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| 00:31.14 | ChannelZ | Call a professional |
| 00:31.22 | ChannelZ | Kappa |
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| 09:13.01 | catphish | i'm not sure if there'll be a solution to this, but i thought it's worth asking, i've got a problem whereby asterisk doesn't know what local IP to use when sending SIP packets |
| 09:13.38 | catphish | for example, if it receives an OPTIONS command, it will respond not from the IP where it received the command, but instead from the default IP that the kernel chooses |
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| 13:00.49 | sparrowvivek16 | hello |
| 13:01.25 | sparrowvivek16 | what is the major difference between digium analogue and digital interface cards? |
| 13:02.17 | sparrowvivek16 | If analogue cards are used to connect PSTN or POTL... then what is the use of Digital? |
| 13:02.47 | sparrowvivek16 | am ijust taking to myself? |
| 13:04.33 | sparrowvivek16 | Hey WIMy... |
| 13:04.45 | sparrowvivek16 | oops WIMPy |
| 13:04.56 | sparrowvivek16 | slaps WIMPy around a bit with a large fishbot |
| 13:05.37 | sparrowvivek16 | sorry for that too... |
| 13:05.47 | sparrowvivek16 | no intention... |
| 13:07.16 | sparrowvivek16 | i don't understand... there are about 50 people here are more... |
| 13:07.17 | catphish | sparrowvivek16: i assume digital interface cards would be for digital lines ie isdn BRI |
| 13:07.41 | sparrowvivek16 | so no PSTN... right? |
| 13:07.56 | catphish | no, analog interface cards are for PSTN, digital are for ISDN |
| 13:08.03 | catphish | i mean, you're correct |
| 13:08.09 | sparrowvivek16 | it's like your LAN card but with more features... |
| 13:08.24 | catphish | well not at all |
| 13:08.47 | sparrowvivek16 | so how do they differ? |
| 13:08.55 | catphish | it says here: https://www.digium.com/products/telephony-cards/digital - "Digium's super-reliable digital line cards connect Asterisk-based communication systems to T1, E1, J1 and ISDN-BRI interfaces." |
| 13:09.15 | catphish | and https://www.digium.com/products/telephony-cards/analog - "Connect your Asterisk system with analog lines, phones, fax machines and other gear with Digium's rock solid analog interface cards." |
| 13:09.22 | sparrowvivek16 | i just saw that |
| 13:09.59 | sparrowvivek16 | oh... so if i need to connect say 50 PSTN lines... what should i do? |
| 13:10.13 | catphish | that sounds like something you wouldn't want to do |
| 13:10.22 | sparrowvivek16 | lol |
| 13:10.30 | sparrowvivek16 | just assuming... |
| 13:10.57 | coppice | you should use 2 E1s or 3 T1s, depending where you live |
| 13:10.58 | sparrowvivek16 | most of my calls will be diverted to PSTN lines and GSM... |
| 13:11.05 | catphish | there's really no sane reason you have 50 PSTN lines, you'd use ISDN PRI instead, where each line can handle 30 calls digitally |
| 13:11.32 | catphish | i'm in europe, so 2 x "ISDN 30" or "E1" |
| 13:11.53 | catphish | using PSTN lines for that would be really messy |
| 13:11.53 | sparrowvivek16 | all calls will be automatically diverted though ISDN? |
| 13:12.08 | catphish | i don't understand what you mean by "diverted" |
| 13:12.37 | catphish | an ISDN line is like a PSTN line, it still carries normal phone calls, except it's digital, and can handle multiple simultaneous calls |
| 13:12.58 | sparrowvivek16 | eg: if someone calls a PSTN Number using softphone and will this digital interface can handle that call? |
| 13:13.24 | sparrowvivek16 | oh... dude... you are a life saver... |
| 13:13.36 | catphish | the ISDN line will have a number, or range of numbers, and will handle incoming calls to those numbers |
| 13:13.44 | catphish | what country are you in> |
| 13:13.46 | catphish | *? |
| 13:13.53 | sparrowvivek16 | India |
| 13:13.57 | sparrowvivek16 | Asia |
| 13:14.19 | catphish | i don't know india, but: https://en.wikipedia.org/wiki/Integrated_Services_Digital_Network#India |
| 13:14.58 | catphish | but basically, you want an "ISDN PRI" line, this can handle 30 simultaneous calls through one line, and you will get one or more numbers that connect to that line |
| 13:15.39 | sparrowvivek16 | good job buddy... |
| 13:15.44 | sparrowvivek16 | you did good |
| 13:15.46 | catphish | it should be cheaper than 30 separate PSTN lines, and will achieve exactly the same, except that you can only plug it into a PBX like asterisk |
| 13:16.11 | sparrowvivek16 | Yeah i have IP PBX running |
| 13:16.28 | catphish | alternatively, consider just using a SIP provider |
| 13:17.13 | sparrowvivek16 | just one more question... What is the differnce bw BRI and PRI? |
| 13:17.48 | sparrowvivek16 | yeah i can but i dont want any SIP provides... i am using IAX2 more flexible... |
| 13:18.09 | catphish | BRI handles 2 channels (calls) |
| 13:18.21 | catphish | PRI handles 30 channels (calls) |
| 13:18.50 | sparrowvivek16 | But pri came first right? |
| 13:18.53 | catphish | for example, this card plugs into 2 x PRI lines, allowing 60 simultaneous calls: https://www.digium.com/products/telephony-cards/digital/dual-span |
| 13:19.18 | catphish | i don't really know what the point of BRI is, probably needs less copper than PRI so cheaper to install |
| 13:20.21 | sparrowvivek16 | alright... what are Hybrids in this section? |
| 13:21.11 | catphish | try reading :) |
| 13:21.17 | sparrowvivek16 | can plug Direct PSTN and ISDN lines? |
| 13:21.45 | catphish | yes, it allows you to mix and match BRI and PSTN modules |
| 13:21.50 | catphish | (not PRI) |
| 13:22.27 | catphish | they're designed mostly for people who have a small setup like 4 analog phones and 2 x BRI lines |
| 13:22.39 | catphish | and then you can do all of that with a single card |
| 13:22.53 | catphish | personally i wouldn't do anything with analog these days, so i see little point |
| 13:23.14 | sparrowvivek16 | Why BRI can only manage only 2 channels while PRI can 30? |
| 13:23.26 | sparrowvivek16 | then whats the use of using BRI? |
| 13:23.31 | catphish | i actually have no idea, wikipedia might know |
| 13:23.42 | sparrowvivek16 | thanks bro... |
| 13:23.45 | catphish | i assume BRI is cheaper to install and works over existing analog copper |
| 13:23.52 | sparrowvivek16 | you did |
| 13:23.55 | sparrowvivek16 | good job |
| 13:24.02 | catphish | https://en.wikipedia.org/wiki/Basic_Rate_Interface |
| 13:24.46 | sparrowvivek16 | see you around J |
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| 16:04.19 | sparrowvivek16 | hello? |
| 16:05.16 | sparrowvivek16 | How to limit any extension say (IAX2 = username: user1) with call balance and info... |
| 16:05.45 | cmendes0101 | Like an account with balance? |
| 16:05.51 | sparrowvivek16 | exactly |
| 16:06.49 | cmendes0101 | Build a database for the account and some code that deducts from there. |
| 16:07.18 | sparrowvivek16 | I am not an expert in these... care to guide me? |
| 16:07.50 | cmendes0101 | Do you know how to use a database? Like Mysql sqlite or something else |
| 16:08.09 | sparrowvivek16 | yes |
| 16:08.33 | sparrowvivek16 | i dont know much about asterisk |
| 16:09.32 | cmendes0101 | If all your setup is mainly in extensions.conf, you can setup odbc that will connect to a database. You can run a query for lets say call balance to make sure user has enough before placing the outbound call, etc |
| 16:09.56 | cmendes0101 | If you look up asterisk and odbc there should be a good number of tutorials |
| 16:11.48 | sparrowvivek16 | do i have to write scripts? |
| 16:13.09 | sparrowvivek16 | i am sorry to trouble you, but can you point me out any single tutorial? |
| 16:13.56 | cmendes0101 | I guess start by installing: http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/installing_configuring_odbc.html |
| 16:14.20 | sparrowvivek16 | ok for billing? |
| 16:15.38 | cmendes0101 | Then you probably don't want just to do a query. You should look into AGI or API. Move the logic out of extensions.conf and into actual scripts to be more powerful |
| 16:16.49 | cmendes0101 | I don't really know what your looking for, but there are some solutions that already have been done for billing: http://www.voip-info.org/wiki/view/Open+Source+Billing+Systems |
| 16:22.08 | sparrowvivek16 | @cmendes0101 i am just looking for a method to limit all users in the extension with time limit say 10mins... after which no one can use that particular extension |
| 16:23.19 | sparrowvivek16 | chris are you there? |
| 16:23.22 | cmendes0101 | Well you mentioned billing. Are you trying to bill? Are you saying all users are limited to 10min? |
| 16:24.02 | sparrowvivek16 | limit 10 minutes |
| 16:25.12 | cmendes0101 | You can just to Set(TIMEOUT(absolute)=seconds) in extensions.conf |
| 16:25.32 | sparrowvivek16 | can this be done in GUI |
| 16:26.06 | cmendes0101 | From a regular asterisk install, no |
| 16:26.44 | sparrowvivek16 | do i have to configure this for every extension? |
| 16:27.09 | *** join/#asterisk azerus (~badass@unaffiliated/badass) |
| 16:27.40 | cmendes0101 | No, you just set in a context. Whats your extensions.conf look like to dialout? |
| 16:27.52 | cmendes0101 | You would just place it before Dial() |
| 16:28.11 | cmendes0101 | Or you can do a parameter in Dial that will nicely warn the user before it hangs up at 10min |
| 16:29.48 | sparrowvivek16 | oh in the user extension page? |
| 16:30.21 | cmendes0101 | page? |
| 16:30.28 | cmendes0101 | What are you using? |
| 16:31.03 | sparrowvivek16 | For FreePBX Administration menu... --> applications -->extensions |
| 16:31.13 | cmendes0101 | You want #freeobx |
| 16:31.20 | cmendes0101 | #freepbx* |
| 16:31.27 | cmendes0101 | not this channel |
| 16:31.49 | sparrowvivek16 | i am running asterisk with free pbx... |
| 16:31.54 | sparrowvivek16 | ok... |
| 16:32.24 | cmendes0101 | Asterisk has no gui |
| 16:32.58 | sparrowvivek16 | there i no in there |
| 16:33.26 | sparrowvivek16 | AsteriskNOW... or that's what it says |
| 16:33.47 | sparrowvivek16 | # freepbx* there is no one... |
| 16:34.12 | cmendes0101 | I typo'ed the first time so don't add the *. Should be #freepbx |
| 16:34.33 | sparrowvivek16 | got in... |
| 16:34.38 | sparrowvivek16 | thank you... |
| 16:34.59 | [TK]D-Fender | <sparrowvivek16> How to limit any extension say (IAX2 = username: user1) with call balance and info... <- FreePBX has no billing system |
| 16:36.04 | sparrowvivek16 | yeah i just learned that... |
| 16:36.18 | sparrowvivek16 | thanks for mentioning it again... |
| 16:36.29 | lnb | if codecs are negotiated, sending trunk has both g279 and ulaw but receiving trunk only allows ulaw, why is sending pbx sending in format receiver cannot handle? |
| 16:37.17 | lnb | receiving pbx: channel.c:5440 set_format: Unable to find a codec translation path: (slin) -> (g729) |
| 16:37.24 | *** join/#asterisk Synthase_ (uid63346@gateway/web/irccloud.com/x-mmnfwsmjnnloghzw) |
| 16:44.44 | [TK]D-Fender | ~book |
| 16:44.44 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
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| 18:04.48 | sparrowvivek16 | hello? |
| 18:05.05 | sparrowvivek16 | tm1000 |
| 18:05.10 | sparrowvivek16 | what did i do? |
| 18:05.52 | [TK]D-Fender | [12:34} <[TK]D-Fender> <sparrowvivek16> How to limit any extension say (IAX2 = username: user1) with call balance and info... <- FreePBX has no billing system |
| 18:06.03 | [TK]D-Fender | Told you this an hour and a half ago |
| 18:06.30 | sparrowvivek16 | well then how else others do it? |
| 18:06.38 | sparrowvivek16 | other voip providers? |
| 18:06.38 | [TK]D-Fender | There is no such thing as an "Account balance". It does not do that. At all. |
| 18:06.56 | [TK]D-Fender | With something else, or they do it themselves in * |
| 18:07.19 | [TK]D-Fender | Or they are using something OTHER than * that may have some of these things more "built in" |
| 18:07.54 | sparrowvivek16 | so basically you all don't even know what my question is? |
| 18:08.15 | [TK]D-Fender | I know what you asked |
| 18:08.16 | sparrowvivek16 | and you all mock me? |
| 18:08.18 | [TK]D-Fender | And I've answered |
| 18:08.36 | [TK]D-Fender | Do it YOURSELF in *, or find some other package that'll configure that for you in some other way. |
| 18:08.40 | [TK]D-Fender | FreePBX does NOT do this |
| 18:09.46 | sparrowvivek16 | last question, have heard or you know anyting about this> |
| 18:09.57 | [TK]D-Fender | About what? |
| 18:10.01 | [TK]D-Fender | Ask a real question |
| 18:10.12 | [TK]D-Fender | *I* could bui;d this myself. |
| 18:10.16 | [TK]D-Fender | *I* know how. |
| 18:10.31 | sparrowvivek16 | can you now help me? |
| 18:10.33 | [TK]D-Fender | No. |
| 18:10.37 | [TK]D-Fender | It is a lot of work |
| 18:10.43 | [TK]D-Fender | and you know nothing about * at all |
| 18:10.45 | sparrowvivek16 | ok guid me... |
| 18:10.51 | [TK]D-Fender | It wouldn't be "helping" you. |
| 18:10.56 | [TK]D-Fender | It would be doing the entire JOB for you |
| 18:11.03 | sparrowvivek16 | thats why you guys are here toHELP |
| 18:11.17 | [TK]D-Fender | You seem to think that you're going to get a DIY by "help" while not knowing how * works at all. |
| 18:11.20 | [TK]D-Fender | That is not going to happen. |
| 18:11.29 | sparrowvivek16 | do have a post or article? |
| 18:11.38 | [TK]D-Fender | Can I learn how to perform brain surgery without knowing even the basics of the rbain? |
| 18:11.42 | [TK]D-Fender | There is no article. |
| 18:11.48 | [TK]D-Fender | Asterisk is PROGRAMMING. |
| 18:12.05 | sparrowvivek16 | see... and they say "ask you shall receive " |
| 18:12.08 | [TK]D-Fender | A biling system is an IDEA. A specific implementation. A collection of rules and data that YOU define., |
| 18:12.08 | sparrowvivek16 | HA HA HA |
| 18:12.34 | [TK]D-Fender | Nobody makes a book specifically on how to mold a statue of a dog out of green Play-Doh. |
| 18:12.36 | [TK]D-Fender | Too specific. |
| 18:12.50 | [TK]D-Fender | The book will teach you how * processes things. |
| 18:13.03 | [TK]D-Fender | The applications & functions all have instructions that tell you what they do |
| 18:13.03 | sparrowvivek16 | say things all you want... |
| 18:13.22 | [TK]D-Fender | it is YOUR job to understand the pieces in order to know what combination you will need to get your job done |
| 18:13.29 | sparrowvivek16 | in the end you are turning your back to me |
| 18:13.40 | [TK]D-Fender | You are asking for a guide that does not EXIST |
| 18:13.44 | [TK]D-Fender | Your quesiotn is bad |
| 18:13.50 | [TK]D-Fender | There is no guide for this |
| 18:13.55 | sparrowvivek16 | You are asking for a guide that does not EXIST? |
| 18:13.56 | [TK]D-Fender | Who would write it? |
| 18:14.03 | sparrowvivek16 | lol |
| 18:14.08 | [TK]D-Fender | NO GUIDE ON WRITING A BILLING SYSTEM FOR ASTERISK |
| 18:14.17 | [TK]D-Fender | There are books that teach how * works |
| 18:14.23 | sparrowvivek16 | nice talking with you... |
| 18:14.30 | [TK]D-Fender | that YOU want to use it to have a calling-card type system is your decision |
| 18:14.37 | [TK]D-Fender | nobody is going to write a book SPECIFICALLY for that |
| 18:14.52 | sparrowvivek16 | just stay here and say this to all incoming users... |
| 18:15.00 | sparrowvivek16 | see ya!!! |
| 18:15.05 | [TK]D-Fender | Do you see a "How to write a spreadsheet application in Turbo Pascal" book out there anywhere? |
| 18:15.06 | [TK]D-Fender | no. |
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| 18:42.59 | xochilpili | hi all |
| 18:43.34 | xochilpili | is there a way to know if my ip is blocked by a sip proxy of my provider? |
| 18:52.27 | [TK]D-Fender | Do you get a message saying you are blocked? |
| 18:52.46 | [TK]D-Fender | is your provider the only provider invloved in the overall call? |
| 18:53.02 | [TK]D-Fender | Is your traffic even actaully making it out to them in the first place? |
| 18:54.12 | *** join/#asterisk sparrowvivek16 (75d800d7@gateway/web/freenode/ip.117.216.0.215) |
| 18:54.21 | sparrowvivek16 | a2billing |
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| 19:03.32 | cmendes0101 | hahaha |
| 19:03.36 | cmendes0101 | did he come back to say that? |
| 19:04.26 | cmendes0101 | I'm still pretty sure I gave him the link for those |
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| 20:58.40 | xochilpili | hi all |
| 21:00.51 | xochilpili | i have 2 asterisk; and i connected each other, asterisk 1 has a freepbx isntalled with a trunk to my provider, then asterisk 2 has just one extension: 100 : the issue is that i want to call from asterisk 2 to asterisk 1 and use that trunk; so, when i make that call, i got this: Failed to authenticate device "100" <sip:100@asterisk2>; tag=... |
| 21:01.10 | xochilpili | in both asterisk i have same extension in dialplan: 100 same secret |
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| 21:11.36 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 22:57.07 | frederific | Anybody about? Banging my head against a brick wall trying to get AnveoDirect to play nicely with PJSIP, if anyone's got a moment? |
| 22:58.22 | WIMPy | ~ask |
| 22:58.28 | 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. |
| 22:58.41 | WIMPy | Is that an ITSP? |
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| 23:03.43 | frederific | @WIMPy apologies, should have known better |
| 23:04.15 | WIMPy | That's what we have infobot for :-) |
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| 23:28.12 | carrar | to repeat the repeatables!! |
| 23:28.27 | carrar | pokes Katty! |
| 23:30.44 | Katty | hugs carrar |
| 23:31.07 | *** part/#asterisk WangDang (~wangdang@108.161.126.113) |
| 23:32.07 | carrar | Ohayoo gozaimasu! |
| 23:34.22 | carrar | or ãã¯ãã |
| 23:34.36 | carrar | if you summport unicode! |
| 23:34.40 | carrar | support |
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