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| 00:22.20 | carrar | MSFT is still in biz? |
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| 04:59.42 | Assid | hey |
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| 05:26.03 | fr0ggie | What's a good place to quickly learn some dialplan magic? |
| 05:26.26 | p3nguin | ~book |
| 05:26.26 | 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://www.asteriskdocs.org/ or see ~buybook |
| 05:27.19 | p3nguin | What are you trying to accomplish? |
| 05:28.21 | slingshot01 | I love 3rd edition, but calculating armor class got really strange compared to second.. |
| 05:28.49 | fr0ggie | p3nguin: just relearn stuff i forgot from not using * in about half a decade and stuff that's probably changed |
| 05:29.03 | p3nguin | Logic remains the same. |
| 05:29.47 | p3nguin | The only real way is going to be to decide on a project and start writing dial plan to do interesting things. |
| 05:30.09 | kaldemar | asterisk cookbook has a bunch of dialplan stuff too. |
| 05:30.42 | p3nguin | Random projects, I presume. Those could be useful. |
| 05:30.57 | fr0ggie | p3nguin: pretty much i've got a couple friends and such and we use asterisk to talk securely |
| 05:31.49 | slingshot01 | it's crazy what you can do with some of the more creative * projects |
| 05:31.55 | fr0ggie | found some annoyances so far in android's SIP UA |
| 05:33.20 | p3nguin | Every once in a while I think of something I would like to be able to do by dialing some numbers from a phone. |
| 05:33.26 | fr0ggie | right now i'm just kinda playing around with using * to make our google voice accounts more friendly |
| 05:34.10 | slingshot01 | I had fun setting up a clinic and voicemails and queues and such |
| 05:34.33 | fr0ggie | is there a way to prevent Dial()ing if theres already a call going outbound from that channel? |
| 05:35.08 | p3nguin | Yes. You'll want to look at GROUP() and GROUP_COUNT(). |
| 05:35.35 | p3nguin | Well... for one channel, there is actually probably something easier. |
| 05:36.37 | p3nguin | Using group count might be more well-suited to more channels, possibly from multiple devices. |
| 05:36.54 | p3nguin | But it could still work fine with one channel and one device. |
| 05:37.38 | p3nguin | For just testing if a phone is in a call already, you could use ChanIsAvail(). |
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| 05:38.48 | fr0ggie | gv seems to drop your first call if you try to make a second outbound call |
| 05:39.40 | p3nguin | I would probably still use group count for that, but that is because I am familiar with using it for limitations such as that. |
| 05:40.21 | fr0ggie | i'll look into it- thanks |
| 05:40.26 | p3nguin | I specifically use it to limit the amount of concurrent inbound and outbound calls. |
| 05:41.30 | p3nguin | Set the GROUP(), then check the GROUP_COUNT(). If the count is equal to or greater than N, congest. |
| 05:42.31 | p3nguin | I use ChanIsAvail() for things like checking if a phone is in use before trying to page it (which does auto-answer). |
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| 05:44.08 | p3nguin | I'm sure there are other ways to check the same criteria, but I wouldn't know how right this moment. |
| 05:44.47 | fr0ggie | cool |
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| 05:51.13 | fr0ggie | whats this same => ? |
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| 06:00.18 | din3sh | mrning ppl |
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| 06:13.36 | fr0ggie | if i have exten => *@gmail.com,1,Playback(hang-on) then later in that context like exten => someuser@gmail.com,1,Playback(locating-user) |
| 06:13.50 | fr0ggie | or such-- would that work how i hope it will? |
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| 08:14.16 | Dovid | hi all |
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| 08:16.37 | Dovid | i have an interesting issue where a carrier of mine is doing NAT detcection so they do not send RTP however the caller is doing the same so there is no rtp and there for no one knows what port to use and then there is no audio. Is there any setting in sip.conf that will send audio (or some sort of rtp) to the requested port (from my carrier) so it see's rtp and then sends back to me? |
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| 09:17.44 | wdoekes2 | Dovid: nat "detection" or nat "assuming" |
| 09:18.27 | wdoekes2 | ? and do you have res_timing_* loaded? |
| 09:18.35 | jww_ | Hello everyone. |
| 09:23.06 | Dovid | wdoekes2: I am using OpenSipS before asterisk and I have direction: active |
| 09:23.23 | Dovid | Customer -> OpenSips -> Asterisk -> carrier |
| 09:25.01 | Dovid | the carrier seems to be doing NAT detection so they send a 183 but still wait for RTP from us and i have nothing to send cause the caller did not send any rtp so i am at a "rtp stalemate". the carrier sends comfort noise so i should send back rtp. but i have nothing to send. i wonder if i can send comfort noise or something else back so they will start sending to me RTP |
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| 09:32.38 | zkn | Hello, I've encountered the following issue when logging call logs to MySQL backend. the "src" column skips spaces when for example I want to write CALLERID(name) there, so instead of Firstname Lastname the data written will be FirstnameLastname, quite annoying... |
| 09:32.50 | zkn | and I have shrinkcallerid=no in sip.conf |
| 09:33.23 | zkn | and column type in MySQL table is varchar |
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| 09:34.40 | zkn | so the question is if Asterisk is skipping spaces when writing to 'src' column ? is this something in Asterisk code since "usually" the data that goes to src column is phone numbers? |
| 09:35.08 | zkn | how could I find this out? |
| 09:42.03 | kaldemar | zkn: what exactly are you using? |
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| 09:45.20 | zkn | kaldemar, Asterisk 1.8.10.1 -> unixODBC 2.2.14 -> MySQL 5.1.61 |
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| 09:46.47 | zkn | and i've noticed that in other columns this anomaly does not happen, only in src column are spaces left out |
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| 09:48.35 | kaldemar | how are you trying to write to the src field? it's supposed to be read-only from dialplan. |
| 09:49.58 | zkn | i mean, when I check MySQL tables, what the CDR backend has logged to my asterisk MySQL database |
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| 09:52.38 | kaldemar | what backend? |
| 09:53.15 | kaldemar | and how are you trying to put something with spaces into src? |
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| 09:53.21 | zkn | for example when I set CALLERID(all)="NAME" <number> before dialing out, then to the src column asterisk usually writes <number>, now when I set CALLERID(all)="number" <Firstname Lastname> then I find recorded in MySQL src column FirstnameLastname, without the space between |
| 09:54.00 | zkn | kaldemar, cdr_adaptive_odbc backend |
| 09:54.01 | kaldemar | stop raping the caller id function. |
| 09:54.09 | zkn | lol |
| 09:54.09 | kaldemar | what do you really want to do? |
| 09:54.41 | zkn | I want to record CALLERID(name) in src column for outbound calls |
| 09:54.50 | kaldemar | why in the src column? |
| 09:55.27 | zkn | because that is the source of the call, the caller |
| 09:55.43 | kaldemar | and it has a specified purpose which you're trying to brake. |
| 09:55.56 | danfromuk | Hi, has anyone seen this before and managed to resolve it? |
| 09:55.58 | danfromuk | PHP Notice: Undefined index: PATH in /var/lib/asterisk/scripts/phpagi.php on line 1693 |
| 09:56.14 | kaldemar | there are other fields too. accountcode, userfield or something that you can define by yourself. |
| 09:57.01 | kaldemar | core show function CDR |
| 09:57.37 | zkn | thought of it too, just could figure out immediately why wouldn't src field work for that |
| 09:57.48 | zkn | could not figure out* |
| 09:58.05 | zkn | but i guess i got my answer |
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| 11:01.37 | rw | hi folks |
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| 11:02.57 | rw | i know mISDN isn't totally on topic here, but: if i compile "isdn4k-utils", shouldn't this give me a new "libcapi20.so", why do i only get a libcapi20dyn.a? |
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| 11:10.30 | Chainsaw | rw: That suggests you have a "static-libs" or equivalent enabled. |
| 11:10.56 | rw | Chainsaw: not intentionally |
| 11:13.16 | rw | i did get isdn4k-utils from the misdn.eu git-repo and just run "make" in "isdn4k-utils" |
| 11:13.35 | Chainsaw | rw: It's generally better to do such things through portage. |
| 11:14.14 | rw | well, it's a debian system |
| 11:15.08 | Chainsaw | rw: Quite. Do they not have it packaged? |
| 11:15.25 | rw | is mISDN v2 supposed to work with libcapi20 from the debian repos or do i have to recompile it, to enable the usage of the misdncapid? |
| 11:15.36 | Chainsaw | rw: If you check ./configure --help, that might have a "static" and a "shared" option. You want the shared library. |
| 11:15.54 | Chainsaw | rw: Debian is not my area of expertise, sorry. |
| 11:16.07 | rw | Chainsaw: i did just recompile with ./configure --enable-static=no --enable-shared=yes; didn't change anything |
| 11:16.22 | rw | s/recompile/reconfigure+compile/ |
| 11:16.46 | Chainsaw | rw: And it's not just that the .a stayed behind from your last effort and the .so has freshly appeared? |
| 11:17.34 | rw | Chainsaw: well, there you go - thanks! |
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| 12:18.27 | invinicible | hello |
| 12:19.17 | *** join/#asterisk F|shie (~chatzilla@39.47.84.180) |
| 12:19.38 | F|shie | hello all, im trying to follow this tutorial, http://ofps.oreilly.com/titles/9781449303822/c03-AudioManipulation_id302347.html#TriggeringAudioViaDTMF , but instead of playing audio into channel i want to record the audio, on the audio extension, is it possible to do so? |
| 12:22.26 | [TK]D-Fender | F|shie, features.conf already has a recording feature under "monitor" and it's a Dial() option. |
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| 12:34.10 | F|shie | yes i am aware of that, but i want to record it under the audio extension, i need to check if i should get audio from chanspy to record |
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| 12:44.06 | [TK]D-Fender | F|shie, If that's the way you insist on doing it, yes |
| 12:45.54 | F|shie | [TK]D-Fender: lol, well i am using record under audio extension, instead of playing digits and i cant seem to record a file |
| 12:46.43 | [TK]D-Fender | F|shie, You aren't showing us what you're doing... |
| 12:48.44 | F|shie | [TK]D-Fender: http://ofps.oreilly.com/titles/9781449303822/c03-AudioManipulation_id302347.html#TriggeringAudioViaDTMF this is the tutorial i am following, it plays audio into the channels being spied on, i just removed the saynumber part, and tried using record |
| 12:49.46 | [TK]D-Fender | F|shie, shouw us your code and your debug. someone elses sample doesn't help us |
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| 12:59.11 | F|shie | [TK]D-Fender: http://pastebin.ca/2142211 |
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| 13:02.30 | F|shie | [TK]D-Fender: record not working here |
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| 13:03.26 | ggtest | anyone has try to stress test asterisk 1.8 or 10 ? |
| 13:03.40 | ggtest | with sipp, I got this in asterisk cli ERROR[3002]: netsock2.c:263 ast_sockaddr_resolve: getaddrinfo |
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| 13:10.52 | [TK]D-Fender | F|shie, show us the failure |
| 13:14.49 | [TK]D-Fender | F|shie, and you should go read the instructions again : "core show application record" |
| 13:14.50 | jeffspeff | I'm missing something that's keeping me from making sense of confbridge... How can you set multiple user profiles to a bridge? and how do you set a particular menu to each user profile? |
| 13:19.46 | F|shie | [TK]D-Fender: there are no failures, the file is generated, but it has no audio |
| 13:20.04 | [TK]D-Fender | F|shie, Your syntax is wrong. Read the instructions again |
| 13:20.45 | F|shie | [TK]D-Fender: i used the same for a normal call, works fine |
| 13:21.09 | [TK]D-Fender | f|I still don't see the failed call... |
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| 13:26.19 | *** join/#asterisk shamelessn00b (~bwahahaha@39.47.242.230) |
| 13:28.38 | shamelessn00b | if I have two users connected using dial, and then use chanspy through a local channel, can I stream that audio outside asterisk, using eagi on local channel |
| 13:29.20 | [TK]D-Fender | shamelessn00b, You can do whatever you want with it |
| 13:29.38 | shamelessn00b | wouldn't chanspy block the local channel? |
| 13:30.17 | [TK]D-Fender | So? What else would it be doing? |
| 13:31.03 | shamelessn00b | ok, so I have a local channel executing chanspy, I want to send that audio out from my asterisk box |
| 13:31.36 | shamelessn00b | would I have to change the application itself, like it has an option to write audio in raw to a file, instead I could modify it to write on a socket |
| 13:31.42 | shamelessn00b | or is their a simpler hack |
| 13:31.48 | shamelessn00b | there* |
| 13:32.25 | _Corey_ | shamelessn00b: Aren't you the guy trying to bridge Asterisk with that translation thing? Didn't MeetMe work out? |
| 13:32.56 | shamelessn00b | I'm just testing for 2 individuals at the moment |
| 13:33.06 | shamelessn00b | here's the scenario |
| 13:33.16 | shamelessn00b | A dials B, I use AMI to originate a local channel that executes chanspy |
| 13:33.25 | shamelessn00b | that hooks to the same call |
| 13:33.27 | [TK]D-Fender | shamelessn00b, You're not thinking about what is on the other end... |
| 13:34.20 | [TK]D-Fender | shamelessn00b, A single Local channel listening on chanspy doesn't lead anywhere. It has to eb bridged to another channel |
| 13:34.36 | _Corey_ | shamelessn00b: I think the suggestion last time you raised this same question was to throw MeetMe/ConfBridge in the middle and join EAGI channel as another participant... |
| 13:34.47 | _Corey_ | shamelessn00b: Have you tried that or not?> |
| 13:36.28 | [TK]D-Fender | No need for meetme/conf |
| 13:37.16 | *** join/#asterisk Tuju (~tuju@176.75.219.213.sta.estpak.ee) |
| 13:37.48 | _Corey_ | [TK]D-Fender: The other half of what he described last time involved one or the other party hearing or not hearing the audio coming back from the EAGI |
| 13:38.00 | shamelessn00b | [TK]D-Fender: originate(local/exten@context,context,exten) |
| 13:38.14 | *** join/#asterisk generalhan (~generalha@about/windows/staff/generalhan) |
| 13:38.28 | [TK]D-Fender | shamelessn00b, And what do you have on each end? |
| 13:38.56 | shamelessn00b | on the local channel I have chanspy, and on the other end I'm trying to run EAGI |
| 13:39.07 | [TK]D-Fender | Good. then you're done |
| 13:39.20 | shamelessn00b | playback works, but it doesn't seem to get any audio from the local channel |
| 13:39.39 | [TK]D-Fender | shamelessn00b, And we're not seeing what you'r actually doing |
| 13:40.19 | shamelessn00b | if I use whisper mode on chanspy, whatever I play back on the other leg of the call ,context,exten I get to hear in the original call, a-B |
| 13:40.44 | shamelessn00b | but whatever A-B speak which I'm supposed to get via chanspy on ,context,exten is not going through |
| 13:40.54 | Faustov | can I somehow retrieve the return value for dundi_lookup, when it fails to look up a number? |
| 13:41.07 | Faustov | the asterisk 1.8 wiki does not mention anything of sorts |
| 13:48.05 | leifmadsen | Faustov: if nothing was found, what are you expecting to be returned? |
| 13:48.51 | Faustov | leifmadsen: I'd like to write a condition depending on if the function succeeded or failed in my dialplan |
| 13:48.57 | leifmadsen | ok... |
| 13:49.16 | leifmadsen | so check to see how many results were returned, if < 1, then error |
| 13:49.34 | Faustov | sounds reasonable, thanks |
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| 14:00.12 | sereal-work | with dialplan patterns, how do you do a else type thing? like if I have _1XXX and I input 1111 and that user doesn't exist, how can I have a fall back? like to have it play a message indicating that is a invalid extension? |
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| 14:03.26 | shadebob | hi |
| 14:03.49 | tnt | Hi. I'm having trouble using a B410P card with asterisk. I compiled the latest libpri, dahdi-utils and dahdi-kernel and asterisk 1.8.11. But all the ports still seem down http://pastebin.com/BSr7vLmx |
| 14:04.22 | [TK]D-Fender | sereal-work, do not mix the concept of dilplan pattern's & devices up. |
| 14:04.52 | sereal-work | err can you explain a bit more what you mean by that? |
| 14:05.08 | [TK]D-Fender | ser"user doesn't exist" is not a concept a pattern match knows or cares about. It's up to you to validate this in your actual diaplan. Preferably by not even creating a non-productive match |
| 14:05.11 | jeffspeff | I'm missing something that's keeping me from making sense of confbridge... How can you set multiple user profiles to a bridge? and how do you set a particular menu to each user profile? |
| 14:05.35 | [TK]D-Fender | sereal-work, "have _1XXX and I input 1111 and that user doesn't exis" dialplan patterna have nothing to do with "users" |
| 14:06.19 | WIMPy | tnt: Doesn't look bad. Are the ports conencted to something alive and do you have the reight signalling configured? |
| 14:07.18 | sereal-work | okay, so if I only have 10 phones, and want to use 10XX block, should is there a way I should handle if if they enter 1011 which doesn't have a sip user behind it? Or should I just create a extension for each user 1001, 1002, 1003, etc |
| 14:07.22 | tnt | WIMPy: The first port is connected. The same hw was running fine with a very old setup using misdn but after updates I switched to dahdi and can't seem to make it work. |
| 14:07.49 | [TK]D-Fender | sereal-work, If you have 10 phones then make 10 extensions. |
| 14:08.01 | [TK]D-Fender | sereal-work, Stop using overkill patterns |
| 14:08.54 | sereal-work | heh alright, that's the way i'm doing it now. I just thought there was a nicer way of doing it. |
| 14:09.14 | [TK]D-Fender | sereal-work, if you are specifically looking for 1001 to 1010 then you can do that in 2 |
| 14:09.14 | sereal-work | so I really should only use patterns for stuff like dialing out? |
| 14:09.26 | sereal-work | oh yeah true... |
| 14:09.27 | [TK]D-Fender | No, you use patterns that are VALID |
| 14:09.45 | [TK]D-Fender | 10XX is 100 valid values |
| 14:10.01 | sereal-work | yes, so I should use 101X and 100X |
| 14:10.12 | [TK]D-Fender | sereClearly not |
| 14:10.14 | tnt | WIMPy: chan_dahdi.conf contains http://pastebin.com/PTKfMQud |
| 14:10.15 | [TK]D-Fender | sereal-work, Clearly not |
| 14:10.22 | WIMPy | tnt: I still prefer misdn. Or maybe even more. But if you got the right settings in chan_dahdi.conf it should work. |
| 14:10.48 | WIMPy | tnt: Is it ptp or ptmp? |
| 14:10.56 | [TK]D-Fender | sereal-work, At least you went from 10 times the entries you want to allow to only DOUBLE |
| 14:11.02 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 14:11.31 | tnt | WIMPy: misdn-init contained te_ptp=1,2 so I guess ptp |
| 14:11.53 | WIMPy | ok |
| 14:11.58 | sereal-work | er I should have a pattern 100X which covered 1-9 and then a pattern 10XX to cover 101-1019 |
| 14:12.28 | sereal-work | but until I have 20 phones I should do 100X and 1010 |
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| 14:12.33 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 14:13.33 | sereal-work | [TK]D-Fender, I guess i'm more asking what the correct/proper way of doing this is. I don't really mind making a extension for ever user I just thought this would be a more elegant way |
| 14:14.03 | [TK]D-Fender | sereal-work, Your pattern is still too big and allowing things outside of what you listed |
| 14:14.19 | [TK]D-Fender | sereal-work, You really aren't loking at your patterns... |
| 14:14.34 | sereal-work | er but wouldn't _100X just cover 1000-1009 ? |
| 14:15.00 | [TK]D-Fender | sereal-work, You said you wanted 10. you'v already expanded to 11+ |
| 14:15.13 | [TK]D-Fender | sereal-work, "pattern 10XX to cover 101-1019" <- and no, that covers 1 HUNDRED values |
| 14:16.01 | sereal-work | err but if I say just had 100X that would only cover 1000-1009 correct? (which is 10 values) |
| 14:16.23 | [TK]D-Fender | yes |
| 14:16.53 | sereal-work | okay, thanks for the help. |
| 14:17.00 | [TK]D-Fender | sereal-work, "Or should I just create a extension for each user 1001, 1002, 1003, etc" <- I don't see you suggesting that 1000 is a valid # |
| 14:17.11 | sereal-work | err yeah slip of my brain |
| 14:17.25 | [TK]D-Fender | sereal-work, Seemed to be you were aiming for 1001-1010 |
| 14:17.39 | sereal-work | err I really ment 1000-1009 |
| 14:18.10 | [TK]D-Fender | sereal-work, Then the first is ok, and your second pattern still needs work. |
| 14:18.11 | sereal-work | if I wanted 1001 to do something else I would just create a extension for 1001 and that would override what 100X does? |
| 14:20.03 | [TK]D-Fender | Should work |
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| 14:21.18 | *** part/#asterisk UQlev (~yuriy@46.251.116.90) |
| 14:25.08 | tnt | WIMPy: any idea how I could diagnose what's wrong ? |
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| 14:44.18 | WIMPy | tnt: Not much. I would have thought that L1 activation has been sorted, but you never know. Have you tried to call out? |
| 14:44.43 | tnt | yes, doesn't work either. |
| 14:44.52 | tnt | The leds at the back of the card are all off. |
| 14:45.51 | *** join/#asterisk brut- (~brut-@uslink-66.173.29-234.uslink.net) |
| 14:46.18 | WIMPy | Off? Not red? Do they work different from the PRI cards? |
| 14:47.14 | tnt | No, they should be red ... (they were in the old system). |
| 14:47.44 | WIMPy | But that was a different driver. So that doesn;t mean anything. |
| 14:49.00 | tnt | true ... dahdi_too show them as "alarm RED" but it's not exactly explicit what's wrong ... |
| 14:49.58 | WIMPy | That surely means no L1. |
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| 14:50.41 | Sheeplet | lo all |
| 14:50.42 | WIMPy | Err, wait. You don't have both drivers loaded, do you? |
| 14:51.00 | tnt | WIMPy: no, the old driver was for an old kernel, can't load it. |
| 14:51.30 | WIMPy | Ok, but the standard kernel also includes drivers. |
| 14:52.01 | brut- | lo folks |
| 14:52.25 | tnt | WIMPy: not here. (it's still a fairly old kernel 2.6.24) |
| 14:53.18 | WIMPy | Ok, that would be the old stuff, but it still contains the driver. |
| 14:55.12 | tnt | WIMPy: there is definitely no misdn modules loaded |
| 14:55.32 | WIMPy | ok |
| 14:55.52 | tnt | http://pastebin.com/649QN8tk |
| 14:56.13 | tnt | it definitely sees the card ... and dahdi_scan shows it. but dahdi_tool shows 'RED' |
| 14:56.49 | WIMPy | What do you have on the other ports? |
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| 14:57.19 | tnt | WIMPy: nothing they're not connected to anything. Only port 1 is connected to the network. |
| 14:57.41 | jeffspeff | when setting up a conference in confbridge, how do you determine if a user is admin or not? |
| 14:58.03 | p3nguin | Did your extension set the admin option? |
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| 14:59.02 | jeffspeff | exten=100,1,ConfBridge(100,General,,menu) |
| 14:59.20 | WIMPy | tnt: Expensive solution for one port. But (so) maybe you should try Digiums support. |
| 14:59.48 | p3nguin | That doesn't indicate admin to me. |
| 15:00.32 | jeffspeff | p3nguin, if the syntax is ConfBridge([confno][,bridge_profile[,user_profile[,menu]]]) then that only leaves one option for user profile. |
| 15:01.20 | p3nguin | Are you using asterisk 10? |
| 15:01.24 | jeffspeff | p3nguin, yes |
| 15:01.32 | tnt | WIMPy: no time for that ... my maintenance window closes in 1h and if it doesn't work by then I have to revert to the old config ... |
| 15:01.55 | p3nguin | That explains why your syntax isn't familiar to me. In 1.8, it's simply ConfBridge([confno][,options]). |
| 15:02.13 | p3nguin | If the a option is set, that enables admin mode. |
| 15:02.22 | p3nguin | 'a' option |
| 15:03.15 | jeffspeff | p3nguin, they rewrote it in 10.x to like a newer/better conferencing app than meetme |
| 15:03.22 | p3nguin | Yeah. |
| 15:04.20 | p3nguin | Someone familiar with ConfBridge in 10 will have to assist you, because I haven't used 10. |
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| 15:09.01 | brut- | has anyone tried to setup asterisk to perform an "on-demand" register before invite on outbound calls before? |
| 15:10.11 | brut- | I'm trying to setup "outbound-only" lines for testing, but I still need the lines to register before they invite, I was hoping to have them register on the fly, then invite, and let the registrations expire when unused to keep resource utilization down |
| 15:12.19 | [TK]D-Fender | brut-, No such option |
| 15:13.31 | Qwell | register has nothing to do with outbound calls |
| 15:13.59 | *** join/#asterisk darkdrgn2k3 (~darkdrgn2@209.90.253.66) |
| 15:14.04 | darkdrgn2k3 | Morning everyon |
| 15:14.29 | darkdrgn2k3 | i never got a chance to really ask, what do you guys thing of Shoretel/Avaya or adiran telephone swiches? |
| 15:14.52 | Qwell | darkdrgn2k3: You really want to ask for opinions of commercial PBXs in a competitors IRC channel? |
| 15:15.04 | darkdrgn2k3 | Qwell: yes actualy |
| 15:15.11 | darkdrgn2k3 | Qwell: this way i get all the dirt on them :-D |
| 15:15.37 | coppice | Qwell: you mean things aren't actually fair and balanced on IRC? |
| 15:16.35 | darkdrgn2k3 | Qwell: Really? i was going to build my whole business case around what is said on irc channel.... |
| 15:16.43 | [TK]D-Fender | brut-, What "resource utilization"? |
| 15:16.54 | darkdrgn2k3 | ...it was said on irc therefore it must be true... |
| 15:17.09 | Qwell | Then why ask? |
| 15:17.28 | darkdrgn2k3 | because im sick and teired of listing to sales ppl try to sell me on krappy hardware... |
| 15:17.37 | darkdrgn2k3 | Id rather ppl who work the trenches try to do that :-P |
| 15:17.40 | coppice | all shades of opinion feed an open mind |
| 15:18.20 | darkdrgn2k3 | But i forgot one little problem.. when asking a question on IRC, you have to spend a few hours defending asking of the questions first... :-P |
| 15:18.57 | _Corey_ | darkdrgn2k3: There are numerous articles, blogs, and other resources elsewhere that explain why people have chosen Asterisk over a proprietary platform. |
| 15:19.14 | brut- | [TK]D-Fender, that's what I was afraid of |
| 15:19.34 | brut- | I want to keep these lines unregistered on the asterisk box with our SBC so they can be used by others for testing |
| 15:19.35 | _Corey_ | darkdrgn2k3: If you have some specific questions, you'll get answers... Taking a survey of everyone here probably won't get very far |
| 15:19.55 | [TK]D-Fender | brut-, And your provider only allows calls if you are registered? You've tested this? |
| 15:19.56 | brut- | these DNs are shared for calling from our serving area NPA/NXXs |
| 15:20.00 | darkdrgn2k3 | ok, first of all i was warned to stay a mile away from avaya because of their warrenty scem.... |
| 15:20.01 | brut- | yes, I am the provider too |
| 15:20.03 | *** join/#asterisk Maliuta (~nobusines@kiev.lusan.id.au) |
| 15:20.17 | brut- | we require registration on the SBC before INVITE |
| 15:20.22 | [TK]D-Fender | brut-, Then go tell your provider to change then :p |
| 15:20.27 | brut- | lol, indeed |
| 15:20.54 | brut- | I thought about changing it, but I was hoping to get around having to look at that option with dynamic registrations from asterisk |
| 15:21.00 | [TK]D-Fender | nope |
| 15:21.08 | brut- | I can do that on Mitel SIP phones, just not with asterisk |
| 15:21.39 | brut- | maybe I could register a SCA of each line dedicated to the asterisk box..., then I could do 2 registrations per DN..., hrm |
| 15:22.13 | brut- | Qwell, in our case, it does, because we require registration before invite on the sbc :) |
| 15:22.28 | Qwell | Then fix your SBC because it's broken |
| 15:22.53 | brut- | it's not broken, that's a security measure we use on it |
| 15:22.59 | Qwell | One that is not valid. |
| 15:23.06 | p3nguin | Asterisk doesn't need a device to be registered to make a call. If it can auth for the INVITE, it can make a call. |
| 15:23.23 | brut- | it's perfectly valid, since the SBC uses ACLs to prevent IPs from registering that aren't allowed :) |
| 15:23.42 | brut- | yeah it's not asterisk that's the UAS in this case, it's a UAC |
| 15:24.17 | [TK]D-Fender | brut-, If you're desperate you could always script a mod to sip.conf, apply your change, iss ue a SIP reload, place your call, then then on expiry, reverse the mod and kil the reg... |
| 15:24.22 | brut- | in any case, I know that it's not an option that asterisk that I can utilize readily |
| 15:24.25 | brut- | so I'll explore other options |
| 15:24.39 | brut- | yeah I was looking at that or realtime sip support via sql |
| 15:24.47 | brut- | either would be viable |
| 15:25.05 | brut- | I saw some chat in the asterisk-dev mailing list for moving registrations down to the peer contexts in sip.conf |
| 15:25.13 | brut- | so that's the first option I was exploring |
| 15:25.16 | [TK]D-Fender | p3nguin, Yes, some switches require it... I consider the practice rather annoying, but it is a reality in a few cases |
| 15:25.21 | brut- | I'll look at realtime |
| 15:25.48 | brut- | lol, annoying - perhaps. but in terms of TISP security, it's recommended |
| 15:26.04 | brut- | at least it's recommended by security experts at SIPNOC |
| 15:26.17 | brut- | then again they change their thoughts on stuff every couple of years, so who knows |
| 15:26.59 | brut- | thanks for the input folks, back to the grind |
| 15:28.51 | WIMPy | tnt: Does the old config still work? |
| 15:30.24 | *** join/#asterisk oej (~olle@h87-96-134-129.dynamic.se.alltele.net) |
| 15:37.03 | tnt | WIMPy: booting the previous image now ... |
| 15:38.31 | tnt | WIMPy: yup, works fine. |
| 15:39.09 | *** join/#asterisk din3sh (din3sh@41.136.244.89) |
| 15:39.14 | tnt | But in the end I managed to have dahdi bring L1 up when running on bare metal rather than via a pci passthrough. |
| 15:39.20 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v002-238.mobile.uci.edu) |
| 15:39.22 | din3sh | hi all |
| 15:39.39 | din3sh | anybody ever hooked up a nortel meridian to *? |
| 15:39.56 | WIMPy | tnt: Ah. So that's the issue. |
| 15:40.23 | tnt | WIMPy: so it seems ... weird because there is no low level errors or anything and other people seem to be using it that way. |
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| 15:56.08 | jaytee | din3sh, I've connected an Option 11C to Asterisk using PRI |
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| 16:07.27 | din3sh | i have managed to hook them up, but for some reason i cannot transfer some calls from nortel extension to asterisk |
| 16:07.48 | din3sh | the asterisk extension rings, the nortel phone is unable to bridge the call |
| 16:08.30 | *** join/#asterisk Azrael808 (~peter@212.161.9.162) |
| 16:10.11 | elanz | what's does the rg directory in asterisk stand for and what's it for |
| 16:11.25 | [TK]D-Fender | elanz, what "rg" directory? Where do you see this? |
| 16:12.04 | *** join/#asterisk bn-7bc (bjarne@macbook-pro.lan-sx.noare-1.holmedal.net) |
| 16:12.06 | WIMPy | din3sh: A bit of pri debug should tell you more. |
| 16:13.51 | elanz | /home/recordings-mp3 |
| 16:17.05 | [TK]D-Fender | elanz, And how is this an Asterisk-related folder? |
| 16:17.30 | [TK]D-Fender | elanz, Is your asterisk.conf pointing there for something? Otherwise this is not some core aspect of * |
| 16:18.20 | elanz | are ARCHIVE CONFERENCE out exten part of * aspect? |
| 16:20.44 | [TK]D-Fender | no |
| 16:21.07 | elanz | yikes |
| 16:21.09 | [TK]D-Fender | elanz, #freepbx <- |
| 16:21.22 | elanz | ok |
| 16:21.45 | elanz | sorry to ask so much but I am just new to all this. |
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| 16:43.00 | *** join/#asterisk n3tn3ws_op (~tripthago@gateway.meteor-web.com) |
| 16:47.05 | n3tn3ws_op | I need a little assistance if anyone here is free to discuss. Thanks |
| 16:47.32 | ChannelZ | ~ask |
| 16:47.32 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 16:52.29 | bmoraca_work | has anyone used one of the AA50 appliances? |
| 16:52.54 | leifmadsen | yes |
| 16:53.02 | n3tn3ws_op | We are trying to set up call routing based on information in our database (i.e. it is a number with a previous order, so send it to channel X). Is there a way to make decisions via a PHP/PERL/whatever script and communicate this back to Asterisk, or a module that exists to fulfill this function? |
| 16:53.14 | Qwell | n3tn3ws_op: look into func_odbc |
| 16:53.19 | leifmadsen | n3tn3ws_op: AGI() or func_odbc |
| 16:53.20 | bmoraca_work | leifmadsen: i'm looking for a personal review here, not a marketing spiel |
| 16:53.27 | leifmadsen | depending on how much logic you need in the script |
| 16:53.42 | Qwell | leifmadsen: I didn't take you as a fan of AGI |
| 16:53.45 | leifmadsen | bmoraca_work: you asked if someone has used an AA50; I answered 'yes'. I personally have not used one. |
| 16:53.58 | leifmadsen | Qwell: well if you just need to do a DB lookup, then func_odbc is significantly easier |
| 16:54.02 | Qwell | leifmadsen: several people have used them, in fact |
| 16:54.06 | leifmadsen | Qwell: indeed |
| 16:55.01 | n3tn3ws_op | thanks |
| 16:55.03 | bmoraca_work | ok, let's try to get an answer that's not from someone with an attitude. is there a demo of the UI availabe for Asterisk Business Edition? |
| 16:55.26 | Qwell | bmoraca_work: It's more or less asterisk-gui, with some tweaks |
| 16:55.37 | Qwell | in fact, the AA50 is why that was written |
| 16:55.48 | bmoraca_work | the old one from the old AsteriskNow? |
| 16:56.04 | bmoraca_work | i'm sure it's improved in the last 3 years |
| 16:56.31 | Qwell | correct |
| 16:56.58 | bmoraca_work | but there's no online demo? |
| 16:57.14 | Qwell | not that I'm aware of |
| 16:57.26 | bmoraca_work | alright, thanks. |
| 16:58.46 | coppice | bmoraca_work: the blackfin boxes all have a weak spot - they run uC Linux, rather than full Linux> if you give them simple jobs they are OK. If you make them run lots of scripts, and generally create complexity in your applications they fragment memory until they fall over |
| 16:59.14 | bmoraca_work | i was actually looking at the S808i box |
| 16:59.41 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 17:00.25 | coppice | its pretty expensive for a fairly simple piece of hardware |
| 17:00.58 | bmoraca_work | true...but it's cheaper than an atom-based server (and easier) |
| 17:01.25 | bmoraca_work | and i only need it to run 4 phones |
| 17:02.13 | *** join/#asterisk irule (~irule@189.195.252.73) |
| 17:04.05 | bmoraca_work | hmm |
| 17:04.14 | bmoraca_work | seems that it's mostly been mothballed |
| 17:04.36 | [TK]D-Fender | AA50 is long-dead |
| 17:04.38 | coppice | isn't it something like $1500? that's pretty expensive for an atom box |
| 17:04.51 | bmoraca_work | coppice: not when you factor in the 8 FXO ports |
| 17:05.06 | coppice | [TK]D-Fender it still seems to be on reseller's sites |
| 17:05.18 | bmoraca_work | unless they're not echo cancelled...in which case, yes, it is expensive |
| 17:05.49 | coppice | you can cancel 8 ports on an atom |
| 17:05.59 | din3sh | bmoraca_work: IP04/IP08 boxes with blackfin/astfin can work |
| 17:06.02 | [TK]D-Fender | bmoraca_work, I can get a 24-port FXO gatwaway and an atom Netbook and have money leftover for that... |
| 17:06.44 | bmoraca_work | i haven't had good luck with non-echocancelled analog lines with the phone lines out here... |
| 17:06.47 | bmoraca_work | they're pretty terrible |
| 17:07.18 | coppice | you certainly need to cancel, but atom software will do it perfectly well for only 8 ports |
| 17:08.18 | bmoraca_work | i would agree, but I've tried numerous times to effectively use software echo cancellation and it's always been miserable. the wiring in this area is abysmal |
| 17:09.00 | coppice | the world needs a replacement for the IP04/IP08 based on a Cortex chip |
| 17:09.23 | coppice | OSLEC gives less trouble than most of the hardware EC modules |
| 17:11.28 | din3sh | oslec works on the blackfin |
| 17:23.18 | *** join/#asterisk HyperNerdV2_ (~HyperNerd@71-95-164-90.static.mtpk.ca.charter.com) |
| 17:25.50 | *** join/#asterisk TimeRider (~steve@188-220-34-144.dsl.cnl.uk.net) |
| 17:33.16 | *** join/#asterisk anonymouz666 (~anonymouz@189-25-40-55.user.veloxzone.com.br) |
| 17:44.53 | *** join/#asterisk AndyHarris2 (~AndyHarri@dsl-217-155-202-52.zen.co.uk) |
| 17:52.00 | *** join/#asterisk DarthExpeditor (~IceChat9@96-42-133-130.static.trcy.mi.charter.com) |
| 17:52.16 | *** join/#asterisk danfromuk (~IceChat77@2.27.26.23) |
| 17:52.47 | DarthExpeditor | I have a customer that needs to be able to do Email to Fax and Fax to Email on a per extension (DID) basis. They are using SIP for their trunk. What would you recommend? |
| 17:53.29 | danfromuk | Hi, i'm currently upgrading from 1.4 to 1.8. Has anything changed with the CDR(accountcode) variable? Its now returning '' when it used to contain the accountcode from the sip peer. |
| 17:54.27 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 17:54.44 | *** join/#asterisk bluregard (~matt@c-71-201-99-216.hsd1.il.comcast.net) |
| 17:57.16 | danfromuk | it appears to be a bug, as listed at issues.asterisk.org |
| 17:58.03 | danfromuk | however the bug issue was closed by someone commenting that its not a bug. :-S |
| 17:58.57 | [TK]D-Fender | DarthExpeditor, I recommend SpanDSP and a provider that supports T.38 |
| 17:59.57 | danfromuk | How do I added patch code to my asterisk build? |
| 18:00.16 | danfromuk | add* |
| 18:05.20 | *** join/#asterisk imox (~imox@91-64-185-199-dynip.superkabel.de) |
| 18:08.16 | coppice | install spandsp, and then install asterisk. no patching is needed |
| 18:10.38 | wdoekes2 | danfromuk: the accountcode bug is fixed in 1.8.12.0-rc2 (r357575) |
| 18:13.37 | danfromuk | ah, thanks i'm using 1.8.11.1. I never used an rc before. Always waited for the full release. |
| 18:19.48 | *** part/#asterisk talntid (~t@173-160-189-58-Washington.hfc.comcastbusiness.net) |
| 18:30.35 | AndyHarris2 | Would anyone know where 'show features' has gone --- trying to debug a *8 issue --- ta |
| 18:31.13 | _Corey_ | AndyHarris2: features show |
| 18:32.21 | AndyHarris2 | @_correy_ yippee ! |
| 18:32.36 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 18:35.45 | *** join/#asterisk coppice (~chatzilla@globbits.tripleone.co.uk) |
| 18:36.14 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 18:40.15 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-147.cablep.bezeqint.net) |
| 18:43.48 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 18:55.27 | eZz | is there a way to debug/troubleshoot an issue with dtmf detection using native ways instead of grep'ing a logfiles ? |
| 18:58.44 | *** part/#asterisk elanz (~elanz@69.169.150.15.provo.static.broadweavenetworks.net) |
| 19:03.18 | pabelanger | eZz: you can use wireshark |
| 19:03.32 | ChannelZ | Is there an RBL out there for CID numbers? |
| 19:03.42 | Marquel | morning |
| 19:04.02 | eZz | pabelanger: i know, i'm asking using native asterisk's methods... |
| 19:04.03 | carrar | Just pay your damn bills!! :) |
| 19:04.18 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 19:04.32 | Marquel | i'm looking for a more complete documentation of DAHDISendCallreroutingFacility() - is there some place aside from asterisk.org which is at best not really helping? |
| 19:04.43 | pabelanger | eZz: then no |
| 19:06.21 | eZz | pabelanger: for some reasons asterisk can't recognize key presses from some users... So the only one way is pcap ? |
| 19:06.42 | pabelanger | ~collectdebug |
| 19:06.42 | infobot | methinks collectdebug is a method of collecting logs allowing others help troubleshoot an issue. Refer to https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information |
| 19:06.58 | pabelanger | eZz: ^ enable DTMF logging for logger.conf |
| 19:07.10 | eZz | already enabled |
| 19:07.22 | fr0ggie | I'm having some problems with using google voice with asterisk-- When a call comes in, the only way i can seem to send the required '1' is by Dial(SIP/targetuser, 180, D(:1)) -- i'd like to play a "hang on while I find that user" message before starting ringdown |
| 19:10.04 | fr0ggie | i tried using SendDTMF, even tried using a /n and /nb Local channel with Answer() but no luck |
| 19:17.16 | *** join/#asterisk nix8n82 (~nate@24.143.10.144) |
| 19:20.10 | *** join/#asterisk greenwolf (cc98c957@gateway/web/freenode/ip.204.152.201.87) |
| 19:20.23 | greenwolf | wondering if anyone has good knowledge with opensips? |
| 19:21.09 | *** join/#asterisk HyperNerdV2 (~HyperNerd@71-95-164-90.static.mtpk.ca.charter.com) |
| 19:21.34 | greenwolf | who here is familiar well with opensips or openSER |
| 19:21.51 | *** join/#asterisk ergodicsum (~micronix@173.221.15.173.nw.nuvox.net) |
| 19:22.21 | ergodicsum | Is there an event that occurs when an agent logs into a queue? Or how can I determine if an agent logs into a queue? |
| 19:23.25 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 19:25.55 | *** join/#asterisk nighty- (~nighty@static-68-179-124-161.ptr.terago.net) |
| 19:27.06 | *** join/#asterisk geogeek (0c477ae3@gateway/web/freenode/ip.12.71.122.227) |
| 19:27.19 | eZz | pabelanger: wireshark was exited with NoMemory exception ;) there is 2.4G of pcap log :-) |
| 19:27.42 | eZz | sorry OutOfMemory or something like that |
| 19:30.27 | fr0ggie | What's a good way to accept an inbound call- play a message, park it, then notify a group of SIP phones that there's a call parked-- with a timeout of 120 seconds after which it goes to a vm box? |
| 19:30.55 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 19:32.26 | [TK]D-Fender | fr0ggie, "core show application park And Annouce" |
| 19:32.33 | [TK]D-Fender | fr0ggie, "core show application parkandannouce" |
| 19:33.01 | bluregard | does anyone have a sound file from Allison that says "payment" and "amount"? |
| 19:33.10 | fr0ggie | hmm my install seems reasonably broken-- my docs arent in the right place |
| 19:33.18 | fr0ggie | [TK]D-Fender: ill look it up, thanks |
| 19:35.41 | Marquel | i'll rephrase my question: what are the possible values for 'reason' in DAHDISendCallreroutingFacility() and their respective meanings? |
| 19:38.38 | fr0ggie | [Apr 30 15:38:27] WARNING[27277]: xmldoc.c:1839 xmldoc_build_field: Couldn't find function IFMODULE in XML documentation |
| 19:38.56 | fr0ggie | ive still got my source i built from-- how do i get these docs in place? |
| 19:39.09 | *** join/#asterisk magnet__ (~magnet@AMontpellier-651-1-178-204.w86-209.abo.wanadoo.fr) |
| 19:39.14 | magnet__ | good evening. |
| 19:39.59 | geogeek | Hey all. Having trouble getting caller id to work...dialing from our nortel phone through asterisk to the pstn (dahdi). Shows up as unknown on receiving party's phone. I can call from one of my SIP phones and the caller id works. Don't see anything about setting caller id in cli. |
| 19:41.38 | magnet__ | I'm trying to debug a sip retransmission problem. I enabled sip debug, and I see this line : o=root 1111478244 1111478245 IN IP4 10.10.10.108 . shoulnd the ip be a public one ? I'm behind a nat both with client and server. |
| 19:42.08 | eZz | look to c= and m= |
| 19:43.41 | magnet__ | it show c=IN IP4 10.10.10.108 and m=audio 19750 RTP/AVP 0 8 101 . |
| 19:44.11 | eZz | yeah, so rtp endpoint is 10.10.10.108:19750 |
| 19:44.32 | *** join/#asterisk asterisk-Tester (~ramy@80.79.159.253) |
| 19:44.48 | magnet__ | I guess it's a problem it should be a public ip. |
| 19:45.51 | *** join/#asterisk shinao1 (~shinao1@41.58.6.235) |
| 19:50.06 | shinao1 | hi guys..im trying to setup dundi over iax2 between two asterisk servers, each behind a pfsense gateway/router. I've setup NAT/firewall rules that redirect UDPports 4569 4520, and 10,000-20,000 on the pfsense firewalls to the asterisk boxes. I can do dundi lookups perfectly, but when I try to make calls i get an "all circuits are busy.ulaw" errors. Please, what am I missing? Any ports left to open? Please advise. |
| 19:51.04 | *** join/#asterisk Hive (~Hive@173-165-205-1-jacksonville.hfc.comcastbusiness.net) |
| 19:51.30 | jaytee | shinao1, what about port 5060 for SIP? that's kind of necessary for most scenarios involving calls |
| 19:52.03 | shinao1 | well since its iax2 trunks i thought it wasnt necessary |
| 19:52.11 | Qwell | Then you don't need the RTP ports open. |
| 19:52.59 | shinao1 | ok |
| 19:53.04 | jaytee | ok, i read through quick and missed the iax2 |
| 19:53.18 | shinao1 | i opened RTP just in case |
| 19:53.55 | [TK]D-Fender | shinao1, that is a soundfile, not an "error" being reported back |
| 19:53.59 | *** join/#asterisk eicto (~eicto@144-71.dsl.aichyna.com) |
| 19:54.04 | [TK]D-Fender | shinao1, Show the actual failure |
| 19:54.09 | shinao1 | ok |
| 20:00.54 | *** join/#asterisk yetanotheruser (~user@p4FDC1FE7.dip.t-dialin.net) |
| 20:01.05 | shinao1 | trying to get the log, its kind of remote |
| 20:02.51 | shinao1 | [TK]D-Fender: should i paste it here or in pastebin? |
| 20:02.59 | [TK]D-Fender | ~pb |
| 20:03.00 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 20:03.02 | [TK]D-Fender | clearly |
| 20:03.09 | shinao1 | ok |
| 20:07.59 | *** join/#asterisk yetanotheruser (~user@p4FDC1FE7.dip.t-dialin.net) |
| 20:09.32 | Hive | so I'm trying to get queue's to work on my system (asterisk 1.8.4.4) and I'm following the guide https://wiki.asterisk.org/wiki/display/AST/Building+Queues . the section that starts "Aha, we're not getting the device state correctly." says that for installs prior to 1.6.0 we have to use 'call-limit' to get device states properly, and in versions after that we should use 'callcounter'. My install is using 'call-limit' but i |
| 20:09.55 | Hive | Should i switch over to 'callcounter' since that is what is documented, or follow the 'if it's not broken...' mentality |
| 20:10.59 | shinao1 | [TK]D-Fender: http://pastebin.com/hmhpJnHY |
| 20:12.28 | [TK]D-Fender | shinao1, Nowhere in there do I see yuo DIAL() |
| 20:12.45 | shinao1 | excuse me |
| 20:12.49 | shinao1 | let me repaste |
| 20:13.03 | [TK]D-Fender | Which I guess is actually due to the got and an assumed sqitch |
| 20:13.06 | [TK]D-Fender | switch |
| 20:13.24 | [TK]D-Fender | [Apr 30 15:33:04] WARNING[2592] chan_iax2.c: No such host: ndcpbx <-------------- |
| 20:13.28 | [TK]D-Fender | Sure seems to be a problem |
| 20:13.47 | shinao1 | ah.. |
| 20:14.04 | shinao1 | but doesnt dundi use DNS? |
| 20:15.50 | shinao1 | thanks a lot.. thats probably the issue |
| 20:16.30 | shinao1 | i used a name ext-ndcpbx.. does dundi not like hypens? |
| 20:17.18 | [TK]D-Fender | Maybe you should be looking at your trunks... |
| 20:17.47 | shinao1 | i have a single dundi trunk and a single iax2 trunk.. |
| 20:20.00 | shinao1 | thanks ill test this out |
| 20:29.53 | *** join/#asterisk irule (~irule@187.139.6.189) |
| 20:30.36 | irule | hi, my spa3102 trunk calls to and from sip sound very low, is there a way within asterisk to pump up the volume? |
| 20:33.05 | Gugge | what about the volume function? |
| 20:37.23 | *** join/#asterisk jamko (~chatzilla@mail.jaguardata.com) |
| 20:40.10 | Hive | does anyone have any experience working with queues in an asterisk install > 1.6.0? |
| 20:40.36 | pabelanger | I'm sure somebody does. Try explaining your issue or what you want to do |
| 20:43.28 | *** join/#asterisk Hive (~Hive@173-165-205-1-jacksonville.hfc.comcastbusiness.net) |
| 20:43.38 | Hive | gah got booted for some reason @_@ |
| 20:44.09 | Hive | anyways, im looking for advice on if i should go in and change all my call-limits or just leave them since they are working ok... im nervous about breaking everything if i switch over |
| 20:47.42 | p3nguin | irule: Increase the gain on the device. |
| 20:49.18 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 20:51.27 | Hive | ok, my coworker just came over and showed me the base code for asterisk. In sip.conf, 'call-limit' and 'callcounter' do essentially the exact same thing. Good little tidbit of information for our noggins :) |
| 20:51.28 | *** part/#asterisk geogeek (0c477ae3@gateway/web/freenode/ip.12.71.122.227) |
| 20:51.50 | *** part/#asterisk asterisk-Tester (~ramy@80.79.159.253) |
| 20:51.56 | Qwell | Hive: not quite |
| 20:52.45 | *** join/#asterisk Russ (foobar@ip68-106-254-4.ph.ph.cox.net) |
| 20:54.40 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 20:54.50 | watchy | today is never going to end |
| 20:56.28 | *** join/#asterisk rossand (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 20:57.00 | Hive | Qwell: care to enlighten me as to how they differ? Or point me in a direction to get my learn on? |
| 20:57.16 | Qwell | Hive: callcounter just sets it to a really really high number |
| 20:58.28 | Hive | oh, so call-limit is not truely deprecated? |
| 20:58.35 | irule | thanks p3nguin |
| 20:59.46 | p3nguin | irule: Did you find the settings? |
| 21:07.23 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 21:07.36 | *** join/#asterisk greenwolf (ad40049b@gateway/web/freenode/ip.173.64.4.155) |
| 21:10.46 | *** join/#asterisk DavidNGonzalez (~dgonzalez@99-158-162-221.uvs.cicril.sbcglobal.net) |
| 21:11.21 | DavidNGonzalez | Is this the right place to ask questions about Asterisk? |
| 21:13.22 | pabelanger | ~ask |
| 21:13.22 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 21:15.20 | DavidNGonzalez | I'm having an issue on an asterisk server, 911 calls call out, but the operator is unable to hear us, the emergency calls are set to go out the same line as the rest of our calls |
| 21:16.51 | *** join/#asterisk Agro (~Agro@108-79-20-223.lightspeed.hstntx.sbcglobal.net) |
| 21:20.05 | *** join/#asterisk bn-7bc (bjarne@macbook-pro.lan-sx.noare-1.holmedal.net) |
| 21:20.08 | p3nguin | davidngonzalez: What kind of "line" are you referring to? |
| 21:22.19 | DavidNGonzalez | The sip trunk |
| 21:25.59 | *** join/#asterisk TimeRider (~steve@188-220-34-144.dsl.cnl.uk.net) |
| 21:33.25 | DavidNGonzalez | I'm not quite sure where to look to resolve the issue, because the emergency calls are set to go out the same trunk line as the regular calls, I don't understand why there would be an audio issue with the 911 calls |
| 21:43.42 | *** join/#asterisk Kobaz (~kobaz@its.kobaz.net) |
| 21:47.32 | *** join/#asterisk jsjc (~Adium@81.Red-80-37-159.staticIP.rima-tde.net) |
| 21:53.39 | pigpen | I quick stupid question: if a "device" does not exist, is the DEVSTATE = Unavailable ? |
| 21:54.07 | pigpen | ie: I setup a hint, that a peer does not exist, then do a core show hints, it shows Unavailable. |
| 21:54.09 | leifmadsen | might be unknown |
| 21:54.22 | pigpen | Yeah, that is what I am bouncing between. |
| 21:54.34 | pigpen | I guess I'll do a real test. |
| 21:55.09 | pigpen | leifmadsen, thanks for the nice write up on SLA. |
| 21:57.23 | pigpen | yeah: UNAVAILABLE |
| 21:57.49 | pigpen | if UNAVAILABLE, no since it attempting to call it. |
| 22:00.34 | *** join/#asterisk nosaj (~jbarinas@gwb.anditel.com.co) |
| 22:16.00 | *** join/#asterisk jeffspeff (~jeffspeff@h215.59.40.162.dynamic.ip.windstream.net) |
| 22:16.39 | *** join/#asterisk elanz (~elanz@69.169.150.15.provo.static.broadweavenetworks.net) |
| 22:16.54 | *** join/#asterisk nix8n82 (~nate@24.143.10.144) |
| 22:17.46 | elanz | quick question. doing a conversion from wav to mp3 file: |
| 22:17.46 | elanz | <PROTECTED> |
| 22:17.46 | elanz | Warning: unsupported audio format |
| 22:17.46 | elanz | why am I getting this warning? |
| 22:18.49 | WIMPy | You probably have a sox sersion without mp3 support. |
| 22:19.37 | WIMPy | Err, lame? Who gave that error? |
| 22:19.44 | elanz | lame |
| 22:19.57 | WIMPy | Oh |
| 22:20.30 | WIMPy | It probably doesn't know the type of wav then. |
| 22:20.53 | *** join/#asterisk SuperNull (~FreeManof@24-148-101-238.ip.mhcable.com) |
| 22:21.10 | SuperNull | Hey guys anyone a ninja at dialplan ? specifically the regexp style for ATAs and the sort. |
| 22:22.06 | WIMPy | SuperNull: If you're referring to the dialplan on the device: That is device specific. |
| 22:22.16 | WIMPy | Or at least vendor specific. |
| 22:22.20 | *** join/#asterisk Agro (~Agro@108-79-20-223.lightspeed.hstntx.sbcglobal.net) |
| 22:22.27 | SuperNull | linksys then ;) |
| 22:22.30 | SuperNull | er. |
| 22:22.36 | SuperNull | 'cisco small business AKA linksys' |
| 22:22.44 | elanz | WIMPy any suggestions on how to make lame know that |
| 22:25.32 | *** join/#asterisk imox (~imox@91-64-185-199-dynip.superkabel.de) |
| 22:26.47 | WIMPy | SuperNull: I made come wild guesses and came up with "[0-9*#].". |
| 22:31.31 | SuperNull | come wild ? |
| 22:31.39 | SuperNull | remembers this for safety reasons |
| 22:32.13 | WIMPy | The default didn't allow to send feature codes. |
| 22:32.49 | SuperNull | odd dude, we have those regular voip phones aka true sip phones, and we have SPA-2102s. |
| 22:33.15 | SuperNull | the 2102s for the life of me could not dial NPA-NXX-NXXX without a 1 in front. it kept short cutting the dialplan. |
| 22:34.13 | WIMPy | These device dialplans are a PITA. |
| 22:34.24 | SuperNull | yeah they are. |
| 22:34.28 | SuperNull | try a Calix GPON ont ;) |
| 22:34.33 | WIMPy | Best to have overlap support so that Asterisk can decide. |
| 22:34.46 | SuperNull | it has like 3 characters.. then it has special .. 'codes' that .. will also re-generate dial tone. |
| 22:34.59 | WIMPy | But At least the SPA962 can't. :-( |
| 22:35.27 | SuperNull | this GPON unit we have is pretty much as dumb as an end point can be. |
| 22:35.54 | WIMPy | That's ok as long as it doesn't try to be clever, I think. |
| 22:36.11 | SuperNull | the SPAs look like.. sorcery compared to the GPON. the irony .. is the features that were so nice in the SPA-2102 made it rediculous to get everything identical on the GPONs as it all had to be moved server side. |
| 22:36.17 | SuperNull | (and should of to begin with anyway) |
| 22:36.36 | SuperNull | Wimpy it does try to be clever with its 'intelligent dialplan' option. |
| 22:36.42 | SuperNull | .. yes.. seriously. |
| 22:36.56 | SuperNull | found out that, it was preventing 911 later.. DETAILS OH DETAILS |
| 22:38.45 | WIMPy | Yes, that's the issue. On other devices, you can just leave the dialplan blank to make it not try to be clever. |
| 22:39.30 | WIMPy | But as I said: The only real solution is overlap support to let Asterisk decide. |
| 22:39.31 | *** part/#asterisk elanz (~elanz@69.169.150.15.provo.static.broadweavenetworks.net) |
| 22:40.11 | SuperNull | i would prefer it did nothing. |
| 22:40.12 | SuperNull | i mean |
| 22:40.18 | SuperNull | the problem then is timeout. |
| 22:40.30 | WIMPy | That's why. |
| 22:40.33 | SuperNull | the 'it takes 5 seconds before it starts dialing at all' type thing. |
| 22:45.41 | *** join/#asterisk s[x] (~sx]@eth589.qld.adsl.internode.on.net) |
| 22:54.01 | jeffspeff | where can I find information about the asterisk database for version 10.x? |
| 22:54.58 | jeffspeff | I'm looking to update the code at http://www.voip-info.org/wiki/index.php?page_id=5614 and need something to reference the DB references. |
| 22:55.02 | jeffspeff | in the macro |
| 22:56.42 | *** join/#asterisk Neptu (~Neptu@213-67-244-97-no100.tbcn.telia.com) |
| 22:59.44 | *** join/#asterisk JuStIcIa_ (~JuStIcIa_@190.167.120.174) |
| 23:05.53 | *** join/#asterisk Russ (foobar@ip68-106-254-4.ph.ph.cox.net) |
| 23:08.07 | [TK]D-Fender | jeffspeff: it hasn't changed from previous versions |
| 23:08.11 | *** join/#asterisk imox (~imox@91-64-185-199-dynip.superkabel.de) |
| 23:13.40 | *** join/#asterisk jpsharp (jsharp@ohno.mrbill.net) |
| 23:18.47 | *** part/#asterisk nosaj (~jbarinas@gwb.anditel.com.co) |
| 23:20.50 | *** join/#asterisk Russ (foobar@ip68-106-254-4.ph.ph.cox.net) |
| 23:21.06 | *** join/#asterisk tekzilla (~jon@hmbg-5f76741e.pool.mediaWays.net) |
| 23:21.23 | *** join/#asterisk FlaPer87 (~FlaPer87@unaffiliated/flaper87) |
| 23:22.36 | jeffspeff | [TK]D-Fender, got a minute to take a look at this? I'm getting stuck on a part, i'm sure it's something that i'm missing because of staring at it too long. You'll see what it's trying to do easily. |
| 23:24.59 | FlaPer87 | hey guys, I'm writing a video streamer using h263-1998 and I'm having some problems. This is the image I'm getting in the client B http://imageshack.us/photo/my-images/843/screenshot20120428at201.png |
| 23:25.04 | jeffspeff | [TK]D-Fender, http://pastebin.com/Qk674AbJ the problem is with line 25 |
| 23:25.15 | FlaPer87 | the frame is broken |
| 23:25.27 | FlaPer87 | I've checked the payload header and it seems right |
| 23:25.47 | FlaPer87 | but I dunno what's happening |
| 23:25.56 | FlaPer87 | Does any of you can give me a hint? |
| 23:26.12 | FlaPer87 | It looks like the first row is correct but the rest of the frame isn't |
| 23:41.12 | [TK]D-Fender | jeffspeff: You're in a regular context and using a generic ARG as though you thought you were in a MACRO |
| 23:41.20 | [TK]D-Fender | jeffspeff: Where do I see what you set it to? |
| 23:41.33 | [TK]D-Fender | jeffspeff: Because on sight I'd venture "never did" |
| 23:43.45 | *** join/#asterisk Naikrovek (~mbluth@unaffiliated/naikrovek) |
| 23:51.46 | *** join/#asterisk file (~file@asterisk/developer-and-muffin-lover/file) |
| 23:51.46 | *** mode/#asterisk [+o file] by ChanServ |
| 23:53.50 | *** join/#asterisk Naikrovek (~mbluth@unaffiliated/naikrovek) |
| 23:55.14 | brut- | jeffspeff, maybe I'm reading it wrong..., but isn't line 19 referencing a macro by the wrong name? ("makeconf" rather than "makeconference") |
