etc |
| 22:47.45 | WIMPy | I have been wondering why such vital information doesn;t show up while we always get informed about TOS values. I doubt there's a reason other that that's the way it happened. |
| 22:49.10 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 22:49.11 | r00f | 11.11.0 |
| 22:49.36 | xochilpili | ok |
| 22:50.13 | xochilpili | trying to explain again less with vague garbage; |
| 22:50.42 | r00f | in channels/chan_sip.c, in handle_response func it hase CASE with differnet responses. and for 404, on INVITE, it has just |
| 22:50.44 | r00f | handle_response_invite(p, resp, rest, req, seqno); |
| 22:50.55 | r00f | i added |
| 22:50.56 | r00f | ast_verb(3, "Got SIP response %d \"%s\" back from %s\n", resp, rest, ast_sockaddr_stringify(&p->sa)); |
| 22:51.01 | r00f | and it works like a charm |
| 22:51.52 | r00f | maybe had to check for new version vefore asking, though |
| 22:51.57 | WIMPy | Open an ticket on issues.asterisk.org. You might be requested to put it up on gerrit. I'm not sure what the exact procedure is now. |
| 22:51.59 | [TK]D-Fender | r00f, 11.18.0 (2015/06/04), <-- |
| 22:52.06 | [TK]D-Fender | you just patched an older version of 11 |
| 22:52.13 | [TK]D-Fender | Which is never where you should start with this |
| 22:52.27 | [TK]D-Fender | You're very likley reinventing the wheel |
| 22:52.32 | xochilpili | clearOS ; wan and pan: lan-> 10.0.1.252. Asterisk: eth0 -> 10.0.1.251; eth1: private pan to a voice proxy |
| 22:52.33 | *** join/#asterisk robl^ (~robl@pdpc/supporter/active/robl) |
| 22:52.45 | WIMPy | Not in this case. |
| 22:52.46 | xochilpili | wan and lan ^ |
| 22:53.49 | xochilpili | http://picpaste.com/IMG_0221-TppsKMt7.JPG |
| 22:53.55 | xochilpili | the schema? |
| 22:54.26 | WIMPy | That is rather incomplete. |
| 22:54.53 | WIMPy | Most IPs and all netmasks are missing and we still don't know where that eth1 is connected to. |
| 22:54.56 | r00f | [TK]D-Fender, just downloaded and checked the 11.18.0, it has same code |
| 22:56.24 | xochilpili | WIMPy, asterisk; eth0 : ipaddr: 10.0.1.251; netmask: 255.255.255.0; gateway: 10.0.1.252 (clearOS) |
| 22:56.56 | r00f | line 24073 - it has just handle_response_invite, no ast_log or ast_verb |
| 22:57.08 | *** join/#asterisk wonderworld (~ww@ip-176-199-166-61.hsi06.unitymediagroup.de) |
| 22:57.50 | r00f | handle_response_invite itself has ast_debug, but no ast_verb |
| 22:58.06 | r00f | maybe it was decided that 404 is not important enough to be shown without debug.... |
| 22:58.12 | [TK]D-Fender | xochilpili, You have not clearly shown what TWO ends of that are even connected to. You haven't provided the IP & netmask ranges either |
| 22:58.13 | *** join/#asterisk pchero (~pchero@0x555140b5.adsl.cybercity.dk) |
| 22:58.23 | [TK]D-Fender | xcWe have no idea what any of that is connected to |
| 22:58.31 | xochilpili | [TK]D-Fender, im on it |
| 23:02.12 | WIMPy | r00f: The best way would probably be to display the result before it gets parsed. |
| 23:02.36 | xochilpili | [TK]D-Fender, http://pastebin.com/gA9gMB3D |
| 23:03.02 | r00f | WIMPy: well my workaround works for me, i guess i am the only one who is bothered if that never been asked so let it be |
| 23:03.27 | r00f | maybe all the other guys run debug when they need it, i just wanted to see it in verbose |
| 23:04.20 | WIMPy | Well, you HAVE TO run debug if you want to know what's going on, and I find that extremely annoying. |
| 23:04.35 | xochilpili | [TK]D-Fender, is that more clear? |
| 23:04.50 | r00f | i always run verbose 5, in 99% it is all i need |
| 23:04.55 | [TK]D-Fender | xochilpili, No |
| 23:04.58 | xochilpili | :( |
| 23:05.24 | [TK]D-Fender | You not telling us what it PLUGGED INTO |
| 23:05.30 | [TK]D-Fender | Draw the damn picture |
| 23:05.37 | [TK]D-Fender | You left LOOSE ENDS on the last one |
| 23:05.42 | WIMPy | Well, at least we found out now that the two networks don't overlap. |
| 23:05.48 | [TK]D-Fender | We don't know what stupid box things LEAD to. |
| 23:06.42 | r00f | gtg, best of lucks |
| 23:07.13 | [TK]D-Fender | r00f, I think I see what you mean though |
| 23:07.22 | [TK]D-Fender | r00f, You wanted SIP DEBUG in basic verbose |
| 23:07.38 | [TK]D-Fender | r00f, which was never something * did there. That's what "sip debug" was for |
| 23:08.56 | WIMPy | Well, at least you can retrieve the final response when Dial()ing via dialplan now. |
| 23:09.40 | WIMPy | Before that chan_sip was completely unsupportable. |
| 23:10.00 | *** join/#asterisk johnny_|_ (~johnny@unaffiliated/johnny-/x-2623418) |
| 23:10.03 | xochilpili | ONT means something to you? |
| 23:11.18 | WIMPy | Ontario? Optical Nwtwort Termination? Something else? |
| 23:11.34 | xochilpili | Optical Network... |
| 23:11.37 | WIMPy | *Network |
| 23:11.59 | WIMPy | And what is it you want to tell us about that? |
| 23:12.03 | [TK]D-Fender | Media means nothing ot me. |
| 23:12.09 | [TK]D-Fender | I care what it LEADS TO. |
| 23:12.15 | [TK]D-Fender | And you aren't telling us |
| 23:12.21 | [TK]D-Fender | WHAT'S AT THE OTHER END |
| 23:12.26 | [TK]D-Fender | Whaet fraffic goes there? |
| 23:19.27 | xochilpili | http://picpaste.com/IMG_0222-tpUqmFwJ.JPG |
| 23:22.30 | xochilpili | is that more clear? |
| 23:22.56 | [TK]D-Fender | No, that brings up all sorts of new questions |
| 23:23.11 | [TK]D-Fender | You say "WAN" and don't give an IP ADDRESS |
| 23:23.11 | WIMPy | Ok, so you have two seperate internet connections. Took only an hour to find that answer. |
| 23:23.19 | *** join/#asterisk alexw (~textual@unaffiliated/alexw) |
| 23:23.27 | [TK]D-Fender | You say that leads to INTERNET, yet your * box talks DIRECT to this toher thing. |
| 23:23.31 | *** part/#asterisk kharwell (kharwell@nat/digium/x-jhndgotneieiaqap) |
| 23:24.08 | xochilpili | yes, clearos provides internet, while the other thing is only for voice, no internet there |
| 23:24.27 | [TK]D-Fender | WTF is "thing"? |
| 23:24.39 | xochilpili | jaja |
| 23:24.40 | WIMPy | Still unanswerd ones are: Do you also use other ITSPs and ar your phones all local? |
| 23:24.42 | xochilpili | you said that! |
| 23:25.01 | xochilpili | yet your * box talks direct to this other thing |
| 23:25.08 | [TK]D-Fender | Why would you need to use the INTERNET to talk to this "ONT" when you can get to it DIRECTLY PRIVATELY? |
| 23:25.21 | [TK]D-Fender | Why do you have TWO ends talking to this same point? |
| 23:25.30 | xochilpili | this other thing is a private lan connected to this ONT to the isp proxy |
| 23:26.10 | [TK]D-Fender | Is this over a completely separate wired connection? |
| 23:26.16 | xochilpili | yes |
| 23:26.20 | WIMPy | And your description of what went wrong was very unclear as well. |
| 23:26.35 | xochilpili | there's 2 wires, one for internet directly to wan on clearos |
| 23:26.47 | xochilpili | and the other is directly to * box |
| 23:26.52 | [TK]D-Fender | So far it doesn't look like it |
| 23:27.14 | [TK]D-Fender | You draw this ONT like it's a BOX on your site |
| 23:27.35 | xochilpili | it is a box in my site |
| 23:27.44 | [TK]D-Fender | ... |
| 23:27.59 | [TK]D-Fender | OThen how is clearOS talking to it via the INTERNET? |
| 23:28.06 | [TK]D-Fender | And why WOULD it? |
| 23:28.30 | [TK]D-Fender | This drawing is trash |
| 23:28.34 | xochilpili | this ont box has 2 ports LAN1; clearOS, and LAN2: private lan |
| 23:28.49 | WIMPy | It's getting internet. |
| 23:28.59 | [TK]D-Fender | There is no INTERNET on ClearOS from what I am seeing here |
| 23:29.04 | xochilpili | lan2 is connected directly to asterisk |
| 23:29.16 | [TK]D-Fender | It is one dumb box at your site plugged DIRECTLY into another dumb box at your site |
| 23:29.19 | xochilpili | Yes! in clearos there's internet |
| 23:29.40 | [TK]D-Fender | Your description is terrible |
| 23:29.49 | xochilpili | ah come on! |
| 23:29.49 | [TK]D-Fender | ONT is AT YOUR LOCATION |
| 23:30.06 | [TK]D-Fender | You can't call it "internet" when its 2 boxes in your building directly plugged into one another |
| 23:30.33 | WIMPy | I *think* I understand what's going on, but the description really is too chaotic to be sure. |
| 23:30.48 | [TK]D-Fender | Nothing make that port "wan" anywhere |
| 23:31.09 | [TK]D-Fender | WAN is a theoretical name now and it's meaning is broken |
| 23:31.13 | Iamnacho | [TK]D-Fender: that should be the quote of the day: "you can't call it 'internet' when its 2 boxes in your building directly plugged into one another" LOL |
| 23:31.16 | WIMPy | assumes the ONT is providing Internet on one port and a VPN on the other. |
| 23:31.23 | [TK]D-Fender | And you have not provided NETWORK and IP' on this |
| 23:31.57 | WIMPy | Or 2 VLANs or whatever. |
| 23:31.58 | [TK]D-Fender | I have 3 internets how I can voip?!?! HAELP!@ |
| 23:34.24 | xochilpili | the ONT is providing Internet on one port and a VPN on the other |
| 23:35.13 | WIMPy | Are you sure about that? |
| 23:35.19 | xochilpili | completly |
| 23:35.30 | WIMPy | And why didn't you tell us that most basic fact in the beginning? |
| 23:35.43 | xochilpili | because i didnt know |
| 23:35.44 | xochilpili | sorry |
| 23:36.24 | [TK]D-Fender | How did you JUST find this out? |
| 23:36.28 | WIMPy | Now to all the other questions. |
| 23:36.51 | [TK]D-Fender | Next, to don tall that port on your ClearOS "internet" |
| 23:36.53 | [TK]D-Fender | It is not. |
| 23:36.54 | xochilpili | [TK]D-Fender, the isp provider told me a few minutes ago |
| 23:37.01 | [TK]D-Fender | Your FIBER is the only link to the internet |
| 23:37.19 | [TK]D-Fender | And that is on THEIR box. |
| 23:37.45 | xochilpili | excuse me, again please. |
| 23:37.47 | [TK]D-Fender | Now go confirm what IP & NETWORK you have on your clearOS port that leads to their box |
| 23:37.56 | xochilpili | My only link to the internet is the fiber. |
| 23:38.07 | xochilpili | oks |
| 23:38.17 | [TK]D-Fender | We've asked half a dozen times already |
| 23:39.19 | xochilpili | ClearOS eth1: IPADDR="187.188.213.132" ; netmask: 255.255.255.128; gateway: 187.188.213.129 |
| 23:39.49 | xochilpili | this is connected to this ont (in the port that provides internet on the ONT) |
| 23:39.57 | WIMPy | \o/ internet! |
| 23:40.03 | [TK]D-Fender | FINALLY |
| 23:40.08 | xochilpili | jaja |
| 23:40.11 | xochilpili | yeah! |
| 23:40.33 | [TK]D-Fender | So you use the VPN connection to your provider for ITSP use, correct? |
| 23:40.35 | WIMPy | Jaja heißt leck mich am Arsch! |
| 23:40.42 | [TK]D-Fender | As in your ISP *is* your ITSP as well? |
| 23:40.59 | xochilpili | really im so sorry, but you know that every time i come here to ask is because, once you trample on me, then i learn... |
| 23:41.23 | xochilpili | [TK]D-Fender, wait |
| 23:41.29 | xochilpili | again? |
| 23:41.32 | xochilpili | ITSP? |
| 23:41.42 | WIMPy | ~itsp |
| 23:41.43 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
| 23:42.08 | xochilpili | [TK]D-Fender, yes |
| 23:42.24 | xochilpili | my isp also is my itsp |
| 23:42.57 | [TK]D-Fender | xochilpili, And you have clients you reach to over the internet via the clearOS box which is on the PUBLIC subnet that is routed to it? |
| 23:43.45 | WIMPy | Still unanswerd ones are: Do you also use other ITSPs and ar your phones all local? |
| 23:44.09 | xochilpili | i think that i understand, you mean; that i can connect from internet to asterisk box? |
| 23:44.32 | xochilpili | WIMPy, no, no all the phones are local |
| 23:44.38 | WIMPy | Do you even want to? |
| 23:44.50 | WIMPy | And what about other ITSPs? |
| 23:45.10 | xochilpili | yes, i want to connect from internet to asterisk box |
| 23:45.13 | xochilpili | that's the issue |
| 23:45.27 | xochilpili | locally i have no issues |
| 23:45.28 | WIMPy | With what? |
| 23:45.33 | [TK]D-Fender | xochilpili, Ok, so if you only have local phones and you only communicate via your ISP which is supposed to go over that VPN link..... show us the call with SIP DEBUG. |
| 23:45.36 | [TK]D-Fender | ~pb |
| 23:45.36 | 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. |
| 23:45.37 | *** join/#asterisk babak (uid19622@gateway/web/irccloud.com/x-vgfjmlqpweijrjci) |
| 23:45.37 | [TK]D-Fender | ^^^ |
| 23:47.05 | xochilpili | [TK]D-Fender, locally i can make calls, that's not the issue, the issue i think is routing from internet to clearos and then asterisk through vpn |
| 23:47.31 | [TK]D-Fender | Then you did your networking wrong on your * box. |
| 23:47.42 | WIMPy | What on the internet are you talking to? |
| 23:47.43 | [TK]D-Fender | You said it was DIRECTLY connected to that other port |
| 23:47.44 | xochilpili | that's for sure |
| 23:47.48 | [TK]D-Fender | Show us the call |
| 23:47.49 | *** join/#asterisk jasonwert (~jasonwert@71.89.137.28) |
| 23:48.24 | xochilpili | oks |
| 23:48.43 | [TK]D-Fender | WIMPy, So far he's not supposed to route directly there. Only out the ETH1 which is on a VPN connection to his provider |
| 23:48.59 | [TK]D-Fender | (directly to the internet that is) |
| 23:49.27 | [TK]D-Fender | Basically, we're not supposed to have to know OR care about that clearOS box AT ALL |
| 23:49.28 | WIMPy | That's why I want to kow what he's doing what wasn't described, yet. |
| 23:50.30 | xochilpili | http://pastebin.com/eu8vGYXZ |
| 23:51.52 | xochilpili | that's a call from zoiper via internet to my home phone, my home's phone rings but once answered there's no audio |
| 23:52.15 | [TK]D-Fender | xcYou just said "no internet" only LOCAL |
| 23:52.19 | [TK]D-Fender | WTF |
| 23:52.23 | WIMPy | So you're using a phone that is not local. |
| 23:52.35 | [TK]D-Fender | You just LIED to us |
| 23:52.39 | xochilpili | no!!! |
| 23:52.48 | WIMPy | yes |
| 23:52.58 | xochilpili | why i lied to you? |
| 23:53.01 | [TK]D-Fender | <xochilpili> WIMPy, no, no all the phones are local <--- |
| 23:53.10 | [TK]D-Fender | zoiper is over the internet!!!!! |
| 23:53.16 | WIMPy | You said all your phones were local. |
| 23:53.17 | [TK]D-Fender | INTERNET != LOCAL |
| 23:53.27 | xochilpili | not all phones are local |
| 23:53.42 | xochilpili | xochilpili> [TK]D-Fender, locally i can make calls, that's not the issue, the issue i think is routing from internet to clearos and then asterisk through vpn |
| 23:54.06 | Cyford33 | is your * behind firewall with private ip address? |
| 23:54.26 | WIMPy | Now we're going form bad descriptions to wrong descriptions. That doesn't work out. |
| 23:55.14 | [TK]D-Fender | <--- Transmitting SIP response (600 bytes) to UDP:187.212.202.164:49012 --->Contact: <sip:10.0.1.251:5060> |
| 23:55.22 | xochilpili | WIMPy, but i said before the issue i have, from internet i have no audio, but locally i have |
| 23:55.36 | [TK]D-Fender | xochilpili, You clearly didn't set up your * server to send the right contact infor while working from behind NAT |
| 23:56.00 | WIMPy | You didn't say if it was from an ITSP or a remote phone. That's why I asked. And you gave the wrong answer. |
| 23:56.58 | [TK]D-Fender | <--- Transmitting SIP response (939 bytes) to UDP:187.212.202.164:49012 ---> |
| 23:56.58 | [TK]D-Fender | SIP/2.0 200 OKo=- 0 2 IN IP4 10.0.1.251 |
| 23:57.15 | [TK]D-Fender | You are giving them a PRIVATE IP to send audio to. They can't do that |
| 23:57.26 | WIMPy | A correct NAT configuration might end up making the ITSP link unusable. So this is going to be an interesting one. |
| 23:57.26 | [TK]D-Fender | Fix your extenIP and localnet settings |
| 23:57.41 | [TK]D-Fender | WIMPy, No, localnet's can handle that easily |
| 23:57.55 | [TK]D-Fender | WIMPy, We see the IP's & network it's on. Easy to indeify |
| 23:57.59 | [TK]D-Fender | identify |
| 23:58.40 | WIMPy | Yes, but it isn't local. So that means all directmedia/reinvite attempts must be disabled. |
| 23:58.59 | [TK]D-Fender | Of course |
| 23:59.12 | [TK]D-Fender | I'm sure he's done none of what's necessay for thiws |
