| 00:03.49 | ddemarteleire | Oh, I think I see |
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| 00:45.26 | lnb | what is a cli command to show something that would require odbc working? |
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| 03:30.43 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: Asterisk 11.11.0 (2014/07/10), 1.8.29.0 (2014/07/10); Standard: Asterisk 12.4.0 (2014/07/10); DAHDI: DAHDI-linux 2.9.0 (2014/01/30), DAHDI-tools 2.9.0.1 (2014/01/31); libpri 1.4.15 (2014/06/16) -=- Visit the official Asterisk wiki: wiki.asterisk.org -=- Read the Code of Conduct bit.ly/1hH6P22 |
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| 04:28.47 | *** mode/#asterisk [+o sruffell] by ChanServ |
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| 04:48.30 | stevePearPear | Hi, |
| 04:55.21 | *** join/#asterisk wonderworld (~ww@ip-62-143-157-238.unitymediagroup.de) |
| 04:55.32 | stevePearPear | I am using MixMonitor to record audio calls in .wav and I realize the file size are quite huge |
| 04:55.42 | stevePearPear | is there a way I could reduce the file size? |
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| 06:18.32 | stevePearPear | Hi, I am using MixMonitor to record audio calls in .wav, I realized the file size is quite huge, is there ayway I could reduce its size? |
| 06:21.39 | [TK]D-Fender | compress it afterwards or record in another format |
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| 07:17.50 | XATRIX | Hi, i'm trying to add localized ivr voice sounds, i took it here http://downloads.asterisk.org/pub/telephony/sounds/ |
| 07:18.17 | XATRIX | Made /var/lib/asterisk/ru directory, and placed files into |
| 07:19.05 | XATRIX | And echo "language=ru" >> /etc/asterisk/sip_general_custom.conf |
| 07:19.33 | XATRIX | Did service restart, but still have en sounds in business... Any ideas how to change it ? |
| 07:20.38 | RadJackson | HI , while chanspying on a channel , i want to remind the spyer to type 9 in order to exit ChanSpy, i want to do that every 9 seconds |
| 07:21.06 | RadJackson | ChanSpy =>[After 9sec] => Playback(File) => ChanSpy=>[After 9sec] => ... etc |
| 07:21.11 | RadJackson | What is the best way to do this? |
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| 07:31.31 | XATRIX | Ok, i've made it |
| 07:32.17 | r00f | XATRIX, were you successfull yesterday? i had to leave, did not see how it ended |
| 07:33.09 | XATRIX | r00f: yes, i fixed my pbx, i had problem with domain. I was using 'parkovka.ua' but my DNS domain for this server is pbx.parkovka.ua |
| 07:33.27 | XATRIX | So when i was trying to connect to pbx it had a conflict |
| 07:33.37 | r00f | i c |
| 07:33.56 | XATRIX | I removed my Organization, created a new one, with a domain pbx.parkovka.ua - and now everything works as a charm |
| 07:34.16 | XATRIX | Now i'm searching a way how to disable voicemail... |
| 07:35.23 | XATRIX | If my recepient is unavailable it offers me to leave a voicemail to. I'd like to disable it |
| 07:36.06 | XATRIX | Globally |
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| 07:37.07 | r00f | well i guess it's somewhere in elastics config. i have no experience with it |
| 07:38.40 | XATRIX | Yea |
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| 08:02.22 | stevePearPear | Hi, I am using MixMonitor to record audio calls in .wav, I realized the file size is quite huge, is there ayway I could reduce its size? |
| 08:02.36 | stevePearPear | earlier Fender mentioned another format, is there any hint which format is smaller? |
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| 08:02.53 | stevePearPear | if I compress it later, does Asterisk have a DialPlan function that I could checkout? |
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| 08:24.41 | WIMPy | Start an external script when it's finished. |
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| 10:08.55 | andycol | Hi Guys, |
| 10:09.01 | andycol | I need some help setting up a tdm410p |
| 10:09.09 | andycol | for some reason dahdi show channels does not show any info |
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| 11:14.54 | nuken | hi guys |
| 11:15.22 | nuken | i have a gateway FXO Planet VIP-480FO, that connects perfect in asterisk 1.6 |
| 11:16.00 | nuken | but in asterisk 1.8 the it can't register |
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| 11:46.00 | andycol | dahdi show channels show no channels |
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| 12:05.45 | h8 | hi everyone |
| 12:06.29 | h8 | I'm trying to create a ConfBridge with multiple rooms, I have the following setup: http://pastebin.com/YT4nw7xX , could anyone look it over if it looks right? I'm sensing that it might not be okay, but I can't explain the problem. Some password are read from the password file, some are not |
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| 13:43.58 | michael_work | hey |
| 13:44.10 | file | hi |
| 13:44.23 | michael_work | anyone here has a good experience with webrtc + dtls + asterisk + any js lib on chrome? |
| 13:44.56 | michael_work | there is some info over internet but i think most of it outdated |
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| 13:45.44 | michael_work | i tried to use sipml5. with ws i get chan_sip.c:10535 process_sdp: Rejecting secure audio stream without encryption details: audio XXXXX UDP/TLS/RTP/SAVPF 111 103 104 0 8 106 105 13 126 error |
| 13:45.51 | michael_work | with wss i faield to register |
| 13:46.02 | michael_work | i'm trying with asterisk 11.11.0 |
| 13:46.50 | file | there's a tutorial at https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5?src=search |
| 13:53.02 | michael_work | thanks |
| 13:53.03 | michael_work | checking |
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| 14:17.24 | Katty | SO. |
| 14:17.28 | Katty | it's someone's birthday. |
| 14:17.31 | Katty | can you guess who it is?! |
| 14:19.13 | file | yup |
| 14:19.17 | file | he's gone though |
| 14:19.22 | Katty | :< |
| 14:19.31 | Katty | post gocart recovery? |
| 14:19.38 | file | he's been gone alllllll week |
| 14:20.48 | Katty | oh. |
| 14:20.52 | Katty | i had no idea! |
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| 14:21.32 | file | Katty, let's surprise him NEXT week |
| 14:21.35 | michael_work | file, thanks |
| 14:21.38 | michael_work | you are my here |
| 14:21.47 | file | michael_work, pass the knowledge on. |
| 14:21.52 | michael_work | found one missing parameter ( ithink that this is one) |
| 14:22.18 | michael_work | it was force_avp |
| 14:22.33 | michael_work | :s/here/hero/gc |
| 14:22.50 | michael_work | no bot :( |
| 14:23.31 | michael_work | not that |
| 14:23.33 | michael_work | hmm |
| 14:23.50 | michael_work | now i need to find what was it i changed to make it work :P |
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| 14:24.44 | michael_work | it seems to be type=peer vs type=friend |
| 14:24.53 | michael_work | it works when i use friend |
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| 14:27.52 | michael_work | file, am i mad? or it can be really realted? as i switch to peer i get the "chan_sip.c:10142 process_sdp: Received SAVPF profle in audio offer but AVPF is not enabled, enabling.... chan_sip.c:10535 process_sdp: Rejecting secure audio stream without encryption details...." |
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| 14:29.27 | file | I don't know if I've tried that but I suppose it is possible... a full sip debug might yield information |
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| 14:33.24 | michael_work | file, i see that other server it is friend and i get only the second part of error. I might say we are not there yet. But as next week i'll try to play with all possible options i'll be more sure and maybe report problems if i find them |
| 14:33.30 | michael_work | file, thanks anyway |
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| 14:33.45 | michael_work | last time i checked the wiki it was outdated |
| 14:34.18 | michael_work | and btw for sipml5 there is dirty hack to turn off dtls and it works (as it worked before dtls) but on chrome only as i understand |
| 14:34.28 | file | and only for now |
| 14:34.33 | file | it will be removed |
| 14:36.41 | michael_work | yeah i know |
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| 14:44.58 | alami | file: i want let poeple chose it's own conference number and pin, it's better to make a phg web page where they can enter this imformation and the script paste it to extensions.conf or can you suggest an idea? |
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| 14:51.39 | RadJackson | Hello ive got a dynamic feature that runs a Macro while a DIal, when the macro exits it sends to another context2 |
| 14:51.52 | RadJackson | I cant catch the H extensions on the context2 , any idea ? |
| 14:51.53 | RadJackson | http://pastebin.com/rfFSC7Ge |
| 14:52.54 | RadJackson | when hanging up all it displays is : == Spawn extension (CONTEXT2, s, 1) exited non-zero on 'SIP/renta2-0000aedf' |
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| 15:01.29 | [TK]D-Fender | RadJackson: macro merge with the calling context and may inhereit. Use a hangup handler from * 11+ |
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| 15:50.38 | RadJackson | [TK]D-Fender how to macro merge ? |
| 15:57.47 | [TK]D-Fender | RadJackson: Given it's triggered from a feature code I'm not sure. The rest is documented. Either way test what I have already recommended |
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| 17:15.24 | *** join/#asterisk bibz (~bibz@router.ais.at) |
| 17:15.27 | bibz | hi guys |
| 17:15.36 | bibz | another day, another question.. |
| 17:15.43 | bibz | after about 6 seconds the call gets hung up |
| 17:16.05 | bibz | the debug says that a timeout has been reached and no response was made to the critical packet |
| 17:16.10 | bibz | I'm behind a NAT |
| 17:16.14 | [TK]D-Fender | sounds like improper NAT setup |
| 17:16.15 | bibz | I've specified this in the sip.conf |
| 17:16.26 | bibz | and forwarded the ports 5060 and 10000-200000 |
| 17:17.12 | [TK]D-Fender | pastebin the call with SIP debug enable |
| 17:17.20 | [TK]D-Fender | the COMPLETE call.. beginning to end |
| 17:17.36 | bibz | ok, hold on just a sec |
| 17:18.25 | *** join/#asterisk wonderworld (~ww@34-231.197-178.cust.bluewin.ch) |
| 17:19.50 | *** join/#asterisk evilman_work (~evilman@87.244.6.228) |
| 17:25.10 | *** join/#asterisk bipolar (~bipolar@offsite.guru) |
| 17:26.01 | bibz | log is too big to copy.. arr |
| 17:28.34 | bibz | [2014-07-17 19:09:20] WARNING[2448] chan_sip.c: Retransmission timeout reached on transmission 43c685da1e5ca618318660d81ec4c48b@212.125.39.78 for seqno 102 (Critical Response) |
| 17:28.34 | bibz | Packet timed out after 6400ms with no response |
| 17:28.53 | bibz | I don't even know that IP.. what the.. |
| 17:35.21 | bibz | http://pastebin.com/HMtctgxJ |
| 17:35.25 | bibz | here is the full call-log |
| 17:42.55 | [TK]D-Fender | bibz: Reliably Transmitting (NAT) to 213.164.25.150:5060: INVITE sip:+436644762522@node1.ipaustria.at SIP/2.0 |
| 17:43.09 | [TK]D-Fender | bibz: Your provider is NOT behind NAT. their peer should be "nat=no" |
| 17:43.35 | [TK]D-Fender | bibz: Contact: <sip:436643024277@192.168.3.217:5060> <-- and you are passing them your private IP, not your public. Your [general] nat settings are wrong |
| 17:46.31 | bibz | you're a god |
| 17:46.58 | bibz | thanks. will change nat=no and review the "externaladdr=" setting |
| 17:47.45 | [TK]D-Fender | bibz: "externaladdr=" <_ does not exist |
| 17:48.10 | bibz | externaddr* |
| 17:48.27 | bibz | I mistyped in only here, in the config its "externaddr" |
| 17:49.00 | *** join/#asterisk sgriepentrog (~sgriepent@nat/digium/x-trwbbqunkyqamzdn) |
| 17:49.00 | bibz | what is the correct syntax of externaddr |
| 17:49.10 | bibz | right now i have > externaddr=IP/NETMASK |
| 17:49.17 | [TK]D-Fender | no mask there |
| 17:49.22 | [TK]D-Fender | that sample shows you |
| 17:49.26 | bibz | oh. thats why it doesn't work.. |
| 17:49.36 | [TK]D-Fender | doubt it. |
| 17:53.12 | bibz | ok, that seems to have solved the problem partially |
| 17:53.19 | bibz | the call won't hangup anymore |
| 17:53.26 | bibz | new problem: no sound on both sides |
| 17:54.36 | [TK]D-Fender | pastebin your [general] and peers masking only the secret |
| 17:54.41 | bibz | ok |
| 18:01.27 | bibz | http://pastebin.com/gRJ11G6X |
| 18:02.12 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 18:02.12 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 18:03.09 | [TK]D-Fender | All of your peers should have "directmedia=no" and not "canreinvite" |
| 18:03.37 | [TK]D-Fender | You have MANY ITSP peers taht you are not specifying "nat=no" in there... |
| 18:03.58 | bibz | all of them to nat=no? |
| 18:04.16 | [TK]D-Fender | No ITSP is likely to be behind NAT and require this. It will eventually screw you over |
| 18:05.21 | bibz | ok, great. |
| 18:07.51 | *** join/#asterisk calum_ (~calum_@cpc67428-harg5-2-0-cust142.7-1.cable.virginm.net) |
| 18:11.42 | bibz | ok, strange |
| 18:12.16 | bibz | the call receiver can't hear me, but I can hear him.. |
| 18:14.00 | *** join/#asterisk wonderworld (~ww@34-231.197-178.cust.bluewin.ch) |
| 18:17.46 | bibz | vim sip.con |
| 18:17.51 | bibz | oh, lol. |
| 18:17.53 | bibz | wrong window |
| 18:29.18 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 18:35.43 | bibz | any ideas why there is only one way sound? |
| 18:35.44 | bibz | :/ |
| 18:36.00 | *** join/#asterisk [Outcast] (~Outcast]@50-195-8-201-static.hfc.comcastbusiness.net) |
| 18:38.48 | [TK]D-Fender | pastebin your new call |
| 18:40.03 | TazzNZ | hey all |
| 18:40.12 | mbowie | Moin TazzNZ |
| 18:40.56 | *** join/#asterisk dysinger (~dysinger@cpe-24-94-73-5.hawaii.res.rr.com) |
| 18:42.38 | bibz | you're back :D |
| 18:42.40 | bibz | ok, hold on |
| 18:45.59 | bibz | http://pastebin.com/eqCfERhA |
| 18:46.19 | bibz | btw, TK-DFender: do you have an BTC address? |
| 18:46.27 | bibz | a* |
| 18:47.02 | [TK]D-Fender | Reliably Transmitting (NAT) to 213.164.25.150:5060: |
| 18:47.04 | [TK]D-Fender | INVITE sip:+436644762522@node1.ipaustria.at SIP/2.0 |
| 18:47.10 | [TK]D-Fender | You still did not fix your ITSP peer |
| 18:47.20 | bibz | i did.. |
| 18:47.32 | bibz | hold on, I'll show you my sip.conf |
| 18:47.58 | [TK]D-Fender | Reliably Transmitting (NAT) to 213.164.25.150:5060: <- no |
| 18:48.12 | [TK]D-Fender | whatever you have certainly has not been applied at the very least |
| 18:48.21 | [TK]D-Fender | Also, always be at verbose 10 for call debug |
| 18:48.24 | [TK]D-Fender | you have none there |
| 18:49.13 | bibz | http://pastebin.com/5tuqd8rG |
| 18:50.14 | bibz | excuse me, http://pastebin.com/5tuqd8rG should work now. I didn't typed in the captcha |
| 18:51.09 | [TK]D-Fender | bibz: NONE of your ATMS, nor sipgate specify it |
| 18:51.28 | bsdice | got bitten recently by Deutsche Telekom's habit of establishing a call from non peer IP |
| 18:51.49 | bsdice | which will not work unless one uses allowguest=yes |
| 18:52.15 | bsdice | with careful parsing of SIP header from/to and then jump into incoming context or denied message |
| 18:52.18 | bibz | ok, I have added to nat=no to every peer |
| 18:52.30 | bsdice | plus iptables filtering to keep the russians and chinese away ;-) |
| 19:01.04 | bibz | here is the new call: http://pastebin.com/gSPmFrPS |
| 19:03.41 | bibz | and the sip.conf (without the secrets :P) http://pastebin.com/yRftL7AS |
| 19:04.57 | [TK]D-Fender | Found peer 'atms-2' for '004366488317786' from 212.125.39.78:5060 <--- Transmitting (NAT) to 212.125.39.78:5060 ---> |
| 19:05.03 | [TK]D-Fender | Why are they still listed as NAT? |
| 19:05.16 | bibz | i have no clue, I have nat=no for every peer.. |
| 19:05.24 | bibz | I also reloaded (after every change I reload) |
| 19:05.52 | bibz | in [general] there is nat=force_rport,comedia .. is this needed? |
| 19:15.04 | bibz | I see I don't have the RTP ports forwarded.. |
| 19:17.51 | *** join/#asterisk theron (~theron@66.220.145.150) |
| 19:18.22 | bibz | NAT is only defined in sip.conf, right? |
| 19:21.13 | bibz | can anyone help me? =( |
| 19:21.43 | *** join/#asterisk netmonk (~af@static-82-85-142-163.clienti.tiscali.it) |
| 19:22.56 | *** part/#asterisk bsdice (~bsdice@meran.embinet.com) |
| 19:23.04 | mbowie | bibz: Might I recommend a little patience... you're getting free assistance on account of the goodwill of a total stranger; don't expect the only thing they're doing today is sitting waiting to respond to you. |
| 19:23.04 | [TK]D-Fender | Have you forwarded the RTP ports now? |
| 19:24.08 | bibz | you're totally right. I'm sorry about that, still learning to be patient.. |
| 19:24.22 | bibz | I have fordwarded the RTP ports |
| 19:25.04 | [TK]D-Fender | What are you using as a NAT/firewall? |
| 19:26.11 | bibz | a TP-Link Switch (TL-ER6020).. |
| 19:26.32 | [TK]D-Fender | ok, retest |
| 19:26.44 | [TK]D-Fender | and make sure it doesn't have any kind of ALG, SIP helper on it |
| 19:27.48 | bibz | ok |
| 19:29.23 | bibz | well, well... every ALG option is ENABLED on the router |
| 19:29.38 | [TK]D-Fender | KILL those |
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| 19:32.46 | marceloamorim | guys, I asked yesterday about the dtmfcidlevel on the chan_dahdi.conf, but I had a problem with this number to receive right cid, In Brazil to receive properly I need to set cidstart=dtmf and cidsignalling=dtmf. Do you guys knows about those trigger and how I should fix power level? |
| 19:33.42 | bibz | ok, still one-way audio |
| 19:34.11 | bibz | will do a trace now |
| 19:39.14 | bibz | here is the new call: http://pastebin.com/Kq8fk1Rv |
| 19:40.16 | *** join/#asterisk D-Boy (~D-Boy@unaffiliated/cain) |
| 19:43.28 | [TK]D-Fender | show screenshots of your forwarding and pastebin rtp.conf |
| 19:43.34 | [TK]D-Fender | picpaste.com <- for SS |
| 19:43.49 | [TK]D-Fender | So far the SIP negotiation side looks fine |
| 19:44.05 | [TK]D-Fender | So it's down to router / ISP interference |
| 19:47.51 | bibz | rtp.conf: http://pastebin.com/7NgQFekx |
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| 19:50.51 | bibz | rules: |
| 19:50.52 | bibz | http://picpaste.com/rules-aqtYKIkO.png |
| 19:52.52 | [TK]D-Fender | mistake... |
| 19:53.16 | bibz | what? where? the forwarding should be alright |
| 19:53.21 | [TK]D-Fender | Your rtp.conf specifies 10000-20000, your forwarding specifies 10001-20000 |
| 19:53.28 | [TK]D-Fender | you left one out right there |
| 19:53.46 | bibz | one single port could result in one-way audio? damn.. |
| 19:53.52 | [TK]D-Fender | Retest and on failure picpaste the ALG page |
| 19:54.40 | [TK]D-Fender | Also those should all be just UDP |
| 19:56.02 | bibz | ok, now no call gets through anymore.. |
| 19:57.17 | bibz | ok, calls working again, still one way audio |
| 19:57.47 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-28-58.cablep.bezeqint.net) |
| 19:59.07 | [TK]D-Fender | What are you using on the client-side? |
| 19:59.17 | bibz | http://picpaste.com/alg-dbaTTbkN.png |
| 19:59.48 | bibz | a mobile phone |
| 20:00.14 | [TK]D-Fender | Testeach end independently now |
| 20:00.15 | bibz | I'm calling a number which is pointed at the asterisk server. the asterisk receives the call and "forwards" it to another mobile |
| 20:00.48 | [TK]D-Fender | just have it reach voicemail to see if the provider side is bad |
| 20:01.29 | bibz | you mean for example turning off one phone and trying to call it? |
| 20:03.40 | bibz | the mobile phones seem to be alright |
| 20:04.53 | [TK]D-Fender | The last thing I saw loked like 1 * calling another. |
| 20:04.57 | [TK]D-Fender | What is that other box? |
| 20:05.59 | bibz | i have only one "box", the asterisk server.. |
| 20:06.00 | bibz | o.O |
| 20:06.44 | [TK]D-Fender | Odd that the US from your inbound call says "Asterisk" as well |
| 20:06.49 | [TK]D-Fender | UA |
| 20:09.43 | *** join/#asterisk ttyS3 (~o@gateway/tor-sasl/omlib) |
| 20:10.02 | bibz | hm.. I'm thinking about connecting the asterisk server directly to the internet |
| 20:10.10 | bibz | so its not behind a NAT |
| 20:10.14 | bibz | maybe NAT is causing this problems |
| 20:11.55 | [TK]D-Fender | msee if you router is randomizing ports. |
| 20:12.29 | [TK]D-Fender | there may be a checkbox for this (you don't want that). Then yes you should at least swap that router as the next test\ |
| 20:13.06 | bibz | âWarning: 399 213.235.212.90 "detected NAT type is symmetric NAT" |
| 20:13.11 | bibz | what does this mean? o.O |
| 20:16.53 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 20:18.51 | [TK]D-Fender | BAD |
| 20:19.08 | [TK]D-Fender | <PROTECTED> |
| 20:19.13 | [TK]D-Fender | swap that thing out |
| 20:25.33 | bibz | thanks, will try that out.. |
| 20:26.54 | bibz | I'm reading, that enabling ALG will solve that.. is that true? you advised me to disable it... |
| 20:30.27 | [TK]D-Fender | it usually screws stuff up bad enough on its own |
| 20:30.31 | [TK]D-Fender | swap & test |
| 20:30.35 | [TK]D-Fender | out for now... |
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| 23:03.37 | mbowie | Forced to use teh Googles... again! |
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| 23:10.44 | file | asteriskdocs is going down temporarily, so if anyone needs it... make it snappy |
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| 23:34.35 | snadge | is a2billing offtopic here? |
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