| 00:01.43 | p3nguin | For one PIN, pin=$(apg -a 1 -n 1 -m 4 -x 5 -M N); echo $pin;echo -n $pin|md5sum |
| 00:03.29 | p3nguin | Now I have to come up with a reason to use this authentication method. |
| 00:04.44 | p3nguin | hlieberman: I think I may have just found a bug. |
| 00:04.55 | p3nguin | mainmenu, extension s... |
| 00:05.19 | hlieberman | Which? |
| 00:05.28 | p3nguin | Goto(1) is going to set the timeout value again. I'm not sure if it will reset the counter from when it was set the first time. |
| 00:06.15 | p3nguin | To fix it, I do something like change exten => s,n,BackGround(welcome) to exten => s,n(start),BackGround(welcome) |
| 00:06.22 | p3nguin | and then Goto(start) |
| 00:07.08 | p3nguin | That way it does not touch the Set()ing of TIMEOUT(absolute) a second time. |
| 00:07.26 | p3nguin | Do I need to change it in the pastebin? |
| 00:08.12 | hlieberman | No, I got it. |
| 00:08.47 | hlieberman | The debian default asterisk install has about a billion config files. |
| 00:08.53 | hlieberman | Following them all is making my brain hurt. |
| 00:08.57 | p3nguin | Move or remove them. |
| 00:09.14 | p3nguin | You should not be using sample configs for many of the modules. |
| 00:09.41 | p3nguin | Some of the samples contain the minimal defaults that will be needed, but sip.conf and extensions.conf should be started fresh. |
| 00:10.30 | hlieberman | I did, and now asterisk isn't logging anything, it seems. |
| 00:11.06 | p3nguin | The logger.conf sample is possibly good enough to use. |
| 00:12.11 | hlieberman | 82 default configuration files. |
| 00:12.13 | hlieberman | I mean, really? |
| 00:12.37 | p3nguin | They're just samples of everything. |
| 00:12.51 | p3nguin | Lots of them you won't need. |
| 00:13.37 | p3nguin | You'll want asterisk.conf, modules.conf, sip.conf, extensions.conf, maybe voicemail.conf... not sure what else. |
| 00:14.57 | p3nguin | I only have 38 config files. |
| 00:15.17 | p3nguin | And I see at least a couple that I don't need/use. |
| 00:15.31 | p3nguin | I'll consider deleting them soon. |
| 00:17.14 | hlieberman | modules.conf? |
| 00:17.38 | p3nguin | You'll want it. |
| 00:19.19 | *** join/#asterisk zerohalo (~zerohalo@74.60.136.128) |
| 00:21.39 | hlieberman | OK. Well, let's see if I can get calls actually being sent to the thing. |
| 00:28.13 | hlieberman | Well, that's weird. |
| 00:30.14 | hlieberman | Oh, no, I'm just stupid is all. |
| 00:32.03 | hlieberman | When I run dialplan show, I should see... stuff, right? |
| 00:32.13 | p3nguin | yes |
| 00:32.18 | hlieberman | Because I don't. |
| 00:33.30 | hlieberman | It doesn't look like it's reading the extensions file. |
| 00:33.45 | *** join/#asterisk CrossWired (~CrossWire@pool-108-33-58-210.tampfl.fios.verizon.net) |
| 00:34.36 | p3nguin | core set verbose 3 |
| 00:34.42 | p3nguin | See anything? |
| 00:34.44 | p3nguin | dialplan reload |
| 00:34.59 | hlieberman | No such command dialplan reload |
| 00:35.19 | p3nguin | module load pbx_config.so |
| 00:35.34 | hlieberman | That did it. |
| 00:35.37 | hlieberman | Whoops. |
| 00:35.38 | hlieberman | Thanks. |
| 00:35.49 | p3nguin | Make sure you have autoload enabled in modules.conf. |
| 00:36.46 | CrossWired | i'm working on outbound messaging, my carrier is set to g711, i want to convert my files to the best format so that little or no transcoding needs to take place after the initial formating, what is the best format? |
| 00:37.07 | p3nguin | ulaw |
| 00:37.40 | CrossWired | that what i thought, thanks agains |
| 00:38.09 | hlieberman | In the inbound context, after Answer(), I should Goto(mainmenu), right? |
| 00:38.22 | *** join/#asterisk eva_02 (~somebody@195.93.191.141) |
| 00:38.40 | CrossWired | g729 is a better codec right, would that also be ulaw? |
| 00:38.42 | p3nguin | You don't need an Answer... just a Goto(mainmenu,s,1) |
| 00:39.03 | p3nguin | g729 is a compressing codec, which has lower sound quality than ulaw. |
| 00:39.31 | CrossWired | whats the bandwidth difference in g711 and g729? |
| 00:40.08 | eva_02 | Guys, tell me please, what i'm doing wrong. I have error "username mismatch, have <123123>, digest has <222222>" with this config http://pastebin.com/raw.php?i=kpdfbSRa |
| 00:40.24 | eva_02 | when i'm call to 222222 |
| 00:41.43 | p3nguin | The bandwidth savings of g729 is substantial when compared to ulaw. |
| 00:42.12 | CrossWired | isn't there licensing concerns for g729? |
| 00:42.23 | CrossWired | like 10 per channel or something like that |
| 00:42.42 | p3nguin | It's a patented codec, so you're supposed to pay to use it. |
| 00:49.02 | CrossWired | so if use g729, is there a better format to have my sounds files in? |
| 00:49.19 | p3nguin | g729 seems sensible. |
| 00:50.12 | CrossWired | does the asterisk file convert, convert to that? |
| 00:50.20 | CrossWired | i guess if its installed as a code it will, duh |
| 00:50.31 | p3nguin | I'm pretty sure it will. |
| 00:52.17 | *** join/#asterisk WIMPy (~wimpy@e183095026.adsl.alicedsl.de) |
| 00:55.11 | *** join/#asterisk zerohalo (~zerohalo@74.60.136.128) |
| 01:26.57 | p3nguin | hlieberman: I did the same thing on that Goto() in the checkin context. Should be Goto(auth) instead of Goto(1). |
| 01:27.22 | hlieberman | I will change it. |
| 01:27.34 | hlieberman | Changed. :) |
| 01:27.44 | hlieberman | Got the first sound file working. Onto the next! God, my voice is terrible. |
| 01:28.17 | p3nguin | I think the reason I did that was because I added the timeouts last, after I had already put in the Goto(1)s. |
| 01:28.42 | *** join/#asterisk darkbasic (~quassel@host37-245-static.119-2-b.business.telecomitalia.it) |
| 01:29.53 | hlieberman | Should I put Hangup() after the playback of vm-goodbye? |
| 01:30.16 | p3nguin | Explicit hangups are never a bad idea. |
| 01:30.27 | p3nguin | I typically do, and I'm not sure why I didn't this time. |
| 01:39.33 | hlieberman | Hm. It shoves me in a loop when I authenticate. |
| 01:40.57 | hlieberman | I think my callerID is not matching correctly. But it's right in the file. |
| 01:41.04 | hlieberman | The debug output says it's failing that check. |
| 01:41.37 | p3nguin | Bugfix: http://pastebin.com/UmQnNgVr |
| 01:42.27 | p3nguin | Do your see your callerid number in the verbose output? |
| 01:43.08 | hlieberman | I see SIP/flowroute-0000011 |
| 01:43.16 | *** join/#asterisk dijib (~nobodysho@bas10-kitchener06-1176001555.dsl.bell.ca) |
| 01:43.26 | p3nguin | That's not the caller id number, though. |
| 01:43.42 | p3nguin | Look at the GotoIf() |
| 01:43.56 | p3nguin | s@checkin:3 |
| 01:44.19 | hlieberman | <PROTECTED> |
| 01:44.35 | dijib | btw penguin the callerid name & number gets passed on to callee's not just number |
| 01:44.57 | dijib | didnt you try and tell me it didnt |
| 01:44.57 | p3nguin | dijib: That's a negative. You cannot pass your caller id NAME to the PSTN. |
| 01:45.10 | p3nguin | Yeah I told you it can't because it can't. |
| 01:45.23 | hlieberman | I can pastebin the whole debug output if you'd like, p3nguin. |
| 01:45.29 | dijib | it does though |
| 01:45.34 | p3nguin | dijib: It doesn't. |
| 01:45.42 | p3nguin | Why? Because it can't. |
| 01:45.46 | dijib | well then how do people get my name? |
| 01:45.50 | p3nguin | Because that is not how CNAM works. |
| 01:45.55 | hlieberman | p3nguin, http://pastebin.com/Z9REN6um |
| 01:45.57 | dijib | negative |
| 01:46.06 | dijib | i can instantaneously do it |
| 01:46.14 | dijib | set it and have it changed... no DB nothing |
| 01:47.22 | p3nguin | You just keep telling yourself that you're a magician. |
| 01:48.11 | p3nguin | If I give you a number to call, will you call it? |
| 01:48.42 | dijib | sure |
| 01:48.54 | dijib | wait before that.... |
| 01:48.58 | dijib | let me make a funn CID |
| 01:49.00 | *** join/#asterisk [Outcast] (~anonymous@pool-108-7-70-195.bstnma.fios.verizon.net) |
| 01:49.23 | p3nguin | Don't tell me what you're setting it to. |
| 01:52.28 | p3nguin | hlieberman: You don't need that explicit Answer() nor the Wait(1). You can use BackGround(silence/1&welcome) instead. |
| 01:53.03 | hlieberman | OK, that's fixed. |
| 01:53.12 | p3nguin | Also, you're going to have to add a line to check your caller id. s,n,Verbose(Call from ${CALLERID(all)}) |
| 01:53.27 | dijib | OK its set, both name & number |
| 01:53.37 | dijib | even referenced the real number |
| 01:53.43 | dijib | for what i set the name as |
| 01:53.57 | p3nguin | You better not have used the real name that is registered to that number. |
| 01:54.07 | dijib | no |
| 01:54.13 | hlieberman | Oh, I see. |
| 01:54.14 | p3nguin | If you did, that's cheating. |
| 01:54.20 | hlieberman | It comes through as caller-id +1<number> |
| 01:54.22 | dijib | its an american goverment agency |
| 01:54.25 | hlieberman | And I have it in the login file as <number> |
| 01:54.27 | dijib | that doesnt do any good |
| 01:54.31 | p3nguin | So you used a random number, and a random name? |
| 01:54.43 | p3nguin | hlieberman: I would strip that off if I were you. |
| 01:54.49 | p3nguin | I can't stand that. |
| 01:55.06 | dijib | no, name of organization & number |
| 01:55.37 | p3nguin | If I can look up the organization by the name you're setting and find the number you're setting, that's not a valid test. |
| 01:55.43 | dijib | calling |
| 01:55.45 | p3nguin | Your claim is that you can set any name you want. |
| 01:55.56 | dijib | it said not in service. |
| 01:55.58 | p3nguin | TOLL FREE TF <8008003855> |
| 01:56.10 | dijib | thats where i came from? |
| 01:56.17 | p3nguin | Set it to a random non-toll-free number. |
| 01:56.22 | dijib | number is right |
| 01:56.26 | p3nguin | And a random name if you want. |
| 01:56.28 | dijib | but the toll free TF is wrong |
| 01:56.34 | p3nguin | It's not wrong. |
| 01:56.47 | p3nguin | You just can't set your name like you think you can. |
| 01:56.49 | dijib | if i phone my brother/mother/father its set. |
| 01:56.57 | p3nguin | because that is not how CNAM works. |
| 01:57.00 | p3nguin | They are using VoIP.ms? |
| 01:57.07 | dijib | no |
| 01:57.27 | p3nguin | Have you had your phone number on another telco before? |
| 01:57.31 | dijib | on is cellular, one is a normal bell landline and the other is cogeco voip |
| 01:57.31 | hlieberman | p3nguin, How do I strip off that +1? |
| 01:57.42 | dijib | no |
| 01:57.53 | dijib | first time with number was with voipms |
| 01:58.03 | dijib | maybe its a canadian thing? |
| 01:58.11 | dijib | nortel? |
| 01:58.11 | p3nguin | s,n,ExecIf($["${CALLERID(num):0:2}" = "+1"]?Set(CALLERID(num)=${CALLERID(num):2})) |
| 01:58.28 | p3nguin | Canadian PSTN isn't going to work differently, I'm quite sure. |
| 01:58.41 | dijib | i would figure not... but... possibly |
| 01:58.51 | p3nguin | But try another random number with an unrelated random name. |
| 01:59.03 | p3nguin | not toll-free this time |
| 01:59.18 | dijib | ok |
| 02:01.12 | p3nguin | unKnown <5555550123> |
| 02:01.13 | dijib | why does your number say its not in service? |
| 02:01.16 | dijib | hmm, |
| 02:01.20 | p3nguin | because it's not in service. |
| 02:01.21 | dijib | weird |
| 02:01.51 | p3nguin | I just wanted you to call a number that I could watch the caller id on when you called. |
| 02:01.53 | dijib | ill test extensively tomorrow or monday |
| 02:02.08 | p3nguin | When you're ready to listen to how CNAM works, I'll tell you. |
| 02:02.09 | dijib | how many DID's you got? |
| 02:02.12 | p3nguin | lots |
| 02:02.24 | dijib | operator |
| 02:02.33 | dijib | i dont know if im ready to listen tonight |
| 02:02.46 | dijib | i still have thanksgiving turkey to make stuffing for tonight |
| 02:02.56 | dijib | and need to make a pie... tomorrow && family |
| 02:03.02 | dijib | so late tomorrow i might be around |
| 02:03.10 | hlieberman | Sweeeet. p3nguin, you got it working! :) |
| 02:03.15 | p3nguin | Nice. |
| 02:03.30 | p3nguin | Did you try it with various caller ID numbers and various PINs that do and do not match? |
| 02:03.42 | hlieberman | Yeah. Fails out unless the callerid matches, 100% of the time. |
| 02:03.44 | dijib | PINs? |
| 02:04.09 | dijib | im positive my mother works |
| 02:04.46 | p3nguin | I don't see any possible way that you can alter your caller id name randomly and make her see it change. |
| 02:04.55 | dijib | i can |
| 02:05.09 | p3nguin | Not unless she is on the same switch that VoIP.ms is sending you call to. |
| 02:05.09 | dijib | and im pretty sure my brother with his cell phone can too |
| 02:05.12 | hlieberman | I'm going to go cook dinner, then investigate FollowMe. Thanks again, p3nguin. I'll be back in ~1h. |
| 02:05.16 | dijib | she isnt |
| 02:05.24 | p3nguin | hlieberman: I'll still be here. |
| 02:05.26 | dijib | possibly indirectly |
| 02:05.48 | dijib | but i thought my father said his works with name, |
| 02:05.52 | p3nguin | I just don't see any way it is possible because CNAM isn't something that is transmitted. |
| 02:06.21 | dijib | anyways i was masqueradeing as the BATF |
| 02:06.30 | p3nguin | Using their phone number? |
| 02:06.33 | dijib | yes |
| 02:06.38 | dijib | the 800800 |
| 02:06.43 | p3nguin | That's why they saw the BATF name, probably. |
| 02:06.59 | p3nguin | You couldn't set the BATF phone number and my name and get them to see my name when you call. |
| 02:07.03 | dijib | it said TOLL FREE BT no? |
| 02:07.09 | p3nguin | Becaues CNAM isn't something that is transmitted. |
| 02:07.15 | dijib | TF |
| 02:07.17 | dijib | rather |
| 02:07.32 | dijib | makes sense |
| 02:07.35 | p3nguin | When I looked up the phone number in my LiDB, that's just what I have for the toll free number. |
| 02:07.51 | p3nguin | Apparently your mother's phone company actually had a CNAM entry for the number. |
| 02:08.00 | p3nguin | But I didn't. |
| 02:08.18 | p3nguin | If you called her from my phone number, she'd see my name calling her. |
| 02:08.36 | dijib | no. not that number, with my mother i set whatever i want as the name and she sees it. |
| 02:08.37 | p3nguin | Because her telco would do a lookup and give her the name associated with teh number. |
| 02:08.45 | dijib | cant go into it any further as this is logged chat |
| 02:09.22 | dijib | no sir.. im 100% certain she gets the names, ive setup different extensions that set the cid name and number and for her they work |
| 02:09.25 | dijib | 100% |
| 02:09.26 | dijib | no joke |
| 02:09.58 | p3nguin | If you aren't setting a random name that is unrelated to the number you're sending, you're doing the test incorrectly. |
| 02:10.24 | dijib | i understand that |
| 02:10.28 | dijib | but sitl works.. . |
| 02:10.38 | p3nguin | Good luck getting anyone to believe that. |
| 02:10.59 | dijib | as ive set it to my girlfriends name and number... .her number not having her name registered to it |
| 02:11.10 | dijib | and this works |
| 02:11.22 | p3nguin | Set it to your girlfriend's name and the police department's phone number. |
| 02:11.26 | p3nguin | See how that works out. |
| 02:11.48 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 02:12.12 | dijib | ill see if my mother can take a picture of her phone with your name 'p3nguin' set as caller ID |
| 02:12.19 | dijib | thats the test |
| 02:12.24 | p3nguin | :) |
| 02:12.32 | dijib | but not tonight... i gotta go shower.... |
| 02:12.38 | dijib | ill be back maybe in a bit |
| 02:16.36 | eva_02 | Guys, please look what's wrong with this config http://pastebin.com/raw.php?i=kpdfbSRa |
| 02:17.13 | eva_02 | i have error username mismatch, have <123123>, digest has <222222> when try to cal on 222222 |
| 02:18.20 | p3nguin | What are you trying to do? That config stuff is pretty crazy. |
| 02:18.39 | dlynes_laptop | eva_02, you didn't copy/paste those config files, either |
| 02:18.54 | dlynes_laptop | eva_02, you manually typed them into pastebin |
| 02:19.26 | eva_02 | hold on |
| 02:19.45 | p3nguin | Tell me what you're trying to do and I'll tell you why it doesn't work. |
| 02:19.53 | dlynes_laptop | eva_02, post the real configs, but scrub your passwords |
| 02:20.45 | eva_02 | i'm try forward all incmoning calls from one sip account to another external network |
| 02:21.16 | eva_02 | look like this: 123@sip.com --> my asterisk --> 999@coolsip.com |
| 02:21.32 | p3nguin | Are you sip.com? |
| 02:22.02 | eva_02 | no |
| 02:22.14 | p3nguin | How does the call from sip.com reach you? |
| 02:22.18 | eva_02 | my asterisk work like sip client for sip.com |
| 02:22.48 | p3nguin | Okay, where's the sip peer entry for sip.com in your sip.conf? |
| 02:22.51 | p3nguin | Don't see it. |
| 02:24.24 | eva_02 | but it work with only one peer |
| 02:27.23 | p3nguin | Do you have two accounts on sip.com? |
| 02:27.40 | eva_02 | here is real config http://pastebin.com/raw.php?i=aHEbrZPg |
| 02:28.20 | eva_02 | it's work as i need for 1234@sip.blabla.com |
| 02:28.23 | p3nguin | Do you have two accounts on sip.com? |
| 02:28.30 | eva_02 | p3nguin: yes |
| 02:28.45 | p3nguin | You'll have to use type=friend to match by username, otherwise you'll never be able to have two separate peers working correctly. |
| 02:29.22 | p3nguin | Using type=peer, it's matching by IP address and port. |
| 02:29.48 | eva_02 | ok, did i need fromuser= ? |
| 02:29.54 | p3nguin | Not likely. |
| 02:30.12 | p3nguin | For incoming calls, that is not relevant. |
| 02:31.40 | eva_02 | with type=friend i still have: check_auth: username mismatch, have <1234>, digest has <2222> |
| 02:31.55 | eva_02 | when call to 2222@sip.blabla.com |
| 02:32.25 | p3nguin | You set type=friend on both entrires, saved, and ran "sip reload" in the asterisk CLI? |
| 02:32.37 | eva_02 | p3nguin: yes |
| 02:32.56 | p3nguin | Then I have no idea how to fix it. |
| 02:34.31 | eva_02 | it work for 2222 when i move exten => 222222 to first in extentions.conf |
| 02:35.43 | p3nguin | The lines in extensions.conf have nothing to do with peer matching. |
| 02:37.55 | beta2k | Would anyone mind looking at my polycom phone configs? I can't figure out why they keep rebooting in the sip application. |
| 02:38.39 | eva_02 | p3nguin: and switch peers and register lines in sip.conf |
| 02:39.07 | p3nguin | The order of the register lines also has nothing to do with peer matching. |
| 02:39.26 | p3nguin | But the order of the peer definitions does when using type=peer. |
| 02:39.42 | p3nguin | But if you use type=friend, I'm sure it doesn't matter. |
| 02:41.28 | eva_02 | p3nguin: i use type=friend for both |
| 02:41.35 | p3nguin | You aren't trying to use both type=peer AND type=friend are you? |
| 02:42.03 | eva_02 | only friend |
| 02:42.27 | p3nguin | Then I have no idea why it doesn't work. type=friend should match username. |
| 02:42.54 | p3nguin | Do you ever need to send calls out to sip.blabla.com? |
| 02:43.03 | p3nguin | only inbound, right? |
| 02:43.56 | p3nguin | Change to type=user in both of them. It will accept inbound calls only, and should match by username not IP address. |
| 02:44.00 | eva_02 | yes only incoming calls from sip.blabla.com |
| 02:46.00 | *** join/#asterisk Maliuta (~nobusines@kiev.lusan.id.au) |
| 02:49.40 | p3nguin | Is that working better? |
| 02:52.08 | eva_02 | hold on, visitors came in library |
| 02:54.14 | eva_02 | p3nguin: YAAAA!!! type=user working! |
| 02:54.23 | eva_02 | p3nguin: thank you very much |
| 02:55.14 | p3nguin | With type=friend you should have seen the same results as type=user. I'm not sure why it wasn't working with friend. |
| 02:58.38 | *** join/#asterisk tzafrir_laptop (~tzafrir@212.179.75.202) |
| 03:03.42 | hlieberman | p3nguin, Looks like followme is realtime as of 1.6 |
| 03:03.51 | p3nguin | That could be useful. |
| 03:04.31 | *** join/#asterisk godmachine-x6 (~godmachin@unaffiliated/godmachine-x6) |
| 03:04.36 | p3nguin | But it's really not needed once you said you can get everyone to register their phone numbers with you. |
| 03:04.58 | p3nguin | When they register the number, you could just create the entry in followme.conf. |
| 03:05.11 | hlieberman | But how do I switch between them when needed? |
| 03:05.40 | p3nguin | You'd create an entry for every phone number that an on-call worker registers. |
| 03:06.29 | hlieberman | Right. |
| 03:06.37 | hlieberman | Oh, and just FollowMe to that entry. |
| 03:07.33 | p3nguin | Once the person calls in and authenticates, you'd just FollowMe() to their number rather than Dial() to their number directly. But you'll have to create a new extension or two for FollowMe() to have a way to get out. |
| 03:07.46 | p3nguin | Want me to write up the changes? |
| 03:08.41 | hlieberman | So I'd have the entry both in FollowMe() and in the authentication file. |
| 03:08.52 | p3nguin | Yes. |
| 03:08.57 | hlieberman | Sure, thanks. :) |
| 03:09.13 | p3nguin | Give me some time and I'll get 'er done. |
| 03:09.42 | hlieberman | Appreciate it. |
| 03:10.40 | p3nguin | Looks like I'm getting hit with a SIP botnet. |
| 03:13.12 | *** join/#asterisk nix8n82-phone (~AndChat@24.143.28.16) |
| 03:13.13 | p3nguin | Do you already have an outbound context? |
| 03:14.16 | *** join/#asterisk godmachine-x6 (~godmachin@unaffiliated/godmachine-x6) |
| 03:18.13 | hlieberman | I do. |
| 03:18.56 | *** join/#asterisk tm1000 (~tm1000@li251-245.members.linode.com) |
| 03:19.39 | p3nguin | http://pastebin.com/kvsaWNNY |
| 03:19.51 | p3nguin | I made a couple other adjustments in the auth section. |
| 03:21.15 | p3nguin | I thought of another sound file that could be helpful to a caller setting his number for callbacks. |
| 03:21.55 | p3nguin | After the GotoIf that checks the callerid, it could be helpful to play a sound file that says, "Your PIN does not match your CallerID number." |
| 03:23.14 | p3nguin | Like so: http://pastebin.com/BgKEPmC4 |
| 03:24.14 | p3nguin | That way if there is a failure when the person knows the PIN is entered correctly, there is an explanation. |
| 03:25.35 | hlieberman | Aha. Good idea. |
| 03:26.23 | hlieberman | It's =, not ==? |
| 03:26.59 | p3nguin | yeah |
| 03:27.12 | p3nguin | But feel free to test == if you'd like to see how it works. |
| 03:27.45 | hlieberman | I trust you. :) |
| 03:29.39 | p3nguin | I'm not sure why I used the reverse logic on that the first time. I have a habit of doing that and then changing it back and forth a few times until I settle on one. |
| 03:30.14 | p3nguin | But I think I like this logic better than the first way. |
| 03:30.47 | p3nguin | "If the callerid matches the accountcode, jump over the next two lines and continue." |
| 03:31.13 | p3nguin | "If it does not match, go to the next line, play the sound, then go back to the authentication line." |
| 03:31.17 | p3nguin | Makes sense to me! |
| 03:34.39 | p3nguin | Now your followme.conf should have an entry for each phone number registered, with the phone number as both the followmeid (found in the square brackets), and the number to dial. |
| 03:35.29 | p3nguin | You may find that you need to edit the followme sound files so it doesn't play certain parts that don't really apply in this case. |
| 03:35.53 | p3nguin | It's not really used for direct dialing like this, but we want to use the features it provides. |
| 03:42.31 | hlieberman | Right. |
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| 05:00.55 | SeRi | p3nguin, you avail? |
| 05:05.29 | SeRi | damn I lost access to my vociemail internally lol |
| 05:06.13 | p3nguin | What does that mean? |
| 05:06.59 | SeRi | If I dial my own ext it wont let me to the voicemail prompt |
| 05:07.13 | SeRi | I just get a bussy tone |
| 05:07.19 | SeRi | not sure what happen |
| 05:08.28 | p3nguin | It should ring your phone, then, if configured to do so, go to your voicemail. |
| 05:10.03 | SeRi | It does that but back before I enable internall calling I was able to dial my own ext and would get the voicemail prompt to enter a vmail box number and password |
| 05:10.37 | p3nguin | It sounds like you had it configured wrong before, and now you've fixed it. |
| 05:10.47 | SeRi | lol. |
| 05:11.15 | SeRi | ok so how can I get to the vmail options? |
| 05:13.17 | p3nguin | Create an extension which runs VoiceMailMain(). |
| 05:13.43 | SeRi | o ok. so it has to be a seperate ext? |
| 05:13.52 | SeRi | ok so i got confused there |
| 05:13.54 | p3nguin | Of course. |
| 05:14.04 | p3nguin | It wouldn't make any sense to have it part of your own extension. |
| 05:14.09 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 05:15.12 | SeRi | Like you said I had it broken before. fixed without knowing and confused my self |
| 05:15.16 | SeRi | what a cluster fuck |
| 05:15.17 | SeRi | lol |
| 05:16.46 | SeRi | is tired today... |
| 05:16.47 | p3nguin | Most people want their own extension to dial their phone, then go to their voicemail if there's no answer. |
| 05:17.36 | SeRi | make sense |
| 05:17.43 | *** join/#asterisk godmachine-x6 (~godmachin@unaffiliated/godmachine-x6) |
| 05:17.53 | SeRi | p3nguin, you ever heard of sipbri.com? |
| 05:18.02 | p3nguin | And that's by using VoiceMail(yourmailbox@voicemailcontext) |
| 05:18.06 | p3nguin | no |
| 05:23.14 | SeRi | got it.... check out sipbri.com when you get a chance and tell me your thoughts. I been thinking about using my google voice account via sip which is what sipbri does for your for free... they are able to connect to your google voice account and create a sip account that links to your google account.... |
| 05:24.01 | SeRi | I been using gv since grand central so I would like to use it at my office as well. |
| 05:24.21 | SeRi | so I been thinking about sipbri |
| 05:25.13 | p3nguin | Why do I need to use that when I can use gv directly on asterisk? |
| 05:27.28 | SeRi | in my case I cant. thats another service that I have issues trying to get out from comcast. |
| 05:28.25 | p3nguin | They're blocking xmpp also? |
| 05:29.45 | SeRi | not sure yet. I have been running wireshark and trying to capture shit at the firewall. |
| 05:29.53 | SeRi | I am building a case with comcast about it. |
| 05:29.56 | SeRi | the same with sip |
| 05:30.06 | p3nguin | That's denial of service. |
| 05:30.49 | *** part/#asterisk JohnTeddy (unstable@tor/regular/sid) |
| 05:39.16 | SeRi | yeap. the swear they are not doing it but they are. I also ask them about there phone service and they told me that there home phone service does not use the same technologies as sip but there proprietary shit.... |
| 05:39.48 | SeRi | which make s me belive that they will block 5060 for even better reasons. |
| 05:51.36 | ChannelZ | is this comcast consumer? |
| 05:54.26 | *** join/#asterisk Naikrovek (~mbluth@unaffiliated/naikrovek) |
| 06:04.36 | linuxgecko | i've been having issues with calling my google-voice line that my asterisk is hooked into. is ther a known issue with calls not getting to asterisk, or being mishandled? sometimes i get sent to g-voice voicemail, and soemtimes i get a non-ringing silence and then the call drops. but neither time, do i get a ring on the sip channel that is in the dialplan for that account. ideas? |
| 06:05.44 | linuxgecko | my asterisk is 1.8.5.0, |
| 06:06.54 | SeRi | ChannelZ, Yes. |
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| 06:59.23 | ChannelZ | linuxgecko: gv doesn't use SIP |
| 06:59.31 | *** join/#asterisk atan (~atan@unaffiliated/atan) |
| 06:59.40 | ChannelZ | You also have to program your dialplan to answer and dial 1 to accept the call |
| 07:00.56 | linuxgecko | ChannelZ: i'm using the g-voice howto on the wiki. it works for incomming calls, intermittently |
| 07:02.54 | ChannelZ | do you see the gtalk call on the console? The rest of your dialplan running? |
| 07:11.22 | linuxgecko | ChannelZ: i've not been able to SEE it fail, because usually it's me calling in from the PSTN, so i'm not at the asterisk box :) it's been working good for a while, and only recently become intermittent. |
| 07:13.33 | *** join/#asterisk tm1000 (~tm1000@li251-245.members.linode.com) |
| 07:25.00 | ChannelZ | Without being able to isolate the problem a little better, I can say Google likes to play with the protocol.. some things were a little busted in 1.8.5 IIRC. You might just try updating. |
| 07:31.24 | linuxgecko | ChannelZ: i might just do that. , and yeah, i've had to play with patches from time to time to keep things working smoothly with google playing musical protocols :-( |
| 07:33.14 | ChannelZ | You get what you pay for I guess |
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| 09:37.51 | mraky | hi is here somebody who is successfully sending sms between 2 sip clients connected to asterisk10 beta ? I am having problems, where in debug i see : [Oct 7 05:00:52] NOTICE[26076]: chan_sip.c:26363 sip_request_call: Asked to get a channel of unsupported format (nothing) while capability is (gsm|ulaw|alaw|h263|testlaw) |
| 09:37.51 | mraky | [Oct 7 05:00:52] WARNING[26076]: app_dial.c:2197 dial_exec_full: Unable to create channel of type 'SIP' (cause 58 - Bearer capability not available) |
| 09:37.51 | mraky | <PROTECTED> |
| 09:39.09 | mraky | or as least, if you can advice some web site with this topic - some examples of sms config in latest asterisk... |
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| 10:03.39 | bootc | hey folks, any Jira admins around? |
| 10:04.04 | bootc | I'd like to reactivate my old account I had on Mantis, but the email address is no longer active so I can't use the form |
| 10:04.54 | bootc | I found and fixed a bug in minivm, so I'd like to report it with an attached patch |
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| 10:19.40 | wdoekes2 | bootc: better wait until the USA is fully awake |
| 10:19.53 | bootc | yeah I get that impression also :-) |
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| 12:20.50 | zamba | i have a question about pattern matching.. in the asterisk book "the definitive guide", it says: "one other important thing to know about pattern matching is that if asterisk finds more than one pattern that matches the dialed extension, it will use the most speficic one (going from left to right)." |
| 12:21.03 | zamba | but doesn't asterisk match the first pattern? |
| 12:21.13 | zamba | even though a better match is found later? |
| 12:21.21 | zamba | potentially could be found later, i mean |
| 12:27.53 | *** part/#asterisk didnot (~didnot@unaffiliated/didnot) |
| 12:41.08 | kaldemar | zamba: define later. |
| 12:42.11 | kaldemar | if the two patterns are in the same context, then the one that is more specific is used. |
| 12:42.26 | *** join/#asterisk hehol (~Adium@ip-95-223-106-142.unitymediagroup.de) |
| 12:42.36 | kaldemar | includes are another story. |
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| 12:50.54 | zamba | kaldemar: ah, i thought it was top to bottom |
| 12:50.57 | zamba | kaldemar: ok, cool |
| 12:59.30 | beta2k | anyone around interested it looking at these configs? I've started over from scratch a couple times now and just can't find anything wrong that would cause my phones to keep rebooting |
| 13:01.22 | beta2k | This is <mac address>.cfg, http://paste.pocoo.org/show/489703/, <mac address>reg.cfg http://paste.pocoo.org/show/489704/ , server.cfg http://paste.pocoo.org/show/489705/ |
| 13:01.29 | beta2k | Oh yeah, their polycom ip430's |
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| 14:13.16 | p3nguin | linuxgecko: I'm having reliability issues on incoming calls, too. Sometimes I don't even get the prompt to press 1 to accept and 2 to go to voicemail, but it sits silently waiting for me to press one of those anyway. |
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| 14:18.53 | p3nguin | seri: I'm thinking you should really do some more network troubleshooting. I really don't believe they are blocking the stuff you think they are blocking. |
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| 14:38.15 | SeRi | p3nguin, you in? |
| 14:38.26 | p3nguin | yes |
| 14:39.15 | SeRi | I got one question... when I am in the phone talking to my brother at some point the conversationgets cut off and i get this |
| 14:39.17 | SeRi | WARNING[27760]: app_dial.c:2041 in dial_exec_full: Unable to create channel of type 'SIP' |
| 14:39.35 | SeRi | gets cut off* |
| 14:40.02 | p3nguin | I'd have to see more evidence that just that one line. |
| 14:40.12 | p3nguin | s/that/than/ |
| 14:40.17 | SeRi | ill turn on sip debug |
| 14:40.27 | SeRi | and try to catch it |
| 14:42.03 | SeRi | I will hit you up in a few.... am going to try and catch it. |
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| 15:14.54 | *** join/#asterisk cusco (~tralala@a79-168-174-232.cpe.netcabo.pt) |
| 15:14.55 | cusco | hi |
| 15:15.09 | cusco | I compilled 1.8.7 |
| 15:15.10 | cusco | root@velhadas:/usr/src/asterisk/asterisk-1.8.7.0# asterisk -vvvvvvvvvvvvvvvc |
| 15:15.10 | cusco | Illegal instruction |
| 15:15.16 | cusco | can someone help me? |
| 15:17.47 | p3nguin | How did you build and install it? |
| 15:18.32 | cusco | well I ./configured it, make menuselect and check that res_jabber was on, disabled nbsstream and morsecode |
| 15:18.35 | cusco | make |
| 15:18.36 | cusco | make install |
| 15:18.38 | cusco | make samples |
| 15:19.08 | p3nguin | And "asterisk -ddddddddvvvvg" does what? |
| 15:19.14 | cusco | hold |
| 15:19.20 | cusco | same |
| 15:19.28 | p3nguin | No other output? |
| 15:19.29 | cusco | want a strace paste? |
| 15:19.32 | cusco | no |
| 15:19.39 | p3nguin | That might show something helpful. |
| 15:20.07 | p3nguin | I've never seen that happen before. |
| 15:20.21 | cusco | neither have I |
| 15:20.30 | cusco | http://paste.debian.net/135234/ |
| 15:20.37 | cusco | however I have something different.. |
| 15:20.40 | cusco | im on a chroot |
| 15:21.10 | cusco | I did mount /dev and /dev/pts and /sys with bind options |
| 15:21.21 | cusco | but I used asteriswk 1.8 before |
| 15:21.31 | cusco | it was a snapshot I think |
| 15:21.39 | cusco | it was 1.8 branch |
| 15:24.32 | p3nguin | My only thought is that some resource that's needed is not available, but I don't really see the strace showing anything like that. |
| 15:26.11 | cusco | :| |
| 15:26.50 | p3nguin | Maybe a backtrace would help. |
| 15:27.48 | p3nguin | Is it dumping a core file? |
| 15:28.01 | cusco | didn't check |
| 15:28.18 | cusco | no |
| 15:29.22 | p3nguin | Can you run it with gdb and see if that shows anything that strace doesn't? |
| 15:29.51 | cusco | im not an experienced debugger lol |
| 15:30.42 | *** join/#asterisk puzzled (~patrick@53533D66.cm-6-4a.dynamic.ziggo.nl) |
| 15:31.05 | p3nguin | ~backtrace |
| 15:31.05 | infobot | backtrace is a debugging tool that is useful when trying to track down asterisk crashes (use -g with asterisk to generate a core, then read doc/backtrace.txt). See also https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace |
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| 16:04.19 | SeRi | p3nguin, it happen again |
| 16:04.29 | SeRi | i have some debug sessions |
| 16:04.41 | SeRi | can you take a look at them? |
| 16:08.25 | p3nguin | pastie pastie |
| 16:08.57 | p3nguin | |
| 16:09.12 | cusco | p3nguin: now I noticed and dmesg shows: |
| 16:09.14 | cusco | err |
| 16:09.20 | cusco | [67031.373199] asterisk[28938] trap invalid opcode ip:52afa3 sp:7fff790a0130 error:0 in asterisk[400000+1ca000] |
| 16:09.30 | cusco | does it sound familiar? |
| 16:10.18 | p3nguin | Not to me. |
| 16:10.34 | cusco | all I see is http://www.mail-archive.com/asterisk-dev@lists.digium.com/msg49953.html but no reply |
| 16:10.41 | SeRi | p3nguin, can I link you to a pb? |
| 16:10.57 | p3nguin | You've used pastebin before, haven't you? |
| 16:11.05 | SeRi | yes |
| 16:11.10 | p3nguin | Then you know how it works. |
| 16:11.15 | SeRi | just asking if you are avil to look at them |
| 16:11.49 | p3nguin | If you never paste them, I can never look at them. |
| 16:11.58 | SeRi | p3nguin, http://pastebin.com/VLkb0SpB |
| 16:14.38 | p3nguin | What's the sip2.rem...vices.net host that you're registering with? |
| 16:14.41 | WIMPy | cusco: I don't see how that is related to asterisk. Either you have a broken lib, build system or hardware. |
| 16:18.49 | SeRi | p3nguin, Thats that sipbri. it does not work. I took all that out... but the drop out happen even before I add it them to the sip.conf |
| 16:21.34 | p3nguin | I'm interested to see your entire sip.conf. Hide your passwords when pasting. |
| 16:22.25 | SeRi | yes sr. |
| 16:26.34 | SeRi | p3nguin, http://pastebin.com/arUdeH78 |
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| 16:27.55 | *** part/#asterisk ArtemMakhutov (~ArtemMakh@ip-95-223-215-71.unitymediagroup.de) |
| 16:29.11 | p3nguin | Your contexts are all jacked up. |
| 16:29.29 | p3nguin | That's not the cause of your dropped calls, but a fact nonetheless. |
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| 16:34.40 | SeRi | ok what can I fix? |
| 16:34.49 | SeRi | or |
| 16:34.55 | SeRi | what do I need to fix? :) |
| 16:35.02 | p3nguin | You're sharing a single context for everything. That's bad. |
| 16:35.43 | SeRi | ooo :/ |
| 16:35.58 | SeRi | didnt know you could separate them |
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| 16:38.27 | Sheeplet | lo lo |
| 16:39.49 | p3nguin | seri: http://pastebin.com/GtbuvHg4 |
| 16:40.09 | p3nguin | crap, messed up something |
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| 16:40.38 | p3nguin | fixed http://pastebin.com/xZZXTcNu |
| 16:41.03 | SeRi | ok |
| 16:41.40 | p3nguin | Ah, need to fix one more thing that I didn't catch was wrong. |
| 16:41.52 | SeRi | p3nguin, context=misc_calls do I leave that as is? or do I put a context in there? |
| 16:41.58 | SeRi | ok |
| 16:42.10 | p3nguin | http://pastebin.com/ndnCsUKm |
| 16:42.18 | p3nguin | CallerID does not have quotes. |
| 16:42.30 | p3nguin | that _IS_ a context. |
| 16:42.42 | SeRi | oo ok |
| 16:42.57 | p3nguin | Make sure you create it in extensions.conf. |
| 16:44.08 | SeRi | do I put my voip.ms plan under it or just leave it empty? sorry for the dumb question just trying to understand the changes. |
| 16:44.20 | p3nguin | One step at a time. |
| 16:44.27 | SeRi | ok |
| 16:44.29 | p3nguin | Look at the voipms peer entry. |
| 16:44.33 | p3nguin | Does it have a context? |
| 16:44.49 | p3nguin | Does the context appear to be a suitable name for the peer? |
| 16:45.17 | SeRi | ah I see. |
| 16:45.34 | p3nguin | Put any extensions that voipms needs to access under that context. |
| 16:45.48 | SeRi | Yes I have that under the dial plan |
| 16:45.55 | p3nguin | The only extensions voipms needs to have access to would be your phone numbers from voipms. |
| 16:46.01 | SeRi | on sec five me one min I will post my dial plan |
| 16:47.47 | SeRi | let me save the changes in sip.conf and ill move to the dial plan |
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| 16:51.02 | SeRi | p3nguin, http://pastebin.com/XwuR6iwW |
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| 16:52.24 | p3nguin | What's the 713 number for? fax only? |
| 16:53.03 | SeRi | yes sr. |
| 16:53.21 | p3nguin | Now I've been over this before, but I'm going to do it one more time. |
| 16:53.56 | SeRi | p3nguin, I saw this one coming :) I know you corrected it before but it did not work. faxes would not deliver for some reason.... |
| 16:55.09 | SeRi | also the include is bad juju I was lazy about that one sorry :( |
| 16:55.20 | *** join/#asterisk luckman212 (~irc@pool-108-41-8-176.nycmny.fios.verizon.net) |
| 16:57.30 | p3nguin | I'm going to fix everything for you. All you have to do is copy what I'm doing and paste it into an empty extensions.conf. |
| 16:57.40 | SeRi | yes sr. |
| 16:59.09 | bootc | any Jira admin available? I'd like to unlock my old Mantis account but the email is no longer valid |
| 16:59.56 | bootc | seraphie: I was told you may be able to help? |
| 17:00.27 | p3nguin | seri: http://pastebin.com/kE6r7gr6 |
| 17:01.02 | SeRi | working on it |
| 17:02.15 | p3nguin | This is a very basic setup. I would improve on a lot of things if it were mine. |
| 17:03.00 | p3nguin | One thing, for example, would be using DIALSTATUS to determine what happens after the call to your phones. You only go to unavailable voicemail, where I would have other choices. |
| 17:04.18 | p3nguin | And now that I know you don't need fax detection because you have a dedicated fax number, you don't need to enable fax detection in sip.conf. |
| 17:04.33 | p3nguin | So I'd go back and disable that. |
| 17:05.01 | SeRi | even if is been recived via asterisk and not a real fax machine? |
| 17:05.30 | p3nguin | Fax detection is for one thing: detect a fax tone on a voice line, and send the call to a fax extension instead of to a phone. |
| 17:06.13 | p3nguin | But you have one voice line which will not have faxes, and one fax line which will not take voice calls. |
| 17:06.18 | p3nguin | Fax detect is not required. |
| 17:06.52 | SeRi | got it |
| 17:07.16 | p3nguin | I've configured the extensions according to those needs. |
| 17:07.28 | p3nguin | And fax detection can be disabled now. |
| 17:08.03 | SeRi | p3nguin, I tryied to dialed out and all I get was a woman saying "that number" |
| 17:08.19 | p3nguin | Did you sip reload and dialplan reload after making all changes? |
| 17:08.36 | SeRi | yes sr |
| 17:08.46 | p3nguin | Then you must have changed something that I pasted. |
| 17:09.13 | SeRi | nothing at all. |
| 17:09.13 | p3nguin | I didn't change the "meat" of your extensions, just the design. |
| 17:09.25 | p3nguin | Show me a log of a failure. |
| 17:09.31 | SeRi | I can recive calls just fine :) |
| 17:09.35 | SeRi | on sec |
| 17:16.33 | SeRi | p3nguin, http://pastebin.com/Vg2GgTXt |
| 17:18.49 | p3nguin | That looks like a successful call to me. |
| 17:19.23 | SeRi | well I dont know what to tell you all of my calls go to a voice that says "that number" and just blanks out |
| 17:19.33 | SeRi | its a womans voice |
| 17:19.33 | p3nguin | core set verbose 3 |
| 17:19.36 | SeRi | ok |
| 17:19.39 | p3nguin | Make a call. Show me. |
| 17:21.29 | SeRi | p3nguin, http://pastebin.com/KvdxjvKZ |
| 17:22.32 | SeRi | ignore the manager part. thats my ami script |
| 17:22.52 | SeRi | did you just try to call me? |
| 17:22.58 | p3nguin | Yes |
| 17:23.09 | p3nguin | Call some other number. |
| 17:23.17 | SeRi | hahaha sorry I dont answer priv numbers |
| 17:23.27 | p3nguin | I didn't need an answer. |
| 17:23.34 | p3nguin | I just wanted to see if the number was bad. |
| 17:23.36 | SeRi | I did I called a shit load of numbers with the same results |
| 17:24.09 | p3nguin | The call is making it all the way to VoIP.ms and it is being bridged. |
| 17:25.02 | SeRi | I have nat enable in voip.ms settings if that matters |
| 17:25.11 | p3nguin | It's wrong. |
| 17:25.12 | p3nguin | They are not behind nat. |
| 17:25.19 | p3nguin | I specifically set it to nat=no for a reason. |
| 17:25.39 | p3nguin | I told you it was copy and paste, drop-in. |
| 17:26.00 | p3nguin | Let me give you a number to call... |
| 17:26.56 | SeRi | o no I meant at voip.ms website |
| 17:27.03 | p3nguin | oh |
| 17:27.07 | *** join/#asterisk Ad-Hoc (~nimbus@athedsl-377528.home.otenet.gr) |
| 17:27.10 | p3nguin | Turn that off. |
| 17:27.11 | SeRi | they have a setting in there site if you are behind a nat router to enable it in the site |
| 17:27.16 | SeRi | ok |
| 17:27.54 | SeRi | p3nguin, its working now |
| 17:28.02 | SeRi | I add it fromuser=myuid |
| 17:28.16 | p3nguin | if you set fromuser it works again? |
| 17:28.20 | SeRi | Yes sr |
| 17:28.28 | SeRi | in the [voipms] context |
| 17:28.28 | p3nguin | That's weird... I do not use fromuser on voipms. |
| 17:28.55 | p3nguin | I checked my own config to make sure you didn't need it. |
| 17:28.56 | SeRi | Mhhhhh weird indeed |
| 17:29.10 | p3nguin | Had I used it, I would have made sure it was in yours. |
| 17:30.27 | SeRi | I know you would. It just weird that I have to use settings that others dont. whats even weirder is that others have told me the same thing |
| 17:30.47 | SeRi | I have settings that others dont use.... |
| 17:30.50 | p3nguin | Maybe there's some other setting in the portal that makes a difference. |
| 17:30.58 | SeRi | ah! |
| 17:31.03 | SeRi | maybe is that nat bs |
| 17:31.26 | *** join/#asterisk luckman212 (~irc@pool-108-41-8-176.nycmny.fios.verizon.net) |
| 17:31.28 | p3nguin | I have my NAT enabled in the portal, but I bet it will still work the same without it. |
| 17:32.06 | SeRi | :/ |
| 17:33.14 | p3nguin | I'll have to test that later. |
| 17:33.56 | SeRi | no sweat. |
| 17:34.07 | SeRi | p3nguin, can you help me stup gv using asterisk? |
| 17:34.17 | p3nguin | maybe |
| 17:34.28 | SeRi | please? :) |
| 17:34.40 | SeRi | well if you have time |
| 17:34.42 | SeRi | :) |
| 17:35.21 | SeRi | p3nguin, what is the default core verbose so I can put it back? |
| 17:35.32 | p3nguin | 0 |
| 17:35.42 | SeRi | Thanks |
| 17:35.49 | p3nguin | Verbosity was 0 and is now 3 |
| 17:36.06 | p3nguin | https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google |
| 17:36.21 | SeRi | Got that link |
| 17:36.27 | p3nguin | Follow that wiki, and when it doesn't work, tell me what part isn't working. |
| 17:36.41 | SeRi | ok |
| 17:37.54 | SeRi | externip is my ip right? |
| 17:38.12 | p3nguin | You'd be better of using externhost and the dynamic host name. |
| 17:38.27 | SeRi | yes sr |
| 17:43.53 | seraphie | bootc: PM me your username and old email address |
| 17:47.03 | *** join/#asterisk irroot (~irroot@197.106.68.2) |
| 17:47.27 | *** join/#asterisk andresmujica (~andres@ubuntu/member/andresmujica) |
| 17:54.51 | depressed | is it possible to run asterisks with 128MB ram |
| 17:55.32 | p3nguin | Asterisk uses about 34MB for me. |
| 17:57.06 | WIMPy | How did you get it that small? |
| 17:57.15 | p3nguin | *shrug* |
| 17:57.23 | p3nguin | I just built it and installed it. |
| 17:57.26 | WIMPy | Min is at 144M. |
| 17:57.28 | WIMPy | +e |
| 17:58.11 | WIMPy | +72M for LCR. And I'm not even using 64-Bit. |
| 17:58.16 | p3nguin | Do you go by RES or VIRT? |
| 17:58.49 | p3nguin | If you go by VIRT, mine is almost 70MB. |
| 17:58.59 | WIMPy | Virt. Res is 114+69 |
| 18:00.17 | p3nguin | When I used 1.4, it was even less. |
| 18:00.31 | WIMPy | Yes, it has become a lot bigger. |
| 18:01.13 | SeRi | p3nguin, http://pastebin.com/iEs9cZQU |
| 18:01.13 | WIMPy | That's the two biggest daemons running here, by far. |
| 18:02.50 | p3nguin | extension not found in context 'google'. |
| 18:02.59 | p3nguin | You do know what that means, right? |
| 18:05.00 | p3nguin | It means you should add exten => _NXXNXXXXXX,1,Goto(1${EXTEN},1) to your google outbound context. |
| 18:09.25 | p3nguin | My largest ones are asterisk, postgres, and fail2ban. |
| 18:11.04 | SeRi | p3nguin, thats what I get for not reading the outgoing plan in google. sorry |
| 18:11.13 | *** join/#asterisk ferdna (~ferdna@67.10.220.35) |
| 18:20.40 | SeRi | hahaha the module is not there buhahahaha |
| 18:20.55 | p3nguin | Which one? |
| 18:21.05 | p3nguin | chan_gtalk, I'd guess. |
| 18:21.08 | SeRi | yes sr |
| 18:21.26 | p3nguin | Go back to your source tree and make menuselect |
| 18:21.42 | p3nguin | Add the channel module, then make |
| 18:21.50 | SeRi | yeap time for some recompilation. ill get the module compiled and move it in place |
| 18:22.20 | p3nguin | If you didn't change anything else, it will just compile the new channel. |
| 18:22.45 | SeRi | yes sr. |
| 18:32.53 | SeRi | p3nguin, ok so I did some further testing on dialing in and out and the calls get cut off |
| 18:33.06 | *** join/#asterisk hlieberman (~hlieberma@about/crypto/member/hlieberman) |
| 18:33.14 | hlieberman | Hello, everyone. |
| 18:33.31 | hlieberman | p3nguin, you still around by any chance, so I can impose on you for more help? :) |
| 18:33.39 | p3nguin | Yes. |
| 18:33.46 | hlieberman | How goes, man? |
| 18:33.50 | p3nguin | Typical. |
| 18:36.07 | hlieberman | Hang on, I may have solved it. One sec. :) |
| 18:39.08 | hlieberman | Nope. I'm getting the error No such extension/context SIP@default while calling Local channel when using FollowMe() |
| 18:40.32 | hlieberman | Which is strange, since there's no entry that even looks close to that anywhere I can find. |
| 18:40.54 | p3nguin | In your followme.conf, you've set context to be your context where your outbound extension is created? |
| 18:40.58 | hlieberman | Do I need an outgoing context in followme.conf as well? |
| 18:41.29 | hlieberman | In my followme.conf, I set the number equal to SIP/<number>@flowroute, which is the way it looks in extensions |
| 18:42.14 | p3nguin | If your outbound context is something like voipms-outbound, in followme.conf, you'd set context=>voipms-outbound |
| 18:42.25 | p3nguin | And you'd set number to be the number you're trying to call. |
| 18:42.37 | p3nguin | [3145551212] |
| 18:42.44 | p3nguin | context=>voipms-outbound |
| 18:42.52 | p3nguin | number=>3145551212,30,1 |
| 18:43.32 | p3nguin | If your outbound context is flowroute-outbound, make the necessary adjustment. |
| 18:43.47 | hlieberman | I can't define context under [general]? |
| 18:44.30 | p3nguin | As far as I know, no. |
| 18:45.26 | hlieberman | Do I then need to define the outgoing context in followme.conf as well as in extensions? |
| 18:45.52 | p3nguin | It should be exactly as I've described. |
| 18:48.08 | hlieberman | Ahh, I see. I was missing the 1 in the number definition. |
| 18:48.52 | p3nguin | The trailing 1 is optional. |
| 18:49.07 | p3nguin | number=>3145551212,30 would be fine. |
| 18:49.36 | *** join/#asterisk irroot (~irroot@197.104.193.116) |
| 18:50.00 | hlieberman | No, I mean, I needed to prepend the number with 1, or it wasn't matching anything in the outgoing context. |
| 18:50.16 | p3nguin | Then you should consider adjusting your outbound extensions to allow 10-digit dialing. |
| 18:50.31 | p3nguin | That's going to cause an inconsistency if you don't. |
| 18:50.46 | p3nguin | Up to this point, we've been focusing on 10-digit NANP numbers. |
| 18:50.46 | hlieberman | I have... exten => _1NXXXXXXXXX,1,Dial(SIP/${EXTEN}@flowroute) |
| 18:50.56 | hlieberman | I just used exactly what my provider gave me. |
| 18:51.12 | p3nguin | exten => _NXXNXXXXXX,1,Goto(1${EXTEN},1) |
| 18:51.15 | p3nguin | add that. |
| 18:51.33 | p3nguin | And you should use SIP/flowroute/${EXTEN}) |
| 18:51.43 | p3nguin | Dial(SIP/flowroute/${EXTEN}) |
| 18:51.47 | hlieberman | What does the N stand for? |
| 18:51.58 | p3nguin | numbers 2-9 |
| 18:52.49 | p3nguin | exten => _1NXXNXXXXXX,1,Dial(SIP/flowroute/${EXTEN}) |
| 18:52.56 | p3nguin | exten => _NXXNXXXXXX,1,Goto(1${EXTEN},1) |
| 18:53.06 | p3nguin | That's what I'd use. |
| 18:53.28 | hlieberman | It also doesn't seem to be hanging up on one end of the call when the other hangs up. |
| 18:54.06 | p3nguin | okay... |
| 18:54.08 | p3nguin | exten => _1NXXNXXXXXX,1,Dial(SIP/flowroute/${EXTEN}) |
| 18:54.10 | p3nguin | exten => _1NXXNXXXXXX,n,Hangup() |
| 18:54.12 | p3nguin | exten => _NXXNXXXXXX,1,Goto(1${EXTEN},1) |
| 18:55.10 | hlieberman | Brilliant. |
| 18:58.32 | *** join/#asterisk wonderworld (~ww@port-92-201-112-174.dynamic.qsc.de) |
| 19:00.40 | hlieberman | Hm. It's still not hanging up on the person that the system calls. |
| 19:01.32 | hlieberman | http://pastebin.com/FxnRVwGR is my outgoing context. |
| 19:02.44 | p3nguin | You almost copied what I wrote for you. |
| 19:03.36 | hlieberman | Almost? I tried to do it exactly. What did I miss? |
| 19:03.46 | p3nguin | You say it's not hanging up the person called... what happens? It just sits there until the callee disconnects? |
| 19:04.11 | p3nguin | Your pattern is skewed slightly. |
| 19:04.22 | p3nguin | _1NXXNXXXXXX |
| 19:04.48 | hlieberman | So, the person calls the system. The system calls the other person and asks if you want to be connected. If you decline, it will tell the original caller that it can't find anyone and hang up on them. But the person that the system calls just gets silence, without a hangup, forever. |
| 19:05.19 | p3nguin | That doesn't seem like normal FollowMe operation. |
| 19:05.27 | hlieberman | No, it doesn't. |
| 19:05.49 | p3nguin | I would think it would go to the next step, which is Hangup, and that would end both legs of the call. |
| 19:06.18 | hlieberman | It hangs up the first one, but not the second. |
| 19:10.14 | hlieberman | Is there anything I can provide that might give you an idea? Verbose and debug to 3, and transcrpt of the output of an entire call? |
| 19:10.54 | *** join/#asterisk Azrael808 (~peter@cpc17-walt12-2-0-cust657.13-2.cable.virginmedia.com) |
| 19:13.28 | *** join/#asterisk Tim_Toady (~fuzzy@178.128.1.13.dsl.dyn.forthnet.gr) |
| 19:16.21 | hlieberman | Or maybe is it some weirdness from the POTS system? |
| 19:16.27 | hlieberman | PSTN, I mean? |
| 19:17.31 | p3nguin | I couldn't say what would cause it. See if the channel is still open after the caller disconnects. "core show channels" and "sip show channels" |
| 19:18.08 | hlieberman | It says there are 20 active channels under core show channels. |
| 19:18.21 | p3nguin | Does that sound accurate? |
| 19:18.35 | hlieberman | No, there should be 0. |
| 19:19.05 | ChannelZ | It's so hard to say goodBYE |
| 19:19.51 | hlieberman | http://pastebin.com/ingzLF2L is a transcript of the complete call. |
| 19:21.39 | hlieberman | Right now, the system is completely inactive except for my testing, so I would expect 0. |
| 19:22.53 | *** join/#asterisk mfairchild (~mfairchil@doublem.xs4all.nl) |
| 19:26.13 | hlieberman | I'm not... well, I don't understand a lot, but I really don't understand why that outgoing class isn't falling through to the Hangup(). |
| 19:26.28 | hlieberman | Maybe it thinks it's still talking, for some reason? Followme is doing something strange? |
| 19:26.39 | p3nguin | Where did the n come from in line 13? |
| 19:26.52 | p3nguin | FollowMe("SIP/flowroute-0000003d", "<ON-DUTY-NUMBER>,n") |
| 19:27.26 | p3nguin | You decided to use option n rather than options a and d? |
| 19:27.33 | hlieberman | Yes. |
| 19:27.35 | p3nguin | okay |
| 19:27.50 | hlieberman | Because we didn't need the name recorded. |
| 19:27.52 | *** join/#asterisk f2knight (~ben@c-76-105-136-103.hsd1.or.comcast.net) |
| 19:28.15 | hlieberman | Do you think that could be the problem? |
| 19:28.22 | p3nguin | no |
| 19:28.46 | p3nguin | I was just curious where it came from, trying to make sure it wasn't an extraneous option that was misplaced. |
| 19:29.19 | f2knight | Q: When using asterisk Realtime, should changing the password in the DB take effect with in say 2 or 3 min? or should it require a restart of asterisk? Seems the only way it is taking is if I restart So I think I might just be missing an option to set |
| 19:32.10 | hlieberman | p3nguin, If it would help, I can set you up with a passcode and in the followme.conf file, and I can have the system ring you. If that would help debugging. |
| 19:33.29 | *** join/#asterisk wonderworld (~ww@port-92-201-112-174.dynamic.qsc.de) |
| 19:35.08 | *** join/#asterisk didnot (~didnot@unaffiliated/didnot) |
| 19:35.26 | p3nguin | I doubt it would make much difference. |
| 19:37.19 | p3nguin | Maybe you can just get people to hang up their phone after they decide to not take the call. |
| 19:37.31 | hlieberman | I can do that, certainly. |
| 19:37.38 | hlieberman | Hm. |
| 19:37.51 | hlieberman | Both calls that remain in core show channels are Local/<on-call number>@default |
| 19:37.59 | hlieberman | One is in state ring, one is in state down. |
| 19:38.00 | p3nguin | I can't imagine someone would get a call, reject it by pressing 2, and then stay on the line. |
| 19:38.21 | p3nguin | hangup those channels. |
| 19:38.55 | hlieberman | How? |
| 19:38.58 | p3nguin | channel request hangup ... |
| 19:40.19 | hlieberman | Is there a way to get the non-truncated name of the channels? |
| 19:40.35 | p3nguin | channel request hangup <tab key> |
| 19:41.05 | p3nguin | Are you needing the name for something else? |
| 19:41.07 | hlieberman | Requested hangup on both, but they are both still active. |
| 19:44.38 | p3nguin | That's irritating when that happens. |
| 19:44.38 | hlieberman | Strange. What part of the extension is calling Local/ anything? |
| 19:44.43 | *** join/#asterisk seraphie (~erin@75.76.38.159) |
| 19:45.11 | p3nguin | FollowMe probably uses a local channel: Local/${number}@${context} |
| 19:45.14 | hlieberman | Yeha. |
| 19:45.16 | hlieberman | Yeah. |
| 19:45.25 | *** join/#asterisk [sr] (~kvirc@pal-213-228-140-150.netvisao.pt) |
| 19:45.31 | [sr] | yellow my friends |
| 19:45.34 | hlieberman | Maybe there's a way to get around that by specifying something else in the followme.conf? Have it hit SIP/flowroute/<number> directly? |
| 19:46.02 | WIMPy | [sr]: What's not green? |
| 19:46.09 | p3nguin | I doubt it. FollowMe() works correctly here, so I don't know. |
| 19:46.26 | p3nguin | Consider upgrading to the most recent version of Asterisk and see how it behaves. |
| 19:46.41 | [sr] | WIMPy: well... 'cause lots of sun here (few hours ago) :p |
| 19:47.05 | p3nguin | If I press 2 when asked to take the call, the call ends for me and returns to dial plan. |
| 19:47.23 | WIMPy | We have some really annoying cold wind. Already feels like winter :-( |
| 19:47.59 | [sr] | we going to have sun till next week |
| 19:48.02 | [sr] | our summer is now |
| 19:48.42 | [sr] | but i miss rain, i admit (not the cold) |
| 19:49.58 | [sr] | anyway we're on the best month of the year... :p |
| 19:51.28 | hlieberman | p3nguin, Eh. No big deal, I suppose. |
| 20:10.12 | *** join/#asterisk darkbasic (~quassel@host37-245-static.119-2-b.business.telecomitalia.it) |
| 20:36.05 | f2knight | hlieberman, p3nguin, just trying to catch up with what the issues is... ? |
| 20:36.20 | hlieberman | It seems to be working pretty well, f2knight. |
| 20:37.03 | hlieberman | I think I managed to fix the problem, p3nguin. I think it's a bug with how the default context is handled in Followme. Looks like specifying the context in each individual... context in followme.conf fixes the problem. |
| 20:37.17 | hlieberman | Well, it fixes the channels hanging about. It still doesn't hang up right, but I think you're right - non issue. |
| 20:37.20 | f2knight | hlieberman, okay .. just noticed you mentioning local channels.. .had some issues a while ago with other context functions not working as 'expected' Lief was kind enough to point me to the /n switch needed in local dial string to make it act as a normal channel. |
| 20:37.34 | hlieberman | The problem is, this is all inside the followme app. |
| 20:37.58 | f2knight | hlieberman, out of curisioty what was the 'issue'? |
| 20:38.37 | hlieberman | The system uses followme to call the on-duty person. If the person calling into the system hangs up, or the on-duty person refuses the call, it disconnects the first leg of the call into the system, but leaves the second one open. |
| 20:39.13 | f2knight | hlieberman, try adding the /n option to the dial command with the local channel |
| 20:39.31 | hlieberman | There isn't a dial command. It's just FollowMe() |
| 20:40.37 | f2knight | https://wiki.asterisk.org/wiki/display/AST/Local+Channel+Modifiers |
| 20:40.44 | f2knight | how are you calling the followme app |
| 20:41.23 | hlieberman | There isn't a dial command. It's just FollowMe() |
| 20:41.26 | hlieberman | er |
| 20:41.32 | hlieberman | FollowMe(${medical},n) |
| 20:42.21 | f2knight | Personsally I do not use follow me app. at all. I just store the value in to be called in astdb and do a dialplan look up with the DB statement. Works every time. And just use a basic macro/context to support 1 to accept 2 to voicemail 3 to forward. |
| 20:43.09 | f2knight | I found lots of things not 'working' in early versions of followme and found that its just as easy to do the same in the dialplan. |
| 20:45.46 | *** join/#asterisk quid246 (~quid246@99.228.252.229) |
| 20:46.03 | f2knight | hlieberman, have you posted your dialplan? |
| 20:46.21 | hlieberman | I've posted an older version a while ago. I can post the latest. Hang on a tic'. :) |
| 20:46.53 | hlieberman | http://pastebin.com/ZLXiJ8hi |
| 20:47.26 | quid246 | Hmm, think I found a bug in 1.6.20... Set(test=$[((0.0046 * 30) / 60) + ((0.0046 * 0) / 60)]) results in correct value but * complains "ast_expr2.y:1458 op_times: overflow". |
| 20:48.37 | quid246 | I think it has to do with the mix of Ints and floats |
| 20:53.38 | f2knight | hlieberman, correct me if I am wrong.. The person checking in supplies a single number for there call back correct.? You are looking up that persons callback and using that 'number' as a variable to lookup the followme context.. what is in the followme.? I presume a list of numbers you are keeping updated by hand? |
| 20:53.50 | hlieberman | Correct. |
| 20:54.17 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 20:56.22 | f2knight | hlieberman, okay so this puzzels me... if your maintaing the followme app by hand (which most people seemed to need to do) then the there is not a lot of reasons to use it. (not that I obviously see anyways.) Depending on what is in the Followme.conf for each of these 'assigned' numbers... |
| 20:56.51 | hlieberman | I'm just using it as a short-cut for the macro/context accept/reject thing. |
| 20:57.23 | hlieberman | And to keep the fallback number for who to try in the event the first person doesn't answer. |
| 21:01.33 | f2knight | http://pastebin.com/8zPFxhbz |
| 21:02.59 | f2knight | The sample I posted a modified line.. is uses the number supplied by the person who signed in .. calls it for 20 seconds. If they do not answer it dials the next line where it recalls the person(in this example) and also calls another number at the SAME time. who ever answers gets the call. |
| 21:03.17 | f2knight | When either party hangs up you will have your hangup caught. |
| 21:06.28 | f2knight | If you want call screening you simply add U(callee-screen,s,1) to the dial options. (where the TtWw are at ) |
| 21:06.59 | f2knight | the U() passes it to a context if you do a M() it passes it to a marcro. I have a very generic macro I use if you want a copy |
| 21:17.27 | *** join/#asterisk ChrisInSydney (~Chris@60-242-81-231.tpgi.com.au) |
| 21:19.41 | hlieberman | I'd love one. |
| 21:30.56 | hlieberman | f2knight, If you've got it, I'd love it. |
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| 21:54.50 | mraky | hello guys - i am having problems to get to work sending SMS message between 2 sip clients connected to asterisk 10beta... is somebody using it ? I want to get some advices, will tell more details... |
| 22:06.31 | beta2k | anyone around interested it looking at these configs? I've started over from scratch a couple times now and just can't find anything wrong that would cause my phones to keep rebooting |
| 22:06.39 | beta2k | This is <mac address>.cfg, http://paste.pocoo.org/show/489703/, <mac address>reg.cfg http://paste.pocoo.org/show/489704/ , server.cfg http://paste.pocoo.org/show/489705/ |
| 22:06.44 | beta2k | Oh yeah, their polycom ip430's |
| 22:07.32 | beta2k | They hit running sip.ld then reboot after a few minutes |
| 22:09.43 | *** join/#asterisk mraky_1 (~mraky@ip-94-112-113-65.net.upcbroadband.cz) |
| 22:12.49 | mraky_1 | or if you can tell me some site with examples/docs describing sip SMS in asterisk10beta ... official docs on www are just few rows... |
| 22:12.51 | mraky_1 | thx |
| 22:23.27 | f2knight | hlieberman, sorry was away I will PB it for you |
| 22:23.40 | hlieberman | Thanks, f2knight. |
| 22:24.25 | f2knight | hlieberman, http://pastebin.com/jz2Tz2tJ |
| 22:25.36 | f2knight | hlieberman, I have moved to writing most my setup in Python using AGI/FastAGI because it is a heck of a lot easier to create logic. but I still use a few macros left in the Dialplan for things like screening and basic call routing |
| 22:28.27 | f2knight | hlieberman, call it on your dialplan like so ... dial(sip/555,40,M(screen)) |
| 22:51.41 | hlieberman | So it would be... |
| 22:52.03 | hlieberman | Dial(SIP/flowroute/${medical},20,TtWwM(screen))? |
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