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| 01:01.54 | chocolate | what should I read, to learn how to config a server to make my home phone, dial using my voip provider? receiving calls in the landline but dialling always using my credits on my voip provider... Do I need to buy a hardware for this? |
| 01:11.04 | *** join/#asterisk Samael28 (~Samael28@176.104.56.91) |
| 01:11.48 | [TK]D-Fender | if you want to use a physical line you have that is obviously hardware |
| 01:12.15 | [TK]D-Fender | What are you intending to speak into and hear out from? That would be hardware too... |
| 01:13.41 | [TK]D-Fender | 2 options for the "home phone" to place calls over your provider (or talk to the PBX in general), A: get an ATA, B: get an interface card with FXS ports. "A" is a lot cheaper and easier |
| 01:13.44 | [TK]D-Fender | ~ATA |
| 01:13.44 | infobot | hmm... ata is Analogue Terminal Adapter which provides an FXS and/or FXO and ethernet, see http://www.voip-info.org/wiki/view/ATA |
| 01:13.52 | [TK]D-Fender | As for learning how to use * : |
| 01:13.53 | [TK]D-Fender | ~book |
| 01:13.53 | 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 |
| 01:20.19 | *** join/#asterisk infobot (ibot@rikers.org) |
| 01:20.19 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.19.0 (2018/01/11), Standard: 15.2.0 (2018/01/11); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
| 01:20.31 | *** join/#asterisk Samael28 (~Samael28@176.104.56.91) |
| 01:20.45 | chocolate | lol [TK]D-Fender loved your reply |
| 01:22.06 | chocolate | I plan to use a normal phone as a home phone but must to dial using the provider |
| 01:22.53 | [TK]D-Fender | then you'll either need something that can "pass through" to your home line, or have another interface to also take yourhome line into your PBX completely. |
| 01:23.33 | [TK]D-Fender | For this I'd recommend something like the SPA-3102 or modern equivalent |
| 01:27.48 | chocolate | that way I can receive by the landline number and make phone calls by the provider inside asterisk? |
| 01:29.32 | chocolate | and where will I configure the voip provider to dial using it? sorry about my ignorance |
| 01:29.50 | [TK]D-Fender | All of this is in the asterisk dialplan |
| 01:29.58 | [TK]D-Fender | that is what processes every call sent to it |
| 01:30.08 | [TK]D-Fender | calls from your phone line. Calls from your attached phone. |
| 01:30.22 | [TK]D-Fender | Calls from things like soft-phones, etc. Calss from your provider. |
| 01:30.39 | [TK]D-Fender | all hit your server and get processed however you tell it to. |
| 01:31.01 | [TK]D-Fender | Dial 12345 on my phone on a tuesday, X happens. On a Wednesday Y happens |
| 01:31.09 | [TK]D-Fender | 4th time in a row? Z |
| 01:31.12 | [TK]D-Fender | Or whatever |
| 01:31.19 | [TK]D-Fender | It's all upp to you |
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| 01:34.45 | chocolate | and can I redirect the landline received calls to my mobile number using this? |
| 01:35.22 | chocolate | to answer calls when I'm not home |
| 01:35.30 | [TK]D-Fender | Anything you want |
| 01:35.40 | [TK]D-Fender | Call comes in and does what you say. |
| 01:36.11 | [TK]D-Fender | Maybe you want you physical phone to ring. Maybe you want to to call out to your cell via your provider. Maybe you want to present them a menu first. |
| 01:36.28 | [TK]D-Fender | Think of any number of options and that's what you can do. |
| 01:48.17 | chocolate | can I make my landline dial through my mobile phone too? |
| 01:48.56 | *** join/#asterisk Samael28 (~Samael28@176.104.56.91) |
| 01:49.05 | [TK]D-Fender | Anything you want |
| 01:49.14 | [TK]D-Fender | call comes in from anywhere goes where you tell it to. |
| 01:49.28 | [TK]D-Fender | Every call is jsut a call and it's up to you to determine where it all goes |
| 01:49.57 | chocolate | but what hardware do I need to make my config dial using my mobile number? |
| 01:50.08 | [TK]D-Fender | maybe you call a special number you configure from your phone and you'll flip a flag you'll check for in processing calls from somewhere |
| 01:51.02 | [TK]D-Fender | if you want * to be able to dial out your phone it MIGHT be supported with a BT dongle and chan_dongle |
| 01:51.32 | [TK]D-Fender | not always a functional idea though. There are other GSM gateways if you want to pick up another SIM card if that's worth it to you |
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| 01:53.39 | chocolate | Because here in my country for example there are 4 providers like AT&T in USA, and each one call for free from themselves... If I buy 4 mobile numbers and be possible to configure to call using one of these 4 options I can always talk for free not having 4 mobile phones... |
| 01:56.15 | [TK]D-Fender | at that point it's probably worth getting a multi-sim gatway, but of course that does start getting expensive when you consider paying for service on those lines, etc |
| 01:58.38 | chocolate | yeah but if be possible to implement I can share with friends the expenses |
| 02:00.01 | chocolate | will it get busy if I try to use the sim 1 and one of my friends be in a call at the same sim? |
| 02:00.08 | *** join/#asterisk Samael28 (~Samael28@176.104.56.91) |
| 02:01.29 | [TK]D-Fender | as I said, its's your pgramming for ALL calls. |
| 02:01.56 | [TK]D-Fender | if your sim is busy, then it's busy |
| 02:02.40 | chocolate | doesn't work so... |
| 02:02.51 | chocolate | thank you for your time, patience and help |
| 02:03.36 | chocolate | anyway for my first plan to configure my home server, what should I ask or google to have a tutorial? I don't even know what to ask |
| 02:03.58 | chocolate | ask to google |
| 02:04.01 | [TK]D-Fender | ~book |
| 02:04.01 | 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 |
| 02:04.04 | [TK]D-Fender | ^^^^^ |
| 02:04.11 | chocolate | thanks |
| 02:04.21 | chocolate | already downloaded |
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| 02:45.42 | me^ | hello anyone used |
| 02:45.42 | me^ | Grandstream-UCM6102-6104-IPPBX- |
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| 02:46.59 | [TK]D-Fender | Why would we? |
| 02:47.37 | me^ | just wondering if that would be a good unit for small business ? |
| 02:48.01 | [TK]D-Fender | Well we don't use or support those devices so... |
| 02:48.16 | [TK]D-Fender | guess you could just read the feature sheet and find something to compare it to... |
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| 03:11.25 | Samot | me^: This is now the second room where you are asking about voice systems that have no relation to the subjects of the rooms. |
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| 16:19.47 | bittis | random question, the regex function in asterisk, is it using some specific regular expression "engine" ? |
| 16:19.58 | bittis | would it expect perl like syntax? |
| 16:20.03 | bittis | for example |
| 16:22.25 | Samot | What do you mean perl like syntax? |
| 16:22.34 | Samot | It's a pattern matching scheme. |
| 16:22.49 | igcewieling | bittis: I'd assume POSIX regex unless otherwise specified. |
| 16:24.23 | igcewieling | bittis: I've not done anything complicated with Asterisk's REGEX function just simple stuff like if (${REGEX("^(180|200)$" ${sm_hangupcause})}) { |
| 16:26.12 | Samot | ^(180|200)$ <<< common in all that use regex |
| 16:26.51 | igcewieling | *nod* That's why I never worried about Perl regex .vs. POSIX regex .vs. EREGEX, etc. |
| 16:32.55 | bittis | thanks guys |
| 16:33.46 | bittis | been looking at some way of getting a substring of the regex as well, but doesn't seem to be an option with ael |
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| 17:22.45 | josefig | good morning, is there a way to use asterisk as gateway between a trunk in IAX (from customer) and terminate the call to SIP (vendor) ? |
| 17:24.33 | file | Asterisk does what it is configured to do, and it can be configured to do that. |
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| 17:40.36 | josefig | file: i appreciate it my friend :) |
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| 17:48.35 | JohnWigley | josefig: YATE (does do IAX trunking) and Freeswitch (doesn't do IAX trunking) can both do that also, so Asterisk isn't the only game in town for that job if you needed for example it to run on *cough* Windows :) |
| 17:50.27 | josefig | ok, JohnWigley let me see it |
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| 17:56.55 | tpdmoore | for the record, asterisk does use POSIX regex.h and does not use PCRE |
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| 18:34.32 | rathjeja | Can anyone point me in the right direction for learning how to interpret the output of the 'pri set debug' command in the freepbx CLI? |
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| 18:49.30 | igcewieling | bittis: look into CUT() and FIELDQTY() functions |
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| 18:56.58 | war9407 | I use a cisco spa3102 with a analog line, sometimes when voicemail picks up, the person will hang up but asterisk keeps on recording what it sounds like when you hang up, is there anyway to only record voicemaeils when someone speaks/says something? the call hangup does not seem to be working properly (in the US and using the US settings) |
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| 20:35.13 | degenerate | Hey guys, can anyone look at this call and tell me why one end could not hear the other? http://termbin.com/211d |
| 20:41.45 | [TK]D-Fender | We'd need to see it with SIP debug |
| 20:44.09 | *** part/#asterisk rathjeja (~rathjeja@50-204-85-14-static.hfc.comcastbusiness.net) |
| 20:44.23 | degenerate | [TK]D-Fender ok thanks. i'll try to figure out how to turn that on. |
| 20:45.29 | [TK]D-Fender | "sip set debug on" |
| 20:46.37 | degenerate | # sip set debug on |
| 20:46.37 | degenerate | -bash: sip: command not found |
| 20:46.59 | degenerate | nm im retarded |
| 20:48.24 | degenerate | [TK]D-Fender ok. so i've got that on now. but is there a way to make it dump to a log file instead of just the consolve? |
| 20:48.29 | degenerate | -v |
| 20:48.57 | degenerate | the call issues are intermittant, so i need to leave it on this debug mode and be able to pull up the log file in the future. |
| 20:49.23 | [TK]D-Fender | Show any call. |
| 20:49.39 | vo1pbx | _in it's entirety_ |
| 20:50.06 | [TK]D-Fender | And scrolling should be irrelevant. Use PuTTY for SSH, and set a large scrollback buffer, and right-click the title bar and "cop all to buffer" |
| 20:50.13 | [TK]D-Fender | you don't have to select anything at all |
| 20:50.49 | vo1pbx | PuTTY, almost twenty years old. |
| 20:51.47 | degenerate | http://termbin.com/pq4l |
| 20:52.11 | [TK]D-Fender | There is no new call in there |
| 20:52.15 | [TK]D-Fender | or no verbose to back it up |
| 20:52.22 | [TK]D-Fender | "core set verbose 10" |
| 20:52.42 | vo1pbx | SIP/2.0 401 Unauthorized |
| 20:52.52 | vo1pbx | fix your credentials |
| 20:53.18 | vo1pbx | ~pb |
| 20:53.18 | infobot | extra, extra, read all about it, 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://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 20:53.20 | vo1pbx | please |
| 20:53.30 | vo1pbx | so we can give line number references |
| 20:54.31 | degenerate | +1 for teaching me about pastebinit |
| 20:54.38 | degenerate | installing. sec |
| 20:55.37 | vo1pbx | installing... ffs. |
| 20:56.09 | degenerate | lol log too long for pastebin |
| 20:56.20 | degenerate | lemme break it into chunks |
| 20:57.07 | vo1pbx | anyways... lobster poutine...bbiab |
| 20:57.32 | degenerate | https://pastebin.com/1BeMb9rJ |
| 20:57.47 | degenerate | https://pastebin.com/L1ETz31C |
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| 21:27.35 | degenerate | ^^ was this enough info? |
| 21:29.00 | *** join/#asterisk bravvve22 (~user@5-196-189-80.cinfuserver.com) |
| 21:33.08 | bravvve22 | hi,i have some asterisk server installed on vps ,on debian machine,i can connect to him from 3G network,and i can"t connect to it from ADSL connection (NAT) there is tcpdump of the server https://pastebin.com/ybARvzZr |
| 21:33.20 | *** join/#asterisk bravvve22 (~user@5-196-189-80.cinfuserver.com) |
| 21:34.29 | *** join/#asterisk bravvve22 (~user@5-196-189-80.cinfuserver.com) |
| 21:34.43 | bravvve22 | hi,i have some asterisk server installed on vps ,on debian machine,i can connect to him from 3G network,and i can"t connect to it from ADSL connection (NAT) there is tcpdump of the server https://pastebin.com/ybARvzZr |
| 21:54.00 | [TK]D-Fender | degenerate, I don't see a call in there |
| 21:54.29 | degenerate | [TK]D-Fender ok one sec lemme try again. i was still trying to figure out why i'm getting all those 401 unauthorized |
| 21:54.35 | degenerate | it seems to be happening for every extension |
| 21:54.52 | degenerate | but all those extensions are "working", i mean they can make and receive calls, just sometimes the quality is crap |
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| 21:59.43 | degenerate | [TK]D-Fender https://pastebin.com/TFJvRyHn |
| 21:59.49 | degenerate | that one has a call i'm pretty sure |
| 22:00.00 | bravvve22 | hi,i have some asterisk server installed on vps ,on debian machine,i can connect to him from 3G network,and i can"t connect to it from ADSL connection (NAT) there is tcpdump of the server https://pastebin.com/ybARvzZr the client wireshark https://pasteboard.co/H6TNOWd.png |
| 22:05.44 | [TK]D-Fender | Reliably Transmitting (NAT) to 216.115.69.144:5060: |
| 22:05.49 | [TK]D-Fender | Flowroute is not behind NAT. |
| 22:06.17 | degenerate | my pbx is? |
| 22:06.25 | [TK]D-Fender | Fix that in your trunk and make sure you have "directmedia=no" |
| 22:06.33 | [TK]D-Fender | "nat=no" for your peer |
| 22:06.43 | [TK]D-Fender | YOU may be NAT'd, but THEY aren't |
| 22:07.11 | [TK]D-Fender | so ensure those 2 lines are right in your peer, and that you ahve your RTP port range forwarded to your server properly, and that should do it |
| 22:08.09 | degenerate | my current peer config: https://pastebin.com/KZJKLARr |
| 22:08.41 | degenerate | i see you're right, im missing directmedia=no |
| 22:08.44 | degenerate | and nat=no |
| 22:08.47 | degenerate | ok i'll try that. |
| 22:09.17 | *** join/#asterisk rwb (~Thunderbi@65.183.151.121) |
| 22:16.28 | degenerate | [TK]D-Fender I am confused about this though, i copied those rules right from flowroute, this is what they gave me: https://i.imgur.com/itui0So.png |
| 22:16.33 | degenerate | why would they give people the wrong thing? |
| 22:18.04 | Samot | Its outdated |
| 22:18.25 | Samot | So if i am reading this right. Audio works but it is crappy? |
| 22:18.51 | degenerate | Samot just some calls. sometimes the calls one end can't hear the other end. but not always. |
| 22:19.16 | Samot | All calls? |
| 22:19.27 | degenerate | no. not all calls. most calls are fine. |
| 22:19.35 | Samot | I mean are the devices on the same network? |
| 22:19.51 | Samot | This is only calls from flowroute? |
| 22:20.08 | degenerate | yes, we are only using flowroute. and all the phones are in one building on the same LAN. |
| 22:20.20 | degenerate | the PBX is on that same LAN as well. |
| 22:20.30 | Samot | So phone to phone never have an issue? |
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| 22:21.08 | degenerate | you mean local calls within the org? dialing another extension? |
| 22:21.15 | Samot | Yes |
| 22:21.19 | degenerate | this very rarely happens the room is very small, they just get up and walk over to eachothers desks |
| 22:21.33 | Samot | Thats not the answer |
| 22:21.43 | degenerate | i really have no clue tbh cause it never happens. |
| 22:21.53 | Samot | So it doesnt |
| 22:22.00 | Samot | Thats what i am asking |
| 22:22.25 | Samot | This only happening for inbound calls from flowroute? |
| 22:22.49 | degenerate | no. mostly outbound dials. |
| 22:22.55 | Samot | Ok |
| 22:23.03 | Samot | Who cant hear who? |
| 22:23.30 | degenerate | the person receiving the call cannot hear my staff. |
| 22:24.15 | degenerate | but i have also seen it where they place a call and get dead air for a bit after being connected. so i guess it happens both ways. |
| 22:24.16 | Samot | So this is an issue with flowroute, more importantly the carrier route they use |
| 22:24.54 | degenerate | Hmmm so i should try another provider perhaps? |
| 22:25.00 | Samot | Well |
| 22:25.26 | Samot | You can collect information like, do all calls to a certain area code or NPANXX have this issue |
| 22:25.29 | Samot | Etc. |
| 22:25.34 | Samot | Present it to Flowroute. |
| 22:25.42 | Samot | They will fix a bad carrier route for you |
| 22:26.09 | Samot | If this is happening on only certain calls, then you need to see what those calls have in common. |
| 22:26.17 | Samot | So you can see if there is a _common_ problem. |
| 22:27.25 | degenerate | I see. so should i still do the directmedia=no, nat=no peer settings? |
| 22:28.06 | degenerate | and I have started compiling a list of numbers with which the issues occur, it seems to be fairly random assortment of area codes so far (we dial all over NA) |
| 22:28.24 | Samot | You need to remove canreinvite |
| 22:28.25 | Samot | It's old |
| 22:28.35 | Samot | directmedia replaces it |
| 22:28.56 | Samot | directmedia=no is pretty much canrevinite=no |
| 22:29.10 | Samot | You should have nat=no however. |
| 22:29.26 | Samot | But again, this may not even be an issue on your side. |
| 22:29.33 | Samot | You need to work with Flowroute at some level. |
| 22:29.38 | degenerate | Ok. thank you. |
| 22:30.02 | degenerate | One more thing, i see that my phones are constantly doing this SIP/2.0 401 Unauthorized |
| 22:30.15 | degenerate | i've checked my flowroute sip passwords and they match. |
| 22:30.27 | degenerate | im not sure why i would get those SIP/2.0 401 Unauthorized messages. |
| 22:30.35 | degenerate | this happens all the time, with every extension, just looping. |
| 22:31.10 | degenerate | this log has some examples: https://pastebin.com/1BeMb9rJ |
| 22:32.15 | Samot | That's normal |
| 22:32.23 | Samot | The first attempt is always challenged. |
| 22:33.00 | degenerate | but should it continue to retry over and over again? |
| 22:33.05 | degenerate | doesn't this add extra load to the server? |
| 22:33.35 | degenerate | these extensions are doing this all night long while nobody is even in the office using them. lol. |
| 22:36.28 | bravvve22 | hi,i have some asterisk server installed on vps ,on debian machine,i can connect to him from 3G network,and i can"t connect to it from ADSL connection (NAT) there is tcpdump of the server https://pastebin.com/ybARvzZr the client wireshark https://pasteboard.co/H6TNOWd.png |
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| 23:50.51 | bravvve22 | hi,am getting qualify:0 with sip show settings whanever i gonfigure it |
| 23:50.57 | bravvve22 | hi,am getting qualify:0 with sip show settings whanever i configure it |
