| 00:02.12 | zamba | so what to do? |
| 00:03.13 | WIMPy | Ok. Got the same issue. |
| 00:03.34 | zamba | so it wasn't murphy after all :) |
| 00:04.01 | WIMPy | Maybe 1.10 is too old for the git mISDNuser indeed. |
| 00:04.27 | WIMPy | Yes, seems indeed to be the case. |
| 00:05.27 | WIMPy | I must have upgraded them in opposite order so I didn;t notice. |
| 00:05.46 | zamba | so downgrade the mISDNuser version? |
| 00:06.08 | zamba | is that the solution? |
| 00:06.34 | WIMPy | There is only one tarball as far as I can see. |
| 00:07.17 | WIMPy | Or none. |
| 00:07.22 | zamba | http://www.misdn.org/downloads/releases/ |
| 00:08.04 | zamba | mISDN is the stuff that's already in the kernel.. so what we need is the mISDNUser part? |
| 00:08.21 | WIMPy | Correct. |
| 00:08.32 | WIMPy | But that looks like the misdn1 stuff. |
| 00:09.10 | WIMPy | The one in the parent dir might be the right one. |
| 00:09.30 | zamba | are you testing? |
| 00:09.51 | WIMPy | yes |
| 00:11.10 | WIMPy | No. That's an old 1_1_9.1. |
| 00:12.03 | WIMPy | Great. Looks like it's gone. |
| 00:12.10 | WIMPy | :-( |
| 00:12.21 | *** join/#asterisk WindBack (~quassel@190.123.120.190) |
| 00:12.28 | zamba | should be in git somewhere |
| 00:12.46 | zamba | just roll back to a previous release there? |
| 00:13.02 | WIMPy | Sure. |
| 00:13.06 | WIMPy | But how many? |
| 00:13.19 | zamba | hehe, i have -NO- idea :) |
| 00:14.02 | WIMPy | That's the issue. |
| 00:14.07 | zamba | but looks like there's active development happening sitll |
| 00:14.17 | zamba | karstein keil checked in some code 6 days ago |
| 00:14.40 | WindBack | [TK]D-Fender: Hi. I want to ask you something: Is there any way of doing this: same => n(dial),Set(SOMEVAR=${DB(someFamily/${ANOTHERVAR})}) |
| 00:15.42 | WIMPy | Always good to have a backup even if it's rather old: http://voice.yeti.dk/mISDNuser-20101024.tar.bz2 |
| 00:16.15 | WIMPy | I hope that's usable. |
| 00:16.43 | zamba | please, please, PLEEEASE compile :p |
| 00:16.54 | zamba | *fingers crossed* |
| 00:17.04 | zamba | well, it compiled :) |
| 00:17.20 | WIMPy | Ok, now lcr 1.10 needs to like it. |
| 00:17.24 | zamba | now let's try lcr |
| 00:17.41 | zamba | please, please, pleeeease like it! :) |
| 00:18.05 | [TK]D-Fender | WindBack: That's close... missing the priority though.... |
| 00:18.07 | zamba | not exactly a clean compile, but so far only warnings |
| 00:18.09 | [TK]D-Fender | Wait... |
| 00:18.10 | [TK]D-Fender | nvm |
| 00:18.15 | zamba | compiled! |
| 00:18.29 | [TK]D-Fender | WindBack: looks about right..... |
| 00:19.42 | WIMPy | Ok, so you got chan_lcr now as well? |
| 00:20.10 | zamba | depends |
| 00:20.18 | WIMPy | o.O |
| 00:20.20 | WindBack | [TK]D-Fender: It doesn't work. Seems that I cant use a variable as key in DB function |
| 00:20.25 | zamba | it has been built, yeah |
| 00:20.27 | zamba | yup |
| 00:20.29 | zamba | got it |
| 00:20.39 | WIMPy | finally. |
| 00:20.46 | zamba | dunno where make install put it, though |
| 00:21.00 | zamba | ah, /usr/lib/asterisk/modules |
| 00:21.09 | zamba | cool, cool |
| 00:21.24 | WindBack | [TK]D-Fender: I was googling and nobady uses variable as key in any example |
| 00:22.02 | [TK]D-Fender | I've seen lots Callback script samples, forwarding flags, etc |
| 00:22.34 | zamba | WIMPy: yup! it's there :) |
| 00:22.47 | WIMPy | WindBack: I'm using variabled there all the time. |
| 00:23.08 | WindBack | WIMPy: can you show me an example please |
| 00:23.37 | zamba | WIMPy: and it's loaded! |
| 00:23.56 | zamba | WIMPy: tomorro will be an interesting day.. will try connecting this baby directly to the line |
| 00:23.59 | wtf911 | [TK]D-Fender: what's a simple way to have a log file of calls recieved/made?...how would i just get a log file saved that's the equivalent of saving "core set verbose 10"? |
| 00:24.02 | WIMPy | exten => setcf,n,Set(DB(cf/${cid}/${type}_d)=${num}) |
| 00:24.19 | WIMPy | WindBack: ^^ |
| 00:26.56 | wtf911 | WIMPy: maybe you can answer my question? |
| 00:27.45 | WindBack | WIMPy: thanks, I will see why it is not working for me |
| 00:27.54 | WIMPy | The first or the second question? Logger.conf or your CDRs. |
| 00:29.18 | wtf911 | well i tried this "/tmp/asterisk.log => notice,warning,error" in my logger.conf file to see if that would do what i wanted but as im sure you know it didn't |
| 00:30.47 | WIMPy | What about "verbose"? |
| 00:31.57 | wtf911 | try adding verbose? "/tmp/asterisk.log => notice,warning,error,verbose" is that what you mean? |
| 00:32.08 | WIMPy | yes |
| 00:32.54 | zamba | WIMPy: and that should basically be it, right? |
| 00:33.14 | zamba | WIMPy: are you familiar with the alcatel 4200 pbx? |
| 00:33.18 | WIMPy | zamba: Yes |
| 00:33.31 | zamba | WIMPy: you are? |
| 00:33.49 | WIMPy | No, didn;t get any Alcatel between my fingers unfortunatly. |
| 00:34.03 | zamba | well, maybe we can ship you this once we've replaced it :) |
| 00:34.36 | zamba | btw.. how is the pin configuration for ISDN cables? |
| 00:34.37 | WIMPy | How much older is it than the 44xx? |
| 00:34.42 | zamba | have no idea |
| 00:34.44 | zamba | absolutely no idea |
| 00:35.03 | WIMPy | http://voice.yeti.dk/Asterisk_vs_ISDN/7 |
| 00:35.07 | zamba | i'm not sure i still have ISDN cables (S0? is that what they're called?) around still.. |
| 00:35.20 | WIMPy | I always wanted to get my hands on a 44xx. |
| 00:35.57 | zamba | oh, so regular cat5 will be fine? |
| 00:36.27 | zamba | sweet deal |
| 00:36.38 | zamba | i'm actually quite exited about this :) |
| 00:37.20 | WIMPy | Gerat. I found a ways to make Asterisk lose channels without hanging up. |
| 00:37.32 | zamba | though i feel quite certain this will be a disappointment :) |
| 00:37.42 | WIMPy | Why? |
| 00:37.49 | WIMPy | And what do you expect? |
| 00:37.54 | WIMPy | Or want? |
| 00:38.06 | zamba | i want to be able to call and receive incoming calls :) |
| 00:38.25 | WIMPy | Might be disappointing, yes. |
| 00:38.33 | zamba | oh? why? |
| 00:38.37 | WIMPy | But there's more you can do :-) |
| 00:38.51 | WIMPy | Just making calls is boring, isn;t it? |
| 00:38.59 | zamba | oh, no.. it isn't :) |
| 00:39.05 | zamba | if this works.. i'll be over the moon |
| 00:40.40 | wtf911 | WIMPy: good stuff! the only thing is i have to set verbosity to say 10 first or it starts at 0 and doesn't log what i want....what would i do to have it start in verbosity 10? (also what would be the minimum verbosity for it to grab the numbers of incoming and outgoing calls?) |
| 00:41.46 | WIMPy | wtf911: you can set debug and verbose leven in asterisk.conf. |
| 00:41.55 | zamba | WIMPy: this should be sufficient to test the dialing, right: http://pastie.org/3933334 ? |
| 00:42.08 | WIMPy | And the numbers aren't explicitly displayed. |
| 00:42.09 | zamba | that's my whole extensions.conf |
| 00:42.21 | zamba | .. so far |
| 00:42.36 | zamba | just so i know i've understood how the channel module works |
| 00:42.37 | WIMPy | zamba: Looking good |
| 00:42.40 | zamba | and how it all ties together |
| 00:42.42 | zamba | sweet |
| 00:42.57 | WIMPy | Err |
| 00:43.08 | WIMPy | Apart from the pattern. |
| 00:43.20 | WIMPy | _X. would work a lot better. |
| 00:43.41 | zamba | good thing someone's still awake :) |
| 00:44.16 | zamba | when looking at lcradmin state.. is there anything there i can use to debug the connection? |
| 00:44.26 | zamba | right now it's of course down, since it's not connected to the NT box |
| 00:44.32 | WIMPy | Yes. All of it :-) |
| 00:44.33 | zamba | but once it is.. will it output anything useful? |
| 00:44.47 | zamba | Ext(port 0: diva201.1) TE ptp l2hold use:0 L2 unkn L1 down |
| 00:44.50 | zamba | that's what it says now |
| 00:45.02 | WIMPy | It might stay down until you actually try to place a call. |
| 00:45.07 | zamba | k |
| 00:45.20 | WIMPy | But you might as well get an up as soon as you plug it in. |
| 00:45.30 | WIMPy | That depends on th lines configuration. |
| 00:45.34 | zamba | ooooh! |
| 00:45.37 | zamba | i'm so exited |
| 00:45.44 | zamba | wish we still had ISDN at home :) |
| 00:45.56 | zamba | though we disconnected that years ago |
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| 00:46.13 | WIMPy | There's an power save option that brings L1 down when the line is not in use. |
| 00:46.20 | WIMPy | But AFAIK that's only used on ptmp lines. |
| 00:46.54 | zamba | and the funny thing is that i actually threw away an eicon diva card in easter.. because i thought i'd never get use for it :p |
| 00:47.03 | zamba | good thing i found one in my basement yesterday :) |
| 00:47.25 | WIMPy | I still got a bunch of the bog standard HFC-s cards. |
| 00:47.37 | zamba | which definitely works? |
| 00:47.47 | zamba | buy you some beers? :) |
| 00:47.52 | WIMPy | Yes. They are the most common ones. |
| 00:49.13 | WIMPy | Actually I once get a very nice bottle of whinsky in a parcel. |
| 00:49.22 | WIMPy | got |
| 00:50.20 | zamba | what's the exchange rate? :) |
| 00:51.12 | WIMPy | That was instead of the initially offered box of beer. |
| 00:51.46 | WIMPy | The beer would have been cheaper, but shipping would probably have been quite expensive. |
| 00:52.14 | zamba | but you're from denmark, and i'm from norway, so anything alcohol-related would be unfair :) |
| 00:52.44 | WIMPy | Actually I'm about 10km further away ind Germany. |
| 00:52.48 | zamba | ah, ok |
| 00:52.52 | WIMPy | So yes, that would be a bad deal. |
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| 01:09.43 | phix | WIMPy: goto germany for cheap beer :p |
| 01:10.36 | WIMPy | That's where I am. |
| 01:10.42 | phix | you might see me there :p |
| 01:10.53 | WIMPy | But I'm not really a fan of beer. |
| 01:11.06 | phix | jager? |
| 01:11.16 | WIMPy | Urgs |
| 01:11.20 | phix | ah |
| 01:11.42 | WIMPy | No. Whiskey or log drinks / girlie drinks :-) |
| 01:11.49 | phix | haha |
| 01:18.58 | wtf911 | WIMPy: i checked a log after a call and it's missing the sip information i need to have the number of who called in...what can i do to get that logged too? :) |
| 01:19.32 | WIMPy | wtf911: What's wrong with the CDRs? |
| 01:20.09 | WIMPy | The easiest way to get obvious information wouild be to put a Verbose() in to your dialplan otherwise. |
| 01:22.15 | wtf911 | CDRs? |
| 01:22.35 | WIMPy | The Call Detail Records. |
| 01:23.04 | WIMPy | /var/log/asterisk/cdr*/* usually |
| 01:24.36 | wtf911 | i have... |
| 01:24.39 | wtf911 | root@unknown:/opt/var/log/asterisk# ls |
| 01:24.39 | wtf911 | cdr-csv cdr-custom event_log messages queue_log |
| 01:26.48 | wtf911 | both cdr folders are empty? |
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| 01:40.48 | WIMPy | No cdr modules loaded? |
| 01:40.58 | WIMPy | module show like cdr |
| 01:41.48 | WIMPy | Also cdr_*.conf |
| 01:44.01 | p3nguin | I sure wish I knew why this googletts.agi thing doesn't give me any sound. |
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| 02:00.18 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 02:01.31 | rspp2k | how do I make a dialplan that makes two calls and connects them together? |
| 02:01.57 | WIMPy | And how do you get there? |
| 02:02.19 | rspp2k | I'm using call files for outgoing calls |
| 02:02.32 | rspp2k | they point at DAHDI/2/5551212 |
| 02:02.43 | WIMPy | You always have to specify both ends of the call there. |
| 02:03.03 | rspp2k | what I'm doing is using the dialplan to test an IVR system. |
| 02:03.13 | rspp2k | I would like to be able to listen in on the outbound call |
| 02:03.28 | rspp2k | on my cell or sip phone |
| 02:03.48 | rspp2k | I could just record the call and listen to the wav, but being able to listen live would be great. |
| 02:04.35 | WIMPy | ChanSpy(), ExtenSpy(), ConfBridge() |
| 02:04.43 | WIMPy | Pich one :-) |
| 02:04.53 | rspp2k | reading man pages now. thanks! |
| 02:05.43 | cusco | set spygroup=something; chanspy(something) |
| 02:06.13 | cusco | p3nguin: what google tts agi? |
| 02:06.24 | cusco | i sort of made one of my own |
| 02:06.37 | cusco | not a agi really but works |
| 02:07.29 | cusco | funny that I called it 'googletts' too |
| 02:10.35 | p3nguin | There's an actual project called google tts. |
| 02:10.44 | p3nguin | It doesn't produce any sounds for me. |
| 02:11.17 | cusco | I made myself a simple php that takes the sentence/string as a argument |
| 02:11.26 | cusco | ad returns only the filename of the sound |
| 02:11.45 | cusco | I use Set(sound=${SHELL} ... |
| 02:11.56 | cusco | and playback(${sound}); |
| 02:12.18 | cusco | php uses sox to convert it from mp3 |
| 02:13.13 | rspp2k | WIMPy cusco: Chanspy fits the bill. you rock! |
| 02:13.36 | cusco | I also added the host translate.google.com to /etc/hosts so php works fast enough |
| 02:13.42 | cusco | resolving that was |
| 02:13.54 | cusco | rspp2k: good for you :) |
| 02:14.35 | p3nguin | If I use the perl script on my desktop, googletts works. |
| 02:15.00 | cusco | as a stand alone? |
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| 02:19.45 | p3nguin | ./googletts-cli.pl -l en -t "Hello world" |
| 02:19.50 | p3nguin | works correctly |
| 02:21.26 | p3nguin | Maybe it is a problem with perl* on the asterisk box. |
| 02:23.20 | cusco | here is my web pagie http://cusco.tretas.eu/scripts/tts/index.phps |
| 02:25.41 | cusco | and the script I call in asterisk: http://paste.debian.net/170070/ |
| 02:34.32 | dijib | i never did get the google tts and others working |
| 02:34.35 | dijib | hows everyone |
| 02:35.10 | carrar | alive |
| 02:36.38 | wtf911 | the reason i thought it would be a good idea to log verbose output is because (as p3nguin knows) i have my caller id script which accepts people entering numbers and i want those entered numbers logged....but when i log just that i don't get who is calling in the first place (just my ipkall set number 808 in my case) i need it to pass the caller id or what not...i get that CDR would do this |
| 02:36.39 | wtf911 | but is there a way to have this information end up in one place not two? something i can add to logger.conf or extensions.conf? |
| 02:36.39 | *** join/#asterisk mynoon (~my@46.20.109.147) |
| 02:38.07 | carrar | why not put it in your own db |
| 02:39.01 | cusco | wtf911: i'm not following |
| 02:39.07 | carrar | me either really |
| 02:39.17 | wtf911 | ok let me rephrase that :) |
| 02:39.18 | carrar | accept digits in the dialplan |
| 02:39.31 | carrar | log them to where ever you want |
| 02:39.39 | carrar | via db calls |
| 02:39.50 | carrar | or agi |
| 02:39.55 | cusco | cdr already logs everything for you, and its output is well structured |
| 02:40.38 | wtf911 | i would like to log ...the caller id of who is calling, the 10 digit number they enter for the caller id to be "spoofed", and the 10 digit number they want to be connected to |
| 02:40.58 | wtf911 | and the time they call would be nice |
| 02:41.05 | carrar | CDR |
| 02:41.25 | carrar | could use custome feild for the spoof number |
| 02:41.34 | carrar | or dst |
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| 02:41.42 | carrar | err not dst sorry |
| 02:42.02 | cusco | lastapp , data |
| 02:42.06 | cusco | whatever |
| 02:42.11 | wtf911 | (and please forgive me for being a noob with asterisk i'm new to it so i really appreciate the help!) |
| 02:42.12 | carrar | lots O places |
| 02:42.12 | cusco | wtf911: so, what is the problem? |
| 02:42.33 | cusco | wtf911: where do you want to log that info? |
| 02:42.40 | carrar | read up on CDR database stuff |
| 02:43.13 | carrar | you can do it |
| 02:43.24 | carrar | YCDI |
| 02:43.32 | carrar | !! |
| 02:43.51 | wtf911 | the problem is i don't know how to accomplish that lol...what i have right now is just logger.conf with "/tmp/asterisk.log => notice,warning,error,verbose" |
| 02:44.03 | carrar | Can you read? |
| 02:44.13 | carrar | Can you google? |
| 02:44.19 | cusco | logger.conf will log asterisk internal handling, not your needs |
| 02:44.24 | wtf911 | so CDR will pick up the digits people enter? |
| 02:44.40 | carrar | if you save it to the CDR |
| 02:44.42 | bdfoster | carrar, do you respect others? are you that heartless? |
| 02:44.44 | cusco | wtf911: cdr picsk up everything, do you have it enable? |
| 02:44.52 | carrar | I am that heartless |
| 02:45.02 | carrar | haha |
| 02:45.13 | carrar | I am trying to help him |
| 02:45.13 | bdfoster | carrar, welcome to #asterisk. you've officially made it. |
| 02:45.15 | carrar | help himself |
| 02:45.25 | wtf911 | ok give me a moment i will have the module load first |
| 02:45.30 | cusco | but you may not need cdr |
| 02:45.45 | carrar | after years of asterisk someone finally welcomes me |
| 02:45.50 | carrar | about time!! |
| 02:45.59 | wtf911 | well if i could just add the caller id of the incoming call to logger.conf i would be set |
| 02:46.12 | bdfoster | carrar, it's not a joke. you need to do it some other way. so, do me a favor and work on the attitude. you're scaring away the new users. |
| 02:46.16 | wtf911 | or get it added to my asterisk.log file |
| 02:46.21 | cusco | sometimes I use a piece of dialplan that validates numbers in mass... I just use System |
| 02:46.32 | cusco | System(echo '${ID}|${EXTEN:4}|${DIALSTATUS}|${HANGUPCAUSE}|' >> /root/validateNumbers/fel.result); |
| 02:46.34 | carrar | wt |
| 02:46.35 | carrar | f |
| 02:46.51 | carrar | who have I scared away? |
| 02:47.29 | bdfoster | ill tell you that i see users coming from both asterisk and freepbx EVERY DAY because they say the community sucks. |
| 02:47.39 | carrar | sure |
| 02:47.40 | bdfoster | and they go to freeswitch. |
| 02:47.42 | carrar | blame all that on me |
| 02:47.45 | carrar | if you need oto |
| 02:47.51 | bdfoster | im not blaming it all on you |
| 02:47.57 | carrar | you're scaring away the new users. |
| 02:47.59 | carrar | seems like it |
| 02:48.00 | bdfoster | it's an epidemic |
| 02:48.23 | cusco | wtf911: see what that System(); did ? |
| 02:48.26 | wtf911 | cdr_custom.so cdr_manager.so ... would i need one, both, or to load some other module i dont know of? |
| 02:48.40 | wtf911 | that would right that info to a file |
| 02:48.47 | cusco | right |
| 02:48.56 | cusco | thats one way of logging it |
| 02:48.58 | wtf911 | could i make it append caller id to my asterisk.log? |
| 02:49.10 | cusco | you just replace those variables for yours |
| 02:49.20 | bdfoster | the funny thing is, id rather have the new users coming from asterisk. i have no vested interest in telling you the reasons why people are switching. |
| 02:49.49 | cusco | say ${EPOCH}|${CALLERID(num)}|${CALLERID(dnid)}|${Variable holding the destination} |
| 02:50.57 | cusco | although cdr automatically logs anything you need in a well structured format |
| 02:51.04 | carrar | please twitter about it and move on |
| 02:51.59 | wtf911 | and so i'm not blind following that...what is epoch for? |
| 02:51.59 | bdfoster | ...you look at my tweets? |
| 02:52.10 | bdfoster | feels loved. |
| 02:52.22 | carrar | no, but that expression works for most people |
| 02:52.35 | carrar | heh |
| 02:52.36 | bdfoster | meh. |
| 02:53.22 | cusco | wtf911: epoch is unix time stamp |
| 02:55.16 | wtf911 | for what it's worth i would say everyone has been very nice to me here...i can say that having jumped into trying asterisk to make a little caller id spoofing thing for myself and a few friends, that no amount of googling would have got me working (at least not for countless hours) that the help i got here did......besides not VNC viewing me everyone was very informative :) |
| 02:56.19 | carrar | !! |
| 02:56.32 | wtf911 | i'll give this a try cusco and let you know how it goes :) ...a one liner to my extensions.conf would be a great solution in place of logging a bunch of not needed stuff |
| 02:59.17 | wtf911 | ah yes one more question ^.^ |
| 02:59.41 | wtf911 | i currently have... |
| 02:59.43 | wtf911 | exten => 808,n,Playback(privacy-prompt) |
| 02:59.44 | wtf911 | exten => 808,n(collect),Read(digito,,10) |
| 03:00.12 | wtf911 | which doesn't accept input for the 10 digits until the privacy-prompt is done playing...is this what Background() is for or something? |
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| 03:30.51 | wtf911 | what does this mean? == Spawn extension (ipkall-inbound, 808, 8) exited non-zero on 'SIP/ipkall-00000000' |
| 03:31.56 | *** join/#asterisk retentiveboy (~retentive@74-95-28-33-Atlanta.hfc.comcastbusiness.net) |
| 03:32.34 | wtf911 | i added this line ... exten => 808,n,System(echo "${DATETIME} - ${CALLERID} - ${digitoa} - ${digitob}" >> /tmp/asterisk.log) |
| 03:40.26 | rspp2k | how do I pass a range of channels to dial? I want to use one of my free 23 DAHDI channels |
| 03:40.56 | rspp2k | something like Dial(DAHDI/1-23/5551212) |
| 03:44.17 | wtf911 | use the & symbol |
| 03:44.23 | rspp2k | got it.. DAHDI/g0/5551212 |
| 03:44.36 | wtf911 | oh nevermind x.x |
| 03:45.03 | rspp2k | forgot about groups. but just saw the & that makes the calls simultaneously. might be useful in my application. |
| 03:45.21 | rspp2k | I am load testing. |
| 03:46.44 | wtf911 | it got quiet in here :( |
| 03:47.52 | *** join/#asterisk wudles (~wudles@gateway.secureinstrument.com) |
| 03:50.24 | kaldemar | wtf911: the spawn line just means that the channel exited the extension. verbosity, nothing more. |
| 03:51.40 | wtf911 | ah help has arrived :D |
| 03:51.55 | wtf911 | i tried this just now... exten => 808,n,System(echo "hi" >> /tmp/asterisk.log) |
| 03:52.03 | wtf911 | and it still disconnects my call when it hits that |
| 03:52.05 | *** join/#asterisk greenwolf (a6f80176@gateway/web/freenode/ip.166.248.1.118) |
| 03:56.36 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 03:57.58 | kaldemar | wtf911: do you have anything after that in the extension? i.e. what is the problem? |
| 03:58.04 | greenwolf | friday night...what to do..what to do |
| 03:58.25 | greenwolf | has anyone here ever used asterisk alongside with opensips as a dispatcher or load balancer? |
| 03:59.45 | wtf911 | kaldemar: it's then supposed to dial a number ...let me paste you everything 1 sec |
| 04:00.17 | wtf911 | http://pastebin.com/zYVPkQsq |
| 04:00.36 | wtf911 | if i take out the System( line it works great |
| 04:05.39 | kaldemar | what does CLI output say for the whole call? |
| 04:07.21 | wtf911 | it goes through normally, says asking user for 10 digits, i enter them, then i get the == Spawn extension .... |
| 04:07.33 | kaldemar | use pastebin |
| 04:07.51 | wtf911 | sure sure 1 sec while i make the call to get the stuffs |
| 04:10.17 | wtf911 | http://pastebin.com/QUc1pYNU |
| 04:13.21 | kaldemar | "dialplan show 808@ipkall-inbound" |
| 04:14.35 | wtf911 | http://pastebin.com/4fWzPukz |
| 04:20.34 | kaldemar | that sure is strange. but so is labeling two different priorities with the same name. remove the "collect" labels. |
| 04:21.35 | wtf911 | like i said it does work fine if i remove the system line...but ok :D |
| 04:22.00 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 04:22.10 | wtf911 | ill try it now with collect removed |
| 04:23.25 | wtf911 | same thing |
| 04:24.20 | wtf911 | :o maybe i dont have a module loaded? |
| 04:24.33 | kaldemar | i tried that on a 10.4.0 test box, works here. |
| 04:24.47 | wtf911 | let me make sure i have app_system.so loaded |
| 04:24.55 | kaldemar | it would complain about a non-registered application. |
| 04:25.03 | wtf911 | and thank you for testing it for me!!! *shakes hand* |
| 04:25.37 | kaldemar | you might want to change the DATETIME stamp to something valid. |
| 04:26.44 | wtf911 | didn't have app_system.so loading so let me see if it works now |
| 04:26.49 | wtf911 | x.x |
| 04:27.06 | kaldemar | like ${STRFTIME(${EPOCH},Pacific/Auckland,"%Y-%m-%d %H:%M:%S")} |
| 04:28.10 | wtf911 | it works now :o |
| 04:28.19 | wtf911 | just had to add app_system.so to modules.conf ...wow |
| 04:28.42 | wtf911 | let me check what the asterisk.log file output was to see if its getting the right info |
| 04:29.34 | kaldemar | what version are you using? |
| 04:29.38 | *** join/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 04:29.38 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 04:29.46 | wtf911 | ah beautiful ... " - BINGHAMTON NY <XXXXXXXXXX> - XXXXXXXXXX - XXXXXXXXXX" except you're right the timedate thing doesn't work |
| 04:29.52 | wtf911 | how do i show the version? |
| 04:30.04 | wtf911 | nvm |
| 04:30.11 | wtf911 | Asterisk 1.6.2.22, Copyright |
| 04:32.02 | wtf911 | i called a second time to make sure it would add calls to a new line and it does so that's good |
| 04:34.44 | kaldemar | 1.6.2 is EOL, do yourself a favor and migrate to a supported version. |
| 04:35.23 | *** join/#asterisk CRCinAU (~CRCinAU@another.bloody.irc.session.from.crc.id.au) |
| 04:35.27 | *** part/#asterisk CRCinAU (~CRCinAU@another.bloody.irc.session.from.crc.id.au) |
| 04:35.33 | *** join/#asterisk CRCinAU_ (~CRCinAU@another.bloody.irc.session.from.crc.id.au) |
| 04:35.37 | wtf911 | :/ i would but i'm just using what i can with optware ...i installed it with "ipkg install asterisk" |
| 04:35.39 | CRCinAU_ | hmm |
| 04:35.51 | CRCinAU_ | with Asterisk 1.8.12.0... I have compiled in res_fax and res_fax_spandsp |
| 04:36.03 | CRCinAU_ | but whenever I try to use ReceiveFax, I get this: |
| 04:36.03 | wtf911 | actually asterisk18 ...so i could have newer |
| 04:36.05 | CRCinAU_ | [May 19 14:34:56] ERROR[4205]: res_fax.c:784 fax_session_reserve: Could not locate a FAX technology module with capabilities (RECEIVE) |
| 04:41.28 | CRCinAU_ | if I understand what should be going on |
| 04:41.42 | CRCinAU_ | res_fax takes over from app_fax in 1.8... but it requires res_fax_spandsp |
| 04:41.55 | CRCinAU_ | which is cool, because its there and all, but it still fails. |
| 04:42.05 | CRCinAU_ | does anyone have any pointers on what it actually needs? |
| 04:47.21 | CRCinAU_ | the bad thing is that it was working fine in 1.6.x :\ |
| 04:57.01 | ChannelZ | hmm actually res_fax is part of FFA and res_fax_spandsp is something else and they don't work together. |
| 04:57.50 | CRCinAU_ | :\ |
| 04:58.40 | CRCinAU_ | it does say in the text on a make menuselect: |
| 04:58.56 | CRCinAU_ | Spandsp G.711 and T.38 FAX Technologies |
| 04:58.56 | CRCinAU_ | Depends on: spandsp(E), res_fax(M) |
| 04:59.07 | CRCinAU_ | thats the entry for res_fax_spandsp |
| 05:00.37 | CRCinAU_ | however, at the CLI: |
| 05:00.38 | CRCinAU_ | asterisk*CLI> fax show capabilities |
| 05:00.42 | CRCinAU_ | 0 registered modules |
| 05:00.59 | CRCinAU_ | so something is missing - in either the docs, the scripts or res_fax |
| 05:03.09 | ChannelZ | my bad, it's res_fax_digium and _spandsp that clash |
| 05:03.19 | CRCinAU_ | ah - that makes sense. |
| 05:03.54 | CRCinAU_ | its like res_fax doesn't even know that res_fax_spandsp exists. |
| 05:04.03 | ChannelZ | Did res_fax_spandsp actually build? Do you get an error if you manually load it? |
| 05:04.05 | CRCinAU_ | but, /usr/lib/asterisk/modules/res_fax_spandsp.so <-- there it is :\ |
| 05:04.32 | CRCinAU_ | hahah |
| 05:05.05 | CRCinAU_ | of course :) |
| 05:05.05 | CRCinAU_ | Command 'module load res_fax_spandsp.so ' failed. |
| 05:05.05 | CRCinAU_ | [May 19 15:04:25] WARNING[14826]: loader.c:398 load_dynamic_module: Error loading module 'res_fax_spandsp.so': libspandsp.so.2: cannot open shared object file: No such file or directory |
| 05:05.05 | CRCinAU_ | that'd do it. |
| 05:05.17 | CRCinAU_ | Asterisk is probably not looking in /usr/local/lib |
| 05:06.30 | CRCinAU_ | which is probably not set in ld.so.conf |
| 05:07.36 | CRCinAU_ | bingo |
| 05:07.36 | CRCinAU_ | <PROTECTED> |
| 05:07.36 | CRCinAU_ | asterisk*CLI> fax show capabilities |
| 05:07.36 | CRCinAU_ | 1 registered modules |
| 05:07.36 | CRCinAU_ | That'll do it. |
| 05:07.36 | CRCinAU_ | ChannelZ: thanks for that prod.... I didn't think of loading the module manually :) |
| 05:09.34 | ChannelZ | glad it's working now |
| 05:10.12 | CRCinAU_ | me too |
| 05:10.33 | CRCinAU_ | I've been expermenting in doing faxing FXS -> FXO with not much success :( |
| 05:11.54 | CRCinAU_ | so I'm going to try a store + forward for faxes... |
| 05:12.07 | CRCinAU_ | as bridging the call from FXS to FXO doesn't seem to work properly |
| 05:14.33 | ChannelZ | hmm.. I do that with a real fax machine |
| 05:15.43 | *** join/#asterisk screenn (~screenn@37.46.237.217) |
| 05:15.44 | CRCinAU_ | hmmm |
| 05:15.55 | CRCinAU_ | I always seem to get COMM ERROR on my fax |
| 05:16.06 | CRCinAU_ | it'll go fine for 2-3 pages, then bomb out |
| 05:16.34 | CRCinAU_ | so I'm going to try to do it the way I prefer... FXS -> Asterisk -> T.38 -> SIP provider |
| 05:16.45 | CRCinAU_ | failing that, I'll go FXS -> Asterisk -> FXO |
| 05:17.07 | wtf911 | that was a tough one lmao...i figured out how to set a timezone in linux by symlinking ///zoneinfo to /etc/timezone ....lol linux i tell ya |
| 05:23.29 | *** join/#asterisk nix8n82 (~none@24.143.10.144) |
| 05:25.45 | wtf911 | right now i have a sound play via Playback() and then Read() 10 digits....the digits can't be entered until the sound has quit playing...how can i have the sound quit playing when digits began to be entered? like how Authenticate() works where i can internupt the sound by beginning to enter my pin |
| 05:26.57 | ChannelZ | well there's Background |
| 05:28.03 | ectospasm | ...but Background won't accept a string of 10 digits. Have you tried passing the playback file directly to Read? See "core show application Read" for details. |
| 05:28.16 | wtf911 | right now i have this... http://pastebin.com/zYVPkQsq |
| 05:28.32 | wtf911 | i'll try using Read() for sound and taking the digits and get rid of Playback() |
| 05:28.36 | wtf911 | will report back after trying |
| 05:28.50 | kaldemar | wtf911: "core show application Read" <-- look at "filename" |
| 05:29.19 | [TK]D-Fender | ectospasm...but Background won't accept a string of 10 digits. <--- sure it can |
| 05:29.24 | [TK]D-Fender | if you have a match for it |
| 05:30.28 | ectospasm | [TK]D-Fender: and assuming there are no _X!. extensions in that context... |
| 05:31.05 | [TK]D-Fender | no |
| 05:31.28 | [TK]D-Fender | Because if you had a 10-digit pattern wuth X! in there as well it would wait until you reached 10 digits |
| 05:31.40 | [TK]D-Fender | or timed out |
| 05:32.12 | [TK]D-Fender | "!" means it has to be distinct. If you're possibly building up to a bigger match it wont just jump in |
| 05:32.53 | *** join/#asterisk screenn (~screenn@37.46.237.217) |
| 05:33.16 | ectospasm | I thought '!' means match as soon as possible. |
| 05:35.01 | [TK]D-Fender | no, it's match as soon as it is distint. |
| 05:35.18 | [TK]D-Fender | Which means if you have 10 as the longest pattern then on the 11th it will cut in. |
| 05:35.36 | *** join/#asterisk ks3 (ks3@tatooine.digitallotus.com) |
| 05:38.37 | wtf911 | hey guys i would like to report back that getting rid of my lines for Playback() and using Read() for playing the sound too has done what i wanted and i may interupt the sound playing by entering numbers |
| 05:39.26 | *** join/#asterisk nanoha-sama (~nanoha-sa@nanoha-sama.freenode.bouncers.smb.curriegrad2004.ca) |
| 05:39.56 | *** join/#asterisk TriJetScud (~TriJetScu@testarossa.smb.curriegrad2004.ca) |
| 05:48.00 | [TK]D-Fender | checkout time, later all |
| 05:55.06 | ectospasm | wtf911: congrats! |
| 05:56.31 | *** join/#asterisk s[x] (~sx]@ppp59-167-157-96.static.internode.on.net) |
| 05:56.34 | wtf911 | yea figuring Read() has the ability to play sound it makes sense it would work good that to be interupted by digit entering :) ....thank you for all the help! |
| 06:02.44 | wtf911 | good night :) |
| 06:12.57 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 06:21.16 | *** join/#asterisk amessina (~amessina@2001:470:1f11:a4:d6be:d9ff:fe8d:7c1e) |
| 06:25.34 | *** join/#asterisk jkroon (~jkroon@dsl-244-23-64.telkomadsl.co.za) |
| 07:04.11 | *** join/#asterisk qakhan (~qakhan@203.130.22.202) |
| 07:04.43 | qakhan | i getting this error when i run asterisk -r |
| 07:04.57 | qakhan | asterisk: error while loading shared libraries: libtds.so.5: cannot open shared object file: No such file or directory |
| 07:19.31 | *** join/#asterisk qakhan (~qakhan@182.185.150.121) |
| 07:19.50 | qakhan | i am getting this error when i run asterisk -r |
| 07:20.04 | qakhan | asterisk: error while loading shared libraries: libtds.so.5: cannot open shared object file: No such file or directory |
| 07:26.54 | Gugge | if you compiled youself, compile again |
| 07:27.04 | Gugge | if you use a package, you are missing some dependencies |
| 07:33.33 | qakhan | i fixed it by uninstall and reinstall |
| 07:35.58 | greenwolf | figures :) |
| 07:41.19 | qakhan | <PROTECTED> |
| 07:41.33 | qakhan | its working fine |
| 07:41.46 | qakhan | but speech speed is fast. can i slow it down? |
| 07:44.10 | *** join/#asterisk jsjc (~Adium@185.Red-83-59-179.dynamicIP.rima-tde.net) |
| 07:48.07 | *** join/#asterisk qakhan (~qakhan@182.185.150.121) |
| 07:57.08 | *** join/#asterisk Tuju (~tuju@176.75.219.213.sta.estpak.ee) |
| 08:05.59 | *** join/#asterisk qakhan (~qakhan@182.185.150.121) |
| 08:11.43 | qakhan | <PROTECTED> |
| 08:11.44 | qakhan | but speech speed is fast. can i slow it down? |
| 08:29.23 | *** join/#asterisk s[x] (~sx]@60-241-151-10.tpgi.com.au) |
| 08:46.45 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 08:58.56 | *** join/#asterisk felimwhiteley (~quassel@089-101-203026.ntlworld.ie) |
| 09:09.45 | CRCinAU_ | Hmmmm |
| 09:14.13 | CRCinAU_ | so I'm trying to do a fax relay |
| 09:14.21 | CRCinAU_ | from FXS -> asterisk -> T38 |
| 09:14.37 | ectospasm | CRCinAU_: with FAX for Asterisk? |
| 09:15.03 | CRCinAU_ | now the fax dials, answers, but then when the fax is complete |
| 09:15.24 | CRCinAU_ | the channel hangs up and the rest of the dialplan never executes |
| 09:15.30 | ectospasm | what version of Asterisk are you using? |
| 09:15.36 | CRCinAU_ | ectospasm: res_fax & res_fax_spandsp on 1.8.12.0 |
| 09:16.00 | ectospasm | I don't know anything about spandsp |
| 09:16.23 | ectospasm | ...but with FAX for Asterisk fax relay support isn't available. |
| 09:16.50 | ectospasm | ...it's supposed to be in Asterisk 10, but I don't know if or when that will be available. |
| 09:17.23 | CRCinAU_ | by relay, I mean asterisk receives it as an endpoint, then starts up a new one |
| 09:17.34 | ectospasm | store and forward, gotcha |
| 09:17.49 | ectospasm | Are you executing things in the h extension? |
| 09:18.03 | CRCinAU_ | sorry, I realise the term relay is a bit off. |
| 09:18.46 | CRCinAU_ | no, I'm trying to keep it on the same one... isn't it possible to get things on the one entry? or does it have to be on the answer / hangup parts of the dialplan? |
| 09:19.23 | ectospasm | CRCinAU_: well, when one side hangs up, Asterisk automatically jumps to the h extension |
| 09:19.37 | ectospasm | ...if it doesn't exist, then there's no more processing of the call in dialplan. |
| 09:19.50 | ectospasm | ...and the call is hung up. |
| 09:21.54 | CRCinAU_ | hmmmmm |
| 09:22.08 | CRCinAU_ | I was hoping I could set a flag so that it continues, but never mind |
| 09:23.30 | CRCinAU_ | ie continue until you tun out of stuff in the current extension path |
| 09:32.42 | *** join/#asterisk greenwolf (~Penguin@173.234.43.202) |
| 09:34.32 | *** join/#asterisk dddh (~dddh@pdpc/supporter/active/dddh) |
| 09:36.47 | ectospasm | that's not the way dialplan processing works. |
| 09:37.28 | ectospasm | what I recommend is that you receive a fax in one context, and in that same context forward stuff in the hangup extension. |
| 09:48.41 | zamba | WIMPy: you around? |
| 09:49.24 | zamba | trying to dial over lcr now, but asterisk returns the following: [May 19 11:48:28] WARNING[1717]: app_dial.c:2218 dial_exec_full: Unable to create channel of type 'LCR' (cause 0 - Unknown) |
| 09:52.49 | *** join/#asterisk jetlag (~jetlag@pool-71-168-245-146.cmdnnj.east.verizon.net) |
| 09:55.33 | zamba | and for incoming calls i get the following error: 19.05.12 11:53:48.741 EP(1): ACTION remote (not available) applicatio asterisk |
| 10:00.02 | zamba | haha, yes! |
| 10:00.05 | zamba | got it working! |
| 10:01.14 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 10:01.30 | CRCinAU_ | ectospasm: yeah - trying that now |
| 10:01.51 | CRCinAU_ | just realised I have to set a variable to hold the fax number too |
| 10:02.02 | CRCinAU_ | as EXTEN gets set to h in the hangup :) |
| 10:06.52 | *** join/#asterisk [sr] (~kvirc@pal-213-228-140-150.netvisao.pt) |
| 10:06.55 | [sr] | hi |
| 10:10.33 | CRCinAU_ | Hmmm |
| 10:10.35 | CRCinAU_ | thats strange |
| 10:10.49 | CRCinAU_ | it starts dialling, then the sip channel dies. |
| 10:14.25 | ectospasm | could be dying for a number of reasons. |
| 10:24.53 | Marquel | morning. is there a way, to convince the outside network a number is unreachable if a certain condition (caller id) is met? |
| 10:25.07 | Marquel | or can i just hang up the call? |
| 10:26.23 | ectospasm | you can always hang up the call with Hangup() |
| 10:27.37 | ectospasm | Marquel: but the answer is yes, should be quite easy to do with the branching features of dialplan. |
| 10:28.12 | Marquel | ectospasm: well, the condition part is no concern. it's the "convince the network" part the question is about ;) |
| 10:28.39 | ectospasm | depends on the technology, really |
| 10:28.43 | ectospasm | (SIP, DAHDI, etc.) |
| 10:29.17 | Marquel | dahdi |
| 10:29.39 | Marquel | dumb question about hangup(): where are the causecodes documented? |
| 10:30.00 | *** join/#asterisk Chinorro (~chino@62.82.228.35.static.user.ono.com) |
| 10:34.51 | ectospasm | in the code |
| 10:35.00 | ectospasm | ...I forget where. |
| 10:35.28 | ectospasm | Marquel: are these analog or digital DAHDI channels? |
| 10:35.44 | Marquel | ectospasm: digital as in ISDN |
| 10:36.00 | ectospasm | look up Q.931 cause codes, they should be the same. |
| 10:36.39 | ectospasm | http://www.cisco.com/en/US/docs/ios/11_3/debug/command/reference/disdn.html |
| 10:36.44 | zamba | WIMPy: please let me know when you're back :) |
| 10:39.03 | *** join/#asterisk samba35 (~shrikant@unaffiliated/samba35) |
| 10:40.16 | WIMPy | Hi zamba |
| 10:40.51 | zamba | WIMPy: oh! lovely :) |
| 10:41.04 | zamba | WIMPy: i had to hack some, to get the routing.. had to change some socket options in options.conf |
| 10:41.31 | zamba | but.. there seems to be a problem with outgoing audio |
| 10:42.00 | zamba | when i do the echo test, i can hear myself for some seconds and then i disappear completely |
| 10:42.02 | WIMPy | Huh? What did you have to change there? |
| 10:42.08 | *** join/#asterisk s[x] (~sx]@ppp59-167-157-96.static.internode.on.net) |
| 10:42.24 | zamba | socketrights, socketuser and socketgroup |
| 10:42.39 | zamba | i guess it was just socketrights that did the trick |
| 10:42.41 | samba35 | i am planning to install asterisk voip server on ubuntu ,which hardware i should use for voip for 1-2 users |
| 10:43.12 | zamba | samba35: for just 1-2 users, whatever hardware should be fine, i guess |
| 10:43.56 | WIMPy | Hmm. A few seconds ok and then failing somehow sounds like a timing issue to me. |
| 10:44.08 | zamba | and i have problems with feedback |
| 10:44.17 | samba35 | and which voip card do you recommend ? for voice i was told i need some special card |
| 10:44.58 | ectospasm | samba35: shouldn't need a special card to do VoIP, that can be done with a standard NIC, and the right software. |
| 10:45.04 | Marquel | ectospasm: the answer is "yes, one can. just use Hangup(1)." - that actually _really_ convinces the network the number isn't allocated... >:-> |
| 10:45.34 | WIMPy | zamba: That is something I experienced recently when trying to adapt cahn_lcr to the current Asterisk SVN version. But never before. |
| 10:45.47 | samba35 | ok and how do i connect phone to system then ? |
| 10:46.02 | ectospasm | samba35: what kind of phone? |
| 10:46.29 | CRCinAU_ | ok |
| 10:46.31 | CRCinAU_ | I don't get it. |
| 10:46.34 | WIMPy | samba35: The phone syste, will be your PC, or is there more to your plans than you mentioned so far? |
| 10:47.08 | CRCinAU_ | the h,1,blah gets hit after the incoming fax has been stored and the fax machine on the FXS has been stored |
| 10:47.22 | samba35 | a phyiscal phone or software will do ? can you please tell me what other software do i need to buy and service to make phone call ? |
| 10:47.30 | zamba | WIMPy: any suggestions to what i can try? |
| 10:47.34 | CRCinAU_ | it does the Dial(SIP/${FAXNUMBER}@sipout) |
| 10:47.35 | WIMPy | Marquel: It certainly won't convince the network, but it probably will convince the caller. |
| 10:47.51 | CRCinAU_ | then it seems to disappear |
| 10:47.58 | CRCinAU_ | sip show channels shows no active channels |
| 10:48.09 | CRCinAU_ | and the next action doesn't dire |
| 10:48.10 | CRCinAU_ | fire even |
| 10:48.34 | ectospasm | samba35: if you mean POTS (Plain Old Telephone Service) phones, you WILL need special hardware to get that working. |
| 10:48.37 | WIMPy | zamba: It was related to file descriptors. So it may still be related to your permision issue. |
| 10:48.45 | samba35 | WIMPy,i only require voip service provider ? to make phone calls |
| 10:49.04 | CRCinAU_ | ie: http://fpaste.org/737H/ |
| 10:49.08 | WIMPy | samba35: You can do it any way you like. |
| 10:49.09 | CRCinAU_ | what am I missing :\ |
| 10:49.34 | samba35 | ectospasm, ok so if i will use software the i don't require any hardware |
| 10:49.46 | Marquel | WIMPy: actually the cell network i used to call the number in question told me the number is not in use and i should call my information service. |
| 10:50.09 | ectospasm | CRCinAU_: I wouldn't use SendFax from the dialplan, make a call file to do it. |
| 10:50.09 | Marquel | WIMPy: that message was from the cell phone network as it used that provider's information service number... |
| 10:50.25 | samba35 | so i can use make voip calls free of cost any where ? |
| 10:50.56 | WIMPy | Marquel: Yes, but if you were able to get debug information, you could see that the information that the number isn't allocated didn't come from the network, but from the called user. |
| 10:51.09 | CRCinAU_ | ectospasm: why would that be required though? :\ |
| 10:51.20 | CRCinAU_ | it seems a bit of a ghetto way to do things |
| 10:51.42 | WIMPy | samba35: Technically yes. Practically you need to know how to rech them. |
| 10:51.58 | WIMPy | See |
| 10:52.01 | samba35 | ok |
| 10:52.01 | WIMPy | ~itsp |
| 10:52.02 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
| 10:52.03 | Marquel | WIMPy: the person i want to reject from that certain number is certainly uncapable of making the necessary distinction not speaking of getting his hands on that kind of information :p |
| 10:52.15 | samba35 | ok |
| 10:52.20 | ectospasm | CRCinAU_: because the call is stuck in Dial and only gets to SendFax (the way you have it) once the outbound Dial is complete... |
| 10:52.55 | samba35 | thanks you all of you |
| 10:53.11 | CRCinAU_ | ectospasm: but isn't that what the g is for? |
| 10:53.31 | CRCinAU_ | oh wait. |
| 10:53.33 | CRCinAU_ | g - Proceed with dialplan execution at the next priority in the current extension if the destination channel hangs up. |
| 10:53.36 | CRCinAU_ | well. |
| 10:53.38 | CRCinAU_ | that's useless |
| 10:53.41 | CRCinAU_ | hangs up. |
| 10:53.52 | CRCinAU_ | so how the hell are you ever supposed to send faxes?! :\ |
| 10:54.09 | ectospasm | CRCinAU_: do something like this: http://paste.debian.net/170096/ |
| 10:54.13 | Marquel | WIMPy: it's just that i have to hope he never gets the idea to suppress his callerid... |
| 10:54.25 | zamba | WIMPy: where is the socket for lcr written? |
| 10:54.37 | samba35 | ~itsp-india |
| 10:54.45 | WIMPy | Marquel: In that case your provider may offer a blacklisting service. |
| 10:55.02 | CRCinAU_ | ectospasm: I understand that... but it just seems wrong :( |
| 10:55.07 | CRCinAU_ | like a broken workflow |
| 10:55.13 | ectospasm | why is it wrong? |
| 10:55.41 | ectospasm | you'd normally initiate an outbound fax with an external program (script) |
| 10:55.46 | CRCinAU_ | I guess it just seems silly that there is no way to Dial, then call SendFax |
| 10:55.47 | Marquel | WIMPy: i'd have to pay money for that. no. |
| 10:56.06 | WIMPy | There's always a catch |
| 10:56.18 | Marquel | WIMPy: i don't like that guy, but i don't like him enough that i don't even think him worth that kind of money. |
| 10:57.00 | CRCinAU_ | hmmm |
| 10:57.08 | WIMPy | I made a CAPTCHA for unknown callers. |
| 10:57.14 | ectospasm | Marquel: TT (Telemarketer Torture) for his CID? |
| 10:57.36 | ectospasm | ...and use TT for blocked/unknown numbers as well. |
| 10:57.45 | CRCinAU_ | hmmm |
| 10:58.01 | WIMPy | zamba: Interesting question. I never cared so far :-) |
| 10:58.09 | CRCinAU_ | ectospasm: so. If I can only call sendfax via a call file, how can I create it so that I can set FAXOPTS |
| 10:58.17 | Marquel | ectospasm: no. just convince him the number he got is out-of-service and be done with it. |
| 10:58.25 | CRCinAU_ | well, even that means that I'll end up having to use Dial or something similar |
| 10:58.33 | samba35 | ~itsp-us |
| 10:58.37 | ectospasm | CRCinAU_: you can set the FAXOPTS in the call file... |
| 10:58.42 | CRCinAU_ | which mean. well, I don't exactly know. |
| 10:58.44 | CRCinAU_ | oh? |
| 10:58.53 | Marquel | WIMPy: how did you do that? |
| 10:58.58 | samba35 | ~itsplist-us |
| 10:58.58 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com |
| 10:59.05 | samba35 | ~itsplist-india |
| 10:59.14 | CRCinAU_ | ohhh |
| 10:59.15 | zamba | well.. kind of disappointing.. |
| 10:59.17 | CRCinAU_ | WAIT a minute.... |
| 10:59.26 | Marquel | WIMPy: just asking out of curiosity, we do not have that much problems with anonymous telemarketers here. |
| 10:59.29 | CRCinAU_ | http://www.voip-info.org/wiki/view/Asterisk+cmd+Originate |
| 10:59.38 | WIMPy | Marquel: Generate two random digits and ask the user to enter them. |
| 10:59.41 | zamba | WIMPy: could there be some missing kernel modules? |
| 10:59.47 | zamba | WIMPy: i had to load mISDN_dsp earlier for instance |
| 10:59.53 | CRCinAU_ | ectospasm: if I'm reading this right, wouldn't that do what I want? ie a combination of Dial+SendFax? |
| 11:00.06 | WIMPy | zamba: Yes, but without that you don't get audio at all. |
| 11:00.07 | ectospasm | Originate is using the AMI, so that would work |
| 11:00.12 | zamba | WIMPy: ah :) |
| 11:00.41 | CRCinAU_ | It seems that is a rather new command |
| 11:00.43 | Marquel | WIMPy: ah, okay. (unfortunately my car is pretty good at recognizing spoken digits even at high levels of background noise, so i guess that won't work for a long time anymore ;) ) |
| 11:00.45 | WIMPy | zamba: But first ok and then failing after a few seconds is new to me. |
| 11:00.50 | CRCinAU_ | hmmm - time to try some re-writing. |
| 11:00.54 | CRCinAU_ | brb |
| 11:01.05 | zamba | WIMPy: it seems like the audio loops somehow |
| 11:01.14 | zamba | WIMPy: and that's also why i get the feedback |
| 11:01.18 | ectospasm | CRCinAU_: but I think it'd be easier to use a call file. Just call a script to generate the call file, then move the call file to the outgoing spool dir |
| 11:01.39 | WIMPy | Marquel: Maybe, but at least they have to willingly circumvent that. Which might be a criminal offence in many places. |
| 11:02.03 | zamba | WIMPy: if i dial outbound from a sip ext over the isdn trunk.. it dials just fine.. but when i pick up the phone in the other end and speak i hear myself in the sip end |
| 11:02.15 | zamba | and nothing from the "far end" |
| 11:02.15 | WIMPy | zamba: Maybe it's something completely different. Can you describe that feedback? |
| 11:02.15 | ectospasm | sounds like echo |
| 11:02.23 | zamba | yeah, echo |
| 11:02.27 | zamba | basically |
| 11:02.33 | zamba | which turns into feedback |
| 11:02.39 | Marquel | WIMPy: agreed :) |
| 11:02.44 | WIMPy | wonders how that could work. |
| 11:03.02 | ectospasm | sounds like acoustic echo. |
| 11:03.12 | CRCinAU_ | ectospasm: does this sound right to you: |
| 11:03.12 | CRCinAU_ | exten => _X.,n,Originate(SIP/${FAXNUMBER}@spin,app,SendFax,"${FAXFILE},dfz") |
| 11:04.01 | ectospasm | yeah, that could work |
| 11:04.03 | WIMPy | zamba: Have tou tried somethign as calling to an MusicOnHold() extension externally? |
| 11:04.07 | CRCinAU_ | It seems like I have to pass on arg 3 in quotes - as if I set it without, arg4 is ignored for an app |
| 11:04.09 | ectospasm | I had to look up the syntax of Originate. |
| 11:04.35 | zamba | WIMPy: if i call an echo test externally i have the problem described earlier.. that i can hear myself for some seconds, and then it dies |
| 11:04.39 | CRCinAU_ | it doesn't mention anything about quoting for multiple args to the app though :( |
| 11:04.42 | ectospasm | CRCinAU_: you should only need the 'z' option to SendFAX if your T.38 provider is broken |
| 11:04.48 | zamba | you can try yourself if you want :) |
| 11:05.03 | WIMPy | zamba: Does it just go silent or will it hang up? |
| 11:05.10 | zamba | it just goes silent |
| 11:05.12 | CRCinAU_ | ectospasm: I haven't even tried T.38 as yet - so I'll force it first to see if it even works, then try it without :) |
| 11:05.15 | zamba | and then reappears at random |
| 11:05.30 | WIMPy | Uh? |
| 11:05.35 | CRCinAU_ | as I haven't been able to dial, I haven't got as far as testing yet |
| 11:05.44 | WIMPy | That's getting even stranger. |
| 11:05.52 | zamba | WIMPy: you can try yourself if you want to :) |
| 11:05.54 | CRCinAU_ | ok - time to try sending one :) |
| 11:06.15 | WIMPy | ja, ok |
| 11:06.48 | zamba | just have to prime the provider first :) |
| 11:07.01 | zamba | seems like it dials two times on each NT box for any given number |
| 11:07.08 | zamba | so it load balances like that |
| 11:07.26 | zamba | ok, it's ready now |
| 11:13.27 | CRCinAU_ | ectospasm: I just realised. I'll still have to do it in the h,1,blah section won't I. |
| 11:13.42 | CRCinAU_ | otherwise when the Hangup is processed, it won't process anything further... |
| 11:14.01 | ectospasm | right. |
| 11:14.08 | CRCinAU_ | facepalms |
| 11:14.10 | ectospasm | ...didn't you learn anything from before? |
| 11:14.35 | CRCinAU_ | yeah - but I had a stupid. ;) |
| 11:14.48 | ectospasm | hehehe |
| 11:15.52 | CRCinAU_ | I got too carried away with what Originate meant ;) |
| 11:17.23 | ectospasm | yeah, don't you need to receive the fax first? |
| 11:17.42 | CRCinAU_ | yeah |
| 11:17.46 | CRCinAU_ | thats no problem |
| 11:18.21 | CRCinAU_ | I have that part down without an issue |
| 11:19.10 | CRCinAU_ | but it bails after the calls bail and tahts all... hence just figuring out the sending part. |
| 11:20.53 | CRCinAU_ | oh great |
| 11:21.05 | CRCinAU_ | now the FXS isn't picking up all the digits dialed :( |
| 11:22.37 | ectospasm | www? |
| 11:32.45 | *** join/#asterisk s[x] (~sx]@ppp59-167-157-96.static.internode.on.net) |
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| 12:08.16 | qakhan | when i use MRCPSynth i get this message |
| 12:08.25 | qakhan | WARNING[3090]: app_unimrcp.c:1008 audio_queue_write: (TTS-6) audio queue overflow! |
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| 12:40.39 | CRCinAU_ | hmmm |
| 12:40.41 | CRCinAU_ | ok |
| 12:40.52 | CRCinAU_ | Originate doesn't support changing the context etc |
| 12:41.13 | CRCinAU_ | it'll try to dial out of the FXS port. which obviously ain't going to wokr |
| 12:41.34 | cusco | suports specifying a context and extension, therefore you can make dialplan to change context |
| 12:41.38 | CRCinAU_ | ie comes in on DAHDI/3, tries to go out again on DAHDI/3 |
| 12:42.09 | CRCinAU_ | well, I tries using SIP/${EXTEN}@sipout |
| 12:42.21 | CRCinAU_ | <PROTECTED> |
| 12:42.29 | CRCinAU_ | where sipout = a SIP peer |
| 12:43.56 | cusco | well that is how I do it too |
| 12:44.01 | cusco | and it dials trough that peer |
| 12:44.50 | cusco | paste the asterisk call ? |
| 12:45.56 | *** join/#asterisk angryuser_laptop (~angryuser@ps444-1-78-224-152-133.fbx.proxad.net) |
| 12:46.51 | CRCinAU_ | this is my dialplan part: http://fpaste.org/qzE2/ |
| 12:47.24 | CRCinAU_ | let me try resending something |
| 12:51.39 | CRCinAU_ | *sigh* |
| 12:51.43 | CRCinAU_ | now its fucking with me: http://fpaste.org/e6wW/ |
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| 12:58.36 | CRCinAU_ | I wonder why ReceiveFax isn't even working now :( |
| 13:01.04 | *** join/#asterisk joelsolanki (~joelsolan@124.125.148.2) |
| 13:01.20 | joelsolanki | Hi All |
| 13:02.57 | joelsolanki | Right now we are using VoIPSwitch software where eyebeam softphone gets registered. All call flow is working but we are having issue with hanged calls(stucked calls) even though callers using eyebeam disconnect the calls, on voipswitch it still shows calls connected and we are charged for this calls too. |
| 13:04.00 | joelsolanki | we are thinking if we can put customers on asterisk. is there any issue like this in asterisk. please note that this are all short duration calls like 1-4 mins calls only max. |
| 13:05.00 | joelsolanki | please suggest |
| 13:15.11 | CRCinAU_ | ok. can anyone spot the problem here: http://fpaste.org/jtE9/ |
| 13:15.20 | CRCinAU_ | for some reason, ReceiveFax isn't creating any output |
| 13:15.59 | CRCinAU_ | the fax machine is saying that it failed. |
| 13:16.20 | CRCinAU_ | ie its failing with COMM ERROR |
| 13:16.33 | CRCinAU_ | and I can't see any reason why :( |
| 13:22.38 | *** join/#asterisk _Corey_ (~chatzilla@64.215.11.114) |
| 13:31.17 | *** join/#asterisk ean (ean@unaffiliated/ean) |
| 13:31.42 | ean | do you guys have any SIP trunk service provider to recommend? |
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| 13:51.41 | CRCinAU_ | Hmmm |
| 13:51.56 | CRCinAU_ | does does the Use Count on res_fax_spandsp increment with each failed fax. |
| 14:11.39 | *** part/#asterisk CRCinAU_ (~CRCinAU@another.bloody.irc.session.from.crc.id.au) |
| 14:14.58 | *** join/#asterisk scgm11 (~Sebastian@r186-52-18-139.dialup.adsl.anteldata.net.uy) |
| 14:16.01 | scgm11 | I have a question regarding compiling asterisk, why if I download the tar.gz sources I can compile ok, but if I download from svn when I want to do make menuselect I get |
| 14:16.02 | scgm11 | make menuselect |
| 14:16.03 | scgm11 | CC="cc" CXX="" LD="" AR="" RANLIB="" CFLAGS="" make -C menuselect CONFIGURE_SILENT="--silent" makeopts |
| 14:16.04 | scgm11 | make: *** menuselect: No such file or directory. Stop. |
| 14:16.05 | scgm11 | make: *** [menuselect/makeopts] Error 2 |
| 14:17.24 | *** join/#asterisk qakhan (~qakhan@182.185.150.121) |
| 14:17.52 | qakhan | i forgot the command to install the pkg-config |
| 14:18.04 | qakhan | on CentOS |
| 14:21.56 | qakhan | please help |
| 14:32.39 | *** part/#asterisk scgm11 (~Sebastian@r186-52-18-139.dialup.adsl.anteldata.net.uy) |
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| 15:53.01 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 16:22.09 | *** join/#asterisk MarkS- (~mark@unaffiliated/mark21) |
| 16:24.18 | MarkS- | Hello, I'm looking for create something that is described on https://support.zabbix.com/browse/ZBXNEXT-667?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel#issue-tabs and http://www.zabbix.com/wiki/howto/config/alerts/notificationsusingasterisk (in other words, if there is a problem call a number using call files with Asterisk). Is there other documentation I should check or is this somewhere available? When it isn't yet available I wou |
| 16:25.28 | [TK]D-Fender | it cut off |
| 16:25.56 | [TK]D-Fender | As for causing * to dial out, there are Call Files, AMI & CLI Originate. |
| 16:26.02 | [TK]D-Fender | 3 options right there |
| 16:26.11 | [TK]D-Fender | Pick whichever you feel like |
| 16:26.24 | MarkS- | [TK]D-Fender: ok, are there things I can check when using call files? |
| 16:26.38 | [TK]D-Fender | like? |
| 16:27.42 | MarkS- | how to get the status back (if the call is accepted or not) or is this something I can/should do in a dialplan? |
| 16:27.58 | [TK]D-Fender | "things"Use dialplan for this |
| 16:35.41 | *** join/#asterisk ChannelZ (channelz@burner.com) |
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| 17:03.55 | *** join/#asterisk dlu (~dlu@ip-62-143-162-249.unitymediagroup.de) |
| 17:04.04 | dlu | hi all |
| 17:05.00 | dlu | tzanger: ar eu alive? |
| 17:05.23 | tzanger | did you mean me, or tzafrir_laptop? |
| 17:05.38 | dlu | no, you :) |
| 17:06.00 | dlu | hop all is fine |
| 17:11.04 | dlu | i have a strange behaviour on a asterisk 10.4 box with libss7 and spandsp_fax, fax show stats didnt remember any call, this on 3 boxes with different distries and kernels. any clue what i am miss? |
| 17:11.45 | dlu | show calls work fine |
| 17:11.47 | *** join/#asterisk oej (~olle@h87-96-134-129.dynamic.se.alltele.net) |
| 17:12.05 | dlu | hi olle! :) |
| 17:16.00 | oej | hi! |
| 17:16.37 | *** join/#asterisk sawgood (~sawgood@173-13-158-29-sfba.hfc.comcastbusiness.net) |
| 17:16.43 | dlu | long time not reading |
| 17:17.06 | dlu | hope all is fine |
| 17:44.12 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 17:50.27 | *** join/#asterisk DelphiWorld (~VoCloud@openvpn/user/DelphiWorld) |
| 17:50.30 | DelphiWorld | hello |
| 17:50.38 | DelphiWorld | this problem maybe be common to everyone |
| 17:50.46 | DelphiWorld | i have a FXO gateway |
| 17:50.51 | DelphiWorld | vobtech |
| 17:51.04 | DelphiWorld | if i call through sip to landline, it's sending 183 then shortly 200 |
| 17:51.10 | bdfoster | DelphiWorld, ? |
| 17:51.19 | DelphiWorld | i don't want it to send 200 if the destination don't answer me, any solution ? |
| 17:51.25 | DelphiWorld | slaps bdfoster around a bit with a large trout |
| 17:51.35 | bdfoster | you haz asterisk? |
| 17:52.25 | ChannelZ | Analog doesn't really have proper call progress. |
| 17:52.48 | ChannelZ | That's just the way it is. |
| 17:53.02 | bdfoster | it depends on your local carrier |
| 17:53.06 | [TK]D-Fender | MarkS-: it can |
| 17:53.11 | [TK]D-Fender | DelphiWorld: rather |
| 17:53.16 | DelphiWorld | [d? |
| 17:53.21 | DelphiWorld | [d? |
| 17:53.23 | DelphiWorld | [TK]D-Fender: ? |
| 17:53.29 | [TK]D-Fender | And no, call progress on analog is tones... |
| 17:53.36 | [TK]D-Fender | "callprogress=yes" |
| 17:53.48 | [TK]D-Fender | Actually.. i |
| 17:53.49 | ChannelZ | YMMV |
| 17:53.50 | bdfoster | DelphiWorld, spandsp takes care of figuring out your tones |
| 17:53.57 | DelphiWorld | [TK]D-Fender: i has it... same issue is the gw sending 200 |
| 17:54.07 | DelphiWorld | bdfoster: i tested both fs/ast |
| 17:54.13 | [TK]D-Fender | if your gateway was actually a SIP device.... (you didn'ts say), then that device sets it's own rules |
| 17:54.24 | [TK]D-Fender | and sending a 200 id its problem, not *'s |
| 17:54.26 | DelphiWorld | the gw is a sip to analog |
| 17:54.31 | DelphiWorld | is not asterisk |
| 17:54.36 | [TK]D-Fender | Then it does what it does |
| 17:54.46 | [TK]D-Fender | Read its manuals to see if you can tell it otherwise. |
| 17:54.50 | [TK]D-Fender | This isn't *'s fault |
| 17:55.00 | DelphiWorld | i know just thinking if someone have same issue |
| 17:55.15 | DelphiWorld | [TK]D-Fender: did i say is asterisk fault ? |
| 17:55.18 | bdfoster | [TK]D-Fender, ill take care of it. DelphiWorld go to #freeswitch |
| 17:57.37 | dlu | i hav a short question about dahdi lastest. i need to compile it on a kernel 2.6.11 box but get a dahdi-base.c:52:25: linux/mutex.h: No such file or directory. it isnt pssible to run it on this box or there are any Makefile hacks to compile it? |
| 17:58.17 | dlu | thnx |
| 18:01.40 | ChannelZ | do you actually have the kernel dev or source package installed in your distro? |
| 18:01.50 | dlu | yes i do |
| 18:01.57 | dlu | old dahdi can be compiled |
| 18:02.32 | dlu | but its from 2009-11 :) the box have 1156 days uptime :| |
| 18:03.02 | dlu | but works fine :] |
| 18:03.20 | ChannelZ | well I don't think mutex.h is "new" by any stretch |
| 18:03.21 | [TK]D-Fender | DelphiWorld: No, but I've just clarified that it isn't |
| 18:03.25 | ChannelZ | sounds like you're missing something. |
| 18:03.52 | DelphiWorld | [TK]D-Fender: :) |
| 18:04.57 | dlu | hmmm |
| 18:05.42 | dlu | asterisk 10.4+libss7-1.0.2 have compiled fine |
| 18:05.49 | ChannelZ | do a 'locate mutex.h' and see if it's somewhere. Maybe you think you have headers installed but they are not for the actual kernel version you are running |
| 18:06.31 | dlu | -bash: locate: command not found :) |
| 18:06.38 | dlu | its a 9.3 suse |
| 18:06.52 | ChannelZ | "running fine" eh? hmm |
| 18:07.51 | *** join/#asterisk brdude (~brdude@c-24-7-76-160.hsd1.ca.comcast.net) |
| 18:07.52 | dlu | find / -name "mutex.h" progress |
| 18:08.13 | dlu | hmm find a lot of mutex.h :( |
| 18:08.45 | dlu | i guess this is the right one usr/include/ptlib/unix/ptlib/mutex.h |
| 18:09.03 | dlu | or usr/include/ptlib/mutex.h |
| 18:09.24 | dlu | but why it will be not found? |
| 18:09.27 | ChannelZ | doubtful, ptlib is something else |
| 18:09.51 | dlu | i have tis one too /usr/src/linux-2.6.11.4-21.9/fs/xfs/linux-2.6/mutex.h |
| 18:10.07 | dlu | thats all |
| 18:10.23 | dlu | i guess its not the right one |
| 18:10.25 | ChannelZ | it should be just /usr/src/linux-2.6.11*/include/linux/mutex.h |
| 18:11.43 | *** join/#asterisk screenn (~screenn@37.46.237.217) |
| 18:15.02 | *** join/#asterisk gusto (~gusto@80.187.231.212) |
| 18:15.13 | dlu | isnt there |
| 18:15.17 | dlu | sooo strange |
| 18:15.46 | [sr] | ChannelZ: can I be ChannelX ? :p |
| 18:16.42 | dlu | :| |
| 18:17.49 | ChannelZ | You can be anything you want to be |
| 18:18.11 | [sr] | just trying to be funny :p |
| 18:18.13 | ChannelZ | though CharlieX would be funnier. |
| 18:18.53 | ChannelZ | (Channel Z = B-52's song; Charlie X = Star Trek episode) |
| 18:19.28 | [sr] | oh i see, wasnt aware of it |
| 18:24.42 | *** join/#asterisk wtf911 (~wtf911@cpe-67-251-103-137.stny.res.rr.com) |
| 18:25.35 | ChannelZ | hmm. "'struct mutex' was introduced in 2.6.16" |
| 18:27.24 | ChannelZ | looks like there's supposed to be alternate support for older kernels but perhaps they've deprecated that for some other reason. |
| 18:27.29 | ChannelZ | Progress is a bitch I guess. |
| 18:32.08 | *** join/#asterisk micols (~t@rlogin.dk) |
| 18:32.10 | dlu | hehe |
| 18:32.37 | dlu | ill try to get all work with the old dahdi |
| 18:41.59 | gusto | hi |
| 18:42.11 | gusto | what's up [sr] ? |
| 18:42.53 | [sr] | hi gusto, not much |
| 18:44.09 | gusto | [sr]: why? |
| 18:44.53 | [sr] | no news... don't know what are you refering exactly... but life its the same it was one month ago :p |
| 18:45.58 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 18:46.20 | gusto | [sr]: you know, i wish life would be the same as it was 4 years ago |
| 18:47.01 | gusto | [sr]: but on the other hand 4 years ago i had no idea what asterisk is and also did not know that i would ever plan a VoIP rollout |
| 18:47.30 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 18:48.28 | DelphiWorld | [sr]: [Sip-Router] ;) |
| 18:49.42 | gusto | well, maybe he meant that |
| 18:51.16 | *** join/#asterisk amessina (~amessina@2001:470:1f11:a4:d6be:d9ff:fe8d:7c1e) |
| 18:51.27 | gusto | hey ppl some of you have ipv6 |
| 18:51.37 | gusto | amessina: is it a tunnel or native? |
| 18:52.11 | amessina | gutso: tunnel -- covad/megapath won't dole out IPv6 to end users yet |
| 18:52.29 | dlu | ChannelZ: got it working |
| 18:52.47 | gusto | i hate this ISP's |
| 18:52.51 | gusto | they are soo dumb |
| 18:53.31 | gusto | i recently heard something about a workshop they did recently and vodafone was showing its project of rolling out IPv6 to LTE and UMTS |
| 18:53.36 | gusto | someone knows about that? |
| 18:54.02 | gusto | and also t-com was there with their DSL test, however, i was not included in that test even though i contacted them that i want it |
| 18:54.48 | dlu | all fine with * 10.4 libss7-1.0.2 ans T.38 gateway in and outbound dahdi version is svn from 2009-03-19 |
| 18:55.23 | dlu | libpri is 1.4.11.2 |
| 18:56.58 | dlu | ok go to tv now l8ter al |
| 18:57.19 | *** part/#asterisk dlu (~dlu@ip-62-143-162-249.unitymediagroup.de) |
| 19:01.34 | [sr] | DelphiWorld: lol no |
| 19:01.44 | [sr] | gusto: i whish i could go back 20years |
| 19:03.00 | gusto | [sr]: why? |
| 19:06.04 | [sr] | clean up tha mess! |
| 19:08.08 | DelphiWorld | [sr]: just a fun joke;) |
| 19:08.35 | DelphiWorld | gusto: you're ipV6ified? |
| 19:09.12 | DelphiWorld | [sr]: pt... what's going on there? :P |
| 19:14.37 | *** join/#asterisk F|shie (~chatzilla@182.177.112.62) |
| 19:17.44 | *** part/#asterisk DelphiWorld (~VoCloud@openvpn/user/DelphiWorld) |
| 19:22.22 | [sr] | del |
| 19:22.24 | [sr] | ops |
| 19:22.27 | [sr] | hes gone |
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| 19:55.26 | *** join/#asterisk eZz (~ez@195.114.6.134) |
| 19:55.33 | eZz | hi |
| 19:55.53 | F|shie | hello all, i am trying to run wait for digit command in an eagi (modified the eagi-test from samples), but cant seem to get it working. Here is the eagi http://pastebin.ca/2150887 and my CLI output http://pastebin.ca/2150889.The warning output is from channel.c, http://www.pastebin.ca/2150893 (line 25). Could anyone explain why this is happening? |
| 19:56.39 | eZz | is it possible to add a header (via SIPAddHeader) BEFORE Progress() ? I'd like to see a custom header in 183... As I see, there is no effect on setting SIPAddHeader. |
| 19:57.22 | [TK]D-Fender | There isn't. That only applies to Dial() |
| 19:57.40 | *** join/#asterisk screenn (~screenn@37.46.237.217) |
| 19:58.29 | eZz | :( |
| 20:02.52 | *** join/#asterisk F|shie (~chatzilla@182.177.86.121) |
| 20:03.38 | F|shie | [TK]D-Fender: hey TK, could you offer an opinion on the issue i mentioned |
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| 20:09.50 | *** part/#asterisk MarkS- (~mark@unaffiliated/mark21) |
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| 20:24.00 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 20:43.20 | gusto | [sr]: what kind of mess do you want to clean up? |
| 20:52.42 | *** join/#asterisk DarthExpeditor (~IceChat9@96-42-133-130.static.trcy.mi.charter.com) |
| 21:05.01 | *** join/#asterisk francisvgarcia (~linux@186.1.77.241) |
| 21:05.37 | francisvgarcia | hi folks |
| 21:06.08 | francisvgarcia | can you please give a little help configuring calling using SIP URI? |
| 21:06.48 | francisvgarcia | I made a copy paste to an example from the Book but It does not work for me |
| 21:11.38 | [TK]D-Fender | Show us and maybe we can.... |
| 21:13.47 | francisvgarcia | Here is the config http://pastebin.com/fLWdvy6r |
| 21:14.47 | [sr] | gusto: things |
| 21:15.20 | francisvgarcia | whenever I call extension@domain I only get the SIP/EXTENSION dialed instead of SIP/EXTENSION@DOMAIN |
| 21:17.06 | gusto | [sr]: what kind of? you do not need to hide anything |
| 21:17.13 | [sr] | lol |
| 21:17.28 | [sr] | gusto: r u a sentimental adviser? :p |
| 21:19.59 | gusto | [sr]: well |
| 21:20.37 | *** part/#asterisk Bullmoose (~Bullmoose@71-37-168-220.bois.qwest.net) |
| 21:24.43 | francisvgarcia | I solved this by adding this (SIP/${EXTEN}@${SIPDOMAIN},60,tT) |
| 21:25.57 | francisvgarcia | but I will need to make a macro for the extensions overlapping |
| 21:26.01 | francisvgarcia | my extensions |
| 21:26.02 | *** join/#asterisk acedia (~garland@unaffiliated/ffs) |
| 21:27.10 | [TK]D-Fender | Sounds like you already have your answer.... |
| 21:27.32 | ChannelZ | still isn't even clear on the question |
| 21:50.48 | francisvgarcia | but now I got an issue calling domain names directly |
| 21:51.40 | francisvgarcia | if by example I call directly to an IP address then It adds the server domain s name as domain |
| 21:52.12 | francisvgarcia | I mean if I call 192.168.1.1 it then appear as 192.168.1.1@serverdomain |
| 21:53.20 | [TK]D-Fender | So you'll have to parse that out too |
| 21:53.33 | [TK]D-Fender | * wasn't made as a SIP proxy. You've got the dialplan to deal with. |
| 21:55.40 | francisvgarcia | I will try to do it |
| 21:55.54 | francisvgarcia | If I have any difficult I will let you know |
| 22:01.07 | ChannelZ | Dial(SIP/fart.com/123) |
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| 23:30.30 | Igneous | when doing 'pri show spans', what exactly does the "Provisioned" in "Provisioned,Up,Active" mean? If I have something that's "Down,Active", is that different somehow from "Provisioned,Down,Active"? |
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