| 00:00.04 | [TK]D-Fender | Read the page |
| 00:00.11 | [TK]D-Fender | Logical operators <--- |
| 00:00.17 | [TK]D-Fender | expr1 | expr2 (Logical OR) |
| 00:00.24 | [TK]D-Fender | bold print |
| 00:00.26 | Cyford33 | i read that |
| 00:00.44 | Cyford33 | my and works |
| 00:00.45 | [TK]D-Fender | Yes and you shoved a SQUARE BRACE right after your in that line |
| 00:00.53 | [TK]D-Fender | Where do you get the idea that's valid? |
| 00:00.55 | Cyford33 | tht is my or |
| 00:01.15 | Cyford33 | it works with AND & |
| 00:01.29 | [TK]D-Fender | you have LOTS of juk charaters in your line |
| 00:01.30 | Cyford33 | i do not see an example for or |
| 00:01.37 | [TK]D-Fender | Logical operators |
| 00:01.37 | [TK]D-Fender | <PROTECTED> |
| 00:01.53 | [TK]D-Fender | You see a sample for AND. |
| 00:01.59 | [TK]D-Fender | the only difference is the SYMBOL |
| 00:02.03 | Cyford33 | yes |
| 00:02.04 | [TK]D-Fender | That should have been enough |
| 00:02.06 | Cyford33 | and workd |
| 00:02.10 | Cyford33 | or does not |
| 00:02.15 | [TK]D-Fender | And your line is full of JUNK extra braces |
| 00:02.30 | Cyford33 | this works ExecIf($[$[ ${strcredit} < 3 ] & [${strtariff} != 13] ] ]?Set(refill=You should refill your Card.)) |
| 00:02.55 | Cyford33 | <PROTECTED> |
| 00:02.59 | [TK]D-Fender | BROKEN |
| 00:03.09 | [TK]D-Fender | {4 <- BAD |
| 00:03.16 | [TK]D-Fender | [$ <- rather |
| 00:03.34 | [TK]D-Fender | You also don't need to enclose that at all |
| 00:03.35 | Cyford33 | what u mean? |
| 00:03.56 | [TK]D-Fender | <PROTECTED> |
| 00:04.07 | [TK]D-Fender | <PROTECTED> |
| 00:04.26 | Cyford33 | ok well that has been working for months |
| 00:04.32 | [TK]D-Fender | DOLLAR-SIGN BRACE, not BRACE DOLLAR-SIGN |
| 00:04.35 | Cyford33 | <PROTECTED> |
| 00:04.41 | [TK]D-Fender | You can see it break in your CLI output |
| 00:04.52 | Cyford33 | the or broke it |
| 00:04.54 | [TK]D-Fender | All your extra chars there are breaking it |
| 00:04.59 | Cyford33 | <PROTECTED> |
| 00:05.03 | [TK]D-Fender | You are not paying attention to syntax |
| 00:07.50 | [TK]D-Fender | Read it all over again slow |
| 00:08.01 | Cyford33 | $[ $[ ${strcredit} < 3 ] & [${strtariff} != 13] ] |
| 00:08.03 | [TK]D-Fender | no |
| 00:08.28 | [TK]D-Fender | Since when is it valid to shove a sqrare brace FIRST right after the "&"? |
| 00:09.03 | [TK]D-Fender | Nothing in Asterisk starts with the brace like that |
| 00:09.19 | Cyford33 | well it has always worked for me |
| 00:09.32 | [TK]D-Fender | <PROTECTED> |
| 00:09.39 | [TK]D-Fender | obvious breakage |
| 00:09.41 | Cyford33 | <PROTECTED> |
| 00:09.53 | Cyford33 | only breaks when i add or |
| 00:09.54 | [TK]D-Fender | maybe the fact it evlauted to true was LUCXKY for a wile |
| 00:09.55 | [TK]D-Fender | while |
| 00:10.01 | [TK]D-Fender | or at least while you were looking |
| 00:10.11 | Cyford33 | only breaks when i add or |
| 00:10.15 | [TK]D-Fender | but wwe can see junk in the eval guranteeing its broken junk |
| 00:10.25 | Cyford33 | i have no problem fixing it |
| 00:10.31 | Cyford33 | <PROTECTED> |
| 00:10.32 | [TK]D-Fender | [ alone is NEVER valid |
| 00:10.43 | Cyford33 | <PROTECTED> |
| 00:10.44 | [TK]D-Fender | you should know what you are trying to do there. |
| 00:10.49 | [TK]D-Fender | because that isn't valid syntax |
| 00:11.09 | Cyford33 | i am not argueing, just looking for the correct way |
| 00:11.23 | Cyford33 | aww u saying change [ into { |
| 00:12.03 | Cyford33 | i got examples from http://www.voip-info.org/wiki/view/Asterisk+cmd+GotoIf |
| 00:12.07 | Cyford33 | <PROTECTED> |
| 00:12.23 | Cyford33 | <PROTECTED> |
| 00:15.18 | [TK]D-Fender | You started with a SOLO BRACE. |
| 00:15.21 | [TK]D-Fender | wjhy? |
| 00:15.24 | Cyford33 | i searched the whole asterisk wiki for & with no results |
| 00:15.52 | [TK]D-Fender | You've got a broken understanding of expressions as a whole |
| 00:16.05 | [TK]D-Fender | https://wiki.asterisk.org/wiki/display/AST/Expressions |
| 00:16.23 | [TK]D-Fender | Again, where do you see a { alone? |
| 00:16.28 | [TK]D-Fender | [ <- or that? |
| 00:16.44 | [TK]D-Fender | $[] <- EXPRESSION |
| 00:16.59 | [TK]D-Fender | ${} <- variable or function |
| 00:17.05 | [TK]D-Fender | [ ALONE = nothing |
| 00:17.21 | [TK]D-Fender | and you have multiple of these |
| 00:17.41 | [TK]D-Fender | <[TK]D-Fender> callingcard:15] ExecIf("SIP/geils-000008a4", "0 | [10 != 16] ]?Set(refill=You should refill your Card.)") in new stack |
| 00:17.48 | [TK]D-Fender | the pipe isn't what broke there. |
| 00:18.03 | [TK]D-Fender | You can see stuff BEFORE the pip shoing you that your parsing was broken before than point |
| 00:18.05 | Cyford33 | aww |
| 00:18.18 | Cyford33 | <PROTECTED> |
| 00:18.20 | [TK]D-Fender | and after the pip you have a [ as well |
| 00:18.27 | [TK]D-Fender | MULTIPLE $ |
| 00:18.30 | Cyford33 | thats the killer |
| 00:18.59 | [TK]D-Fender | You did it at the start and didn't double-check your work |
| 00:19.08 | Cyford33 | 1 big expression with 3 little ones inside it |
| 00:19.29 | [TK]D-Fender | And those sub expressions separated by | don't NEED to be in $[] |
| 00:20.25 | [TK]D-Fender | $[ ${x} > 5 | "${name}" = "john" ] |
| 00:23.49 | Cyford33 | ExecIf($[ ${strcredit} < 3 ] & $[${strtariff} != 13] ] | $[${strtariff} != 16]?Set(refill=You should refill your Card.)) |
| 00:23.57 | [TK]D-Fender | no |
| 00:24.32 | [TK]D-Fender | ExecIf($[ ${strcredit} < 3 ] <- that will end up witha 1 or 0 and TEHEN hit the & sign OUTSIDE of an expression |
| 00:24.48 | [TK]D-Fender | You aren't following very well. |
| 00:25.08 | [TK]D-Fender | read my sample again closely |
| 00:26.21 | Cyford33 | ExecIf($[ ${strcredit} < 3 & ${strtariff} != 13 | ${strtariff} != 16]?Set(refill=You should refill your Card.)) |
| 00:28.06 | [TK]D-Fender | much better looking |
| 00:28.29 | [TK]D-Fender | Though I'm not 100% sure how it prioritizes the fact you have 2 things going on there |
| 00:28.41 | [TK]D-Fender | if the & happens first, or the | |
| 00:29.00 | [TK]D-Fender | THAT you can put into a sub-expression to be sure |
| 00:32.35 | Cyford33 | yeah no more error, however when a user calls in with tarrif =16 the expression gives a 1 |
| 00:33.02 | Cyford33 | gonna try the and last |
| 00:35.48 | [TK]D-Fender | encasing it would be a gaurantee |
| 00:42.10 | Cyford33 | aww |
| 00:42.15 | Cyford33 | <PROTECTED> |
| 00:43.51 | Cyford33 | i was trying tariff != 13 OR tariff != 16 so if tariff =16 it is still true... i should of been using & and not OR |
| 00:47.23 | Cyford33 | Thanks for the help tk :) |
| 00:47.59 | *** join/#asterisk italorossi (~Adium@179.234.138.92) |
| 00:51.13 | [TK]D-Fender | 32nd times the charm! |
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| 01:00.14 | Cyford33 | no man ur example took 2 tries lol |
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| 01:01.00 | Cyford33 | those links dont discribe anything with & or | |
| 01:01.10 | [TK]D-Fender | And it took you a pile to get to mine ;) |
| 01:01.13 | [TK]D-Fender | Yes, they both do |
| 01:01.21 | [TK]D-Fender | it's right on the one you gave me |
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| 01:01.28 | [TK]D-Fender | I even copy/pasted it |
| 01:01.42 | Cyford33 | i searched the page for & no results |
| 01:01.43 | [TK]D-Fender | <[TK]D-Fender> Read the page |
| 01:01.43 | [TK]D-Fender | <[TK]D-Fender> Logical operators <--- |
| 01:01.43 | [TK]D-Fender | <[TK]D-Fender> expr1 | expr2 (Logical OR) |
| 01:02.19 | Cyford33 | that lines doesnt give a live example, |
| 01:02.30 | [TK]D-Fender | exten => s,n,GotoIf($[ $[ "${FOO}" = "1" ] & $[ "${BAR}" = "2" ] ]?labelTrue:labelFalse) |
| 01:02.32 | [TK]D-Fender | YES THEY DO |
| 01:02.33 | Cyford33 | and shows no brackets |
| 01:03.24 | Cyford33 | aww, i suppose i didnt know what an expression was then lol |
| 01:03.35 | [TK]D-Fender | REALLY broken understanding |
| 01:03.43 | [TK]D-Fender | $[] <- EXPRESSION |
| 01:03.50 | [TK]D-Fender | the crap IN there gets evaluated |
| 01:04.21 | [TK]D-Fender | You can nest expressions in there as well like this or not do that at all |
| 01:04.27 | Cyford33 | u have 2 expressions inside another though |
| 01:04.39 | [TK]D-Fender | Correct. Nothing stopping you |
| 01:04.47 | [TK]D-Fender | but it's typically wasteful |
| 01:04.51 | [TK]D-Fender | Sometimes useful |
| 01:05.12 | [TK]D-Fender | As in the case of wondering what operator precedence is with * |
| 01:05.37 | Cyford33 | if u replace the & with a | u will get error though |
| 01:05.42 | Cyford33 | <PROTECTED> |
| 01:06.28 | Cyford33 | <PROTECTED> |
| 01:07.10 | phix | [TK]D-Fender!!! WIMPy!!!!!!!!!!!!!!!!!!!11111 |
| 01:07.32 | [TK]D-Fender | that loks fine |
| 01:08.25 | Cyford33 | :) |
| 01:10.11 | [TK]D-Fender | tiny bit wasteful, but whatever |
| 01:10.37 | phix | [TK]D-Fender: I still don't have phones ;( |
| 01:10.59 | [TK]D-Fender | phix, Go get phones! |
| 01:11.10 | [TK]D-Fender | <-- PROBLEM SOLVER |
| 01:11.18 | phix | :D |
| 01:11.20 | phix | Cheers |
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| 02:07.13 | janicez | Shoot me |
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| 02:12.38 | [TK]D-Fender | gave up on explosive ballistic weapons long ago |
| 02:36.27 | phix | janicez: pew pew! |
| 02:36.34 | janicez | phix: hi |
| 02:36.41 | janicez | phix: do we know each other? :p |
| 02:37.01 | phix | nope |
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| 03:19.50 | zopsi | In Asterisk 13 I'm having an issue where the caller cannot hear the prompt to say your name and a SLN with size of 0 bytes is placed into priv-callerintros. However, when I answer the phone and press 1 we can both talk to each other fine. It appears it is just before it is connected to me. Any ideas? Any tips? |
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| 03:52.25 | ChannelZ | none of that means a lot to us not knowing what your dialplan is doing. Random guess: make sure you Answer() the channel first |
| 04:23.40 | drmessano | Geesh someone fixed that Moe guys issue |
| 04:24.14 | drmessano | I made dip this time |
| 05:03.07 | cryptic | in it possible to use Monitor to record only one side of the call? |
| 05:09.44 | cmendes0101 | cryptic: https://wiki.asterisk.org/wiki/display/AST/Application_Monitor |
| 05:09.52 | cmendes0101 | options i and o I believe |
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| 05:30.17 | snadge | what would cause an asterisk 1.4 pbx to respond to an incoming call with 407 proxy authentication required? |
| 05:30.36 | snadge | everything looks right. .it permits the ip range that the call is coming from, it shouldn't be trying to challenge it |
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| 05:50.23 | drmessano | Pastebin the peer config |
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| 05:57.48 | ChannelZ | drmessano, which one? Last I knew he was having Skype haxx0r problems or something |
| 05:58.31 | drmessano | I think he finally blacklisted 2/3 of the internet and it included the offending IP |
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| 11:29.21 | Fira_ | Hey guys |
| 11:29.59 | Fira_ | I'm playing around with Asterisk, and suddenly i'm getting consistant segfaults |
| 11:30.03 | Fira_ | On 13.4 |
| 11:30.55 | Fira_ | I'm not sure what I might have changed that caused this, it was working like a charm yesterday, and today as soon as I bridge 3-4 channels together, it segfaults in format handling |
| 11:31.35 | Fira_ | I think I had this point on 13.0.1 at some point aswell |
| 11:31.38 | Fira_ | bug* |
| 11:34.03 | Fira_ | Nailed it, it occurs as soon as i try to bridge 3+ channels in a mixing_bridge... aka. as soon as there's transcoding to do rather than simple passthrough |
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| 11:34.18 | Fira_ | That's problematic for a conferencing app :p |
| 11:34.35 | Fira_ | Definitely ran into this bug a while back on 13.0.1 |
| 11:34.52 | Fira_ | Shall i just JIRA it up ? |
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| 11:39.49 | file | a backtrace is needed to know what is going on - https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace |
| 11:44.31 | Fira_ | Yeah i'll include the backtrace |
| 11:44.40 | Fira_ | basically ast_format_get_name() gets called with a null format |
| 11:44.51 | Fira_ | and when it tries to grab format->name... bam, segfault |
| 11:45.44 | file | then you'd also need to include configuration, console output, and configuration |
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| 11:47.12 | Fira_ | Alright |
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| 12:34.33 | Fira_ | well, opened #25269 concerning the crash, and #25270 for the yesterday issue while i was at it :p |
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| 12:56.03 | Fira_ | lol, nailed the crash |
| 12:56.16 | Fira_ | it happens when you're an idiot like me and enable both VAD and PP_VAD in Speex |
| 12:56.39 | Fira_ | ... oh |
| 12:56.49 | Fira_ | or when you just enable PP_VAD, i guess it's just broken right now |
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| 13:25.11 | ak77 | hello all |
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| 13:32.31 | ak77 | how would one apply changes to pjsip (pjsip.conf or pjsip_wizard.conf or ?) without restarting asterisk? |
| 13:35.16 | file | module reload res_pjsip.so, module reload res_pjsip_config_wizard.so |
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| 13:46.03 | ak77 | file: tnx, but res_pjsip doesn't reload transports and res_pjsip_config_wizard doesn't support reloads. |
| 13:46.39 | file | res_pjsip is unlikely to ever really fully support changing transports on reload |
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| 14:01.07 | ak77 | file: thank you. |
| 14:03.15 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 14:04.42 | ak77 | where can I lookup which method to use to subscribe to ari event using ari-py. how to subscribe to Dialed ? ( to Channel* events one can use client.on_channel_event) |
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| 14:27.23 | *** join/#asterisk acro458 (0cab5efe@gateway/web/freenode/ip.12.171.94.254) |
| 14:29.21 | acro458 | can asterisk make calls itself? |
| 14:30.38 | acro458 | I need a solution where I have software call a POTS phone using voip |
| 14:31.37 | acro458 | It is all going to be local (the POTS phones are even in the same room as the server) |
| 14:32.58 | [TK]D-Fender | sure |
| 14:33.07 | ak77 | acro458: that call has to be triggered (originated) somehow. |
| 14:33.09 | [TK]D-Fender | Though you should make your description clearer |
| 14:33.37 | acro458 | I want to program an application that goes through a bunch of phone numbers and calls them one at a time |
| 14:33.50 | *** join/#asterisk azerus (~badass@unaffiliated/badass) |
| 14:33.57 | [TK]D-Fender | thats's a POTS LINE, not a POTS PHONE |
| 14:34.13 | acro458 | Dont know the difference |
| 14:34.23 | acro458 | Its a phone connected to a pots line |
| 14:34.34 | [TK]D-Fender | at my home I have a LINE from the telco |
| 14:34.36 | ak77 | acro458: you can use ARI for that (https://github.com/asterisk/ari-py/blob/master/examples/originate_example.py) |
| 14:34.50 | [TK]D-Fender | I own a PHONE that I plug into that line |
| 14:35.09 | acro458 | What type of equipment do i need? |
| 14:35.17 | [TK]D-Fender | You can connect Asterisk to a LINE via an interface so that it ACTS line a phone and can dial out and get calls in. |
| 14:35.18 | *** join/#asterisk bkruse (~Adium@64.89.97.113) |
| 14:35.37 | [TK]D-Fender | Or you can get an interface to plug a PHONE into so that the phone can talk to Asterisk |
| 14:35.50 | [TK]D-Fender | Totally differnt things |
| 14:35.59 | [TK]D-Fender | You want to us a LINE that you have to call out? |
| 14:36.17 | acro458 | I want to program an application that goes through a bunch of phone numbers and calls them one at a time |
| 14:36.26 | [TK]D-Fender | And you have a LINE currently? |
| 14:36.28 | [TK]D-Fender | What kind? |
| 14:36.59 | acro458 | All of the phones are connected using POTS lines |
| 14:37.05 | [TK]D-Fender | huh? |
| 14:37.31 | acro458 | I do not know what you mean by "LINE" |
| 14:37.38 | [TK]D-Fender | You the WIRE in your wall |
| 14:37.45 | [TK]D-Fender | you plug a phone *INTo* a line |
| 14:38.34 | acro458 | The phones have their own local and 800 numbers |
| 14:38.38 | [TK]D-Fender | You don't seem to understand the different from the dumb piece of plastic and metal you hold to you head and talk to ... from the WIRE that a telephone company runs to your home and gives you SERVICE |
| 14:38.42 | acro458 | I need to dial the phone numbers |
| 14:38.45 | [TK]D-Fender | PHONES don't have numbers. |
| 14:38.49 | [TK]D-Fender | LINES have numbers |
| 14:39.09 | acro458 | This seems irrelevant |
| 14:39.16 | [TK]D-Fender | No, because you are not being clear |
| 14:39.26 | acro458 | The lines are connected to the phones. THey are not going to bounce around |
| 14:39.39 | [TK]D-Fender | and since you can't get your wording right we can't tell for sure that you don't means something quite the opposite of what you actually want |
| 14:39.50 | acro458 | Right now, I dial an 800 phone number, and a phone rings |
| 14:39.55 | *** join/#asterisk rmudgett (rmudgett@nat/digium/x-fzhjbmevjvvcvkpd) |
| 14:39.56 | acro458 | I want a server to call |
| 14:40.02 | [TK]D-Fender | that call comes in on a LINE. |
| 14:40.05 | acro458 | instead of me calling on my cell phone |
| 14:40.29 | [TK]D-Fender | so you have a phone line provided by a telco already? |
| 14:40.56 | acro458 | Yes |
| 14:41.01 | [TK]D-Fender | what KIND of line? |
| 14:41.01 | acro458 | the phones all work |
| 14:41.17 | acro458 | What are my options... |
| 14:41.31 | acro458 | POTS line...4 pins... |
| 14:41.41 | acro458 | from AT&T |
| 14:41.46 | [TK]D-Fender | ok, so you need some kind of FXO interface to call out with |
| 14:42.10 | [TK]D-Fender | three are PCI cards, USB interfaces, and VoIP ATA's that Asterisk can use to let you use your analog line |
| 14:42.36 | *** join/#asterisk gustopn (~gustik@ec2-52-11-23-173.us-west-2.compute.amazonaws.com) |
| 14:42.39 | [TK]D-Fender | How many separate lines do you intend to use? |
| 14:43.03 | acro458 | How many lines am i calling? Hundreds |
| 14:43.08 | [TK]D-Fender | no |
| 14:43.11 | [TK]D-Fender | you call NUMBNERSS |
| 14:43.17 | gustopn | hey, I just added a shellscript to /etc/shells putting a user directly into asterisk -r , does someone use this as well? |
| 14:43.32 | gustopn | and what are your ideas about how "secure" it is, ha? |
| 14:43.32 | [TK]D-Fender | how many WIRES do you have connected to the TELEPHONE COMPANY |
| 14:43.52 | *** join/#asterisk azerus (~badass@unaffiliated/badass) |
| 14:44.09 | acro458 | I have no idea. We have a phone system |
| 14:44.16 | gustopn | LOL |
| 14:44.20 | acro458 | No clue how it works. Dont really care |
| 14:44.20 | gustopn | a phone system |
| 14:44.29 | acro458 | "phoneswitch" |
| 14:44.31 | gustopn | so why do you need asterisk_ |
| 14:44.34 | gustopn | ha? |
| 14:44.35 | [TK]D-Fender | you need to determine how many you want to use then so you can pick a proper interface to support that many |
| 14:45.11 | gustopn | what do you ean with interface? like hardware or chan_ driver? |
| 14:45.16 | doop | acro458: now might be the time to go look at it with your eyes. maybe snap a photo of what it looks like. |
| 14:45.30 | acro458 | Do you know what Twilio is? |
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| 14:45.53 | [TK]D-Fender | yes, an ITSP |
| 14:46.18 | acro458 | With that, I can dial a phone number and get a recording of the conversaion |
| 14:46.27 | acro458 | I am trying to do the exact same thing |
| 14:46.36 | doop | easy enough to do |
| 14:46.36 | acro458 | But not through them |
| 14:46.45 | [TK]D-Fender | acro458: All fine |
| 14:47.15 | acro458 | I can dial 1 phone number per second with them |
| 14:47.40 | acro458 | Cant remember how many concurrent calls I can have, but it was over 30, which is ok with me |
| 14:48.10 | [TK]D-Fender | acro458: so you have some number of analog lines.... AND service with Twilio. |
| 14:48.34 | acro458 | No, we used Twilio to test with |
| 14:48.38 | acro458 | we just used the API |
| 14:48.56 | acro458 | We just used it to make the phone calls and record the conversation |
| 14:49.03 | [TK]D-Fender | ok, so taht gave you "inspiration" |
| 14:49.11 | acro458 | It was a failed test |
| 14:49.20 | acro458 | It didnt do exactly what we wanted |
| 14:49.21 | [TK]D-Fender | So you want to pick up a phone in your building, and have Asterisk call out and record the call? |
| 14:49.53 | acro458 | I want asterisk to call a phone number, and when someone picks up, record the call. |
| 14:50.17 | WIMPy | Do you want to talk as well? |
| 14:50.18 | [TK]D-Fender | more than that you want to have aPHONE in your hand connectweds so YOU are talking to the other party, no? |
| 14:50.31 | acro458 | No, one way |
| 14:50.45 | acro458 | HOWEVER, I may want asterisk to "talk" back |
| 14:50.49 | [TK]D-Fender | because wehn you say "asterisk call out" alone.. that means the COMPUTER is ta"talking" with the other end. so far "you" are not in that description. |
| 14:50.56 | WIMPy | Or to start at the beginning: Do you want to record calls as currently made with your existing phone system or do you want to use your PC to make calls? |
| 14:50.56 | acro458 | Correect |
| 14:50.57 | [TK]D-Fender | ok, be clearer on this |
| 14:51.16 | acro458 | You are looking more into it than required |
| 14:51.33 | acro458 | ALL I want is it to call a phone number |
| 14:51.36 | doop | acro458: are you robo-calling with recordings or something |
| 14:51.40 | [TK]D-Fender | acro458: So lewts start over. You want your server to make AUTOMATED calls, and then interact with the user once they answer? So a ROBO-CALLER? |
| 14:52.07 | acro458 | something like that, yes |
| 14:52.14 | [TK]D-Fender | acro458You are looking more into it than required <- you are using vagure wording and it makes it hard to be sure what you want. Stop being so vague |
| 14:52.31 | acro458 | I dont have a clue what words to actually use... |
| 14:52.41 | acro458 | I never dealt with phone systems in my life |
| 14:52.42 | [TK]D-Fender | So far this is all doable |
| 14:53.38 | [TK]D-Fender | acro458: saying "I want my server to make automated call-outs and them interact with the person once they answer". Those are the words. |
| 14:54.13 | [TK]D-Fender | its makes it clear that you are not having a conversation between 2 parties. It's Server > Callee. |
| 14:54.30 | [TK]D-Fender | acro458: So yes, you can do this |
| 14:54.40 | acro458 | what equipment do I need? |
| 14:54.42 | [TK]D-Fender | Now you need to confirm exactly what you are going to call USING. |
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| 14:55.00 | [TK]D-Fender | You said PTS lines. And are currently unsure of HOW MANY you intend to use |
| 14:55.15 | [TK]D-Fender | Figure this number out so we can recommend appropriate hardware for you |
| 14:55.32 | WIMPy | Do you want to use your existing lines? You could use an ITSP as well. |
| 14:55.48 | acro458 | I have no idea how it works. I figured the server could make the calls. How it ends up being translated to a POTS line, I have no idea |
| 14:55.53 | WIMPy | And if you want to use your lines, how many simultaneous calls do you want to make? |
| 14:56.07 | acro458 | 30 simultaneous calls |
| 14:56.34 | WIMPy | No way you're going to do that with POTS. |
| 14:56.37 | doop | acro458 sounds like you're better off doing the ITSP thing unless you have 30 POTS lines coming in |
| 14:56.39 | [TK]D-Fender | acro458: You said "POTS phone" |
| 14:56.41 | doop | which seems unlikely |
| 14:56.50 | [TK]D-Fender | you brought up the ambiguity that you had PHONE LINES there you wanted to use |
| 14:56.56 | WIMPy | For non-US that would be one PRI or using VOIP. |
| 14:57.01 | acro458 | Server -> Ethernet -> ????? -> POTS line -> Phone I do not know what the ??? is that I need to make it work |
| 14:57.04 | [TK]D-Fender | Seems you are still not being clear bout HOW you want your server to be able to call out |
| 14:57.17 | [TK]D-Fender | I have a dumb phone line at home |
| 14:57.18 | doop | acro458: that ??? is an ITSP |
| 14:57.21 | [TK]D-Fender | I *could* use that |
| 14:57.29 | [TK]D-Fender | I have a cell phoe. I *could* use that. |
| 14:57.39 | WIMPy | ~itsp |
| 14:57.39 | 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. |
| 14:57.45 | [TK]D-Fender | There are ITSP's. Tiwilo is just ONE. I c*could* use one of those. |
| 14:57.52 | acro458 | That is my question. HOW do I get the server to call out |
| 14:57.58 | [TK]D-Fender | acoI just listed ways |
| 14:58.04 | [TK]D-Fender | acro458: I just listed ways |
| 14:58.27 | [TK]D-Fender | I can get piece of hardware similar to a MODEM and plug it into a PHONE LINE. |
| 14:58.35 | acro458 | But it cant do 30 calls concurrently, can it? |
| 14:58.45 | [TK]D-Fender | I could have my server talk to my cell-phone and use THAT things service to call out |
| 14:58.55 | acro458 | I would need something with 30 lines (out) ? |
| 14:59.05 | [TK]D-Fender | I could use an INTERNET PHONE COMPANY service. Twilio is one these kinds of things |
| 14:59.13 | [TK]D-Fender | DO you need 30 simultaneous? |
| 14:59.17 | acro458 | Yes |
| 14:59.33 | [TK]D-Fender | No, a boring analog line cannot do 30 simultaneous. Or even 2 |
| 15:00.04 | [TK]D-Fender | then you need either a large digital phone link to your telco (T1), which is expensive. Or you could use an ITSP like Twilio |
| 15:01.20 | WIMPy | One T1 isn't even enough. 1x E1 or 2x T1. |
| 15:02.29 | [TK]D-Fender | Yes, technically for 30 |
| 15:03.02 | [TK]D-Fender | WIMPy: I suppose if your telco supported something like G.726 on it you could fit more, but I've never seen any that do. |
| 15:04.45 | WIMPy | That's not going to happen on a PRI. G.711 or G.722. You can use less but the number of channels stays te same. |
| 15:05.12 | WIMPy | There is equipment that splits up channels, but I'm pretty sure no telco offers that. |
| 15:05.18 | Cyford33 | unless he was connecting 30 pot phones via ata's and using sip to trunk out |
| 15:05.20 | [TK]D-Fender | WIMPy: PRI doesn't support &.26? Only possible on CAS? |
| 15:05.25 | [TK]D-Fender | G.726 |
| 15:06.07 | Cyford33 | he is probably using ata's to asterisk and need a sip provider |
| 15:06.34 | [TK]D-Fender | Cyford33: No. |
| 15:06.39 | [TK]D-Fender | Cythere is no PHONE in this equation |
| 15:06.40 | WIMPy | Well, technically it doesn't really care, you get a bit transparent channel with 64kbps. It's only of interest if the contents have to be decoded. |
| 15:06.54 | [TK]D-Fender | Cyford33: He's setting up a robo-calling system |
| 15:07.08 | [TK]D-Fender | CyAnd it's all about how he plans to get to the PSTN |
| 15:07.16 | Cyford33 | aww |
| 15:07.40 | sekil | telcos which give PRI/ISDN in Europe mostly enable only PCMA...there are stories that some can do G.722 |
| 15:08.28 | [TK]D-Fender | sekil: I've heard many support G.722, esp as it's the same bitrate |
| 15:08.47 | [TK]D-Fender | sekil: But I was wondering if you could squeeze more out of PRI using G.726 at half instead. |
| 15:08.50 | WIMPy | G.722 has been part of the spec right from the beginning. But indeed there are telcos where you have to tell the that you want it enabled. |
| 15:09.03 | sekil | well I have yet to see one |
| 15:09.15 | sekil | it is true about bitrate..shouldn't be a problem |
| 15:09.31 | WIMPy | Some PBXs do half channels with G.726 for DECT bases. |
| 15:09.40 | sekil | but everything is moving to voip anyways |
| 15:11.28 | sekil | btw..not sure if libpri supports g.722 over pri/e1 |
| 15:11.55 | WIMPy | It doesn't. |
| 15:12.12 | WIMPy | Neither do any of the other channeltypes. |
| 15:12.39 | Cyford33 | doesnt pri voice take 1 channel no matter the codec? |
| 15:13.07 | sekil | Cyford33: two actually...one for sync..and one for signalling |
| 15:13.12 | WIMPy | Yes. That's the basic idea. |
| 15:13.21 | WIMPy | But there are ... deviations. |
| 15:14.06 | Cyford33 | but compressing a codec doesnt save u bandwith does it? |
| 15:14.48 | WIMPy | Well, there are not so standard ways to do it. |
| 15:15.05 | Cyford33 | oic |
| 15:15.31 | WIMPy | GSM bases also use quarter channels. |
| 15:15.55 | WIMPy | But that's not rellt a PRI. |
| 15:16.28 | Cyford33 | thats a 1.5 pipe right |
| 15:16.45 | WIMPy | A what? |
| 15:16.48 | *** join/#asterisk coppice (~chatzilla@123203240102.ctinets.com) |
| 15:16.57 | *** join/#asterisk pchero_work (~pchero@109.70.54.56) |
| 15:17.26 | Cyford33 | alot of isp here sell a 1.5mb bandwith pipe they call t1's but they are not channel driven |
| 15:17.30 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 15:17.48 | Cyford33 | and not 1.544 |
| 15:18.26 | WIMPy | No, it's still E1, but not PRI. |
| 15:18.45 | [TK]D-Fender | \o/ CAS |
| 15:19.09 | WIMPy | Surely not. |
| 15:20.21 | Cyford33 | e1's have 24 channels correct? |
| 15:20.46 | [TK]D-Fender | 31 |
| 15:20.50 | WIMPy | 32 timeslots with 31 usable channels. |
| 15:21.07 | sekil | e1 is just 2048 kbit/s transport |
| 15:21.20 | Cyford33 | then i think of t1's lol |
| 15:21.27 | Cyford33 | <PROTECTED> |
| 15:21.41 | sekil | it can be used as framed or unframed |
| 15:21.42 | [TK]D-Fender | They aren't |
| 15:21.43 | WIMPy | No. T1 are similar to J1. |
| 15:21.56 | [TK]D-Fender | First clue is they aren't the same name. |
| 15:22.25 | Cyford33 | By definition, a T1 provides 1.544 Mbps, full duplex. (The equivalent European specification is the E1, which provides 2.048 Mbps.) However, it is divided into 24 channels, and they can be aggregated in any number, so you can get a "partial" T1 in many different bandwidths |
| 15:23.01 | coppice | [TK]D-Fender: J1 isn't the same name as T1M, yet they are the same |
| 15:23.07 | [TK]D-Fender | ~isdn |
| 15:23.07 | infobot | i heard isdn is (Integrated Services Digital Network) This is a digital line that is often used to connect to the Internet. It generally come in two flavors: one is a 56 Kbps version, which in actuality only uses half of the ISDN line's bandwidth; the other is the 128 Kbps version, which uses both the 56 Kbps channels on the line. However, that's only 112 Kbps--the other 16 Kbps are an 8 Kbps back channel of each line. "It Still Does Nothing" |
| 15:23.16 | [TK]D-Fender | ~T1 |
| 15:23.16 | infobot | [~T1] T1 is the basic digital telephony circuit used in North America. T1 runs at 1.544 Mbps. It can be an unstructured channel for data. It can be channelized, to provide 24 time slots of voice or data, each of 64kbps. Time slot 24 is used for D-Chan when used with PRI signalling. |
| 15:23.20 | [TK]D-Fender | ~J1 |
| 15:23.21 | Cyford33 | yes usa have many differnt names for what the rest of the word calls things |
| 15:23.30 | [TK]D-Fender | hrm, I never bothered with that one... |
| 15:23.33 | WIMPy | They are not exactely the same. |
| 15:23.57 | coppice | which are not exactly the same? |
| 15:24.09 | WIMPy | Wow. That is a really bad explanation of ISDN. |
| 15:24.25 | [TK]D-Fender | yes, a bit off for sure |
| 15:24.52 | WIMPy | T1 and J1 are not exactely the same. |
| 15:25.08 | coppice | T1M and J1 are exactly the same |
| 15:25.28 | WIMPy | Never heard of T1M. |
| 15:25.56 | Cyford33 | oc1 and t1 ARE Same |
| 15:26.14 | coppice | T1 modified. Its what people mostly call J1 now, although its still mostly called T1M in Taiwan |
| 15:27.36 | mirela666 | as there is a saying in my country: "lets not mix frogs and grandmothers now" :D |
| 15:28.18 | coppice | Cyford33: where are OC1 and T1 the same? |
| 15:28.24 | sekil | Cyford33: OC-1 is not T1 |
| 15:29.02 | sekil | Cyford33: you can compare OC-x and STM-x |
| 15:29.49 | Cyford33 | ocx is the bandwidth of tx |
| 15:30.15 | coppice | OC-1 is 52Mbps. T1 is 1.5Mbps |
| 15:30.16 | Cyford33 | opps i mean dx |
| 15:30.29 | Cyford33 | d1 = t1 d3 - t3 |
| 15:30.45 | Cyford33 | d1 = t1 d3 = t3 |
| 15:30.48 | WIMPy | OC/STM neither mixes well with Tx not Ex. |
| 15:31.25 | Cyford33 | <PROTECTED> |
| 15:31.28 | WIMPy | DS1 |
| 15:32.29 | Cyford33 | T1 and T3 corresponding to DS1 and DS3, respectively |
| 15:33.09 | WIMPy | yes |
| 15:33.16 | coppice | does the US ever actually use T2? |
| 15:33.45 | WIMPy | I think E2 have never been seen in the wild, either. |
| 15:33.57 | coppice | E2 used to be widely used |
| 15:34.16 | coppice | At least they made a lot of the E2 ports we designed |
| 15:34.19 | WIMPy | Where? for what? |
| 15:34.53 | coppice | exchange to exchange links in metro areas in the 80s |
| 15:35.22 | coppice | we put a lot of E2 in the city of london for financial circuits |
| 15:35.26 | WIMPy | I have only ever seen E1 or E3. |
| 15:36.00 | *** join/#asterisk hackeron (~hackeron@gentoo/user/hackeron) |
| 15:36.03 | coppice | i doubt there is much E2 today, but I don't remember ever seeing T2 stuff |
| 15:36.06 | sekil | there were E2 links here |
| 15:36.20 | sekil | and of source nxSTM-1 |
| 15:37.50 | sekil | for example...ISP needs many PRIs for dial-in ....so telco brings up fiber optics and couple of muxes to give channelized STM-1 |
| 15:38.31 | sekil | but that all changed with IP/L2TP tunneling |
| 15:38.40 | WIMPy | Yes, I remember those. |
| 15:40.16 | coppice | all those E1 and E2 links we used to make for exchange to exchange use, and now I have 2 gigabit fibres at home |
| 15:42.23 | sekil | coppice: comes from living near seaside |
| 15:42.57 | coppice | sekil: do we need more bits near the sea? |
| 15:43.24 | sekil | coppice: not much cabling needed..straight from the sea :) |
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| 15:44.38 | coppice | sekil: I believe the fibre actually runs over the sea through the deck of this https://encrypted-tbn0.gstatic.com/images?q=tbn:ANd9GcTVYC1ZPGhST756xbn6kIZFn6_Yk8xp_mv2vxpkUz_So-IQLR64Gw |
| 15:50.09 | Cyford33 | In order for the hungry man to dish out more bytes.. he has to to be givin more bits |
| 15:50.45 | Cuzner | https://www.youtube.com/watch?v=oLERL4_SveI |
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| 15:57.44 | Cyford33 | if it last that long, an does itsjob good.. i would keep it lol.. todays machines are mostlly crap and requires repair or maintance atleast every 3 years |
| 16:03.43 | Cuzner | oh crap, i didn't even mean to paste that into this channel, sorry boys... fun story though. |
| 16:12.22 | *** join/#asterisk superscrat (asanders@nat/digium/x-pmfbpnclnyqobxay) |
| 16:21.28 | *** part/#asterisk cheche (~cheche@47.Red-83-60-7.dynamicIP.rima-tde.net) |
| 16:37.05 | *** join/#asterisk dhawani (6a33848a@gateway/web/freenode/ip.106.51.132.138) |
| 16:37.44 | dhawani | hi, can anyone help me with sangoma usbfxo to get caller id? |
| 16:38.40 | dhawani | i am getting following in logs |
| 16:38.44 | dhawani | [2015-07-21 21:58:34] VERBOSE[9224][C-00000002] sig_analog.c: Starting simple switch on 'DAHDI/2-1' [2015-07-21 21:58:34] DEBUG[9224][C-00000002] sig_analog.c: Receiving DTMF cid on channel DAHDI/2-1 [2015-07-21 21:58:34] DEBUG[9224][C-00000002] sig_analog.c: analog_exception 2 [2015-07-21 21:58:34] DEBUG[9224][C-00000002] sig_analog.c: Exception on 23, channel 2 [2015-07-21 21:58:34] DEBUG[9224][C-00000002] sig_analog.c: __analog |
| 16:39.17 | *** join/#asterisk zogm (~zogm@208.167.254.26) |
| 16:41.23 | zogm | has anyone had any luck virtualizing asterisk servers w primary use being meetme conferences ? (100+ simultaneous ongoing calls) |
| 16:43.03 | drmessano | zogm, plenty of use cases |
| 16:43.07 | drmessano | Whats the concern? |
| 16:43.11 | drmessano | oh |
| 16:43.15 | zogm | dahdi timings and performance in general |
| 16:43.16 | drmessano | Dont use MeetMe |
| 16:43.22 | drmessano | MeetMe is dead |
| 16:43.30 | zogm | I wish it was lol |
| 16:43.32 | drmessano | Use a modern Asterisk |
| 16:43.33 | zogm | not in my world (vicidial) |
| 16:43.39 | drmessano | and confbridge |
| 16:43.57 | drmessano | No one is continuing to fool with MeetMe and virtualizing DAHDI |
| 16:45.39 | dhawani | how can i tweak asterisk to wait till DTMF for CID is complete |
| 16:45.40 | dhawani | ? |
| 16:45.57 | drmessano | Uh, what? |
| 16:47.04 | zogm | DTMF for CID??? 0_o |
| 16:47.28 | drmessano | Yeah thats a new one |
| 16:47.31 | dhawani | i m getting incoming call but i m not getting caller id |
| 16:47.39 | zogm | guessing it's some robocalling setup ? |
| 16:47.42 | zogm | oh |
| 16:47.50 | zogm | so what's DTMF got to do with it? |
| 16:48.08 | drmessano | Asterisk doesnt "wait" for CID... it's either passed or not |
| 16:48.15 | drmessano | If its not passing, call your provider |
| 16:48.31 | zogm | ^ 100% |
| 16:49.01 | zogm | or if you have a softswitch proxying ur calls back and forth for billing and what not - check that it's not stripping CIDs |
| 16:49.23 | dhawani | i see in logs that caller id is coming late |
| 16:49.24 | dhawani | [2015-07-21 21:58:34] DEBUG[9224][C-00000002] sig_analog.c: Begin DTMF digit: 0x39 '9' on DAHDI/2-1 [2015-07-21 21:58:34] DEBUG[9224][C-00000002] chan_dahdi.c: Begin DTMF digit: 0x39 '9' on DAHDI/2-1 [2015-07-21 21:58:34] DTMF[9224][C-00000002] channel.c: DTMF begin '9' received on DAHDI/2-1 [2015-07-21 21:58:34] DTMF[9224][C-00000002] channel.c: DTMF begin passthrough '9' on DAHDI/2-1 |
| 16:49.31 | drmessano | ~pb |
| 16:49.31 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 16:49.34 | dhawani | between second ring |
| 16:50.06 | drmessano | What does a DTMF 9 have to do with CID? |
| 16:51.13 | drmessano | CID is sent between the first and second ring, standard |
| 16:51.15 | drmessano | It's not late |
| 16:52.23 | dhawani | please look this http://pastebin.com/5uSRwvZg |
| 16:52.44 | drmessano | [2015-07-21 21:58:34] DEBUG[9224][C-00000002] sig_analog.c: CID got string '' <----- |
| 16:52.55 | drmessano | [2015-07-21 21:58:34] DEBUG[9224][C-00000002] sig_analog.c: CID got string '' |
| 16:52.55 | drmessano | [2015-07-21 21:58:34] DEBUG[9224][C-00000002] callerid.c: No cid detected |
| 16:52.55 | drmessano | [2015-07-21 21:58:34] DEBUG[9224][C-00000002] sig_analog.c: CID is '', flags 8 |
| 16:53.03 | drmessano | Boom |
| 16:53.19 | dhawani | ya but in same logs u see them coming as DTMF tones |
| 16:53.24 | dhawani | am i missing some thing? |
| 16:56.25 | zogm | CID isn't being passed |
| 16:56.46 | zogm | if u referring to this |
| 16:56.48 | zogm | [2015-07-21 21:58:34] DEBUG[9224][C-00000002] sig_analog.c: Receiving DTMF cid on channel DAHDI/2-1 |
| 16:57.07 | zogm | when u say "DTMF cid" and that it's coming in late, that's not it at all |
| 16:57.32 | dhawani | [2015-07-21 21:58:34] DEBUG[9224][C-00000002] sig_analog.c: Begin DTMF digit: 0x39 '9' on DAHDI/2-1 <-- this is first digit of number i m calling from |
| 16:58.31 | zogm | what device are you calling in from (cell phone/skype/another PBX system/landline etc..)? |
| 16:59.01 | dhawani | cell phone |
| 16:59.57 | zogm | and first digit of ur cellphone number is a 9? |
| 17:00.10 | dhawani | btw, what should be cidstart and cidsignalling values in this case? |
| 17:00.11 | dhawani | yes |
| 17:01.14 | zogm | what's ur asterisk version |
| 17:01.43 | dhawani | 13.4.0, |
| 17:02.07 | zogm | and what telephony hardware r u using ? (Sangoma cards or digium or what not and model #s) |
| 17:02.15 | zogm | basically walk us through ur setup |
| 17:02.55 | dhawani | sangoma usbfxo u100 |
| 17:03.22 | dhawani | dahdi -> 2.5.0.2 |
| 17:03.44 | dhawani | analog line is connected to this fxo |
| 17:04.55 | *** join/#asterisk midsandhighs (~midsandhi@li296-126.members.linode.com) |
| 17:05.40 | dhawani | and chan_dahdi config is http://pastebin.com/szTDCgte |
| 17:07.39 | zogm | k |
| 17:08.41 | zogm | u r in india? |
| 17:08.46 | dhawani | yes |
| 17:09.11 | zogm | and calls are coming in from where and to which country's phone number ? |
| 17:10.29 | dhawani | from my cell phone gsm (Tata Docomo) to analog line (Airtel) connected to usb fxo |
| 17:11.24 | zogm | which country are your DIDs assigned to ? Are those US DIDs ? And are you calling from a US number ? |
| 17:11.48 | dhawani | Nothing right now |
| 17:11.55 | dhawani | now US config right now |
| 17:13.23 | zogm | uhmmm |
| 17:13.27 | zogm | could u clairfy |
| 17:13.38 | zogm | so US DID and calling in from a US phone number? |
| 17:15.44 | *** join/#asterisk tdonahue (~tdonahue@vonmail.vonworldwide.com) |
| 17:15.45 | dhawani | no indian celphone/gsm calling to pstn line |
| 17:15.48 | tdonahue | hi all |
| 17:18.13 | zogm | ok, what's the full caller ID u r dialing from (u can replace some of the digits, I just need to know the proper format) |
| 17:19.38 | dhawani | 9591XXX4X0 |
| 17:20.16 | tdonahue | we are having an issue with call distribution in our call center, we have all our queues setup with random call distribution but are not gett |
| 17:21.32 | tdonahue | * but are seeing the same agent get bunches of calls (i.e. 3 calls to agent A then 1 call to agent B then 2 calls to agent 007( |
| 17:22.17 | tdonahue | is there any way to have asterisk distribute calls in a round robin across multiple queues |
| 17:22.28 | *** join/#asterisk superscrat (asanders@nat/digium/x-bprrkoubdbzywmsn) |
| 17:24.46 | *** join/#asterisk lvlinux (~ruel@unaffiliated/lvlinux) |
| 17:28.47 | zogm | dhawani, r u sure they are all Xs and not Ns ? |
| 17:29.02 | zogm | can u give me an actual caller ID example, not ur asterisk dialplan entry |
| 17:29.37 | dhawani | 9591N00N40 |
| 17:30.07 | dhawani | so now only one digit is missing fromast_set_flag |
| 17:30.10 | zogm | (>á) |
| 17:31.12 | dhawani | so now only one digit is missing from my actual number |
| 17:31.17 | dhawani | * |
| 17:31.22 | zogm | * is a digit??? |
| 17:31.31 | zogm | -_- |
| 17:32.26 | dhawani | no ;) |
| 17:33.01 | *** join/#asterisk superscrat (asanders@nat/digium/x-xvregqrpphztqhdu) |
| 17:44.54 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 17:46.17 | dhawani | @zogm ? |
| 17:51.06 | Milenco | tdonahue, yes you can define your own queue strategy |
| 17:51.17 | zogm | u rn't answering my question nor are u providing me w any details and I don't have the time to play 100 questions, sry |
| 17:51.18 | Milenco | check out voip-info.org on queues, its all explained there |
| 17:52.23 | dhawani | i gave my entire number |
| 17:52.35 | dhawani | and the pastebin also has my full number in DTMF |
| 17:58.51 | *** join/#asterisk carrar (~tim@troutlake.net) |
| 17:58.59 | carrar | moof |
| 18:02.11 | Milenco | foom |
| 18:02.16 | *** join/#asterisk sparetire_ (~sparetire@unaffiliated/sparetire) |
| 18:11.15 | *** join/#asterisk hfp (~hfp@70.52.83.163) |
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| 18:16.21 | tdonahue | Milenco, do you have a link to that page? I can't seem to find anything on custom strategies on the pages i've found |
| 18:16.53 | Milenco | sorry i mispoke. you can change the strategy to another one, as random is..pretty random |
| 18:17.33 | Milenco | http://www.voip-info.org/wiki/view/Asterisk+call+queues was what i meant |
| 18:18.11 | tdonahue | yeah, the problem i seem to be having is the system seems to track each queue individually, which would be fine if we had a bunch of members in a couple queues but we have the opposite situation |
| 18:18.27 | Milenco | hmm, indeed |
| 18:18.52 | Milenco | shared_lastcall is the only setting i know for multiple queues |
| 18:19.01 | Milenco | but thats for the wrapuptime |
| 18:19.02 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
| 18:19.45 | *** join/#asterisk hfp (~hfp@65.93.185.19) |
| 18:20.52 | Milenco | this might sort of what you'd be looking for tdonahue: https://issues.asterisk.org/jira/browse/ASTERISK-17732 |
| 18:21.01 | tdonahue | "shared_lastcall will make the lastcall and calls received be the same in members logged in more than one queue." hmm... if it does keep the lastcall value synced across the queues, then that might work for me |
| 18:21.27 | *** join/#asterisk mjordan (mjordan@nat/digium/x-zcwyuvqspdmyuklo) |
| 18:21.27 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 18:21.36 | Milenco | yes but i'm not sure if it would affect the strategy, think not |
| 18:21.42 | Milenco | but be free to test it and let me know |
| 18:22.33 | *** join/#asterisk babak (uid19622@gateway/web/irccloud.com/x-aydrvklnafbqvyru) |
| 18:22.34 | Milenco | as for that ticket, i dont think queue groups have been introduced into normal Asterisk, but you can probably pull that code from any of these links and compile yourself, with some work.. |
| 18:22.37 | tdonahue | i'm hoping if the lastcall is kept consistent across the queues, i can use the leastrecent strategy, which is what our call center manager wanted anyway |
| 18:23.03 | Milenco | yeah leastrecent here as well, but all our agents are only in a single queue |
| 18:23.26 | tdonahue | if only we were that simple here :-) |
| 18:23.36 | Milenco | although we are in process of introducing multiple queues because a recently aquired callcenter has a special VIP line |
| 18:23.46 | tdonahue | yeah, that diff is against 1.8, so it would probably be a pain to get to apply to 11.whatever we are running |
| 18:24.03 | Milenco | still running 1.8 :< |
| 18:24.16 | *** join/#asterisk hfp_ (~hfp@70.52.83.163) |
| 18:24.18 | Milenco | but i'm hoping to upgrade to 11 soon, since Debian Jessie ships 11 by default now |
| 18:36.09 | Cuzner | at least you're not still running 1.4s :P |
| 18:38.17 | *** join/#asterisk hfp (~hfp@MTRLPQ0736W-LP140-04-1176426031.dsl.bell.ca) |
| 18:43.41 | Milenco | lol true that :P |
| 18:49.56 | *** join/#asterisk happy-dude (uid62780@gateway/web/irccloud.com/x-clahmdrrmgptznoe) |
| 18:56.08 | *** join/#asterisk tzafrir (~tzafrir@bzq-179-40-172.cust.bezeqint.net) |
| 18:58.39 | *** join/#asterisk HeN (uid3747@gateway/web/irccloud.com/x-smnukmtrtoakaopr) |
| 19:03.47 | *** join/#asterisk xochilpili (~xochilpil@unaffiliated/xochilpili) |
| 19:03.49 | xochilpili | hi all |
| 19:04.22 | xochilpili | is it possible to delete all family from database ; something like database del AMPUSER 100/* ? |
| 19:08.40 | xochilpili | found it! thanks |
| 19:11.03 | drmessano | Dont delete your family |
| 19:11.07 | drmessano | Love them |
| 19:11.09 | drmessano | Work with them |
| 19:15.46 | *** join/#asterisk brad_mssw (~brad@66.129.88.50) |
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| 19:38.40 | *** join/#asterisk monsterco (~monsterco@70.50.209.119) |
| 19:39.33 | monsterco | any recommendation for simple but opensource autodialers for Asterisk? I would like to be able to pump 2000 - 3000 calls and the ability to send calls back to reception - of course detection of voicemail would be nice. I am looking for something I can install on same asterisk server that I have already |
| 19:40.18 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 19:40.32 | zogm | vicidial |
| 19:40.40 | zogm | or goautodial or any of those |
| 19:40.48 | zogm | simple ? not so much, very cumbersome |
| 19:41.10 | zogm | and requires a lot of tuning, but when setup right - it gets the job done |
| 19:48.58 | *** join/#asterisk josecapurro (~jcapurro@190.52.189.6) |
| 19:50.54 | monsterco | vicidial was nasty last time I checked it in terms of configs etc |
| 19:51.21 | monsterco | is goautodial same? I see they only have an .iso version : |
| 19:51.23 | monsterco | :( |
| 19:52.16 | zogm | when is the last time u tried vicidial? |
| 19:52.23 | zogm | they have a pre-built iso vicibox |
| 19:53.52 | Cuzner | equires: The server elements require an Asterisk server(we support Asterisk versions 1.2, 1.4 and 1.8[1.4 is preferred]) |
| 19:54.33 | xochilpili | how can i change the language of a user ? |
| 20:13.34 | *** join/#asterisk Micc (~Micc@static-50-125-113-34.frr01.both.wa.frontiernet.net) |
| 20:30.05 | [TK]D-Fender | xochilpili: "core show function CHANNEL" |
| 20:31.13 | xochilpili | [TK]D-Fender, yes? and then? |
| 20:31.35 | xochilpili | channel set language es? |
| 20:32.13 | [TK]D-Fender | to whatever code you have prompts for |
| 20:32.17 | [TK]D-Fender | this is all documented. |
| 20:32.32 | xochilpili | core function CHANNEL set language es? |
| 20:32.45 | [TK]D-Fender | yes |
| 20:32.58 | [TK]D-Fender | with the proper syntax |
| 20:33.13 | [TK]D-Fender | and then you need es equivalents of all your sound finles to be put in the right folders |
| 20:33.40 | [TK]D-Fender | The base stuff is already availabl from digium |
| 20:33.40 | [TK]D-Fender | heads home |
| 20:45.05 | *** part/#asterisk josecapurro (~jcapurro@190.52.189.6) |
| 20:45.06 | *** join/#asterisk [Outcast] (~outcast@173.38.117.87) |
| 20:51.02 | luckman212 | good afternoon ! does anyone have a favorite desk phone model? I need to buy some new phones and its been years since I looked at "new" stuff. I have a Polycom VVX 500 on my desk that is nice but it's about 4 years old and I was wondering if there's anything newer you guys like |
| 20:57.34 | *** join/#asterisk [Outcast] (~outcast@173.38.117.87) |
| 21:00.42 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
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| 21:25.12 | dan_j | Hi, has anyone used a Panasonic KX-UT13x with Asterisk? I find that they respond very slowly to incoming calls when a large number of BLF keys have been programmed. Has anyone experienced this? |
| 21:33.10 | *** join/#asterisk newtonr (RustyNewto@nat/digium/x-vbredvpmrovqbctl) |
| 21:33.10 | *** mode/#asterisk [+o newtonr] by ChanServ |
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| 21:35.11 | *** part/#asterisk ehsjoar (~jonas@c-24-9-87-13.hsd1.co.comcast.net) |
| 21:35.26 | dan_j | Is there any way to get asterisk to send the CANCEL sip packet before any BLF NOTIFY packets are sent? |
| 21:36.01 | WIMPy | May the source be with you. |
| 21:36.17 | WIMPy | But at least you've got two sip channels to try now. |
| 21:37.20 | dan_j | oh great. thats not good. |
| 21:37.24 | dan_j | thanks anyway |
| 21:38.36 | WIMPy | But what's going on there? There should only be one subscription updated when a call ends. |
| 21:40.11 | dan_j | The phone is monitoring multiple extensions. An incoming call is sent to all extensions, so all the BLF keys on this phone light up. Therefore the phone is receiving around 16 NOTIFY packets to update the BLF state. |
| 21:40.36 | WIMPy | Oh, so it's about ringing calls. |
| 21:40.52 | [TK]D-Fender | dan_j, "chan_pjsip resource list" <----------- |
| 21:40.58 | dan_j | That seems to be slowing down the phone so it doesnt started ringing instantly that the packet comes in... and it doesnt stop ringing instantly when the CANCEL packet comes in |
| 21:41.03 | WIMPy | Doesn't sound terribly clever to do it in that order. |
| 21:41.07 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 21:42.54 | dan_j | [TK]D-Fender: does chan_pjsip have to be installed somewhere? I dont get any results from the CLI with that. |
| 21:43.19 | WIMPy | Off course it has to be loaded. |
| 21:43.39 | [TK]D-Fender | dan_j, Are you on * 13? |
| 21:43.40 | dan_j | I dont see it listed in menuselect as a module |
| 21:43.51 | WIMPy | And you probably can't load it without propper configuration. |
| 21:43.51 | [TK]D-Fender | because it doesn't exist in 11- |
| 21:43.53 | dan_j | Not yet. Still on 11 |
| 21:43.57 | [TK]D-Fender | FAIL |
| 21:44.02 | [TK]D-Fender | stop looking and upgrade |
| 21:44.02 | dan_j | nuts |
| 21:44.19 | dan_j | Hate upgrades. |
| 21:44.33 | dan_j | Ok. Thanks for the pointer. I'll upgrade and then look at it. |
| 21:44.56 | [TK]D-Fender | readup on it in the meantime |
| 21:45.01 | [TK]D-Fender | I gave you the terms |
| 21:45.07 | WIMPy | Or fix chan_sip. There's a good chance it's easy. |
| 21:46.44 | [TK]D-Fender | WIMPy, To have * try to reprioritize? Wouldn't put odds on it. |
| 21:47.08 | [TK]D-Fender | so much better to move forward... and if that doesn't cut it ... ditch that troublesome phone |
| 21:47.39 | dan_j | Why is chan_pjsip better than chan_sip? Just more up to date or does it do something significantly different? |
| 21:48.03 | WIMPy | Might be a verry simple case of reordering two blocks of code. Or maybe just a line. |
| 21:48.27 | WIMPy | It relies on an external lib. |
| 21:48.31 | dan_j | Never mind. I got my answer from the wiki. thanks. |
| 21:48.36 | WIMPy | But it definitely has new features. |
| 21:49.17 | Milenco | so WIMPy and [TK]D-Fender, how are you affiliated with Asterisk? I see you answering questions here all the time :) |
| 21:49.34 | Milenco | mostly in a good cop/bad cop style :D |
| 21:49.43 | WIMPy | I just hate it. |
| 21:50.01 | Milenco | hated it so much you dedicated most of your life to it! |
| 21:50.24 | WIMPy | Yes. The others are even worse. |
| 21:50.28 | Milenco | :P |
| 21:50.52 | Milenco | I assume you do this voluntary? |
| 21:51.05 | WIMPy | The chat? Yes. |
| 21:51.06 | [TK]D-Fender | #lesserevil |
| 21:51.18 | [TK]D-Fender | Milenco, massochism |
| 21:51.29 | Milenco | i noticed :P |
| 21:51.41 | [TK]D-Fender | s/ss/s/ |
| 21:51.53 | WIMPy | Side effects of using Asterisk. |
| 21:52.00 | Milenco | but youre employed at digium then? |
| 21:52.13 | [TK]D-Fender | is pleased his regex is an anagram |
| 21:52.24 | WIMPy | nonono |
| 21:52.24 | [TK]D-Fender | Milenco, Neither of us is |
| 21:52.32 | mjordan | WIMPy: don't be *quite* so adament |
| 21:52.39 | mjordan | OH DEAR GOD NOT THOSE PEOPLE |
| 21:52.40 | Milenco | haha :) |
| 21:53.10 | Milenco | I've been here for months and always see you two helping people out :) |
| 21:53.10 | [TK]D-Fender | mjordan, don't worry, we've all booked time in the de-con chamber ;) |
| 21:53.16 | Milenco | and wondered what the backstory is :) |
| 21:53.27 | mjordan | They're good people who like to help others? |
| 21:53.43 | robmal | That's never the answer. |
| 21:53.50 | Milenco | Yes it's not that i find it hard to believe it or anything :) |
| 21:53.58 | robmal | I'm sure they're slave labour from China. |
| 21:54.00 | Milenco | Just wondered what their background/drive is |
| 21:54.07 | WIMPy | We already clarifies the masochism thing. |
| 21:54.29 | WIMPy | s/d/d/ |
| 21:54.31 | [TK]D-Fender | luckman212, Polycom > All |
| 21:54.33 | Milenco | :P |
| 21:54.43 | WIMPy | No/ s/s/d/ |
| 21:55.09 | Milenco | Anyway, kudos for you! :) |
| 21:55.54 | robmal | Yes, i, too, would like to express my gratutide for your patience. |
| 21:56.02 | robmal | You're awesome! |
| 21:56.07 | Milenco | +1 |
| 21:57.33 | file | wobbles |
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| 21:58.01 | robmal | file: You're awesome too ;-) |
| 21:58.24 | Milenco | https://www.youtube.com/watch?v=9cQgQIMlwWw |
| 22:03.01 | *** join/#asterisk zogm_ (~zogm@gateway/vpn/privateinternetaccess/zogm) |
| 22:04.05 | file | I don't exist, therefore I can't be awesome |
| 22:04.05 | *** join/#asterisk Robotman321 (~brad@50-194-126-9-static.hfc.comcastbusiness.net) |
| 22:05.14 | WIMPy | You don't exist? That proves you're god! |
| 22:05.31 | [TK]D-Fender | file, if you don't exist then your awesomeness is theoretical and possibly infinite ;) |
| 22:05.37 | file | mmm infinite |
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| 22:41.54 | luckman212 | [TK]D-Fender: ok thanks that's what I figured but never hurts to ask! |
| 22:42.10 | luckman212 | i have a Yealink at work and it's "ok" but feels cheap / cheesy |
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| 23:00.56 | *** join/#asterisk learath (~learath@static-71-246-219-35.washdc.fios.verizon.net) |
| 23:01.31 | learath | Hi, I set username, authuser and fromuser in my sip.conf, however my sip uri shows none of them |
| 23:04.24 | [TK]D-Fender | Show us |
| 23:04.28 | [TK]D-Fender | ~pb |
| 23:04.28 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 23:06.37 | learath | http://pastebin.com/MiX3k7wG |
| 23:07.03 | learath | ugh. stupid notepad. |
| 23:07.59 | [TK]D-Fender | And the config |
| 23:08.50 | *** join/#asterisk ehsjoar (~jonas@c-24-9-87-13.hsd1.co.comcast.net) |
| 23:08.56 | [TK]D-Fender | Keep in mind that if you are using a separate "register =>" directive that it has NOTHING to do with a peer entry you have added these values to |
| 23:08.56 | learath | hmm. I wonder... |
| 23:09.10 | [TK]D-Fender | the peer is not involved in the creation of regtration attempts |
| 23:09.24 | WIMPy | It could be. |
| 23:09.34 | [TK]D-Fender | they are only used if you use the peer-based value to tell it to register |
| 23:09.50 | learath | so, I'm doing both. |
| 23:10.15 | WIMPy | Both is one to many. |
| 23:10.32 | learath | http://pastebin.com/kvyKAuaA |
| 23:10.38 | zogm_ | what advantages does ConfBridge have over MeetMe? Any performance improvements? Are there any downsides? What are the technical differences? |
| 23:11.12 | learath | so I should comment out the register? |
| 23:11.23 | newtonr | zogm_, https://wiki.asterisk.org/wiki/display/AST/ConfBridge |
| 23:11.24 | learath | I thought I needed it. |
| 23:11.39 | [TK]D-Fender | learath, You are not doing both |
| 23:11.44 | [TK]D-Fender | learath, Nor should you |
| 23:11.48 | WIMPy | learath: sp far you are ONLY using register. |
| 23:11.52 | learath | good to know. |
| 23:12.30 | [TK]D-Fender | learath, comment out that register and add "callbackextension=john" to your peer and apply |
| 23:12.32 | WIMPy | You can add 'callbackextension' to your peer and remove the register line. |
| 23:12.44 | WIMPy | Or you can add the neccessary info to the register line. |
| 23:13.18 | learath | unchanged. |
| 23:13.27 | *** part/#asterisk ehsjoar (~jonas@c-24-9-87-13.hsd1.co.comcast.net) |
| 23:13.27 | [TK]D-Fender | <PROTECTED> |
| 23:13.49 | [TK]D-Fender | that has the user & authuser there as well |
| 23:14.08 | [TK]D-Fender | Though it'd probably be cleaner and simpler to let the peer break it up for you. |
| 23:14.09 | learath | the user and authuser are the same FWIW |
| 23:14.10 | [TK]D-Fender | Your call. |
| 23:14.24 | [TK]D-Fender | try as we've suggested |
| 23:14.31 | learath | I already did. No change. |
| 23:14.35 | [TK]D-Fender | Show us |
| 23:14.40 | [TK]D-Fender | the full set |
| 23:14.48 | learath | eh. ok. |
| 23:14.58 | [TK]D-Fender | you might have to restart * entirely to flush any caching of the request from its prior state |
| 23:15.39 | WIMPy | At least unload/load chan_sip. |
| 23:15.50 | learath | I've been running "reload" |
| 23:15.59 | [TK]D-Fender | yes, that should do as well |
| 23:16.04 | [TK]D-Fender | not good enough |
| 23:16.27 | learath | what isn't good enough? |
| 23:16.39 | WIMPy | reload |
| 23:16.41 | learath | http://pastebin.com/fVzHJGz4 |
| 23:16.52 | learath | ok. I'll fully restart |
| 23:17.17 | learath | unchanged with a full restart. |
| 23:17.41 | [TK]D-Fender | <learath> Hi, I set username, authuser and fromuser in my sip.conf, however my sip uri shows none of them |
| 23:17.46 | [TK]D-Fender | I see them in there now... |
| 23:17.51 | learath | eh? |
| 23:18.01 | learath | uri="sip:sip.phonepower.com", |
| 23:18.05 | learath | ? am I an idiot? |
| 23:18.12 | *** join/#asterisk italorossi (~Adium@179.234.138.92) |
| 23:19.33 | [TK]D-Fender | Show us the config... |
| 23:20.48 | learath | ... http://pastebin.com/7Niz4J9f ... |
| 23:20.57 | learath | the register is, as requested, commented out., |
| 23:23.02 | [TK]D-Fender | comment out the callbackexten |
| 23:23.13 | [TK]D-Fender | then reload chansip and see if it disappears first |
| 23:23.15 | zogm_ | @newtonr thanks |
| 23:23.16 | [TK]D-Fender | make sure it's clean |
| 23:24.11 | learath | uhm. ok |
| 23:26.30 | [TK]D-Fender | Hrm. |
| 23:26.33 | [TK]D-Fender | Well... you CAN. |
| 23:26.38 | [TK]D-Fender | I now see part of the issue though |
| 23:26.41 | learath | ? |
| 23:26.45 | [TK]D-Fender | That number is NOT the ID I was referring to. |
| 23:26.57 | [TK]D-Fender | I missed that you only see the # some of the time |
| 23:27.05 | [TK]D-Fender | and not even on-screen when you edit it |
| 23:27.39 | [TK]D-Fender | Whcih I supposed should NOT be desructive. |
| 23:27.45 | [TK]D-Fender | I am now wondering why the field isn't there... |
| 23:28.00 | [TK]D-Fender | for which MySQL is still the fix in-hand for you for now. |
| 23:28.04 | [TK]D-Fender | Which should probably do it |
| 23:28.07 | learath | I think you are in the wrong window |
| 23:28.11 | [TK]D-Fender | oops |
| 23:28.16 | [TK]D-Fender | dangit |
| 23:28.35 | learath | you had me really confused for a minute there. |
| 23:29.28 | learath | so, as far as I can tell, callbackexten has no effect on uri="stuff" |
| 23:29.36 | learath | nor, as far as I can tell, does anything else. |
| 23:31.41 | WIMPy | It causes registrations to be sent. |
| 23:32.02 | *** part/#asterisk kharwell (kharwell@nat/digium/x-ntuddphcmfzbhzcm) |
| 23:36.01 | [TK]D-Fender | <[TK]D-Fender> comment out the callbackexten |
| 23:36.01 | [TK]D-Fender | <[TK]D-Fender> then reload chansip and see if it disappears first |
| 23:36.04 | [TK]D-Fender | Do this |
| 23:36.11 | [TK]D-Fender | lets make sure it's really taking changes |
| 23:36.17 | [TK]D-Fender | and not just keeping the very first ones |
| 23:36.33 | learath | I did, and I saw no difference whatsoever. How about I change the fromuser? |
| 23:37.07 | learath | well. |
| 23:37.16 | [TK]D-Fender | it should DISSAPEAR if you kill everything that should cause it to try |
| 23:37.19 | [TK]D-Fender | yet it doesn't? |
| 23:37.26 | learath | no change. |
| 23:37.32 | learath | even restarted asterisk. |
| 23:38.36 | [TK]D-Fender | So it's still there though it should be gone? |
| 23:38.51 | learath | so. I restarted asterisk again, and now it is gone. which makes me wonder what I did wrong. |
| 23:39.03 | learath | only my local phone still shows up |
| 23:39.35 | learath | and I'm out of time. Thanks for your help, I'll have to shoot this later. |
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