| 00:02.11 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 00:04.26 | *** join/#asterisk r33dtard (~r33dtard@gateway/tor-sasl/r33dtard) |
| 00:22.55 | *** join/#asterisk r33dtard (~r33dtard@gateway/tor-sasl/r33dtard) |
| 00:23.48 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 00:29.22 | *** join/#asterisk tm1000 (~tm1000@li251-245.members.linode.com) |
| 00:34.54 | p3nguin | fleixius: It's some type of an ID, not the conference room number. |
| 00:42.16 | *** join/#asterisk CrossWired_ (~chatzilla@65.210.186.34) |
| 01:02.59 | *** join/#asterisk tm1000 (~tm1000@li251-245.members.linode.com) |
| 01:09.19 | SeRi | p3nguin, you here? |
| 01:09.25 | p3nguin | Yes. |
| 01:09.39 | SeRi | I took my iphone and made it a sip phone |
| 01:09.39 | *** join/#asterisk tm1000 (~tm1000@173.255.241.245) |
| 01:09.42 | SeRi | works preatty well |
| 01:09.56 | SeRi | You will like it if thats what you are trying to do |
| 01:10.05 | SeRi | using siphon |
| 01:10.23 | p3nguin | I haven't used that one... I've tried all the other phones on my iPod touch. |
| 01:12.09 | *** join/#asterisk moy (~moy@173.239.155.74) |
| 01:12.29 | *** join/#asterisk r33dtard (~r33dtard@gateway/tor-sasl/r33dtard) |
| 01:12.51 | SeRi | You need to be jailbroken to be able to use it |
| 01:13.28 | *** join/#asterisk Rholk (~Rholk@bny92-1-82-67-178-101.fbx.proxad.net) |
| 01:21.17 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 01:21.41 | p3nguin | drdigital: It'd be really great if you'd fix your shit. |
| 01:22.35 | autofsckk | is it normal to be a 2-3 seconds delay within extensions? |
| 01:22.47 | p3nguin | What does that even mean? |
| 01:22.53 | p3nguin | "within extensions" |
| 01:22.55 | p3nguin | *shrug* |
| 01:24.27 | p3nguin | Do you mean between two phones which are bridged? |
| 01:25.00 | autofsckk | p3nguin: so sorry, english isnt my language, so sometimes i dont know how to express exactly what i mean, im running some tests, so i conected 2 softphones locally and i noticed that it is like 2 or 3 seconds late the sound trough the headphones, and it has a lot of echo too |
| 01:25.20 | p3nguin | That's not really considered normal. |
| 01:25.49 | p3nguin | Do an echo test from one phone, then the other. |
| 01:34.26 | autofsckk | p3nguin: it has a lot of echo with the echo test |
| 01:34.42 | p3nguin | It's supposed to echo back to you. |
| 01:34.55 | p3nguin | The main thing is to listen to the latency. |
| 01:35.14 | p3nguin | Does it echo back to you after .1 seconds... or after 1 second? |
| 01:36.01 | *** join/#asterisk r33dtard (~r33dtard@gateway/tor-sasl/r33dtard) |
| 01:36.03 | autofsckk | the echo i mean is that it repeats what i say but it says it many times, like an echo |
| 01:36.25 | p3nguin | Oh, that's not good. |
| 01:39.20 | autofsckk | i think so, how can i find what is going wrong, or why it has like a whistle too :( |
| 01:40.15 | p3nguin | Echo is often a result of the speaker volume being too loud and feeding back into the microphone. |
| 01:41.26 | autofsckk | p3nguin: yes i noticed that, im using headphones on both softphones, downloaded the volume and it was a little better, but it still sounds like shit :( |
| 01:57.33 | *** join/#asterisk r33dtard (~r33dtard@gateway/tor-sasl/r33dtard) |
| 01:59.07 | SeRi | p3nguin, how can I enable call transfer? |
| 01:59.23 | p3nguin | Press the transfer button on the phone. |
| 01:59.34 | SeRi | rofl |
| 01:59.36 | SeRi | nice |
| 02:00.07 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 02:00.15 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 02:00.28 | SeRi | so is there a way I can transfer one extension to the other? |
| 02:00.50 | WIMPy | Goto |
| 02:00.54 | p3nguin | Do you mean you want to transfer a call to some extension? |
| 02:01.20 | SeRi | yes sr. |
| 02:01.24 | p3nguin | Press transfer. Dial the extension. Wait for an answer. Press transfer again. |
| 02:01.43 | SeRi | I dont have transfer in my phones |
| 02:01.54 | p3nguin | What kind of phone? |
| 02:02.00 | DrDigital | here ill fix it this way? i mean i got a laptop and i disconnect and reconnect from going from one place to another threw the day |
| 02:02.04 | DrDigital | i don't reallyt need to be here |
| 02:02.05 | *** part/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 02:02.39 | SeRi | p3nguin, just a regular analog phone going threw my ata |
| 02:02.52 | p3nguin | What kind of ATA? |
| 02:02.54 | SeRi | I want to transfer the call to my sip phone |
| 02:02.59 | SeRi | linksys PAPT2 |
| 02:03.02 | WIMPy | collects a few bit for a tunnel for drdigital |
| 02:03.59 | p3nguin | What happens if you do a hook-flash, get a new dial tone, dial a new exten, get an answer, then do a hook-flash again? |
| 02:04.27 | SeRi | not sure. havent tried that |
| 02:04.29 | SeRi | one sec |
| 02:04.44 | p3nguin | If you can't answer the other phone because you are not near it, use call parking. I'll help you with that if you need it. |
| 02:05.18 | SeRi | ok one sec going to try now |
| 02:06.17 | *** join/#asterisk ChrisInSydney (~Chris@60-242-81-231.tpgi.com.au) |
| 02:15.05 | SeRi | p3nguin, It worked. Thank You very much! |
| 02:15.35 | SeRi | Though call parking sounds nice :) but cant play with it now since I am on the phone. |
| 02:15.44 | p3nguin | Sure you can! |
| 02:16.04 | SeRi | really? Awesome! |
| 02:16.51 | SeRi | me and my family are loving voip.ms |
| 02:17.22 | p3nguin | Have you checked the CDR to see the lack of charges? :) |
| 02:17.31 | SeRi | Yes and you are right is all free! |
| 02:17.38 | SeRi | Outbound to Customer |
| 02:17.42 | SeRi | Thats how they have it |
| 02:17.49 | SeRi | :D |
| 02:17.49 | p3nguin | Minutes 432, Charge 0.000000 |
| 02:18.15 | SeRi | lmao |
| 02:18.17 | SeRi | yes! |
| 02:18.20 | SeRi | Its awesome! |
| 02:18.26 | SeRi | we love it |
| 02:18.49 | p3nguin | And none of that jacked up CallCentric crap either. |
| 02:19.02 | *** join/#asterisk r33dtard (~r33dtard@gateway/tor-sasl/r33dtard) |
| 02:19.15 | SeRi | Yes. God do they suck |
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| 02:33.48 | *** join/#asterisk Naikrovek (~mbluth@unaffiliated/naikrovek) |
| 02:35.49 | SeRi | p3nguin, do you any type of backup provider? |
| 02:35.55 | SeRi | use* |
| 02:36.21 | *** join/#asterisk godmachine-x6 (~godmachin@unaffiliated/godmachine-x6) |
| 02:38.28 | *** join/#asterisk r33dtard (~r33dtard@gateway/tor-sasl/r33dtard) |
| 02:41.08 | p3nguin | No. |
| 02:41.41 | SeRi | ok |
| 02:41.58 | SeRi | p3nguin, How can I do call parking? |
| 02:42.10 | p3nguin | Open up features.conf |
| 02:42.33 | SeRi | ok |
| 02:42.43 | p3nguin | Make sure you have suitable values for parkext, parkpos, and context. |
| 02:42.53 | p3nguin | The defaults are usually good enough. |
| 02:43.28 | SeRi | parkext => 700 |
| 02:43.35 | SeRi | parkpos => 701-720 |
| 02:43.42 | SeRi | context => parkedcalls |
| 02:43.45 | p3nguin | In your phones context, include => parkedcalls |
| 02:43.53 | p3nguin | save, dialplan reload |
| 02:43.59 | p3nguin | Oh, wait... |
| 02:44.04 | p3nguin | In extensions.conf... |
| 02:44.06 | p3nguin | In your phones context, include => parkedcalls |
| 02:44.09 | SeRi | :) |
| 02:44.13 | SeRi | I knew that one! |
| 02:44.15 | p3nguin | Sorry, didn't mean to confuse there. |
| 02:44.17 | SeRi | I am learning! |
| 02:44.22 | p3nguin | save, dialplan reload |
| 02:44.53 | SeRi | done\ |
| 02:44.56 | p3nguin | When using transfer on an ATA, I don't know if it will announce correctly, so you'll have to test it to let me know... |
| 02:45.02 | p3nguin | Transfer a call to extension 700. |
| 02:45.20 | p3nguin | It should say "seven zero one" and exit. |
| 02:45.26 | SeRi | like I did before right? |
| 02:45.41 | p3nguin | Yeah, attended transfer to exten 700. |
| 02:46.52 | p3nguin | It should report 701 when you do it. |
| 02:46.57 | SeRi | do I know hang up? |
| 02:47.07 | p3nguin | If it said 701, hang up. |
| 02:47.13 | SeRi | ok done |
| 02:47.32 | p3nguin | That means you've put the call in parking space 701. You've parked the call. |
| 02:47.50 | SeRi | lol! cool |
| 02:47.51 | p3nguin | Now go to any other phone and call extension 701 to pick up that call. |
| 02:48.04 | SeRi | It started ringing back at me |
| 02:48.13 | p3nguin | You waited 45 seconds. |
| 02:48.16 | SeRi | when I put it in park it dialed back to me |
| 02:48.28 | p3nguin | parkingtime => 45 ; Number of seconds a call can be parked for |
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| 02:48.33 | SeRi | ah |
| 02:48.35 | SeRi | ok |
| 02:48.36 | SeRi | lol |
| 02:48.39 | SeRi | cool |
| 02:48.40 | p3nguin | Park it again. |
| 02:48.43 | SeRi | ok |
| 02:48.53 | p3nguin | Go to a different phone, and dial 701 to pick it up. |
| 02:49.11 | SeRi | sweet! it worked! |
| 02:49.13 | SeRi | :D |
| 02:49.15 | SeRi | awesome |
| 02:49.21 | p3nguin | This sort of emulates a key system. "Hey Bill, you've got a call on line 3." |
| 02:49.52 | SeRi | This is awesome! With hold on music and everything |
| 02:56.18 | p3nguin | Using the stuff in features.conf, you can also provide some other functionality. DTMF transfers instead of hook-flash, call pickup, etc. |
| 02:56.30 | SeRi | nice |
| 02:56.50 | p3nguin | I don't allow DTMF transfers for any phone that has a transfer key. |
| 02:57.14 | SeRi | ah I see |
| 02:57.37 | SeRi | p3nguin, when my brother dials to me I see the extension that I setup in my dial plan for speed sial. |
| 02:57.54 | p3nguin | I'm not sure what you mean. |
| 02:58.20 | SeRi | for example if I dial him on hes caller ID he can see my name. |
| 02:58.30 | p3nguin | yeah |
| 02:58.53 | SeRi | but if he dials to me I see the extension I setup on my dial plan for speed dial |
| 02:58.58 | p3nguin | What is his callerid number as configured on his SIP peer entry? |
| 02:59.06 | SeRi | yes sr |
| 02:59.17 | SeRi | If he calls my cell I can see hes cid |
| 02:59.30 | p3nguin | *headscratch* |
| 02:59.33 | p3nguin | What is his callerid number as configured on his SIP peer entry? |
| 02:59.58 | *** join/#asterisk r33dtard (~r33dtard@gateway/tor-sasl/r33dtard) |
| 03:00.06 | SeRi | none. I have the cid define in the outbound route |
| 03:00.12 | SeRi | for him |
| 03:00.43 | p3nguin | You must be setting it to the number you're seeing. |
| 03:00.52 | p3nguin | or you're setting it on the inbound call on your side. |
| 03:01.32 | SeRi | Thats what I was thinking because 100 is only define in my dial plan |
| 03:01.35 | SeRi | Mhhh hold on |
| 03:01.40 | SeRi | I think is hes cid |
| 03:01.52 | p3nguin | You show me config, I'll tell you why it happens. |
| 03:02.30 | p3nguin | Ideally, I would want his callerid number to show up as the number you would dial to reach him. |
| 03:02.31 | SeRi | yes sr. one sec. I am finishing hes firewall setup |
| 03:02.55 | p3nguin | If you dial 1003 on your phone to reach him, you should probably set the callerid number to 1003 on the speed dial entry on his side. |
| 03:03.49 | p3nguin | Like I said I prefix my clients' account numbers to reach them, I prefix their callerid with the account number when they call me. |
| 03:03.50 | SeRi | I think thats my problem |
| 03:04.01 | p3nguin | That allows me to dial the same number they call me from to reach them. |
| 03:04.24 | p3nguin | So if account 187 extension 206 calls me, the callerid number that shows up on my phone is 187206. |
| 03:04.44 | SeRi | ahhhh ok than is right |
| 03:04.53 | p3nguin | What number do you use to reach him? |
| 03:05.09 | SeRi | 100 and hes cid for the ext is also 100 |
| 03:05.19 | SeRi | so is correct |
| 03:05.25 | p3nguin | What number does he use to reach you? |
| 03:05.38 | SeRi | none. He dials my full number |
| 03:05.42 | p3nguin | oh |
| 03:05.45 | SeRi | I ahve not setup the ext on hes side |
| 03:05.49 | SeRi | have* |
| 03:06.20 | p3nguin | What number shows up on your cid when he calls you? |
| 03:06.44 | SeRi | 100 |
| 03:06.48 | *** join/#asterisk ks3 (~ks3@cpe-184-57-153-87.cinci.res.rr.com) |
| 03:06.55 | *** join/#asterisk TFGBD (~refhunetg@c-98-225-193-141.hsd1.pa.comcast.net) |
| 03:07.06 | p3nguin | I don't know if it was coincidence or what, but that seems okay to me. |
| 03:07.41 | SeRi | Yes. Thats right. I got confused because I thought he had a different ext number but they are the same :) |
| 03:08.05 | p3nguin | So it was coincidence rather than good configuration. |
| 03:08.21 | TFGBD | Okay, stupid question: Do any mobile/embedded PBXs or SIP servers exist? |
| 03:08.50 | TFGBD | IE, something that you could run on a PDA and then connect to asterisk. |
| 03:08.50 | SeRi | p3nguin, come on give me some credit! LOL |
| 03:09.11 | p3nguin | That would be a soft phone, not a PBX nor a SIP server. |
| 03:09.20 | TFGBD | Uh, no. |
| 03:09.23 | SeRi | TFGBD, They have elastix on a usb stick |
| 03:09.28 | SeRi | also free pbx |
| 03:09.36 | SeRi | but you would have to ask them |
| 03:09.39 | TFGBD | There are plenty of softphones for PDAs.. |
| 03:09.44 | TFGBD | I want an actual SIP server. |
| 03:09.44 | p3nguin | exactly |
| 03:09.54 | p3nguin | Like kamailio? |
| 03:09.56 | TFGBD | And even better if it can use the built in phone hardware. |
| 03:11.05 | SeRi | your only solution at that point is a self compile of uclinux/asterisk |
| 03:11.13 | SeRi | something like AstLinux |
| 03:11.27 | SeRi | small enough to fit in embedded systems |
| 03:11.32 | *** join/#asterisk cyborg-one (1000@85-238-108-99.broadband.tenet.odessa.ua) |
| 03:12.14 | SeRi | freepbx/elastix are some what bloated but still doable. again I know nothing about them |
| 03:12.53 | TFGBD | I don't want linux. That would be too easy. Asterisk already runs on embedded linux. |
| 03:13.52 | TFGBD | Freepbx is more bloated than asterisk? |
| 03:14.11 | p3nguin | ... |
| 03:14.17 | SeRi | freepbx is asterisk with a gui |
| 03:14.18 | p3nguin | You realize what you're asking, right? |
| 03:14.33 | p3nguin | No, FreePBX is just the GUI. |
| 03:14.37 | SeRi | p3nguin, o loard... here we go |
| 03:14.42 | TFGBD | Sorry, I misread. |
| 03:14.50 | TFGBD | I was thinking of FreeSWITCH |
| 03:15.15 | p3nguin | FreeSWITCH is a soft switch, which somehow handles calls in a different manner. |
| 03:15.19 | p3nguin | How, I really do not know. |
| 03:15.30 | SeRi | #freeswitch |
| 03:15.33 | SeRi | :) |
| 03:15.41 | p3nguin | Never touched FS, but a friend uses it for his stuff. |
| 03:15.59 | SeRi | TFGBD, look in to astlinux. Its an embedded plataform with asterisk |
| 03:16.08 | SeRi | Probably would do what you want |
| 03:16.11 | SeRi | no gui though |
| 03:16.20 | SeRi | the gui only help you edit conf files |
| 03:16.33 | p3nguin | s/edit/break/ |
| 03:16.40 | TFGBD | As I said, I there are plenty of asterisk linux ports. |
| 03:16.41 | p3nguin | s/break/ruin/ |
| 03:18.01 | p3nguin | It seems so weird to NEED something like FreePBX to configure things. |
| 03:18.25 | SeRi | lol |
| 03:18.58 | *** join/#asterisk r33dtard (~r33dtard@gateway/tor-sasl/r33dtard) |
| 03:18.59 | TFGBD | It looks like some older version of freeswitch had Windows CE builds. I'll look into it. |
| 03:19.15 | TFGBD | Though, I doubt they'd be able to access the radio. |
| 03:19.31 | SeRi | TFGBD, good luck with your project. |
| 03:20.02 | TFGBD | I was kind of hoping for something proprietary that also supported multiple mobile platforms. |
| 03:21.54 | TFGBD | I just figured people with more experience setting up SIP PBXes who use the most popular one may have heard of something over the years. |
| 03:22.29 | TFGBD | I guess there is always bluetooth. |
| 03:26.22 | TFGBD | I'll keep searching, though. |
| 03:42.57 | *** part/#asterisk TFGBD (~refhunetg@c-98-225-193-141.hsd1.pa.comcast.net) |
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| 04:24.30 | SeRi | lol |
| 04:25.09 | SeRi | my brother hates linux. I guess we all sort of dont like waht we do not understand... and not willing to understand it... |
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| 04:49.09 | rhl | Hi, I am a linux user who has just been gifted a large amount of fiber to his home for _free_ I was wondering how and if I could setup a phone system that works over the network in my home. Is that something do-able with asterisk? What if anything are the recurring costs, is this process documented somewhere, so that I might read about it? |
| 04:53.53 | SeRi | http://ofps.oreilly.com/titles/9780596517342/ |
| 04:54.01 | SeRi | rhl, ^^ |
| 04:54.07 | SeRi | asterisk book |
| 04:54.09 | SeRi | ~book |
| 04:54.09 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/ or ~buybook |
| 04:55.25 | rhl | SeRi, thanks a lot! can you give me a short answer to what kind of recurring costs I could achieve if I wanted to be able to do this on the cheap? |
| 04:55.31 | rhl | aside from hardware that is.. |
| 04:56.00 | SeRi | getting a provider will cost you any wheres from 4 dollars a month to 6.... |
| 04:56.26 | SeRi | depends on the plan |
| 04:56.48 | rhl | and thats generally for unlimited minutes? |
| 04:56.53 | rhl | in and out? |
| 04:57.06 | SeRi | and thats about as much you will have to spend. some providers bill you for each min out bound and free incomming calls |
| 04:57.25 | SeRi | It all depends on which provider you use. |
| 04:57.30 | rhl | SeRi, any provider you recommend? that maybe does unlimited in and out? |
| 04:57.46 | rhl | I guess I could make google call me or something... |
| 04:57.48 | SeRi | the inital setup will also cost you a few bucks |
| 04:58.02 | SeRi | rhl, google is a good start for free |
| 04:58.13 | rhl | for sure.. I am assuming I could run asterisk on my desktop box here |
| 04:58.16 | SeRi | asterisk supports the google protocol "jabber" |
| 04:58.42 | SeRi | but for a serious provider imo voip.ms |
| 04:58.57 | SeRi | but do your research and make your own decision |
| 04:59.21 | SeRi | I am still learning my self how asterisk all works. |
| 05:03.06 | rhl | thanks :) |
| 05:03.31 | SeRi | np |
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| 06:04.03 | joako | Does anyone know a good Asterisk MRTG script? |
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| 07:13.28 | BugKhaM | Anyone know how to add IAX2 configs in server A and B so that incoming CallerIDs can be transferred? |
| 07:14.28 | BugKhaM | adding no username/secret just doesn't work |
| 07:19.59 | kaldemar | how have you configured the boxes now and what makes you think caller id is not delivered? |
| 07:32.05 | BugKhaM | kaldemar: I added IAX friends in both servers + authenticated by IP addresses only i.e. no username+secret |
| 07:33.12 | kaldemar | how do you expect to authenticate without secrets? by "how" i mean show the configs. use pastebin. |
| 07:34.17 | BugKhaM | kaldemar: I normally did it in SIP and it worked fine |
| 07:35.30 | kaldemar | and it works in IAX just fine when configured properly. |
| 07:35.41 | BugKhaM | kaldemar: I'm using this IAX friend as a trunk and want CallerID to be transferred to another server not the IAX friend's CallerID |
| 07:35.54 | kaldemar | in fact, you have to take extra measures to prevent caller id from getting delivered. |
| 07:36.24 | kaldemar | are you going to show your configurations in iax.conf and extensions.conf? |
| 07:39.22 | BugKhaM | kaldemar: iax.conf -> http://pastebin.com/UHCaSTb4 |
| 07:43.03 | BugKhaM | kaldemar: extensions.conf -> http://pastebin.com/A0jLQiuM |
| 07:45.54 | kaldemar | canreinvite, dtmfmode and cancallforward are not iax options at all. |
| 07:46.17 | BugKhaM | kaldemar: and output from iax2 debug -> http://pastebin.com/Mf3LC87V |
| 07:46.43 | BugKhaM | kaldemar: i see, I just coppied from sip settings actuammy |
| 07:47.33 | kaldemar | cancallforward is not a sip option either. |
| 07:47.35 | BugKhaM | kaldemar: the "CALLING NUMBER : 6900002 |
| 07:48.14 | kaldemar | yes? |
| 07:48.25 | BugKhaM | is what I want to transfer to server B but it just said "No such context/extension |
| 07:48.52 | kaldemar | i thought you had a problem with caller id. |
| 07:49.08 | BugKhaM | kaldemar: isn't that a callerid? |
| 07:49.43 | BugKhaM | kaldemar: i see what you mean |
| 07:49.45 | kaldemar | isn't what a caller id? "No such context/extension" has nothing what so ever to do with caller id. |
| 07:50.49 | kaldemar | show a whole call |
| 07:51.26 | BugKhaM | kaldemar: OK, initially I set it up with proper username/secret and the call got through pretty fine but with IAX friend's callerid |
| 07:52.04 | BugKhaM | kaldemar: I just decided to add it this way and now the context isn't found |
| 07:52.09 | kaldemar | the call doesn't match the friend unless you use a username. |
| 07:52.53 | BugKhaM | kaldemar: but in SIP friend I didn't need a username to get it working |
| 07:53.01 | kaldemar | this is not SIP. |
| 07:53.30 | BugKhaM | kaldemar: hmm, I thought you said this would work OK should I set it up properly |
| 07:53.45 | kaldemar | what? |
| 07:54.06 | kaldemar | configuring IAX like SIP does not equal to configuring properly. |
| 07:54.23 | BugKhaM | kaldemar: OK |
| 07:55.37 | BugKhaM | kaldemar: what about a peer? will it require a matched peer name to go to the right context? |
| 07:56.11 | kaldemar | a friend is like a shortcut that creates a user and a peer. |
| 07:56.40 | kaldemar | and yes, you need to configure a username. do you have a problem with that? |
| 07:57.27 | BugKhaM | kaldemar: the server A's friend username will become a callerid everytime I sent the calls from server A to B |
| 07:58.19 | kaldemar | show me. |
| 07:58.44 | kaldemar | and i thought you didn't have a friend configured on server A, but a peer. |
| 07:58.56 | BugKhaM | kaldemar: as you can see from my extensions.conf that I 'm using Dial(IAX2/gbl01-serv-conn/${EXTEN},45,RrL(3600000)) |
| 07:59.14 | BugKhaM | kaldemar: OK, hold on |
| 08:00.23 | *** join/#asterisk timahvo1 (~rogue@41.223.57.75) |
| 08:00.59 | *** join/#asterisk xpot (~xpot@166-70-100-198.ip.xmission.com) |
| 08:03.47 | BugKhaM | kaldemar: hmm, I tried again and this time it seems to work |
| 08:07.28 | BugKhaM | kaldemar: OK, still not what I need |
| 08:10.21 | ChannelZ | I'm not scrolling back pages to decypher... what isn't working? |
| 08:12.01 | kaldemar | ChannelZ: still no idea. |
| 08:12.50 | ChannelZ | something about CallerID |
| 08:13.38 | kaldemar | or abut peer matching or dialplan. |
| 08:14.39 | BugKhaM | kaldemar: got it working now, thanks for your time |
| 08:42.23 | *** join/#asterisk frk2 (~faraz@zivios/member/fkhan) |
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| 10:49.34 | tokozedg | hello, which is best app for displaying active call to specific IVR extension? |
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| 12:41.36 | *** join/#asterisk autofsckk (~autofsckk@unaffiliated/autofsckk) |
| 12:52.50 | autofsckk | good morning, whats the best codec ? im reading about g729 privative, and theres an open source, whats the preferred option talking about codecs? |
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| 13:08.45 | *** join/#asterisk nwxi (~nwxi@host86-178-56-18.range86-178.btcentralplus.com) |
| 13:08.59 | nwxi | does asterisk 1.6.0 have CEL? |
| 13:12.57 | nwxi | ive created cel.conf and cel_custom.conf as required, but it doesn't do anything |
| 13:36.33 | *** join/#asterisk Rufus (Rufus@unaffiliated/rufus) |
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| 13:58.17 | k3asd` | hi |
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| 15:01.51 | devil_evoxxx | hi all |
| 15:07.13 | *** join/#asterisk Tim_Toady (~fuzzy@178.128.1.13.dsl.dyn.forthnet.gr) |
| 15:09.54 | autofsckk | hi |
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| 15:12.42 | devil_evoxxx | hi, have you got any idea because i received this |
| 15:12.58 | devil_evoxxx | Ignore potential DTMF echo from '94.XX.XX.XX:10167' |
| 15:16.28 | *** part/#asterisk wesphillips (~wphill04@adsl-99-161-156-160.dsl.hstntx.sbcglobal.net) |
| 15:20.36 | dlynes_laptop | devil_evoxxx, that's a bad thing? |
| 15:22.15 | devil_evoxxx | dlynes_laptop: ..yes..in 2 month i've got to work with two quescom |
| 15:22.25 | devil_evoxxx | and in both cases, i've got a lot of problem with dtmf |
| 15:22.52 | devil_evoxxx | behind there is ast 1.8.7 |
| 15:23.15 | devil_evoxxx | but the problem is the quescom..and the hints used to make work in the last work, is not valid now.. |
| 15:23.22 | devil_evoxxx | i'm going to be mad |
| 15:43.31 | dlynes_laptop | devil_evoxxx, quescom? |
| 15:44.56 | dlynes_laptop | ah...ic...seems to be a european solution for voip |
| 15:45.44 | dlynes_laptop | devil_evoxxx, is there a particular reason you're using the latest bleeding edge version of asterisk in a production system? |
| 15:46.57 | *** join/#asterisk Linuturk (~linuturk@unaffiliated/linuturk) |
| 15:50.24 | dlynes_laptop | devil_evoxxx, i'm just suggesting you might want to use the latest 1.6.2, instead, or if you need 1.8 for a particular reason, find out from someone what the most stable release of 1.8 currently is |
| 15:51.25 | dlynes_laptop | devil_evoxxx, Asterisk releases don't generally start to become production stable until they get into double digit point releases |
| 15:56.12 | devil_evoxxx | dlynes_laptop: i've found this where i current work..is not my choice |
| 15:56.44 | devil_evoxxx | dlynes_laptop: you have a ast 1.6 or 1.8 for a little test for another problem? |
| 15:57.23 | devil_evoxxx | anyway i'm using 1.8 for testing WaitFax applicaiton |
| 16:00.58 | devil_evoxxx | p.s. usually i prefer to use digium pri card |
| 16:01.12 | devil_evoxxx | but in this case..i've already found this crazy surprise |
| 16:02.48 | p3nguin | 1.8.7.0 is just as stable as whatever 1.6.2 version you're wanting to use. |
| 16:03.03 | p3nguin | AND... it's LTS. |
| 16:03.31 | devil_evoxxx | lol :) |
| 16:05.08 | devil_evoxxx | p3nguin: you use 1.8.X too? |
| 16:05.14 | p3nguin | I use 1.8.7.0 |
| 16:05.40 | p3nguin | http://www.asterisk.org/downloads |
| 16:06.46 | devil_evoxxx | i use 1.8.7 on two test machine..and if the test gone ok in the few weeks i migrate my 1.4 ast boxes |
| 16:06.53 | devil_evoxxx | so, i've only a question if you use 1.8 |
| 16:07.25 | devil_evoxxx | 1.8.7 sorry..you are able to see the rtp stream in cli if you set "rtp set debug ip ip-of-your-phone' |
| 16:07.32 | devil_evoxxx | ? |
| 16:07.32 | p3nguin | 1.8.7. |
| 16:07.33 | p3nguin | 1.8.7.0 |
| 16:07.42 | p3nguin | I'll check. |
| 16:08.19 | p3nguin | RTP Debugging Enabled for address: 192.168.192.69:0 |
| 16:08.26 | devil_evoxxx | because i've noticed that if i set only ip ( and asterisk say to me RTP Debugging Enabled for address:IP:0 |
| 16:09.05 | p3nguin | Nope, nothing shows up. |
| 16:09.18 | devil_evoxxx | if you make a call, you can see rtp stream? in my case the phone are bhind nat and asterisk box is with public ip. If i set the public ip of the phone i can not see the stream |
| 16:09.26 | p3nguin | How could I know what port I need to add? |
| 16:09.41 | devil_evoxxx | the only way that i've found is sniff with tcpdump |
| 16:09.46 | p3nguin | I thought it was pretty much random and negotiated in SIP at the time of the call. |
| 16:11.10 | devil_evoxxx | but, in the 1.4 this work fine.. |
| 16:11.10 | p3nguin | or in my case, SCCP. |
| 16:11.22 | devil_evoxxx | i dont'know if is noious bug |
| 16:11.39 | p3nguin | Someone said the other day if you specify the port, you'll see the rtp stream. |
| 16:11.52 | devil_evoxxx | yes, i know..i try and it'work |
| 16:12.04 | p3nguin | But I don't know how you'd know the port without doing other debug first. |
| 16:12.41 | devil_evoxxx | but i hope it's a work-around because for finding the port i've got to dump packet with tcpdump |
| 16:12.54 | devil_evoxxx | and..in the same dump, i can directly see the rtp stream |
| 16:13.04 | p3nguin | Do you often have to debug rtp? |
| 16:13.22 | p3nguin | I don't have to debug rtp but maybe a couple times in a year. |
| 16:13.36 | p3nguin | And zero times since I have moved to 1.8 branch. |
| 16:13.53 | devil_evoxxx | no :) but sometimes ..someone change my config and have audio-problem (rtp related) |
| 16:14.14 | p3nguin | People can't change configs without access to the system. |
| 16:14.30 | devil_evoxxx | we are 4 people working on this server |
| 16:14.35 | devil_evoxxx | i'm a consulent |
| 16:14.49 | devil_evoxxx | and the boss want to "see the cli".. |
| 16:15.40 | p3nguin | I had to tell someone that I wasn't going to continue working for them if they were going to keep touching stuff. |
| 16:17.45 | devil_evoxxx | eheheheh :) fantastic |
| 16:20.09 | p3nguin | I can only take so much of someone screwing up what I've been working on. |
| 16:20.29 | WIMPy | You're simply not professional. |
| 16:20.47 | p3nguin | Maybe I'm not. |
| 16:21.32 | p3nguin | But when I submitted the bill to the owner for time spent doing the work and then redoing the work after his minion changed stuff, I was told to lock him out so he'd quit messing with things. |
| 16:21.54 | devil_evoxxx | so, next to this noious bug in "rtp debug" i've to fix dtmf trought Quescom Device.. I think who projected this device is not "user friendly" |
| 16:22.32 | p3nguin | If you think the rtp debug thing is a bug, submit an issue for it. |
| 16:22.43 | p3nguin | If it's a bug, they'll hopefully work on it. |
| 16:24.05 | devil_evoxxx | i can submit on jira, rihgt? |
| 16:24.08 | p3nguin | yes |
| 16:24.19 | devil_evoxxx | fine :) now il'do |
| 16:27.08 | p3nguin | I don't see anything about it already submitted. |
| 16:27.22 | p3nguin | Either no one else knows about it, or no one else thinks it is a bug. |
| 16:29.35 | *** join/#asterisk CrossWired__ (~chatzilla@65.210.186.34) |
| 16:31.48 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 16:34.34 | devil_evoxxx | p3nguin: for "affected application" can i select Applications/app_rtp |
| 16:34.38 | devil_evoxxx | ? |
| 16:35.38 | WIMPy | That should be res_rtp, if that exists. |
| 16:35.44 | WIMPy | probably |
| 16:37.08 | p3nguin | If you can't figure out what to select, leave it and someone else will select it for you. |
| 16:40.38 | *** join/#asterisk MarKsaitis (~MarKsaiti@cpc4-rdng22-2-0-cust932.15-3.cable.virginmedia.com) |
| 16:41.36 | devil_evoxxx | ok :) |
| 16:47.44 | SeRi | p3nguin, whats going on. |
| 16:47.53 | devil_evoxxx | i can confirm that this affect 1.8.5.0, 1.8.6.0,1.8.7.0 |
| 16:47.59 | p3nguin | Just Youtubing with the kid. |
| 16:48.41 | SeRi | p3nguin, hehehe I do that from time to time. :) my 4yr old likes the ABC channel |
| 16:53.06 | SeRi | p3nguin, voip.ms working ok for you today? I been getting sudden busy tones in the middle of conversations. |
| 16:53.14 | *** join/#asterisk saliak (~kailas@ip68-9-228-184.ri.ri.cox.net) |
| 16:53.23 | SeRi | "disconnects" |
| 16:53.50 | p3nguin | I'm not on the phone today, so I don't know. |
| 16:54.25 | *** join/#asterisk Dovid (~Dovid@213.8.121.90) |
| 16:57.20 | devil_evoxxx | p3nguin: issue opened, I hope it is helpful to the * comunity :) |
| 16:58.49 | p3nguin | Good description of the problem. |
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| 17:44.45 | *** join/#asterisk ChannelZ (channelz@burner.com) |
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| 18:29.53 | *** join/#asterisk beta2k (~Beta2K@d24-36-128-84.home1.cgocable.net) |
| 18:29.57 | beta2k | Hello all |
| 18:30.09 | beta2k | Anyone around running polycom phones? I moved from a Cisco 3524XL to a 3550-PWR for the PoE and now I can't get the phones to boot all the way |
| 18:30.17 | beta2k | They sit at running sip.ld for a while then reboot |
| 18:30.46 | beta2k | weird bit is I can see traffic from the phones on the PBX, but nothing trying to grab their application config files |
| 18:30.47 | p3nguin | Do you have the sip application on the tftpd that it's trying to load it from? |
| 18:31.13 | *** join/#asterisk airborn (~airborn@host114.router2.tygrys.net) |
| 18:31.18 | beta2k | Yes, one even updated it's bootrom and sip application after installing the new switch (hadn't brought that phone up yet) |
| 18:31.21 | airborn | hell,o |
| 18:32.03 | p3nguin | I guess if it updated it is bootrom and sip application, it should be starting up normally. |
| 18:32.10 | beta2k | I'm running a voice vlan so I tried removing that from one phones port and putting the port in the voice vlan instead but still no love.. |
| 18:32.38 | beta2k | I don't see the sip application trying to read it's config from the tftp either |
| 18:33.01 | p3nguin | If it can't read it is config, that would be a problem. |
| 18:33.10 | beta2k | the bootrom yes, but nothing from the application |
| 18:33.24 | beta2k | No failed read, no attempt at all |
| 18:33.46 | p3nguin | So it's not reading its config... it's not even trying to read its config. |
| 18:34.09 | beta2k | Nope |
| 18:34.12 | p3nguin | Can you upgrade its sip application version? |
| 18:34.22 | beta2k | It's at the latest it can run |
| 18:34.36 | p3nguin | Try downgrading it one version. |
| 18:34.44 | beta2k | Ok |
| 18:37.48 | beta2k | Should I downgrade the bootrom as well? |
| 18:41.13 | p3nguin | No, it sounds like it was booting correctly. |
| 18:44.17 | beta2k | Just downloaded the new application |
| 18:45.01 | p3nguin | If it loads up correctly with the lower version number, then you can upgrade it back to the other newer version later. |
| 18:46.08 | beta2k | Nope :) |
| 18:46.28 | p3nguin | Same result with the lower version? |
| 18:46.51 | beta2k | So far, it's been sitting at running sip.ld for a couple minutes now |
| 18:47.10 | p3nguin | Leave it sit for at least 15 minutes before you try something else. |
| 18:47.18 | [TK]D-Fender | If it boot-cycles then that is a common side effect of corrupted configs. often by upgrading the firmware and not having matching configs. |
| 18:48.23 | beta2k | [TK]D-Fender: I overwrote the sip.cfg and phone1.cfg from the firmware archive just to be sure :) |
| 18:48.35 | *** join/#asterisk netmax (~netmax@is.linux-administrator.com) |
| 18:48.53 | beta2k | the per-phone and local sip config files are fairly simple |
| 18:49.24 | p3nguin | Maybe that's why it doesn't work -- you overwrote the good files from the firmware archive with bad/incompatible files. |
| 18:49.50 | p3nguin | The ones from the firmware archive are pristine, without a doubt. |
| 18:50.06 | beta2k | I just did it again when I put the lower version application on the tftp |
| 18:50.17 | beta2k | from a fresh archive |
| 18:50.58 | p3nguin | While watching the tftp server log, do you see the phone fetching appropriate files? |
| 18:53.22 | beta2k | I see it grabbing the bootrom, bootrom config, and application, but not sip.cfg, phone1.cfg nor the local config or phone config files |
| 18:58.40 | beta2k | <PROTECTED> |
| 18:59.11 | beta2k | Those are the config files I've modified |
| 18:59.29 | beta2k | I haven't touched sip.cfg or phone1.cfg |
| 18:59.45 | [TK]D-Fender | boot from a fresh folder from a stock provisioning pack |
| 19:00.59 | beta2k | will once I get the baby back to bed :) |
| 19:21.05 | *** join/#asterisk brunner (~chris@pdpc/supporter/gold/brunner) |
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| 19:52.43 | Dovid | helllo ev1 |
| 19:59.48 | f2knight | hi |
| 20:16.50 | *** join/#asterisk rampage73 (~rampage73@184.2.147.197) |
| 20:19.12 | rampage73 | i set up asterisk (basic) off of Jeremy McNamara website http://jeremy-mcnamara.com/asterisk/how-to-configure-asterisk/ and all seems well except voicemail when calling in hangs up before greeting so if i call extension 100 from 200 100 rings then click instead of going to voicemail. anyone have that problem or know how to solve it? |
| 20:19.45 | rampage73 | oh asterisk version is Asterisk 1.6.2.5-0ubuntu1.4 |
| 20:19.59 | rampage73 | running on ubuntu 10.04 lts |
| 20:20.49 | f2knight | Q: Is anyone having issues answering Google Voice calls? |
| 20:33.23 | *** join/#asterisk nix8n82-phone (~AndChat@18.sub-174-253-163.myvzw.com) |
| 20:55.09 | *** join/#asterisk rdahlin_1 (~rdahlin_1@2001:16d8:cc97:1:14f3:e64a:a119:7d8b) |
| 21:00.05 | beta2k | [TK]D-Fender: Fresh tftpboot directory, extracted one version down (4.3.2) into it and still no complete boot |
| 21:02.30 | f2knight | beta2k, Cisco phones |
| 21:02.31 | f2knight | ? |
| 21:12.07 | beta2k | f2knight: polycom |
| 21:12.14 | beta2k | ip430's |
| 21:12.45 | f2knight | beta2k, same thing, yea I have always had a damned hard time with them esp used ones... You have done a factory reset right? |
| 21:15.09 | beta2k | Yeah, just did that again on one |
| 21:15.25 | beta2k | I changed switches and it all went downhill from there :) |
| 21:15.53 | f2knight | I feel you. Wish I could be of more help. |
| 21:16.10 | f2knight | Think I might have two 301's sitting around |
| 21:19.58 | f2knight | Anyone know how I can see what password is being attempted for authentication? |
| 21:22.10 | beta2k | ethereal? |
| 21:22.44 | beta2k | I wonder if I should switch from tftp to ftp... |
| 21:22.55 | ChannelZ | except that you're just going to see a hash right? |
| 21:23.04 | beta2k | Not with ftp |
| 21:23.13 | ChannelZ | oh I missed you were talking about FTP |
| 21:23.35 | beta2k | Yeah I assumed it :) |
| 21:23.45 | ChannelZ | (although there can be obfuscation on that too but it's not particularly common) |
| 21:24.04 | carrar | tftp doesn't use passwords FYI |
| 21:24.11 | ChannelZ | tftp no |
| 21:24.57 | carrar | and if you switch to FTP and use FTP passwords to poll your config, that password should be different then your register password |
| 21:25.10 | carrar | for SIP Registration |
| 21:25.37 | ChannelZ | you're no fun |
| 21:25.45 | carrar | heh |
| 21:27.04 | *** join/#asterisk ketas-li (~ketas@ketas-laptop6.si.pri.ee) |
| 21:27.04 | carrar | Though you could modify dhcp to hand out a unique ftp password for everyone phone |
| 21:27.50 | carrar | have to list out every phone mac address in the dhcpd.conf |
| 21:33.18 | *** join/#asterisk puzzled (~patrick@53533D66.cm-6-4a.dynamic.ziggo.nl) |
| 21:34.44 | *** join/#asterisk hlieberman (~hlieberma@about/crypto/member/hlieberman) |
| 21:35.11 | hlieberman | Hello everyone. I'm wondering if anyone can give me a hand - I'm trying to setup an IVR phone tree to redirect calls to some set number for Occupy Boston. |
| 21:35.35 | hlieberman | Normally, I'd give it a shot myself and ask you'all questions, but it |
| 21:35.47 | hlieberman | It's a kind of a rush-job, and my asterisk experience sucks. |
| 21:37.29 | ChannelZ | have you done any dialplan work at all or are you having a specific problem or.. |
| 21:38.26 | f2knight | beta2k, I am talking about the sip authentication sorry. |
| 21:39.20 | hlieberman | I have essentially zero experience, so what I have now is... well. Far from what needs to be done. I guess one question would be, I'm essentially going to need to have an option to check-in and change one of the redirect numbers, as people come on and off shift. How would you recommend storing that? |
| 21:39.23 | ChannelZ | in that case they are hashed then |
| 21:39.53 | ChannelZ | hlieberman: AstDB is probably the easiest |
| 21:40.40 | ChannelZ | via the DB function |
| 21:41.15 | hlieberman | And just maintain a simple 1:1 relationship, and just recreate it every time the check-in happens. Makes sense. |
| 21:42.52 | ChannelZ | FollowMe might help you too |
| 21:48.23 | hlieberman | Can I use the DB function within FollowMe? |
| 21:49.23 | p3nguin | rampage73: Show me extension 100. |
| 21:50.30 | *** join/#asterisk ketas-av (~ketas@217.146.67.53) |
| 21:59.16 | p3nguin | hlieberman: Sure. |
| 21:59.58 | hlieberman | Any ideas how I can actually manage the setting of the number? I think I have the basic dialplan and the call-out functionality (thanks for the FollowMe tip). |
| 22:00.26 | p3nguin | Describe how you want the procedure to work, and I'll help you with creating the dial plan logic to make it happen. |
| 22:02.57 | hlieberman | You dial into a number and are presented with a handful of options. Let's call it three. The first two are to reach the on-duty person for one of two positions. Once you hit this, it will call whoever is on duty for n amount of time, then fall back to someone else, and connect the two calls. The other option is a submenu to check-in to one of the positions and put what phone number (numbers?) you want it to try. |
| 22:03.55 | p3nguin | Sounds easy enough. Do you need all three options set up, or do you just want to focus on option three for now? |
| 22:04.09 | hlieberman | I pulled together http://pastebin.com/MbwDDS0X so far, which I think will work for #1 and #2. |
| 22:04.39 | rampage73 | p3nguin, http://pastebin.com/DfABTaTR |
| 22:04.41 | hlieberman | Which is just using FollowMe and then the DBGet() to define what number that followme should look for. |
| 22:04.56 | hlieberman | Haven't tested it at all, though. ;) |
| 22:05.33 | p3nguin | There are some things wrong with that, so I don't expect it to work out well. |
| 22:05.42 | hlieberman | I'm sure there are. |
| 22:05.59 | p3nguin | rampage73: I said extension 100. |
| 22:05.59 | hlieberman | Again, no asterisk experience. I'm pulling random crap together out of my ass. I'd appreciate any help you can provide. |
| 22:07.03 | rampage73 | p3nguin, ?I am confused that is extension 100 |
| 22:07.33 | p3nguin | rampage73: What you've shown me is a sip peer entry from sip.conf. I asked for extension 100. Extensions are found in extensions.conf. |
| 22:07.55 | rampage73 | p3nguin, oh crap sorry (newb here) |
| 22:08.37 | p3nguin | hlieberman: When someone calls in to option 3 to set a phone number, that number is reflected in options 1 and 2 on subsequent calls? |
| 22:09.37 | hlieberman | Yes, it should be. |
| 22:09.57 | hlieberman | That is, the number that is called in options 1 and 2 should be whatever was last set by extension 3. |
| 22:10.51 | rampage73 | p3nguin, sorry about that here ya go http://pastebin.com/CNRsw3nC |
| 22:12.02 | p3nguin | rampage73: Where is the MACRO_EXTEN value being set? |
| 22:12.06 | f2knight | hlieberman, ... just trying to follow your logic here... someone calls in dials option 3 and enters a phone number to recieve a call back at? That callback number is what is used by option 1 and 2? |
| 22:12.38 | hlieberman | Let me take a step back and tell you the problem I'm trying to solve rather than my attempt at a solution. |
| 22:12.44 | hlieberman | Maybe that will be easier to understand. |
| 22:12.48 | p3nguin | It's more like someone calls in and sets his phone number so that other people calling in and pressing 1 or 2 can reach the person who set his phone number. |
| 22:13.41 | hlieberman | Right. We have on-call people who might need to be reached. Someone who is a client can call in and reach someone in either of the two on-call positions. Someone who is on-call can set what number they want to be reached at. |
| 22:13.41 | p3nguin | I understand the problem, but I need a few minutes to put it together in a way that makes sense to you and that you can put into your system. Concept would be easy, but if you don't understand the concept, you may not be able to use it. |
| 22:15.32 | p3nguin | Will options 1 and 2 always call the same person, or will option 3 need to allow two different people to set two different numbers? |
| 22:15.33 | hlieberman | Sure. I appreciate your help - both of you. I wish I was more competent and could put it together myself, but I'm not going to let my stubbornness let this not get done. |
| 22:15.42 | hlieberman | No, options 1 and 2 are two separate people. |
| 22:15.42 | f2knight | p3nguin, I hate to point out the obvious, but please add an authenticate line on it. |
| 22:15.54 | p3nguin | f2knight: I will when I get to that part. |
| 22:16.00 | rampage73 | p3nguin, to be honest I do not know I took for granted that it was an asterisk program variable, when I searched the /etc/asterisk directory the only place it shows up is in extensions.conf under macro-phone |
| 22:16.15 | f2knight | p3nguin, good. Would hate to see someone change it.. for spite. |
| 22:16.32 | f2knight | hlieberman, are the phone numbers of the people who can call in known? |
| 22:17.03 | hlieberman | Not in advance, no. |
| 22:17.04 | f2knight | hlieberman, might be good to kind of check against the CallerID in addition to an authentication. (kind of double authentication) |
| 22:17.05 | p3nguin | rampage73: I don't use macros, so I will have to check to see how it is set. It could be set where the macro is being run from, so in that case it would be extension 100. I'll check into it soon. |
| 22:17.41 | f2knight | hlieberman, with all the occupy's going on I have noticed heavy hack attempts.. |
| 22:17.58 | f2knight | I am hosting some numbers for Occupy PDX, |
| 22:18.02 | rampage73 | p3nguin, no problem I appreciate any and all help and am happy to wait! I see you are busy |
| 22:18.04 | p3nguin | If the caller will be calling from the number he wants to receive calls on, we could play back something like "You're calling from 2123435454." Is this the number where I can reach you?" |
| 22:18.14 | hlieberman | That would work. |
| 22:18.37 | p3nguin | If yes, set the number. If no, ask for a new one. |
| 22:18.51 | p3nguin | Or force them to call from the number they want to be called. |
| 22:19.05 | hlieberman | f2knight, Oh, yeah? Well, a hello from your comrades east of you. :) |
| 22:19.21 | f2knight | hlieberman, ditto back at you. |
| 22:19.40 | hlieberman | p3nguin, I wouldn't mind that additional protection. I considered having the system call whatever number was set for a confirmation, but I gave up quickly. That, at least, I know I'm not competent enough to do. |
| 22:20.05 | hlieberman | Most people are going to be giving their cell phones, so forcing them to call from the number they are checking in with isn't a huge burden. |
| 22:21.50 | f2knight | hlieberman, p3nguin, I would suggest forcing the authenticating numbers to be from designated phones. and each have a unique authentication code. If you do that its not to hard to add a new number and authentication code. If someones phone gets lost you can disable the one phone and not disable everyone else on the system with permission. |
| 22:22.22 | p3nguin | I'm going to start out basic, then we can build up from there. |
| 22:22.37 | f2knight | hlieberman, callerID's can be forged.. but its less likely to have a forged caller ID and authentication code unless someone knows one of the two already. In which case you have an inside job. |
| 22:22.55 | hlieberman | Which is fine. This isn't going to be used for any of the direct action stuff anyway. |
| 22:23.28 | hlieberman | It's just so that the street medics can get in touch with a more experienced medical or mental health provider if they feel they need advice. |
| 22:24.39 | hlieberman | Well, that's the start use-case. I'm sure it will grow from there. |
| 22:26.14 | f2knight | hlieberman, always does. we just started using it for inner communications, but quickly found that every one else was wanting to hook in |
| 22:26.44 | hlieberman | Yup. |
| 22:37.34 | p3nguin | hlieberman: Here's some basic conceptual design: http://pastebin.com/jZ3py6Zj |
| 22:38.06 | p3nguin | It's not completed... just wanted you to see where I'm going with it. |
| 22:39.09 | p3nguin | I'll work on adding phone numbers on call-in now. |
| 22:39.23 | hlieberman | Yeah. That makes sense. Didn't know you can actually use names as an extension. Makes things a bit simpler. |
| 22:39.31 | hlieberman | What is extension T? |
| 22:39.39 | hlieberman | Timeout? |
| 22:39.52 | p3nguin | Yes, timeout when the absolute timeout value has been reached. |
| 22:40.00 | hlieberman | Makes sense. |
| 22:40.03 | p3nguin | It keeps the main welcome message from playing for ever. |
| 22:40.24 | p3nguin | It could also be done with a count and a loop, but the timeout was quicker for now. |
| 22:40.59 | p3nguin | For my own system, I play a loop of the main message three times and then when it times out the third time, it says sorry you're having trouble, and hangs up. |
| 22:42.47 | hlieberman | Otherwise WaitExten would just keep shoving you back into the main menu and let someone eat up all your time. |
| 22:43.28 | p3nguin | WaitExten() just sits and waits until you dial an extension. |
| 22:43.54 | p3nguin | WaitExten(10) waits for 10 seconds before going on to the next step. |
| 22:44.01 | p3nguin | which is to Goto(1). |
| 22:44.16 | hlieberman | Whoops. That was a mistake. Learned - thanks. |
| 22:44.30 | p3nguin | 10 seconds may be too much time, so you can tune it as needed. |
| 22:44.52 | f2knight | I would say 3 - 5 seconds is usually enough |
| 22:45.24 | p3nguin | Some people are slow, though. |
| 22:48.33 | f2knight | p3nguin, this is true... |
| 22:48.43 | f2knight | p3nguin, ever use kamailio? |
| 22:53.12 | f2knight | had a brain dead hour. Just spent last hour cussing my self left and write because database changes were not being reflected... Reason.. Was on the wrong server.. Doh!!!! |
| 22:55.09 | *** join/#asterisk saliak (~kailas@ip68-9-228-184.ri.ri.cox.net) |
| 22:56.53 | p3nguin | hlieberman: http://pastebin.com/xqgDEAPK |
| 22:56.57 | p3nguin | f2knight: No. |
| 22:58.50 | hlieberman | Makes sense. |
| 22:59.02 | hlieberman | Need to look up the Read() syntax to understand it fully. Give me a sec. |
| 22:59.37 | hlieberman | Yeah, OK, makes sense. |
| 23:00.26 | p3nguin | It's going to save the number entered to MedicalNumber, play the beep sound when it's ready to read the number, limited to a 10-digit phone number, give the caller two chances to get it right, and they need to get it done in 20 seconds or less. |
| 23:01.11 | hlieberman | Right. I assume it works entirely off of the length of the string, not with a #, correct? |
| 23:01.29 | hlieberman | Should I craft the sound file to remind people to start with the area code, not with +1? |
| 23:01.40 | hlieberman | Otherwise it will truncate. |
| 23:02.02 | p3nguin | When they press 9 at the main menu, they will be prompted for authentication. In this concept, you'll have to populate the password file with appropriate information. |
| 23:02.33 | p3nguin | Yes, using this design, it needs 10 digits, without prepending a 1. |
| 23:02.59 | p3nguin | And then the 1 for dialing is written in the Dial() command on options 1 and 2 in the main menu. |
| 23:03.26 | p3nguin | Assuming you need the 1... you can remove it if you don't need to dial it. |
| 23:03.59 | hlieberman | WHen you use Dial(), does the person connected hear the DTMF of dialing the other number before it patches the calls together? |
| 23:04.13 | p3nguin | no |
| 23:04.33 | p3nguin | They just hear ringing (if things are working correctly). |
| 23:04.33 | hlieberman | Good. |
| 23:04.45 | p3nguin | You could make it play music while they wait, even. |
| 23:05.40 | hlieberman | And as soon as the other end picks up, it will connect the two? Is there a way to require confirmation, so we don't get these calls dumped into people's VMs? Or is it better to just use a timeout and hope their VM pickup is set further than n number of rings? |
| 23:06.40 | p3nguin | Someone mentioned FollowMe() earlier... that provides the "press 1 to accept the call or 2 to reject it" on the callee's phone. |
| 23:07.03 | hlieberman | And I can substitute using that and just use your Set() and DBGet() in the followme.conf file. |
| 23:07.27 | p3nguin | I don't think it would be quite that simple. |
| 23:07.59 | hlieberman | Hm. |
| 23:08.27 | hlieberman | If it would be much more complicated, I'll leave it the way it is for testing, and let the end users determine if we need to try and invest the time in making FollowMe work. |
| 23:08.40 | p3nguin | And it's just DB(), not DBGet(). |
| 23:09.01 | p3nguin | (unless you're using something really old with a DBGet app in it.) |
| 23:11.07 | hlieberman | Nope. |
| 23:11.08 | p3nguin | FollowMe() has a config file which you have to set a number, such as a person's extension, and a number to follow. You'd be able to enter the number from that menu, but you would have to specify all possible phone numbers to follow in the conf ahead of time. |
| 23:11.42 | p3nguin | Or maybe there is a way to use static realtime followme, but I'm not aware of it. |
| 23:12.12 | hlieberman | Oh, FollowMe() can't be done on the fly? |
| 23:12.28 | p3nguin | If you can use FollowMe() from a database, you could edit the DB on the fly, then it would work similar to how I've thought it out so far. |
| 23:12.46 | p3nguin | I just don't know if FollowMe works in realtime. |
| 23:13.03 | p3nguin | I run an embedded appliance, so using realtime for me would be a downgrade. |
| 23:13.30 | hlieberman | Well, I appreciate your help, p3nguin. You've been great. This will give me something workable now, rather than what I was doing. |
| 23:13.37 | hlieberman | Yeah, I can imagine. |
| 23:13.54 | p3nguin | It's a start, but there are certainly improvements that could be made. |
| 23:14.25 | hlieberman | I'm sure. |
| 23:14.32 | hlieberman | Any suggestions where? |
| 23:14.39 | p3nguin | The authentication procedure could be changed to make it a bit more secure, for one. |
| 23:15.14 | p3nguin | The entering of a number could be changed to callerid num like we discussed earlier. |
| 23:15.21 | p3nguin | Let me re-write that part now. |
| 23:15.25 | hlieberman | How would you lock down the authentication part further? |
| 23:15.38 | hlieberman | Perhaps also with callerid? Callerid specific authentication? |
| 23:15.50 | hlieberman | Though that can also be faked, it increases the difficulty of guessing. |
| 23:16.28 | p3nguin | Without putting more thought into it and maybe putting it into testing, I'm not sure how I'd change the authentication. |
| 23:16.43 | hlieberman | Nothing is static. |
| 23:17.43 | p3nguin | Why would I want to call in using someone else's phone number as CID, enter in someone else's PIN that I've managed to guess, and then get a callback to that spoofed number? |
| 23:18.06 | p3nguin | It would make more sense to use my own phone number, guess the PIN, and get the calls to my own number so I can answer them. |
| 23:18.45 | hlieberman | But if we pre-authenticate the numbers in the authenticate file, you can't do that unless your number is already in the file. |
| 23:18.50 | hlieberman | And then you can only set it to your own number. |
| 23:19.06 | p3nguin | One way to improve that authentication would be to get a list of all people who will use the callback system. |
| 23:19.19 | p3nguin | Make them "register" their phone numbers. |
| 23:19.51 | hlieberman | I can do that. |
| 23:19.55 | p3nguin | If you get that, that would also help to set up the FollowMe instead of direct dial. |
| 23:20.38 | p3nguin | Let me make some changes to that concept real quick. |
| 23:20.56 | hlieberman | Sure. |
| 23:21.01 | hlieberman | Thanks. :) |
| 23:26.53 | p3nguin | hlieberman: http://pastebin.com/y7G5M8Xz |
| 23:28.23 | p3nguin | Then you need to get them to register the phone number and PIN "key pair" and enter it into the file used for authentication. |
| 23:29.07 | p3nguin | So now they can only use the PIN for the phone number registered when calling from the phone number registered. |
| 23:29.57 | p3nguin | If I give you phone number 3145551212 and PIN 1234, and I call from 3149691077, PIN 1234 will not work right. |
| 23:30.34 | hlieberman | Good. That's awesome. |
| 23:30.44 | p3nguin | PIN 1234 would expect 3145551212, and then the GotoIf() would verify it and fail, sending me back to Authenticate() again. |
| 23:31.00 | hlieberman | Awesome! :) |
| 23:31.05 | hlieberman | Thank you so much, p3nguin. |
| 23:31.30 | p3nguin | I'm not sure that any two entries can share a PIN, though. |
| 23:31.43 | p3nguin | Since it would try looking for the PIN, two matches could cause a problem. |
| 23:32.22 | p3nguin | You could even assign a PIN when they register their phone number. |
| 23:32.34 | p3nguin | That way you would have the control to never duplicate a PIN. |
| 23:32.50 | p3nguin | I think I like that better anyway. |
| 23:32.59 | hlieberman | Agreed. That way, no 1234. |
| 23:33.00 | p3nguin | You give me the phone number, I give you a PIN. |
| 23:33.32 | p3nguin | You could use 1234, and it would only work when called from the phone number specified. |
| 23:33.57 | p3nguin | It's much harder to guess a PIN when you also have to guess (or determine ahead of time) the phone number where the PIN works. |
| 23:34.04 | hlieberman | Right. |
| 23:34.30 | p3nguin | But even if someone guesses the phone number and the PIN, all it would do is send calls to that phone number which you've registered in the system. |
| 23:34.39 | p3nguin | Not beneficial to the person doing it. |
| 23:34.53 | hlieberman | Yup. |
| 23:35.02 | p3nguin | I kind of like this method. |
| 23:35.11 | p3nguin | The more I explain it, the better it seems. |
| 23:35.50 | p3nguin | In order to register phone numbers into the system, how would you do it? |
| 23:36.06 | p3nguin | Manually or via some app you'll write, for examples? |
| 23:36.55 | hlieberman | Manually to start. There won't be that many people, so I don't expect the burden to be too high. If it becomes so, then I'll write a little PHP frontend and then dispense username/passwords to people authorized to edit that file. |
| 23:38.04 | p3nguin | The format of the password file called by Authenticate() needs to be, one per line, phone-number:hashedPIN |
| 23:38.18 | p3nguin | Do you know how to hash it to enter it manually? |
| 23:38.19 | hlieberman | Where hash is straight md5. |
| 23:38.23 | hlieberman | Yup. |
| 23:39.01 | *** part/#asterisk rampage73 (~rampage73@184.2.147.197) |
| 23:39.02 | p3nguin | Make sure you don't echo the newline character when doing it. |
| 23:39.17 | hlieberman | echo -n. Roger. |
| 23:39.39 | p3nguin | That's a pretty common reason that the PINs don't work from a file. |
| 23:41.52 | *** join/#asterisk darkbasic (~quassel@host37-245-static.119-2-b.business.telecomitalia.it) |
| 23:45.49 | p3nguin | I guess you have pretty much a drop-in solution now. |
| 23:45.55 | hlieberman | Indeed, it seems I do. |
| 23:45.59 | hlieberman | Thank you so much for your help, p3nguin. |
| 23:46.22 | hlieberman | You helped me far more than I expected. It's thanks to you that I'll be able to get this up and running in time for the rush on Monday. |
| 23:46.32 | p3nguin | I hope it works out. |
| 23:46.42 | p3nguin | Test it thoroughly. |
| 23:47.54 | hlieberman | I will. |
| 23:48.03 | p3nguin | Always check the file to make sure the PIN hash doesn't already exist before you put a new one in. |
| 23:48.49 | p3nguin | I'd probably use apg to whip out a few random PINs for me. |
| 23:49.12 | beta2k | Ok, moved my ip430's from tftp to ftp for provisioning and now they just reboot faster :) |
| 23:51.19 | *** join/#asterisk vinhdizzo (~vinh@pool-173-60-112-55.lsanca.fios.verizon.net) |
| 23:52.37 | p3nguin | apg -a 1 -n 20 -m 4 -x 5 -M N |
| 23:54.05 | p3nguin | for pin in `apg -a 1 -n 20 -m 4 -x 5 -M N`; do echo -n $pin|md5sum;done |
| 23:54.33 | p3nguin | haha |
| 23:54.37 | p3nguin | Now you don't know what PIN it was! |
| 23:55.33 | hlieberman | That's security for you. ;) |
| 23:55.59 | p3nguin | for pin in `apg -a 1 -n 8 -m 4 -x 5 -M N`; do echo $pin;echo -n $pin|md5sum;echo;done |
| 23:56.04 | p3nguin | Yeah, for sure. |
| 23:58.14 | p3nguin | That ought to work. |
| 23:59.09 | p3nguin | I have no idea what the odds of getting a duplicate would be, but I bet it's in your favor. |