| 00:11.19 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 00:19.26 | *** part/#asterisk kharwell (kharwell@nat/digium/x-bcfwsxztgdmuxdyf) |
| 00:31.52 | *** join/#asterisk slav3_sergal (~frankthet@unaffiliated/slav3-kitten/x-0866809) |
| 00:42.22 | *** join/#asterisk jasonwert (~jasonwert@71.89.137.28) |
| 00:47.29 | *** join/#asterisk carrar (~tim@troutlake.net) |
| 00:55.56 | *** join/#asterisk roler (~textual@unaffiliated/roler) |
| 00:57.33 | roler | We have an asterisk box that worked great with Time Warner. We recently moved offices and switched to AT&T. The outbound line goes over 23ch T1. Incoming audio works, but cuts out about every 3 or 4 seconds. Internal office calls work perfect (desk phone to desk phone) so I seriously doubt it is our network. Is there any PRI settings I can change to try to fix this? |
| 00:59.40 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 01:02.16 | *** join/#asterisk superscrat (~asanders@173-21-89-217.client.mchsi.com) |
| 01:13.21 | carrar | roler: Did you confirm with AT&T that they settings are even the same as they where with TW? |
| 01:14.14 | roler | carrar; nope. but I did have a guy from India call me and "verify" that it was working |
| 01:14.51 | carrar | Sounds like you got yout money worth on that one :) |
| 01:19.48 | roler | carrar; just to confirm, if calls internally in the office work perfectly, then it is the outbound service right? |
| 01:20.09 | roler | carrar; my question comes from the thought of, could the phones be using difference codecs or something outbound than they are internally |
| 01:21.57 | carrar | Could be but your system should transcode that |
| 01:22.11 | carrar | and you could be having issues on your server |
| 01:22.16 | carrar | maybe miss configured pri |
| 01:22.34 | carrar | could still be network related |
| 01:22.48 | carrar | maybe even AT&T issues |
| 01:23.34 | carrar | If nothing has changed you need to look at what has changed |
| 01:24.10 | carrar | that is if you have changed nothing by the PRI, you probably need to focus on that |
| 01:24.36 | carrar | verify your config is correct for the settings AT&T requires |
| 01:24.59 | carrar | Could have AT&T come out and test their service |
| 01:28.37 | *** join/#asterisk roler (~textual@unaffiliated/roler) |
| 01:28.46 | roler | sorry. I did see your messages |
| 01:29.12 | roler | sounds good |
| 01:44.53 | *** join/#asterisk jasonwert (~jasonwert@71.89.137.28) |
| 02:39.48 | *** join/#asterisk jasonwert (~jasonwert@71.89.137.28) |
| 03:04.36 | *** join/#asterisk azerus (~badass@unaffiliated/badass) |
| 03:26.56 | *** join/#asterisk vader- (~Adium@38.126.217.136) |
| 03:27.36 | *** join/#asterisk infina (~infina@unaffiliated/infina) |
| 03:36.37 | *** join/#asterisk crocodilehunter (~Thunderbi@CPE-121-211-223-68.hhui7.cht.bigpond.net.au) |
| 03:40.09 | *** join/#asterisk mac_ified (~mac_ified@67-9-150-210.res.bhn.net) |
| 04:11.23 | *** join/#asterisk DragonAzul (~DragonAzu@187.208.37.198) |
| 04:51.33 | *** join/#asterisk saratogga (~saratogga@ip98-165-139-238.ph.ph.cox.net) |
| 05:09.53 | *** join/#asterisk vader- (~Adium@38.126.217.136) |
| 05:46.44 | *** join/#asterisk gerhard7 (~gerhard7@77-172-82-111.ip.telfort.nl) |
| 05:56.35 | *** join/#asterisk azerus (~badass@unaffiliated/badass) |
| 06:05.17 | *** join/#asterisk crocodilehunter (~Thunderbi@CPE-121-211-223-68.hhui7.cht.bigpond.net.au) |
| 06:13.09 | *** join/#asterisk evil_gordita (robert@ip70-188-63-173.rn.hr.cox.net) |
| 06:25.58 | *** join/#asterisk elitas (~elitas@213.226.135.203) |
| 06:45.27 | *** join/#asterisk tparcina (~tomo@212.92.200.41) |
| 06:56.54 | elitas | hello, i need to register one number and set him a joke, how should write my sip.conf and extensions.conf |
| 06:57.19 | wyoung | set him a joke? |
| 06:57.49 | wyoung | Try writing that sentence again. |
| 06:58.16 | elitas | sorry, set a joke for calling to my register number |
| 06:58.39 | WIMPy | And what does "register a number" mean? |
| 06:59.41 | wyoung | elitas: you would like a dedicated extension to playback a joke? |
| 07:00.11 | wyoung | elitas: What is your native language? perhaps someone here speaks it |
| 07:00.11 | elitas | register => number:pass@server/user |
| 07:01.30 | WIMPy | I'm neither sur what yu're up to nor what your question is. |
| 07:01.58 | wyoung | oh and hi WIMPy!!! :) |
| 07:05.54 | WIMPy | Well, the "user" in your example is the extension to be called. |
| 07:07.22 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 07:25.11 | *** join/#asterisk crocodilehunter (~Thunderbi@CPE-121-211-223-68.hhui7.cht.bigpond.net.au) |
| 07:28.37 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 07:34.20 | *** join/#asterisk defsdoor (~andy@cpc73037-sutt4-2-0-cust62.19-1.cable.virginm.net) |
| 07:46.12 | *** join/#asterisk hehol (~hehol@gatekeeper.loca.net) |
| 07:48.23 | *** join/#asterisk bulkorok (~Adium@89.245.151.228) |
| 08:00.57 | *** join/#asterisk pchero_work (~pchero@109.70.54.56) |
| 08:02.03 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 08:06.24 | *** join/#asterisk Zogot (~Adium@185.21.52.255) |
| 08:18.41 | *** join/#asterisk evil_gordita (robert@ip70-188-63-173.rn.hr.cox.net) |
| 08:26.44 | *** join/#asterisk juned (~juned@202.131.119.122) |
| 08:28.36 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 08:33.23 | *** join/#asterisk beardy (~beardy@unaffiliated/beardy) |
| 08:48.53 | *** join/#asterisk elitas (~elitas@213.226.135.203) |
| 08:50.19 | *** join/#asterisk Panther_Modern (~Panther_M@unaffiliated/panther-modern/x-6168176) |
| 08:50.35 | Panther_Modern | Are there any other channels for enterprise VOIP, or is this the main hangout? |
| 08:54.32 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 08:56.18 | Panther_Modern | I'm interested in putting together a redundant / hot-failover PBX solution that can live in ESXi or HyperV. Recommendations? |
| 09:00.00 | *** join/#asterisk azerus (~badass@unaffiliated/badass) |
| 09:17.26 | *** join/#asterisk aberrios (~antonio@185.99.164.76) |
| 09:18.09 | aberrios | Hello, anyone chucking asterisk SIP debug packets to rsyslog/syslog to store in centralised logging Graylog/Elasticsearch? |
| 09:18.29 | aberrios | or do I need something like Homer |
| 09:38.18 | *** join/#asterisk mastahh|work (~mastahh@89.238.175.130) |
| 09:54.09 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 09:54.57 | *** join/#asterisk cyberfawkes (CyberAssas@gateway/shell/panicbnc/x-yryhmmvodhgzsfkb) |
| 09:55.18 | cyberfawkes | oi, I have a few questions that I can't seem to find concrete answers too. If anyone has a second. |
| 09:58.56 | aberrios | cyberfawkes: just throw 'em out there,, see if anyone responds ;) |
| 10:00.17 | cyberfawkes | I'm interested in getting a Cisco 7940G phone. However I don't know how to go about setting it up to work with Asterisk(which I can easily do with a few simple google searches). But what I'm questioning is do I need an actual phone line? I live in an apartment that lacks a traditional phone line. |
| 10:01.01 | *** join/#asterisk azerus (~badass@unaffiliated/badass) |
| 10:01.37 | aberrios | cyberfawkes: you can connect asterisk to a SIP provider, Twilio. Any particular reason for a Cisco for home use? |
| 10:01.48 | aberrios | so no you do not need a phone line |
| 10:01.56 | aberrios | pstn line* |
| 10:04.20 | *** join/#asterisk sparetire_ (~sparetire@unaffiliated/sparetire) |
| 10:06.34 | cyberfawkes | aberrios: other than personal preference, not really. I use the Cisco 79xx phones quite frequently, and I really like them. |
| 10:07.26 | aberrios | fair enough, seems expensive/overkill unless you had access to some 'cheap' ones ;).. guess it is personal preference |
| 10:08.11 | aberrios | so as long as you have internet then you can connect asterisk to a SIP provider such as Twilio (other brands are available), so you do not require a pstn line |
| 10:11.08 | cyberfawkes | aberrios: my company actually has a few extra 7940G's laying around. |
| 10:13.26 | cyberfawkes | They're going to sell me one for $10. |
| 10:15.00 | cyberfawkes | So I could in theory set up an Asterisk Server on a Raspberry Pi, connect it to my 7940G and use a service like Twilio and be good to go? |
| 10:15.12 | aberrios | yep! |
| 10:15.26 | cyberfawkes | Awesome! |
| 10:15.50 | cyberfawkes | And people are still paying for "Home Phone Service". |
| 10:15.58 | aberrios | Extra bonus points for setting up SRTP phone to provider ;) |
| 10:16.00 | cyberfawkes | And spending ridiculous amounts of money for it... |
| 10:16.07 | *** join/#asterisk skrusty (~skrusty@168.63.14.171) |
| 10:16.09 | cyberfawkes | SRTP? |
| 10:16.20 | aberrios | encrypted RTP (Media) |
| 10:16.36 | aberrios | here in the UK I have to have a phone line to have the internet |
| 10:17.04 | aberrios | it sucks, just an extra expense |
| 10:18.28 | cyberfawkes | So you said Twilio right? Where would I find it on their page? SIP Trunking? |
| 10:18.41 | aberrios | yeah that's right |
| 10:20.11 | cyberfawkes | thoughts on the best provider? |
| 10:20.16 | cyberfawkes | price wise? |
| 10:20.23 | cyberfawkes | Twilio seems decent in the price area. |
| 10:21.08 | cyberfawkes | Also any specific reason to avoid Cisco devices other than cost? |
| 10:22.03 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 10:22.58 | *** join/#asterisk CeBe (~CeBe@a81-14-224-229.net-htp.de) |
| 10:23.44 | *** join/#asterisk ruied (~ruied@46.50.53.13) |
| 10:23.51 | *** join/#asterisk karelk (~karel@31.10.145.202) |
| 10:27.07 | ruied | Hello! When I palce a call to PhoneA, I would like that if a second call is made to PhoneA the dial plan dials to the second alternative (PhoneB). I have added the second Dial(Sip/301). Butit keeps ringing at PhoneA (Second line). Do I need to restrict it at the phone to just receive one call, or can I make this direct with the Dial command? |
| 10:33.52 | ruied | I would like that If a phone is on a call and someone rigns to it than the dialplan rings do the second option. exten => _0,n,Dial(sip/600) |
| 10:58.58 | *** join/#asterisk bluemerlin (~bluemerli@office-nat.westpoint.gradwell.net) |
| 11:02.30 | *** join/#asterisk azerus (~badass@unaffiliated/badass) |
| 11:17.26 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 11:20.20 | *** join/#asterisk areski (~areski@80.174.128.6.dyn.user.ono.com) |
| 11:29.02 | *** join/#asterisk ThomasKeller (~Thomas@vmx.ethz.ch) |
| 11:30.51 | ThomasKeller | tcptls.c: == Problem setting up ssl connection: error:14094418:SSL routines:SSL3_READ_BYTES:tlsv1 alert unknown ca |
| 11:30.51 | ThomasKeller | tcptls.c: FILE * open failed! |
| 11:31.23 | ThomasKeller | I am getting the above errors, when I connect with a softphone client (linphone) to my asterisk |
| 11:34.13 | *** join/#asterisk [NC] (~nc@rv1.sabius.net) |
| 11:37.15 | *** join/#asterisk znf (~ibm86@unu.card-sharing.eu) |
| 11:46.24 | *** join/#asterisk [NC] (~nc@rv1.sabius.net) |
| 11:56.37 | *** join/#asterisk eofster (~eofster@213.61.153.26) |
| 11:58.23 | *** join/#asterisk ruied (~ruied@31.33.136.95.rev.vodafone.pt) |
| 12:01.56 | *** join/#asterisk italorossi (~Adium@187.60.66.11) |
| 12:04.33 | *** join/#asterisk azerus (~badass@unaffiliated/badass) |
| 12:09.31 | *** join/#asterisk znf (~ibm86@unu.card-sharing.eu) |
| 12:40.49 | *** join/#asterisk yokel (~yokel@unaffiliated/contempt) |
| 12:46.50 | *** join/#asterisk yokel (~yokel@unaffiliated/contempt) |
| 12:51.03 | *** join/#asterisk vinrock (~vin@unaffiliated/vinrock) |
| 12:52.39 | aberrios | sorry cyberfawkes, to be honest only ever avoided cisco coz I found them too complicated, or rather overly complicated. Polycoms/Grandstream/Yealink were all easier cheaper.. Polycom is my personal favourite (don't work for Polycom) |
| 12:53.15 | aberrios | cyberfawkes: there may be other SIP providers in your country, do a g search. |
| 12:53.55 | aberrios | ThomasKeller: have you given your softphone access to the CA cert? |
| 12:54.14 | aberrios | ThomasKeller: could be it can't read it,, perms issue? |
| 12:54.40 | aberrios | or rather Asterisk can't read the necessary certs |
| 12:54.45 | *** join/#asterisk gringo (~gringo@unaffiliated/gringo) |
| 12:58.47 | *** join/#asterisk Panther_Modern (~Panther_M@unaffiliated/panther-modern/x-6168176) |
| 12:59.38 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 13:06.56 | *** join/#asterisk azerus (~badass@unaffiliated/badass) |
| 13:16.08 | *** join/#asterisk mac_ified (~mac_ified@67-9-150-210.res.bhn.net) |
| 13:19.12 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 13:28.37 | *** part/#asterisk juned (~juned@202.131.119.122) |
| 13:50.17 | *** join/#asterisk Draecos (~Draecos@203-121-194-11.e-wire.net.au) |
| 14:03.46 | *** join/#asterisk zaf (~zaf@76.72.92.37) |
| 14:03.55 | *** join/#asterisk saratogga (~saratogga@ip98-165-139-238.ph.ph.cox.net) |
| 14:08.26 | *** join/#asterisk azerus (~badass@unaffiliated/badass) |
| 14:08.35 | dan_j | Hi. How do I set a SIP header before dialling when using PJSIP? I've tried PJSIP_HEADER but i'm using it within a Local channel. The following line after the set PJSIP_HEADER, I issue the dial the PJSIP channel. But I'm getting an error saying that I can only set PJSIP_HEADER from within a PJSIP channel. |
| 14:09.23 | dan_j | I used to do it this way with chan_sip, but pjsip doesnt seem happy with that. |
| 14:09.24 | *** join/#asterisk brad_mssw (~brad@66.129.88.50) |
| 14:11.40 | *** join/#asterisk babak (uid19622@gateway/web/irccloud.com/x-hxogndxbbbnfcaow) |
| 14:16.07 | babak | Hi,Is it possible to enable VAD in Asterisk? How? |
| 14:16.09 | *** join/#asterisk ntt (c33cbe9c@gateway/web/freenode/ip.195.60.190.156) |
| 14:16.48 | ntt | Hi, i'm trying to use asterisk with freepbx 13.0.15 but the webinterface is really slow. Someone can help me please^ |
| 14:16.49 | ntt | ? |
| 14:20.15 | *** join/#asterisk jasonwert (~jasonwert@75-134-81-98.static.aldl.mi.charter.com) |
| 14:20.18 | [TK]D-Fender | babak: "core show application amd" |
| 14:20.49 | [TK]D-Fender | ntt: ask in #freepbx . It is only supported for this in their channel |
| 14:21.03 | ntt | thanks |
| 14:25.02 | babak | [TK]D-Fender: thx I mean VAD=voice active detection for reducing bandwith |
| 14:25.37 | [TK]D-Fender | babak: No. |
| 14:30.25 | *** join/#asterisk vader- (~Adium@65.199.222.133) |
| 14:38.08 | *** join/#asterisk eofster (~eofster@213.61.153.26) |
| 14:43.59 | *** join/#asterisk mjordan (mjordan@nat/digium/x-homljiauziyrblhq) |
| 14:43.59 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 14:44.37 | Ickmund | dan_j, you're probably calling PJSIP_HEADER before you are making the dial call? So you're working on a-leg, while PJSIP will be on b-leg. Check out the note on https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Function_PJSIP_HEADER |
| 14:44.41 | igcewieling | babak: Asterisk does not support VAD |
| 14:50.02 | Ickmund | bakak: oej have a branch that supports it |
| 14:50.24 | *** join/#asterisk wonderworld (~ww@ip-84-119-184-180.unity-media.net) |
| 14:51.49 | *** join/#asterisk gerhard7 (~gerhard7@77-172-82-111.ip.telfort.nl) |
| 14:58.47 | *** join/#asterisk kharwell (kharwell@nat/digium/x-ikbfflqrbjsbshvv) |
| 15:01.23 | *** join/#asterisk azerus (~badass@unaffiliated/badass) |
| 15:07.46 | *** join/#asterisk saratogga (~saratogga@ip98-165-139-238.ph.ph.cox.net) |
| 15:08.54 | *** join/#asterisk Zogot (~Adium@185.21.52.255) |
| 15:15.11 | *** join/#asterisk vinrock (~vin@unaffiliated/vinrock) |
| 15:15.11 | *** join/#asterisk mirton (~jazz@mirtoff.ru) |
| 15:23.57 | *** join/#asterisk igcewieling1 (~ewieling@ip98-170-211-145.pn.at.cox.net) |
| 15:30.35 | *** join/#asterisk puzzled (~puzzled@2001:982:1097:1:95d7:9209:52a6:719) |
| 15:31.38 | *** join/#asterisk newtonr (RustyNewto@nat/digium/x-fsokwmmvzlaphhdk) |
| 15:31.38 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 15:34.41 | *** join/#asterisk DavidHuebner (~hubie@ip-69-178-218-71.far.ideaone.net) |
| 15:35.31 | *** join/#asterisk azerus (~badass@unaffiliated/badass) |
| 15:36.05 | *** join/#asterisk zerohalo (~zerohalo@2601:199:4200:d92e:dd91:c587:cfcf:9584) |
| 15:38.29 | babak | igcewieling1:thx Ickmund : thx , oej ? do you have a link ? |
| 15:41.22 | zaf | looking for an analog TA with many ports. like >4, up to 24 if possible. is that a thing, or do you need some other type of device? |
| 15:41.54 | [TK]D-Fender | Tons of gateways like this |
| 15:42.37 | [TK]D-Fender | I'd be surprised if you failed to find any if you actually looked already |
| 15:43.08 | igcewieling1 | zaf: I like Adtran |
| 15:43.43 | zaf | i've looked a little and found a few gateways, just wasn't sure if that was the right terminology |
| 15:43.55 | Ickmund | bakak, http://svn.digium.com/svn/asterisk/team/oej/roibos-cng-support-1.8/ |
| 15:43.56 | [TK]D-Fender | http://www.telephonydepot.com/Catalog/Analog-Gateways |
| 15:44.03 | zaf | thanks |
| 15:44.16 | [TK]D-Fender | Ickmund: That's a real fail unfortunately. 1.8 = DEAD |
| 15:44.32 | *** join/#asterisk ruied (~ruied@31.33.136.95.rev.vodafone.pt) |
| 15:44.36 | [TK]D-Fender | And as I've been informed chan_sip as a whole is no longer under core support |
| 15:44.41 | Ickmund | [TK]D-Fender, not gonna disagree with you |
| 15:44.58 | *** join/#asterisk cyford (~support@96.80.63.53) |
| 15:54.16 | Zogot | [TK]D-Fender: surely still for 11 since 11 is lts also though no? |
| 16:01.09 | igcewieling1 | Zogot: since PJSIP is not in Asterisk 11 it would be logical to assume chan_sip is supported in Asterisk 11. |
| 16:02.40 | Zogot | igcewieling1: that is also what id expect, a point to bring up though when I see the phrase "chan_sip as a whole is no longer under core support" but I dont follow the internals much or so, could of missed something |
| 16:05.34 | file | chan_sip was changed to extended support in 13 and above |
| 16:05.38 | file | it remains core supported in 11 |
| 16:08.30 | [TK]D-Fender | file: so only non-core in 13+ ? |
| 16:08.35 | file | correct |
| 16:09.24 | [TK]D-Fender | So is that to say no bug fixes 13+ as well? or does it get bug+sec? |
| 16:10.44 | file | this is a support level from Digium, it doesn't control what bug fixes will go in or where |
| 16:10.56 | file | if a community member fixes a bug in chan_sip in 13+, then it can still go in |
| 16:14.06 | *** join/#asterisk pjensen00 (~per@ip-69-178-218-71.far.ideaone.net) |
| 16:22.04 | *** join/#asterisk areski (~areski@80.174.128.112.dyn.user.ono.com) |
| 16:23.52 | *** join/#asterisk bulkorok (~Adium@89.245.151.228) |
| 16:26.43 | *** join/#asterisk rmudgett (rmudgett@nat/digium/x-oipzidqtvxptlhki) |
| 16:27.59 | dan_j | Ickmund: thanks for that. didnt spot the note when i was looking at the wiki |
| 16:33.00 | *** join/#asterisk DragonAzul (~DragonAzu@187.208.50.52) |
| 16:47.50 | *** join/#asterisk JonathanD (~JonathanD@freenode/staff/jonathand) |
| 16:57.02 | rrittgarn | seeing a lot of [2015-11-18 10:55:02] WARNING[20256]: db.c:288 db_execute_sql: Error executing SQL (COMMIT): database is locked in my logs but am not sure which table the system is trying to write to transactionally. Not seeing it in debug either - anybody know which module might be trying to commit? |
| 16:57.21 | rrittgarn | looks like its every 40-60 seconds - guessing chan_sip, but am not sure how to check |
| 16:58.32 | rrittgarn | Also reading it closer i see func_cdr - but there can be no calls active and i'm getting it |
| 17:07.09 | *** join/#asterisk cmendes0101 (~cmendes01@pool-98-112-114-3.lsanca.fios.verizon.net) |
| 17:10.19 | rrittgarn | or im crazy... (most likely) going to main/db.c line 288 is a sqlite3 call - i'm not using sqlite - is the astdb sqlite? |
| 17:12.50 | file | yes |
| 17:14.46 | rrittgarn | ok so the comment directly above it saying it doesn't lock on purpose basically tells me this message is really not helpful and i should turn down the verbosity |
| 17:15.17 | igcewieling1 | rrittgarn: could it be a permissions problem? |
| 17:15.42 | rrittgarn | thought so too, so I'm actually running as root to rule that out (i know bad, bad, bad) |
| 17:23.05 | *** join/#asterisk pchero (~pchero@109.70.54.56) |
| 17:32.21 | *** part/#asterisk WangDang (~wangdang@24.140.224.98) |
| 17:40.20 | *** join/#asterisk drab (~spike@23-24-198-190-static.hfc.comcastbusiness.net) |
| 17:41.17 | *** join/#asterisk WangDang (~wangdang@24.140.224.98) |
| 17:41.46 | drab | hi, is there any way to setup patterns in voicemail like you do in extensions? I've used a pattern to match internal phones so that one line is enough to deal with a whole group, but I can't do that with voicemail |
| 17:42.50 | drab | I could just script the creation of all the lines, but it'd be nicer if I could just use $EXTEN and match the callerid in voicemail.conf. however docs don't seem to say that's possible |
| 17:43.14 | [TK]D-Fender | huh? |
| 17:43.22 | [TK]D-Fender | there is no "callerid" in voicemail |
| 17:43.36 | rrittgarn | what about different passwords per mailbox? if everything were the same then you would have issues |
| 17:43.46 | rrittgarn | or a lack of security i guess |
| 17:45.39 | *** join/#asterisk mac_ified (~mac_ified@67-9-150-210.res.bhn.net) |
| 17:48.01 | drab | rrittgarn: fair point, altho it's not much of a concern for folks here, I guess I shouldn't encourage that. |
| 17:48.08 | drab | I'll just script gen the lines, thanks |
| 17:50.29 | *** join/#asterisk bluemerlin (~bluemerli@109.236.171.1) |
| 17:54.35 | *** join/#asterisk areski (~areski@80.174.128.112.dyn.user.ono.com) |
| 17:55.10 | igcewieling1 | it is easy to point sip devices to a custom mailbox |
| 17:56.21 | igcewieling1 | that takes are of monitoring for new messages. |
| 18:05.51 | *** join/#asterisk vane- (~vane-@static-71-176-211-9.rcmdva.fios.verizon.net) |
| 18:07.23 | drab | igcewieling1: I lost you there, would you mind expanding a little on it? thank you |
| 18:10.01 | [TK]D-Fender | You actual question's wording was very incoherent |
| 18:10.14 | [TK]D-Fender | Rephrase what you actually looking to do. |
| 18:16.06 | drab | ok. I have about a dozen phones configured in sip.conf. In extension.conf instead of having one entry per phone I just have a pattern matching on 4 digits that dials $EXTEN. That way with 3 lines I take care of all the dozen phones: exten => _1XXX,1,Dial(SIP/${EXTEN},20) , same => n,VoiceMail(${EXTEN}@office,u) , hangup |
| 18:16.23 | drab | when the second line hits and the call gets routed to voicemail it looks for a line corresponding to that specific extension |
| 18:16.31 | drab | (at least that's my current understanding) |
| 18:16.48 | drab | therefore I should have a dozen lines, one per extension, to set that up |
| 18:16.58 | drab | besides the PIN, all other attributes would be the same |
| 18:17.25 | rrittgarn | anyone know if res_parking supports realtime? Can't seem to find one way or another in the wiki |
| 18:17.36 | drab | so I was wondering if I could do something similar to what I did in extension.conf to catch all voicemails at once |
| 18:18.27 | drab | [TK]D-Fender: is that more coherent? |
| 18:18.42 | [TK]D-Fender | Your sip.conf doesn't magically work. You have an entry for each there already |
| 18:19.03 | [TK]D-Fender | They aren't created automatically because of a dial pattern |
| 18:19.15 | drab | I know, I'm trying to avoid having other places where I need one entry per phone, that's all |
| 18:19.16 | [TK]D-Fender | you PICK which one based on what is dialed by you still had to define it in sip.conf |
| 18:19.28 | [TK]D-Fender | you obviously have to do the same for voicemail |
| 18:19.39 | drab | so that if I add another phone in sip.conf I don't have to go to extension.conf and voicemail.conf and add them there too |
| 18:19.49 | drab | that's all, trying to avoid room for mistakes by forgetting stuff |
| 18:19.56 | drab | that was the intention |
| 18:20.11 | drab | again, I can script I guess and solve it that way, was just asking |
| 18:20.14 | [TK]D-Fender | not happening |
| 18:20.17 | drab | ok, np |
| 18:20.19 | drab | thank you |
| 18:20.20 | [TK]D-Fender | Everything has to be defined |
| 18:23.20 | igcewieling1 | drab: what specific mailbox do you want for everyone? |
| 18:23.34 | [TK]D-Fender | He still want's one each |
| 18:23.52 | igcewieling1 | [TK]D-Fender: terribly confused isn't he? |
| 18:24.08 | igcewieling1 | drab: the short answer is "no, you cannot do that" |
| 18:24.11 | [TK]D-Fender | no, just a poorly worded question and one unrealistic idea |
| 18:25.16 | [TK]D-Fender | just because you can use a dialplan to match and toss to a defined SIP peer didn't mean you didn't ahve to define the peers individually. He somhow thought there'd be a way not to have to define each box . Since he couldn't with SIP it makes little sense with VM either. |
| 18:26.02 | igcewieling1 | [TK]D-Fender: sounds like he didn't read the book. |
| 18:38.23 | [TK]D-Fender | The sample config really says it all. |
| 18:40.32 | *** join/#asterisk CeBe (~CeBe@dhcp-212-159.vpn.tu-berlin.de) |
| 18:49.36 | *** join/#asterisk ruied (~ruied@bl12-163-167.dsl.telepac.pt) |
| 18:50.24 | *** join/#asterisk Dpunkt (~Dpunkt@p5DE5ED45.dip0.t-ipconnect.de) |
| 18:50.36 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 18:53.50 | *** join/#asterisk blooregard (~matt@50.153.153.15) |
| 18:54.03 | *** join/#asterisk slav3_kitten (~frankthet@unaffiliated/slav3-kitten/x-0866809) |
| 19:03.40 | dan_j | I'm trying out pjsip but when the endpoint approaches expiration and tries to re-register, it gets told that there are too many endpoints and therefore the endpoint loses its registration. Any idea whats wrong? |
| 19:03.41 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 19:04.59 | [TK]D-Fender | It didn't wait until actually expired and I suppose it doesn't count it as "the sameZ". |
| 19:05.10 | [TK]D-Fender | you could probably get around this by limiting to 2 |
| 19:05.55 | *** join/#asterisk saratogga (~saratogga@wsip-98-174-232-77.ph.ph.cox.net) |
| 19:06.23 | dan_j | If the expiry time is set differently on the endpoint and on asterisk, do they not negotiate a mutually acceptable expiry/re-register time? |
| 19:06.59 | [TK]D-Fender | Pretty sure eithe side can ignore the other... |
| 19:08.36 | dan_j | Ok. So if asterisk is set to 3600 and the endpoint is set to 300, it's possible that max_contacts=2 could not be enough. |
| 19:09.13 | dan_j | In which case I need to use remote_existing=1 ? |
| 19:10.17 | *** join/#asterisk areski (~areski@80.174.128.67.dyn.user.ono.com) |
| 19:11.34 | [TK]D-Fender | Logically I would say yes... |
| 19:11.45 | [TK]D-Fender | Things could get messy |
| 19:12.11 | [TK]D-Fender | Actually you should do a "remove"\ |
| 19:12.24 | [TK]D-Fender | that would allow it to just kill it's not-yet-expired entry |
| 19:16.17 | *** join/#asterisk areski (~areski@80.174.128.74.dyn.user.ono.com) |
| 19:21.37 | Dpunkt | how can i permit some users to register/invite only from certain subnets using pjsip. i think acls should help me, but i have not found any documentation which seems to help me |
| 19:23.09 | *** join/#asterisk hackeron (~hackeron@gentoo/user/hackeron) |
| 19:29.17 | *** join/#asterisk jeffspeff (~jeff.clay@12.49.160.131) |
| 19:29.55 | igcewieling1 | Dpunkt: you'll find far more docs on chan_sip, can you use that? |
| 19:31.23 | [TK]D-Fender | Dpunkt: ACL's are clearly described in the sample config |
| 19:35.27 | Dpunkt | i got how i create an acl, defining my private subnets, but how to they map to my endpoints? |
| 19:37.45 | *** join/#asterisk MadHatter42 (~MadHatter@unaffiliated/madhatter42) |
| 19:48.47 | Dpunkt | "The ACL module used by res_pjsip. This module is independent of endpoints and operates on all inbound SIP communication using res_pjsip." so i cant permit one endpoint explicit (as would have been posible with chan_sip)? |
| 19:49.41 | dan_j | [TK]D-Fender: Whats the purpose of having max_contacts = more than one ? Can you call one endpoint and it ring multiple devices? |
| 19:49.59 | [TK]D-Fender | yes |
| 19:53.37 | file | Dpunkt, not currently. |
| 19:53.38 | *** join/#asterisk crocodilehunter (~Thunderbi@CPE-121-211-223-68.hhui7.cht.bigpond.net.au) |
| 19:54.25 | dan_j | [TK]D-Fender: Quite cool. Shame I cant rely on max_contacts without possibly causing re-registration issues. |
| 19:54.39 | Dpunkt | ok, thank you file |
| 19:55.43 | file | for a re-registration it should refresh the existing registration, if it's not doing that then it would be considered a new registration |
| 19:56.22 | [TK]D-Fender | Can't restrict individual endpoints? GAH |
| 19:57.33 | dan_j | file: it doesnt appear to be re-registering because the CLI says that max_contacts has been reached when its time to re-register and then the endpoint instantly says unregistered. |
| 19:59.23 | igcewieling1 | file: any idea when chan_pjsip will reach feature parity with chan_sip? |
| 19:59.51 | file | never probably, as there is some stuff that chan_sip has that few people use |
| 20:00.02 | igcewieling1 | dan_j: any chance your NAT router times out the translation so when the phone registers again it gets a new remote port? |
| 20:00.20 | igcewieling1 | file: I doubt acls for peers is one of those never used features. |
| 20:00.30 | file | I never said it was. |
| 20:00.48 | file | Perhaps one day someone will implement it. |
| 20:00.51 | dan_j | igcewieling1: not sure. i'll have to check. i doubt it. it's a decent router. |
| 20:00.54 | igcewieling1 | file: any idea when chan_pjsip will reach reasonable feature parity with chan_sip? 8-) |
| 20:01.09 | file | No idea. |
| 20:01.13 | igcewieling1 | *nod* |
| 20:01.21 | dan_j | igcewieling1: it's not there yet? |
| 20:01.33 | igcewieling1 | dan_j: you've not been following the conversation? |
| 20:01.45 | file | the definition of reasonable is dependent on the deployment and what people need, it's being used already |
| 20:02.11 | dan_j | igcewieling1: sorry. i was afk. what time did you start? |
| 20:02.21 | igcewieling1 | file: *nod* Hopefully most of that stuff is fixed in Asterisk 15 or before. |
| 20:02.29 | file | dan_j, he's referring to ACLs at an endpoint level |
| 20:02.38 | igcewieling1 | dan_j: pjsip does not support having ACLS on devices. |
| 20:03.14 | [TK]D-Fender | it really has to.... this is basic stuff chan_sip has pretty much always had |
| 20:03.35 | file | then why has noone done it? |
| 20:03.43 | igcewieling1 | almost every one of our peers has an acl limititing it to the static IP of the endpoint |
| 20:04.17 | igcewieling1 | file: I thought EVERYONE used it |
| 20:04.24 | file | used what? |
| 20:04.28 | igcewieling1 | only an idiot doesn't limit their endpoints with ACLs |
| 20:04.51 | igcewieling1 | file: I am referring to permit / deny settings per peer in sip.conf |
| 20:05.01 | dan_j | igcewieling1: our clients access our servers from around the world, making ACLs difficult to implement. |
| 20:05.04 | file | I don't see it that often |
| 20:05.22 | file | and since PJSIP really started getting used I think it's been brought up maybe 5 times |
| 20:05.33 | dan_j | either way, surely a firewall rule is better than using ACLs? |
| 20:05.48 | igcewieling1 | dan_j: *nod* All of our customers are on static IPs. Using registration makes things a lot more complicated. |
| 20:06.28 | dan_j | igcewieling1: nice when you can get it. how do you deal with it if a customer decides he wants to take his endpoint on holiday with him? |
| 20:06.31 | [TK]D-Fender | I always lock my local LAN based devices peers to the local subnet for instance |
| 20:06.42 | [TK]D-Fender | And provider-lock "home" users |
| 20:06.55 | igcewieling1 | Dpunkt: you should be able to hack up some ACL-like stuff in the dialplan. |
| 20:07.42 | igcewieling1 | I use 1) iptables 2) permit/deny 3) dialplan based ACL. |
| 20:07.53 | file | and I don't want to make it seem as though they aren't a good thing in some deployments, ACLs on an endpoint level, but the feedback I've read hasn't shown that a large number of people want them on PJSIP - so they didn't get done |
| 20:08.42 | dan_j | file: if asterisk is not using the existing endpoint for the re-registration, how can I debug it prior to create a bug report? Just simply turn on debug and see what it shouts about? |
| 20:08.45 | igcewieling1 | file: *nod* Perhaps they will be done by the time we consider switching. |
| 20:09.24 | file | dan_j, look at the Contact of the original REGISTER and then the Contact of the supposed re-registration, if they differ then they appear as separate registrations and not a re-register |
| 20:09.25 | dan_j | igcewieling1: we are looking at switching because of the RLS feature |
| 20:09.36 | igcewieling1 | dan_j: RLS? |
| 20:09.48 | dan_j | igcewieling1: https://wiki.asterisk.org/wiki/pages/viewpage.action?pageId=30278158 |
| 20:10.32 | dan_j | igcewieling1: Some clients are using Panasonic KX-UT133 handsets. They seem to struggle when there are too many BLFs set up monitoring peers. |
| 20:10.40 | igcewieling1 | dan_j: interesting. More of a PBX feature than a switch feature, but still cool. |
| 20:11.54 | [TK]D-Fender | Yeah I remember us looking at that a long while back |
| 20:12.00 | Dpunkt | igcewieling1 implementing a acl in the dialplan could be possible but sounds just a little bit better then nothing |
| 20:12.15 | dan_j | [TK]D-Fender: Correct. |
| 20:12.51 | igcewieling1 | Dpunkt: I'd not use it as the only protection. |
| 20:14.31 | Dpunkt | Using iptables to limit traffic to "local" subnets and the providers seems to be a good idea to me now |
| 20:14.56 | igcewieling1 | Dpunkt: (using chan_sip terms) for example, you could compare CHANNEL(peerip) to a database of ACLs |
| 20:15.01 | *** join/#asterisk slav3_kitten (~frankthet@unaffiliated/slav3-kitten/x-0866809) |
| 20:18.24 | [TK]D-Fender | Dpunkt: workable if you don't have roaming users |
| 20:18.30 | Dpunkt | im not sure if that is realy necessary after limiting the traffic with the firewall. till now i split between intern&external hosts but when i limit external to only a handfull I should not have to worry about the whole internet.... |
| 20:18.54 | igcewieling1 | Dpunkt: someday your firewall will be down. |
| 20:20.00 | Dpunkt | copying the firewall rules to global asterisk acl should work then |
| 20:20.18 | dan_j | igcewieling1: surely if a firewall is down, no traffic will route anyway. |
| 20:20.27 | dan_j | igcewieling1: as long as it's a decent firewall |
| 20:20.49 | igcewieling1 | dan_j: in your setup maybe, but doesn't have to be. |
| 20:21.11 | dan_j | igcewieling1: agreed |
| 20:21.27 | igcewieling1 | I don't know of a single asterisk box I manage which does NAT as part of the firewall, most of them are on Public IPs |
| 20:22.03 | Dpunkt | will work on implementing that, thanks for your input |
| 20:55.27 | *** join/#asterisk ChkDigit (~u388mw@74.3.144.66) |
| 20:56.47 | *** join/#asterisk pjensen00 (~per@ip-69-178-218-71.far.ideaone.net) |
| 20:57.26 | *** join/#asterisk jzaw (~jzaw@loki.dzki.co.uk) |
| 21:15.34 | *** join/#asterisk crocodilehunter (~Thunderbi@CPE-121-211-223-68.hhui7.cht.bigpond.net.au) |
| 21:35.09 | *** join/#asterisk infobot (ibot@170-72-44-209.vivintwireless.net) |
| 21:35.09 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.6.0 (2015/10/09), 11.20.0 (2015/10/09); Standard: 12.8.2 (2015/04/08); 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 |
| 21:40.00 | *** join/#asterisk karelk (~karel@31.10.145.202) |
| 21:49.25 | *** join/#asterisk CeBe1 (~CeBe@a81-14-224-229.net-htp.de) |
| 22:07.21 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 22:14.43 | *** join/#asterisk tzafrir (~tzafrir@bzq-179-40-172.cust.bezeqint.net) |
| 22:29.16 | *** join/#asterisk [NC] (~nc@rv1.sabius.net) |
| 22:30.24 | *** join/#asterisk crocodilehunter (~Thunderbi@CPE-121-211-223-68.hhui7.cht.bigpond.net.au) |
| 22:43.43 | *** join/#asterisk MrFidget (~MrFidget@27-32-85-66.static.tpgi.com.au) |
| 22:51.23 | jeffspeff | not sure if this is the best place to ask, but maybe someone can point me in the right direction... how can i monitor an audio stream for a specific match and then kick off a script? |
| 22:54.56 | igcewieling1 | jeffspeff: that is a very cool idea. something like Wildfire? |
| 22:56.39 | igcewieling1 | jeffspeff: Off the top of my head, look into the dialplan app ChanSpy and connect a speech recognition engine of some sort to it |
| 22:56.57 | igcewieling1 | I have no idea if it can be done, but there is a direction to investigate |
| 22:59.34 | jeffspeff | igcewieling1, actually i was thinking of some way to give myself an advantage with radio give aways. you hear them all the time where i live where you have to be caller # X after you hear some specific sound or jingle. i figure i'd automate it and use a call file to connect me to the station. |
| 22:59.57 | jeffspeff | if you know that you're caller 1 - 20 then you're going to win. |
| 23:04.10 | *** join/#asterisk jasonwert (~jasonwert@71.89.137.28) |
| 23:06.08 | *** join/#asterisk thinkgnu (~matt@216.67.183.210) |
| 23:08.09 | WIMPy | You thonk they really count calls? |
| 23:08.53 | jeffspeff | count calls? |
| 23:09.14 | WIMPy | callers |
| 23:09.41 | jeffspeff | like look at a list of caller-id numbers? |
| 23:09.57 | jeffspeff | i figured they just answered line 5 if they're looking for the 5th caller |
| 23:10.04 | WIMPy | No just really count to caller # X. |
| 23:11.35 | WIMPy | And the other calls are not answered? |
| 23:13.18 | jeffspeff | when i've called into them before. sometimes they just ring and ring and i never get through only to hear that someone did and was caller X. other times, the station answers and says "sorry you're caller 3, not 5" or whatever. |
| 23:14.11 | WIMPy | So if they answer, how do you pick the correct channel? |
| 23:14.13 | jeffspeff | my thought is that if the station only has let's say 10 lines and you're instantly on all 10 lines and they're looking for caller # 5 then you win. |
| 23:14.56 | jeffspeff | just have * create several .call files and if they answer it bridges out to your cell # or something. |
| 23:15.19 | *** join/#asterisk elcontrastador (~textual@c-67-182-43-55.hsd1.ca.comcast.net) |
| 23:15.23 | WIMPy | So it fails when they answer all calls. |
| 23:15.23 | jeffspeff | you might answer the phone and hear them say "sorry you're caller 3" then click over and hear them say "you've won!" |
| 23:15.34 | elcontrastador | Greetings |
| 23:16.05 | jeffspeff | WIMPy, when the station answers the call then it dials out to your phone |
| 23:16.16 | WIMPy | And while you hear the sorry message, you hera on the radio that the winner doesn't seem to answer :-) |
| 23:16.27 | jeffspeff | LOL |
| 23:16.33 | jeffspeff | ok, then meetme room |
| 23:17.28 | WIMPy | Do you think you win when they hear their own sorry message? |
| 23:17.44 | elcontrastador | Can I get a quick real world opinion of Asterisk 13 with video phones? Is this something that is performant and robust with, say, Grandstream phones? |
| 23:17.49 | jeffspeff | it calls you and tosses you into a conf room and then each answered call from the call file gets a meeting room |
| 23:17.55 | jeffspeff | LOL |
| 23:18.27 | jeffspeff | WIMPy, this is good. i'm liking this back-and forth. there's a way to do this... |
| 23:19.40 | jeffspeff | elcontrastador, the phone is going to have its own quality etc. regardless of which system you use. asterisk limitations will be with the bandwidth, memory and cpu available on the server which is running it. |
| 23:20.57 | elcontrastador | yes, assuming network and server are peak...Meant Asterisk software functionality and support |
| 23:21.04 | jeffspeff | elcontrastador, you could use direct media and just take * out of the equation and have the call quality depend solely on the the resources available at the two end-points. |
| 23:21.12 | elcontrastador | not professional support... |
| 23:21.24 | jeffspeff | video=yes |
| 23:21.49 | elcontrastador | ah...but dm drops some critical features if i recall. Correct? |
| 23:22.12 | *** join/#asterisk dadrc (~quassel@unaffiliated/dadrc) |
| 23:23.04 | jeffspeff | after the call is established * steps out of the way and doesn't come back into the picture. if by critical features you mean ways to manipulate hangups and redirect callers, etc. then yes. if you don't care what happens after the call is established then no |
| 23:24.02 | elcontrastador | ok, good. Thank you for info. |
| 23:24.42 | elcontrastador | Has anyone used the Grandstream phones well with Asterisk? Performance, quality, and configuration solid? |
| 23:27.57 | lvlinux | elcontrastador: I haven't personally used them but from what I've heard they are fine. They are lower end phones, so the quality isn't up there with some of the more respected brands, but many people like and use them. |
| 23:28.58 | elcontrastador | ok...great. That's good to hear. Thx |
| 23:29.09 | file | is biased to Digium phones |
| 23:29.42 | WIMPy | How much better is the video quality on the Digium phones? |
| 23:29.45 | WIMPy | scnr |
| 23:30.02 | elcontrastador | lol...digium phones dont do vid |
| 23:30.21 | elcontrastador | oh, got it...joke |
| 23:30.50 | WIMPy | Do people actually use video finally? |
| 23:31.17 | elcontrastador | i'm goign to lab this all up and check it out |
| 23:31.42 | WIMPy | So far it has bee a complete fail every time. |
| 23:31.47 | WIMPy | +n |
| 23:32.29 | elcontrastador | got 2k of diff phones coming tomorrow in addition to many digium, polycom, and cisco (skinny) phones. New to v13,, so getting arms around pjsip now |
| 23:32.39 | file | WIMPy, I have no evidence as such... but not from desk phones |
| 23:32.45 | elcontrastador | WIMPy: How so? |
| 23:33.01 | *** join/#asterisk vader- (~Adium@65.199.222.133) |
| 23:33.08 | file | mobile phone and computer moreso I'd say |
| 23:33.11 | WIMPy | Each marketing campaign ended i noone using it. |
| 23:33.33 | elcontrastador | thinking for k12 edu clients. |
| 23:33.40 | file | the experience on a computer is just more feature rich and useful imo |
| 23:33.40 | WIMPy | Yes, on things like skype or chatroulette prorbly. |
| 23:34.16 | WIMPy | Although todays phones are just computers, really. |
| 23:34.43 | *** join/#asterisk jasonwert (~jasonwert@71.89.137.28) |
| 23:35.22 | elcontrastador | the browser on the grandstream could be hopefully manip'd so we can send messages thru a web front end like 'Vistior on campus' or something..gonna see how programmable it is |
| 23:37.14 | elcontrastador | thinking notification system via xmpp and autoanswer paging, etc. I'll just hack around for fun and see what's possible |
| 23:46.26 | *** part/#asterisk kharwell (kharwell@nat/digium/x-ikbfflqrbjsbshvv) |