| 00:03.39 | *** join/#asterisk Draecos (~Draecos@203-121-194-218.e-wire.net.au) |
| 00:18.30 | *** join/#asterisk thiagoc (~thiagoc@unaffiliated/thiagoc) |
| 00:21.48 | *** join/#asterisk Spengler (~spengler1@pool-100-16-54-188.bltmmd.fios.verizon.net) |
| 00:22.08 | thiagoc | anyone know why my internet goes down when I connect the same line on the FXO card? |
| 00:22.31 | thiagoc | when I plug on a telephone it doens't goes down |
| 00:29.17 | Spengler | anyone have any luck with using Asterisk behind a Comcast SMC business gateway? specifically connecting external sip clients? |
| 00:46.47 | [TK]D-Fender | thiagoc, If you ar talking about ADSL on an analog line, then you should be using a filter |
| 00:47.04 | [TK]D-Fender | And thinking that you can get away without it is a mistake, it's why the invented the filters |
| 00:48.44 | thiagoc | [TK]D-Fender, so I should use a filter on both lines (modem and FXO port)? |
| 00:49.29 | *** join/#asterisk AviiNL (~AviiNL@h178213.upc-h.chello.nl) |
| 00:49.41 | thiagoc | I mean, both ends |
| 00:50.23 | [TK]D-Fender | You obviously don't put a filter in front of the modem |
| 00:50.34 | [TK]D-Fender | You want ADSL to get INTO the modem, don't you? |
| 00:50.44 | [TK]D-Fender | You filter the CARD |
| 00:50.53 | [TK]D-Fender | The same as you should any other phone |
| 00:53.28 | thiagoc | [TK]D-Fender, thanks! |
| 01:00.06 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 01:01.08 | *** join/#asterisk happy-dude (uid62780@gateway/web/irccloud.com/x-apeezlmxtofoklly) |
| 01:05.51 | *** join/#asterisk Stevie-O (~Stevie-O@pool-96-227-82-58.phlapa.fios.verizon.net) |
| 01:06.24 | Stevie-O | greetings |
| 01:09.13 | pankid | Hell-O Stevie-O |
| 01:13.31 | Stevie-O | hello, pankid |
| 01:15.00 | Stevie-O | hrm |
| 01:15.10 | Stevie-O | my outgoing caller ID is borked |
| 01:16.58 | *** join/#asterisk Draecos (~Draecos@203-121-194-218.e-wire.net.au) |
| 01:21.20 | Stevie-O | wtf |
| 01:21.33 | Stevie-O | func_callerid.c:748 party_number_write: Unknown number presentation '(redacted)', value unchanged |
| 01:22.57 | [TK]D-Fender | you should probably be SHOWING us the actual attempt, not just a singular message with redactions |
| 01:23.37 | [TK]D-Fender | Want an autopsy? Gives us a body. Oh, and please don't fuck it. #nonecrophiliaplz |
| 01:24.05 | Stevie-O | okay |
| 01:24.27 | Stevie-O | num-pres isn't what I want |
| 01:25.48 | Stevie-O | all I can find online is large lists of arguments to CALLERID() but no details on what the various options control or what constitute valid arguments |
| 01:26.32 | Stevie-O | HAH |
| 01:26.34 | Stevie-O | IT WORKS |
| 01:31.34 | *** join/#asterisk Mango45 (~Mango45@d142-59-245-22.abhsia.telus.net) |
| 01:31.40 | Mango45 | My friends. |
| 01:32.20 | Mango45 | Anyone know how to add a custom SIP header to a 200 OK message? |
| 01:33.14 | Stevie-O | mmmmh |
| 01:33.20 | Stevie-O | an interesting challenge |
| 01:33.38 | Stevie-O | what version |
| 01:33.51 | Mango45 | Last I checked it can't be done with chan_sip, but I haven't yet used res_pjsip. Currently using 11 but am open to upgrading if I could do this with it. |
| 01:33.52 | Stevie-O | (of Asterisk) |
| 01:33.54 | [TK]D-Fender | change the source for that module |
| 01:34.01 | [TK]D-Fender | that's it |
| 01:34.10 | [TK]D-Fender | There is no free-form getting to add whatever you want |
| 01:34.23 | Mango45 | Ok, same deal for res_pjsip? |
| 01:34.25 | [TK]D-Fender | as far as * goes |
| 01:34.45 | [TK]D-Fender | same deal period |
| 01:34.48 | Stevie-O | there's a function called pjsip_header |
| 01:34.57 | Stevie-O | "Gets, adds, updates or removes the specified SIP header from a PJSIP session." |
| 01:35.04 | Stevie-O | https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Function_PJSIP_HEADER |
| 01:35.07 | [TK]D-Fender | that adds to an INVITE |
| 01:35.10 | Stevie-O | oh |
| 01:35.13 | [TK]D-Fender | just like the chan_sip equiv |
| 01:35.19 | [TK]D-Fender | You don't get to tage specific stuff in a 200 |
| 01:35.26 | Stevie-O | mmmh |
| 01:35.52 | Stevie-O | ok then |
| 01:36.00 | [TK]D-Fender | Nothing in * will enable this. So either mod the source, or run a proxy that can do this, etc |
| 01:36.52 | Mango45 | Think I could do it in a couple of hours? |
| 01:46.14 | [TK]D-Fender | I have no idea what your skill-set it |
| 01:46.30 | Mango45 | Could YOU do it in a couple of hours? |
| 01:46.31 | [TK]D-Fender | Or what it would take to use something external to add this conditionally |
| 01:47.23 | [TK]D-Fender | I'm not a raw coder like that. I could probably figure something out in a certain # of hours as an ugly hack |
| 01:47.29 | [TK]D-Fender | (guessing) |
| 01:47.31 | Mango45 | Cool |
| 01:47.33 | Mango45 | Thanks. |
| 01:47.36 | [TK]D-Fender | depending |
| 01:47.41 | Mango45 | Of course. |
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| 03:03.52 | Stevie-O | how can I act on SMS messages coming from a peer? |
| 03:07.40 | [TK]D-Fender | They hit the dialplan |
| 03:08.01 | [TK]D-Fender | Go lookup "sip sessage" on the official wiki |
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| 03:16.11 | ZorinOS | oi |
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| 03:20.36 | Stevie-O | that was weird |
| 03:29.16 | Stevie-O | okay, so there's something I don't quite grasp here |
| 03:32.54 | Stevie-O | okay cool |
| 03:33.44 | Stevie-O | so I have a phone number |
| 03:33.50 | Stevie-O | a DID phone number |
| 03:34.23 | Stevie-O | so apparently the the extension <mynumber> is activated in the [lesnet-incoming] section when someone dials that DID |
| 03:34.45 | Stevie-O | how do I have the user enter an extension to dial after that |
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| 14:08.18 | *** join/#asterisk Guuest45818 (6dc99aa0@gateway/web/freenode/ip.109.201.154.160) |
| 14:08.24 | Guuest45818 | Hi |
| 14:08.36 | *** join/#asterisk sparetire (~sparetire@unaffiliated/sparetire) |
| 14:09.56 | Guuest45818 | how can i receive incoming calls with easybell ? |
| 14:12.29 | Guuest45818 | pls help me |
| 14:13.05 | jwpierce3 | I'm getting a "Username/auth name mismatch" failed inside asterisk when i have different values for "Address" & "Authentication User ID". I'd like my address for each of my phones to be different (obviously), while maintaining a single username/password. Everythiug works well when these fields are the same. Any ideas? |
| 14:14.14 | jwpierce3 | I'm using Polycom 550's |
| 14:22.00 | *** join/#asterisk Dpunkt (~Dpunkt@p57A4FD67.dip0.t-ipconnect.de) |
| 14:23.41 | [TK]D-Fender | You can't use the same name on multiple devices |
| 14:23.49 | [TK]D-Fender | They will fight over registration, etc |
| 14:23.58 | [TK]D-Fender | Not with chan_sip anyway |
| 14:31.17 | jwpierce3 | [TK]D-Fender, at this point I only have 1 phone. Are you telling my that I have absolutely have to make these two fields identical within my 550? |
| 14:32.34 | jwpierce3 | It seems though, the answer is yes |
| 14:35.03 | [TK]D-Fender | You don't have 2 people saying "I'm John" |
| 14:36.55 | jwpierce3 | That should be the function of "Address" (within polycom) [address] (within sip.conf) |
| 14:38.44 | jwpierce3 | username shouldn't have anything to do with this. If this isn't the case, then why differentiate username & address? |
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| 16:15.11 | *** mode/#asterisk [+o cresl1n] by ChanServ |
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| 16:37.21 | dougquaid | Is there a tutorial on installing asterisk with TLS/RSTP on Ubuntu? |
| 16:37.34 | dougquaid | I haven't found a good one yet in my google searches |
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| 17:50.11 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.7.2 (2016/02/05), 11.21.2 (2016/02/11); DAHDI: DAHDI-linux 2.10.0 (2014/08/13), DAHDI-tools 2.10.0 (2014/08/13); libpri 1.4.15 (2014/06/16) -=- Asterisk wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
| 17:57.36 | *** join/#asterisk happy-dude (uid62780@gateway/web/irccloud.com/x-elwididhrepycdof) |
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| 18:33.00 | dougquaid | Is there a VPS provider that works the best with asterisk or are they all pretty much the same? |
| 18:49.52 | newtonr | I don't know the answer, but when talking with them you might ask them if they do or can prioritize voice traffic on their network. |
| 18:50.06 | newtonr | SIP/RTP, etc |
| 19:06.51 | *** join/#asterisk drab (~administr@23-24-198-190-static.hfc.comcastbusiness.net) |
| 19:11.36 | drab | mmmh, I was tcpdumping to try and figure out my problems with MWI, but in doing so I found something else that looks odd to me |
| 19:12.04 | drab | the SUBSCRIBE packet in wireshark shows with SRC address of my phone and DST address of the asterisk server |
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| 19:13.45 | drab | but in the message header the "From" header it says From: "$extension"<sip:$extension@$*_server_ip>" |
| 19:14.10 | drab | why does the sip part mentions the @server_ip ? |
| 19:14.21 | drab | I'd expected that to be be the ip of the phone |
| 19:15.15 | drab | but pretty much all the "from/to" headers I'm seeing are all the same, $extension@$*_server_ip |
| 19:25.41 | joako | Is there a better alternative to fail2ban? It's detecting the login attempts but has issues actually blocking the IP addresses |
| 19:26.35 | *** join/#asterisk DivideBy0 (~DivideBy0@unaffiliated/divideby0x0) |
| 19:31.25 | ChannelZ | That's more of a config issue with fail2ban than it being broken and wanting to find something else |
| 19:31.41 | ChannelZ | What action do you have it doing? |
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| 21:34.43 | [TK]D-Fender | It shouldn't have "issues". That's either lacking information to do it, or misconfiguration |
| 22:23.25 | *** join/#asterisk stkoch (~stefan@p5DDA0E6B.dip0.t-ipconnect.de) |
| 22:24.11 | stkoch | hi |
| 22:24.28 | stkoch | I have the following in extensions: |
| 22:24.31 | stkoch | exten => s,n,GotoIf(${BLACKLIST()}?8) |
| 22:24.38 | stkoch | exten => s,8,Set(DIALSTATUS=BUSY) |
| 22:24.45 | stkoch | exten => s,9,Hangup() |
| 22:25.06 | stkoch | The blacklisted caller get's declined message |
| 22:25.14 | stkoch | but I want to set message BUSY |
| 22:26.45 | stkoch | I see only SIP/2.0 603 Declined |
| 22:26.56 | stkoch | but not something else with BUSY |
| 22:27.01 | stkoch | Have you an idea? |
| 22:27.19 | [TK]D-Fender | If you want to send "busy" ... then us BUSY() |
| 22:27.24 | [TK]D-Fender | Not "hangup()" |
| 22:27.41 | [TK]D-Fender | because sets nothing and processing and resulting in nothing = "just declined. No because" |
| 22:29.04 | stkoch | I have replaced s,8 |
| 22:29.06 | stkoch | exten => s,8,Busy() |
| 22:29.46 | stkoch | Asterisk sends now SIP/2.0 486 Busy Here |
| 22:29.50 | [TK]D-Fender | Things you should know: Dialstatus isn't jsut something you set. That is somethinset BY Dial() |
| 22:30.24 | [TK]D-Fender | And the value isn't what sets the response code. That is carried from Dial itself, not somethin twisted by the var value |
| 22:30.28 | stkoch | But the caller will not hear BUSY, it gives an net error |
| 22:30.55 | [TK]D-Fender | "net error"? |
| 22:31.04 | [TK]D-Fender | You should be clearer about who your caller is exactly |
| 22:31.19 | stkoch | The caller is a mobile phone that is blacklisted for test purposes |
| 22:31.38 | stkoch | It shows "net error", not busy |
| 22:31.59 | stkoch | After timeout from 10 seconds |
| 22:32.42 | stkoch | Another phone would show Dial... up to 10 seconds, and get's disconnected then |
| 22:32.59 | stkoch | Could it be that die SIP provider does not support the 486 Busy Here ? |
| 22:33.15 | [TK]D-Fender | "mobile phone" isn't clear about how it is calling |
| 22:34.08 | [TK]D-Fender | please be complete in your description |
| 22:34.49 | stkoch | mobile phone calls a phone at an asterisk server (that is registered at 1&1 SIP) |
| 22:35.16 | stkoch | The asterisk ist registered at 1&1 sip, the phone is directly at the asterisk server |
| 22:35.28 | stkoch | The mobile phone calls the 1&1 SIP number |
| 22:35.35 | [TK]D-Fender | So to be clear : Cell phone -> Cell provider > PSTN > ITSP (1&1) > Asterisk ? |
| 22:36.01 | stkoch | yes |
| 22:38.26 | [TK]D-Fender | Ok, guess that's how the cell provider responds. |
| 22:38.40 | [TK]D-Fender | Try Congestion() instead and see if it likes that better |
| 22:43.56 | stkoch | Busy() works with Sipgate, but not with 1und1 |
| 22:45.46 | stkoch | Congestion() does not work with 1und1, too |
| 22:46.22 | *** join/#asterisk Oatmeal (~Suzeanne@75-103-145-152.ccrtc.com) |
| 22:51.18 | *** join/#asterisk lorsungcu (sid65806@gateway/web/irccloud.com/x-wzdnwjfjuduwdzlw) |
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| 22:56.32 | stkoch | Is it possible to set busy codes manually |
| 22:56.40 | stkoch | Like set SIP 600 instead of BUSY() |
| 22:56.41 | [TK]D-Fender | Those are the key apps you have. Hangup() can also be passed a cause code |
| 22:56.54 | [TK]D-Fender | there is a table showing the mapping so you can try a few different values |
| 23:01.37 | stkoch | Is this a valid call:? |
| 23:01.45 | stkoch | Hangup(AST_CAUSE_BEARERCAPABILITY_NOTAVAIL) |
| 23:02.14 | [TK]D-Fender | not AFAIK. |
| 23:02.19 | [TK]D-Fender | there is a NUMBER value that maps to that |
| 23:14.36 | stkoch | Busy(5) works with 1und1 -> mobile phone shows busy |
| 23:15.12 | *** join/#asterisk d00gster (~d00gster@unaffiliated/d00gster) |
| 23:17.13 | stkoch | Busy() works now, too |
| 23:17.21 | stkoch | I don't know why... |
| 23:18.40 | stkoch | [TK]D-Fender: thanks |
| 23:18.57 | [TK]D-Fender | You're welcome |
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| 23:41.22 | me_meee37 | hi, can anyone assist with asterisk on the RPi? |