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| 02:18.19 | EdwinGrubbs | ~itsplist-us |
| 02:18.19 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.broadvoice.com, http://www.jnctn.com, http://www.sipstation.com, http://vitelity.net, http://voip.ms and http://flowroute.com |
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| 04:31.57 | Arkanos-Morph | How can one prevent brute force attack on asterisk server if using iptables is not an option? |
| 04:37.35 | scv | use secure passwords and acl functionality |
| 04:37.39 | scv | and ignore the brute force |
| 04:37.43 | scv | fuck |
| 04:37.47 | scv | er damn wrong window |
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| 10:02.31 | wyoung | hi |
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| 12:53.58 | elyob_ | Ooops, don't ask question having just called when it drops all connections :) |
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| 13:06.34 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.8.2 (2016/04/20), 11.22.0 (2016/03/29); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.5.0 (2016/03/28) -=- Asterisk wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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| 14:21.45 | davlefou | Bonjour, j'ai vue des irv ou l'utilisais la voix pour naviguer et non les touches du téléphone. |
| 14:22.41 | [TK]D-Fender | davlefou, Yes, and the various speach recognition engines are well documented and easily google-able. |
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| 14:24.10 | davlefou | [TK]D-Fender, auriez vous un exemple à me donner? |
| 14:26.11 | [TK]D-Fender | davlefou, https://www.google.ca/#q=asterisk+speech+recognition <- SAMPLE GOOGLE SEARCH |
| 14:29.52 | davlefou | [TK]D-Fender, merci. |
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| 15:54.26 | syadnom | hi all. I've got a client connected to my hosted asterisk. They are getting some higher latency and jitter. I already have jbenable=yes, jbimpl=adaptive in sip.conf to try to compensate, and I'm using g.711 |
| 15:54.44 | syadnom | any advice on increasing jitter buffer or using a different codec to improve the audio? |
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| 16:20.00 | Lope | I've got a call coming in, I'm playing a greeting, then putting them into a queue, then afterwards offering them to leave a voicemail. I'd like to record only the queue call, just the call between the caller and the member, none of the sound file greetings, announcements etc. What do you suggest for that? |
| 16:20.24 | Lope | Can someone perhaps share a mixmonitor example config for queues.conf that would do that? |
| 16:20.58 | [TK]D-Fender | Use itt OUTSIDE of the queue and tell it to only record bridged calls |
| 16:22.03 | Lope | does bridged mean the caller is connected to a member? |
| 16:22.12 | Lope | and not bridged is the caller connected to * only? |
| 16:22.34 | [TK]D-Fender | correct |
| 16:23.34 | Lope | I see the sample file for asterisk 13 suggests I can just specify a monitor-format and then MixMonitor will automatically be started and stopped during the bridged period. |
| 16:24.02 | [TK]D-Fender | And the result is....? |
| 16:24.31 | Lope | I'll try it tomorrow, I've already disconnected from my * box for the night. |
| 16:24.36 | Lope | Thanks for the tip :) |
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| 16:29.43 | Lope | Do you know what's going on with all of these brackets and symbols? Set(MONITOR_OPTIONS=av(<x>)V(<x>)W(<x>)) |
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| 16:33.57 | Lope | Is there any way to rename the file such that I can embed the members username into the MONITOR_FILENAME? |
| 16:34.21 | Lope | I'm going to try setting this before the queue starts: Set(MONITOR_FILENAME=${STRFTIME(,,%Y%m%d%H%M%S)}-${user}-${extension}@${ctx}.gsm) |
| 16:34.32 | xp-joncaum | dusts off a kitchen towel and slaps it at xp-joncaum |
| 16:35.43 | Lope | I'll also set this envvar, for a test: Set(MONITOR_EXEC=mv /var/spool/asterisk/monitor/^{MONITOR_FILENAME} /tmp/^{MONITOR_FILENAME}) |
| 16:36.39 | Lope | But that's all happening before the queue starts, and I expect that MONITOR_EXEC command will run as the queue exits. So I don't see any way to inject the member name into the filename? |
| 16:38.13 | morsing | What's the 'enter' key for typing in your mailbox password? I've tried nothing, # and * but neither does anything |
| 16:39.13 | Lope | morsing: confirm that your DTMF's are actually getting through, |
| 16:40.27 | Lope | try putting this in [general] of sip,conf and under all your softphones and peer defs: dtmfmode=rfc2833 and set it in your softphone options if you can, |
| 16:41.41 | Lope | morsing: to test your DTMFs are going through, try this in your extensions.conf same => n,Read(choice,,1,,1,7) same => n,SayDigits(${choice}) |
| 16:42.18 | Lope | It should speak your digit that you press to you. |
| 16:43.49 | morsing | Lope: Thanks, will give it a go |
| 16:45.22 | xp-joncaum | I've seen an issue on 3 servers now where Asterisk 13.8.1 will segfault: segfault at 154 ip 00007f1468524a4c sp 00007f144a6ca1c0 error 4 in app_queue.so[7f1468511000+39000] |
| 16:45.45 | xp-joncaum | What info can I gather that will be helpful in determining what is causing this segfault more specifically? |
| 16:46.16 | xp-joncaum | Or is this question better for asterisk-dev? |
| 16:46.29 | [TK]D-Fender | xp-joncaum, you should upgrade immediately and confirm against current |
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| 16:47.09 | Lope | TKD: any ideas regarding my question? |
| 16:47.58 | Lope | I'd like to inject the queue member name into the mixmonitor filename? |
| 16:48.02 | [TK]D-Fender | Where do you actually see those? You pasted some fragment of something... with no context |
| 16:48.39 | xp-joncaum | Ok, will do. Thank you! I do see 13.8.2 available in repos. |
| 16:49.11 | xp-joncaum | I pasted that info from /var/log/messages, btw. |
| 16:50.58 | Lope | I got the idea for those from the queues.conf sample of asterisk 13.6. They suggest that if I set the env vars: MONITOR_FILENAME MONITOR_OPTIONS and MONITOR_EXEC before running the queue, then set "monitor-format=x" inside queues.conf mixmonitor will automatically save the bridged call to my filename and rename it afterwards etc. |
| 16:53.32 | rrittgarn | for realtime, is res_odbc or res_config_mysql preferred? Having issues with odbc in 13 and am wondering if i'm doing something non-standard. I did try the Maria odbc connector but it fails on certain calls crashing asterisk - so i may need to update my mysql server, just trying to isolate issues. |
| 16:53.38 | Lope | So what I'd like to know is: 1. How can I get the member name after the Queue app completes? I don't see a channel variable for it? https://wiki.asterisk.org/wiki/display/AST/Asterisk+standard+channel+variables |
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| 16:54.52 | Lope | 2. Will MONITOR_EXEC run after the queue app, or after Hangup() ? |
| 16:58.11 | Lope | I'd like to somehow associate the recording with the queue member that the caller spoke to? |
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| 17:06.43 | Lope | I see theres a channel var called QUEUEMEMBER but it contained '' after the queue app finished. |
| 17:09.05 | Lope | I see the Queue app has AGI macro and gosub, each could potentially give me the member name or whatever. Can anyone help with this please? |
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| 17:28.17 | tulga | I see "getaddrinfo("nagios", "51403", ...): Name or service not known" in console. what it mean? I not found anything for nagios on my server. |
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| 17:30.28 | mubbashir | Hi All. My asterisk is sending CANCEL request in a response to 180 ringing packet. can any one please help |
| 17:32.49 | [TK]D-Fender | mubbashir, pastebin the full call with SIP debug enabled |
| 17:32.54 | [TK]D-Fender | and verbose 10 |
| 17:33.15 | [TK]D-Fender | tulga, same with you |
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| 17:37.13 | tulga | [TK]D-Fender: http://pastebin.com/dsK9M7m1 |
| 17:37.45 | [TK]D-Fender | Clearly no SIP debug which I requested |
| 17:37.58 | [TK]D-Fender | [2016-05-03 02:35:18] WARNING[20430]: chan_sip.c:18392 check_via: Could not resolve socket address for 'nagios:47394' |
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| 18:00.31 | [TK]D-Fender | 1 down.... |
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| 18:05.29 | jge | hey all, what's your recommendation for asterisk distros? asteriskNow or FreePBX |
| 18:05.47 | jge | i'm intersted in easy deployment and upgradability |
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| 18:11.33 | acidfu_ | jge look at XiVo |
| 18:13.05 | [TK]D-Fender | FreePBX. |
| 18:13.19 | [TK]D-Fender | It's fully maintained by their project. |
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| 18:14.48 | Guest6910 | Hi, is there a way in dialplan to trigger an action if the agi reaches its 2000ms limit? If the asterisk does not recieve an answer of an agi script. |
| 18:15.30 | Guest6910 | The point is to handle these errors res_agi.c: FastAGI connection to '...' timed out after MAX_AGI_CONNECT (2000) milliseconds. |
| 18:19.36 | Kobaz | having a really weird issue with polycom vvx's |
| 18:20.24 | Kobaz | if i do an 'all page'... basically do a sip add header to auto answer, and then Page(30 endpoints), 3-4 phones will go unreachable after the page |
| 18:20.39 | Kobaz | and this only happens with tcp sip |
| 18:20.49 | Kobaz | with udp sip, i have other problems |
| 18:21.15 | Kobaz | anyone else experience anything similar?... heh |
| 18:21.17 | jge | [TK]D-Fender: thank you, I've been reading the team behind FreePBX is also involved in the AsteriskNow project since the cetos redhat purchase took place.. do these projects have any relation? |
| 18:25.12 | [TK]D-Fender | Which? |
| 18:25.51 | Kobaz | [TK]D-Fender: i'm thinking this is a firmware issue... i've been updating to every new release when they come out because of all the bugs |
| 18:26.13 | Kobaz | BLFs were pretty broken a few releases back on VVX |
| 18:35.38 | *** join/#asterisk netman (~netman@185.94.249.77) |
| 18:36.30 | *** join/#asterisk MajesticFudgie (~thomas@81.135.231.20) |
| 18:36.34 | MajesticFudgie | o-o |
| 18:36.42 | MajesticFudgie | \o/ |
| 18:36.59 | MajesticFudgie | If there's anyone about, I've followed the Asterisk docs on setting up voicemail and such |
| 18:37.09 | *** join/#asterisk Arkanos-Morph (~Arkanos-M@61.12.72.138) |
| 18:37.09 | MajesticFudgie | The docs say about entering the voicemail number AND password. |
| 18:37.09 | *** join/#asterisk tzafrir (~tzafrir@bzq-179-40-172.cust.bezeqint.net) |
| 18:38.21 | MajesticFudgie | Is there a way to skip this. I've noticed pudding VoiceMailMain(6002@default) in extensions.conf for voicemail works for accessing mailbox 6002 where 6002 is also set as my extension number |
| 18:38.35 | MajesticFudgie | But this would mean all users would end up trying to access mailbox 6002 |
| 18:38.53 | MajesticFudgie | I've tried ${CALLERID(num)}@default and it get ignored |
| 18:45.01 | [TK]D-Fender | that will take the CALLERID number from the call and pass that on.... |
| 18:45.19 | [TK]D-Fender | So if it isn't there.. then you didn't set the callerID for your device definition |
| 18:45.33 | MajesticFudgie | hmkay |
| 18:45.42 | MajesticFudgie | Is there a way to get the extension of the caller |
| 18:45.55 | MajesticFudgie | atm Im just using SIP stuff |
| 18:46.00 | [TK]D-Fender | no such thing as "extension of the caller" |
| 18:46.07 | [TK]D-Fender | that words are not actually a "thing" |
| 18:46.24 | MajesticFudgie | oh okay |
| 18:46.38 | [TK]D-Fender | A device description has the PEER name, and there is a callerid as well. There could be other variables set (at elast in chan_sip) |
| 18:46.44 | MajesticFudgie | ah |
| 18:46.59 | [TK]D-Fender | CallerID is perfectly fair to use... butt you do have to set it |
| 18:47.05 | MajesticFudgie | ah |
| 18:47.11 | MajesticFudgie | How would I set it? |
| 18:47.14 | [TK]D-Fender | in your peer |
| 18:47.25 | [TK]D-Fender | it's a basic parameter which is shown in the cample config. |
| 18:47.27 | [TK]D-Fender | sample* |
| 18:47.31 | [TK]D-Fender | callerid= |
| 18:48.22 | MajesticFudgie | Which config am I looking to edit? |
| 18:48.26 | [TK]D-Fender | SIP |
| 18:48.33 | [TK]D-Fender | that is what defines the device |
| 18:48.45 | MajesticFudgie | ah |
| 18:48.53 | MajesticFudgie | I'll pastebin what I've got now |
| 18:49.16 | MajesticFudgie | http://pastebin.com/2E5vbXXb |
| 18:49.34 | MajesticFudgie | I assume under each firneds_internal section I'd set the callerid there? |
| 18:50.18 | MajesticFudgie | friends* |
| 18:51.23 | [TK]D-Fender | no |
| 18:51.31 | *** join/#asterisk theron (~theron@2620:10d:c091:200::c:b90) |
| 18:51.51 | [TK]D-Fender | because that is obviously shared amongst all of those devices using it since it is a template |
| 18:52.02 | [TK]D-Fender | You would want it unique to the device |
| 18:52.06 | MajesticFudgie | I was referring to the sub sections |
| 18:52.13 | MajesticFudgie | e.g. under [thomas](friends_internal) |
| 18:52.31 | [TK]D-Fender | then that is under [thomas] specifically, so yes |
| 18:52.37 | MajesticFudgie | ah |
| 18:52.39 | MajesticFudgie | Ill try that |
| 18:53.53 | MajesticFudgie | Right I've done this, would the variable I were using work now? |
| 18:54.31 | MajesticFudgie | YAS! it's working |
| 18:55.27 | MajesticFudgie | While I'm here, is there somewhere I can find a list of default sounds to play using Playback |
| 18:55.45 | MajesticFudgie | e.g. I use Playback(vm-nobodyavail) for one extension |
| 18:56.27 | [TK]D-Fender | The sound files are all in the sounds folder right there for you to see |
| 18:56.32 | [TK]D-Fender | it's not a hidden list. |
| 18:56.37 | MajesticFudgie | Where can I find the sounds folder |
| 18:56.55 | [TK]D-Fender | under the VARLIB folder you have configured |
| 18:57.07 | [TK]D-Fender | "core show settings" will confirm it to you |
| 18:57.35 | MajesticFudgie | ah, ty |
| 18:58.14 | MajesticFudgie | hm, I dont see any, /var/lib/asterisk/sounds only has an empty directory named custom |
| 18:58.15 | [TK]D-Fender | so [varlib]/sounds |
| 18:58.35 | [TK]D-Fender | is that what core show settings tells you? |
| 18:58.38 | MajesticFudgie | yup |
| 18:58.46 | [TK]D-Fender | you may not have installed them at all then |
| 18:58.50 | MajesticFudgie | I used apt-get to install asterisk btw, idk if that does something different |
| 18:59.05 | [TK]D-Fender | IIRC debian packages those all separately |
| 18:59.13 | [TK]D-Fender | asterisk-soundsXXXXXXX |
| 18:59.19 | [TK]D-Fender | by language & format |
| 18:59.35 | MajesticFudgie | ah |
| 18:59.50 | *** join/#asterisk tulga (67395c2e@gateway/web/freenode/ip.103.57.92.46) |
| 19:01.08 | MajesticFudgie | Wheres it pulling "vm-nobodyavail" from at the moment? |
| 19:02.01 | [TK]D-Fender | not sure offhand. Are you infact hering it? |
| 19:02.09 | MajesticFudgie | yup |
| 19:02.31 | [TK]D-Fender | PB the settings dump, folder file list, etc. |
| 19:02.33 | [TK]D-Fender | ~pb |
| 19:02.36 | infobot | A "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://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 19:02.38 | [TK]D-Fender | ^^^ |
| 19:02.42 | tulga | Hi [TK]D-Fender , please check this nagios issue http://pastebin.com/XMHZEz2t |
| 19:03.26 | [TK]D-Fender | Contact: <sip:checksip@nagios:51156> |
| 19:03.36 | [TK]D-Fender | that is clearly a garbage CONTACT it is passing |
| 19:03.39 | [TK]D-Fender | fix it |
| 19:04.05 | MajesticFudgie | http://paste.ubuntu.com/16194817/ - Should be a default untouched config |
| 19:04.07 | tulga | is it configured in our extension's phone? or in server? |
| 19:05.22 | [TK]D-Fender | tulga, that is NAGIOS. |
| 19:05.25 | [TK]D-Fender | it is not a "phone" |
| 19:05.37 | [TK]D-Fender | unless something funky is going on. |
| 19:05.50 | [TK]D-Fender | And there is no such thing as "default untouched config" |
| 19:06.03 | MajesticFudgie | Well, rephrase. I've not touched it :P |
| 19:06.09 | *** join/#asterisk zapata (~zapata@2a02:b18:581:10:25ff:ad95:b8bb:db90) |
| 19:06.17 | tulga | but I haven't any nagios in my network. I find whole day, nothing found in my server |
| 19:07.15 | [TK]D-Fender | tulga, And you aren't paying attention. You said "not on your netwrok" as if that was something you had to search for. |
| 19:07.15 | tulga | <--- SIP read from UDP:202.131.226.137:51156 ---> this IP sending it to me? |
| 19:07.15 | [TK]D-Fender | tulga, <--- SIP read from UDP:202.131.226.137:51156 ---> <--- the packet dumps TELLS you where it's coming from |
| 19:07.34 | [TK]D-Fender | So if that's the outside... then if you don't know what that IP is then it is "jus tsomeone" out there |
| 19:07.56 | [TK]D-Fender | MajesticFudgie, PB the actual command I gave you. |
| 19:08.16 | MajesticFudgie | The settings command? |
| 19:08.21 | [TK]D-Fender | MajesticFudgie, because the (!) on there actual DEACTIVATES all those settings below |
| 19:08.29 | [TK]D-Fender | "core show settings" |
| 19:08.32 | tulga | Thanks [TK]D-Fender , I will find him |
| 19:08.37 | [TK]D-Fender | that's why I never told you to look at asterisk.conf at all. |
| 19:08.50 | MajesticFudgie | lol I assume settings config |
| 19:08.54 | MajesticFudgie | assumed* |
| 19:09.00 | MajesticFudgie | http://pastebin.com/dUUPyxg5 |
| 19:09.44 | [TK]D-Fender | <PROTECTED> |
| 19:09.50 | [TK]D-Fender | well that is a confirmation |
| 19:10.01 | [TK]D-Fender | so lets see the folder in full and the call in full |
| 19:11.13 | *** join/#asterisk infernix (nix@unaffiliated/infernix) |
| 19:12.11 | MajesticFudgie | okay |
| 19:13.32 | MajesticFudgie | http://pastebin.com/d07naEEv - Thats the call that plays the message |
| 19:13.40 | MajesticFudgie | The only purpose of the call is to play a message and hangup |
| 19:14.19 | MajesticFudgie | https://surl.im/i/25tgs.png is the /var/lib/asterisk dir I've tried to list all the directories I deemed important |
| 19:14.38 | MajesticFudgie | Ignore I'm logged in as root, I'm still in the middle of setting it all up |
| 19:16.00 | [TK]D-Fender | "ls -la /var/lib/asterisk/sounds" |
| 19:17.03 | MajesticFudgie | https://surl.im/i/k5aag.png |
| 19:17.44 | [TK]D-Fender | that is curious.... |
| 19:17.54 | [TK]D-Fender | I'm wondering myself now if this is the same system.... |
| 19:18.09 | MajesticFudgie | lol |
| 19:18.23 | [TK]D-Fender | you sure this is actually the same box? |
| 19:18.26 | [TK]D-Fender | (just in case) |
| 19:18.30 | MajesticFudgie | Yup |
| 19:18.36 | [TK]D-Fender | I have seen people mix up multiple sessions before... |
| 19:18.41 | MajesticFudgie | I've done that before |
| 19:18.46 | MajesticFudgie | Only one box for me to use :P |
| 19:18.47 | [TK]D-Fender | do a "find" on itt... |
| 19:19.04 | MajesticFudgie | I tried find -name "vm-nobodyavail.slin" |
| 19:19.06 | MajesticFudgie | and it came up empty |
| 19:19.45 | *** join/#asterisk Arkanos-Morph (~Arkanos-M@61.12.72.138) |
| 19:19.58 | tulga | [TK]D-Fender: these nagios IPs are public IPs. Should I block them by iptables? |
| 19:20.14 | tulga | I dont understand why this ISP sends nagios request to me |
| 19:21.10 | [TK]D-Fender | If it's your ISP and you don't understand why ... then you should ask them |
| 19:22.56 | MajesticFudgie | Oh, while Im here. I'm using Zoiper on Windows. It's showing 0 voicemail messages when there is one. What am I missing to make Asterisk tell Zoiper that I've unread messages |
| 19:23.39 | [TK]D-Fender | in the SIP peer itself. You have to tell it to announce for a box. |
| 19:23.41 | [TK]D-Fender | mailbox= |
| 19:24.21 | tulga | okay |
| 19:24.22 | MajesticFudgie | ah |
| 19:25.32 | tulga | [TK]D-Fender: what mean "res_musiconhold.c:719 monmp3thread: poll() failed: Interrupted system call"? I dont configure any streaming url and MOH files are only wav format. |
| 19:25.54 | MajesticFudgie | Would I be able to put mailbox=${CALLERID(num)}@default in the [friends_internal](!)? |
| 19:25.55 | [TK]D-Fender | tulga, And I have nothing to comment on yet.... |
| 19:26.01 | MajesticFudgie | Or would I have to do it manually for each peer |
| 19:26.07 | [TK]D-Fender | MajesticFudgie, No, you can't do variables in configs like that. |
| 19:26.15 | MajesticFudgie | ah okie |
| 19:26.21 | [TK]D-Fender | MajesticFudgie, You'll have to do it specifically |
| 19:28.54 | MajesticFudgie | hm |
| 19:28.55 | MajesticFudgie | [May 2 20:28:51] NOTICE[27342][C-00000000]: chan_sip.c:25450 handle_request_invite: Call from 'thomas' (192.168.1.149:35454) to extension 'asterisk' rejected because extension not found in context 'from-internal'. |
| 19:29.02 | MajesticFudgie | When I clicked the voicemail notification in zoiper |
| 19:29.24 | [TK]D-Fender | Your peer poitned to that context... and there is nothing to match "asterisk" as an extension |
| 19:29.36 | MajesticFudgie | Would I need to make an extension named that? |
| 19:29.51 | [TK]D-Fender | Why would you be dialing a "word" like that? |
| 19:29.52 | MajesticFudgie | Or is there a setting that reports the voicemail extension? |
| 19:30.04 | MajesticFudgie | I have no idea ._. |
| 19:30.15 | [TK]D-Fender | It's your call... |
| 19:30.19 | [TK]D-Fender | why is it calling? |
| 19:30.32 | MajesticFudgie | idk |
| 19:30.36 | [TK]D-Fender | Where do you think thatt word came from as far as the call goes? |
| 19:30.45 | [TK]D-Fender | You don't know why your phone just placed a call? |
| 19:30.58 | MajesticFudgie | I click the voicemail icon and it calls asterisk |
| 19:31.01 | [TK]D-Fender | I'm pretty sure it wouldn't place calls out of thin air with no trigger whatsoever |
| 19:31.12 | [TK]D-Fender | then what you set for that button is the issue |
| 19:31.19 | *** join/#asterisk Arkanos-Morph (~Arkanos-M@61.12.72.138) |
| 19:31.30 | [TK]D-Fender | If you set up an extension it should call then you'll have to tell the phone to use that |
| 19:31.59 | MajesticFudgie | hm It's not setup to call "asterisk" for voicemail |
| 19:32.14 | MajesticFudgie | I'm trying to make it use 1 which is the voicemail extension |
| 19:32.16 | [TK]D-Fender | go see where you can set what it will use |
| 19:32.16 | MajesticFudgie | but it ignores it |
| 19:32.41 | [TK]D-Fender | set "vmexten" in your peer as well then |
| 19:32.51 | [TK]D-Fender | and "1" is kinda lazy as #'s go.. |
| 19:33.04 | [TK]D-Fender | I'd recommend something a little more sane much like the CLASS standards for this |
| 19:33.07 | [TK]D-Fender | *98 for example |
| 19:33.08 | MajesticFudgie | Ah |
| 19:33.58 | MajesticFudgie | I notice a lot of providers use 901 |
| 19:34.51 | [TK]D-Fender | Think of something that feels more sane |
| 19:34.57 | [TK]D-Fender | And set that value for tthe peer |
| 19:35.04 | [TK]D-Fender | if you can't do it direct in your phone |
| 19:35.11 | MajesticFudgie | It's working now |
| 19:35.14 | MajesticFudgie | I've changed it to 901 |
| 19:35.49 | *** join/#asterisk Qwell (~north@asterisk/developer/Qwell) |
| 19:35.50 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 19:36.14 | MajesticFudgie | All sorted |
| 19:36.19 | MajesticFudgie | Many thanks for that |
| 19:36.37 | MajesticFudgie | My last question is, I'm wanting to allow outside calling through a landline |
| 19:36.54 | MajesticFudgie | I've read I need a FXS or FXO Card to do so |
| 19:37.20 | MajesticFudgie | How would I make asterisk forward calls when starting a 11 digit number with 9 |
| 19:37.32 | MajesticFudgie | I've setup a simple extension which just plays a message and hangs up |
| 19:38.35 | MajesticFudgie | exten => _9XXXXXXXXXXX,1,Answer() |
| 19:38.42 | MajesticFudgie | Is the extension number I'm using |
| 19:39.09 | *** join/#asterisk theron_ (~theron@2620:10d:c091:200::c:b90) |
| 19:40.45 | *** join/#asterisk Synthase_ (uid63346@gateway/web/irccloud.com/x-snphcaxbdjazknmw) |
| 19:42.27 | *** join/#asterisk Arkanos-Morph (~Arkanos-M@61.12.72.138) |
| 19:42.41 | [TK]D-Fender | Sounds like you already have a match then |
| 19:45.01 | MajesticFudgie | mhm |
| 19:45.33 | MajesticFudgie | How would I forward the call to the landline? |
| 19:48.32 | *** join/#asterisk Arkanos-Morph (~Arkanos-M@61.12.72.138) |
| 19:55.35 | *** join/#asterisk Arkanos-Morph (~Arkanos-M@61.12.72.138) |
| 20:00.56 | [TK]D-Fender | You'd Dial() a device that was configured to call out on it |
| 20:01.03 | MajesticFudgie | ah |
| 20:01.28 | MajesticFudgie | btw if I connect a VOIP Phone to my network, I'd need to log the phone into an extension wouldnt it? Or don't the VOIP Phones do that? |
| 20:02.05 | [TK]D-Fender | Phones AUTH based on a username+password. |
| 20:02.13 | [TK]D-Fender | These get matched against your SIP peer definitions. |
| 20:02.25 | [TK]D-Fender | the word "extension" does not even exist here yet |
| 20:02.33 | MajesticFudgie | ah |
| 20:02.59 | [TK]D-Fender | that is adevice that is identifiable and whose calls are allowed to be processed being sent into a context in your dialplan which determines what they are allow3ed to dial, and processes them accordingly |
| 20:03.28 | [TK]D-Fender | the only time you should use the word "extension" is in reference to the dialplan structure you create to match and process what they actually dial. |
| 20:03.34 | [TK]D-Fender | You do not "connect" to an extension. |
| 20:03.34 | MajesticFudgie | ah okie |
| 20:03.37 | [TK]D-Fender | You dial an extension |
| 20:04.19 | [TK]D-Fender | You "connect" (call) as an authorized device and are calling a specific number. |
| 20:04.36 | MajesticFudgie | ah |
| 20:04.53 | MajesticFudgie | So, the next quiz for me is. If someone calls to the landline device |
| 20:05.06 | MajesticFudgie | FXS/FXO Card in this case |
| 20:05.13 | MajesticFudgie | What happens next o-o |
| 20:06.14 | MajesticFudgie | Judging by what I'm reading, it forwards the call to all numbers in an extension which in turn calls all peers? |
| 20:07.41 | [TK]D-Fender | that device also sends its calls into the dialplan. |
| 20:07.58 | [TK]D-Fender | what happens is all dialplan, therefor exactly what you tell it to do |
| 20:08.05 | MajesticFudgie | ah |
| 20:08.12 | [TK]D-Fender | Which could be anything |
| 20:08.37 | [TK]D-Fender | You could play messages. You could look things up and make decisions based on that. Anything at all |
| 20:08.41 | MajesticFudgie | yeah |
| 20:09.01 | MajesticFudgie | extensions.conf is the dialplan isnt it |
| 20:09.04 | [TK]D-Fender | "John is calling after 9PM again? GTFO". |
| 20:09.10 | [TK]D-Fender | correct |
| 20:09.11 | MajesticFudgie | lmao |
| 20:09.16 | [TK]D-Fender | all call processing = extensions.conf |
| 20:09.22 | MajesticFudgie | So someone calls into the dialplan |
| 20:09.28 | MajesticFudgie | Where does the call go currently |
| 20:09.39 | [TK]D-Fender | depends on what's doing the calling |
| 20:09.42 | [TK]D-Fender | your phones dial #'s |
| 20:09.50 | [TK]D-Fender | those extensions handle it |
| 20:10.05 | MajesticFudgie | Well for example, someone calls my landline on their landline |
| 20:10.19 | [TK]D-Fender | FXo channels don't tend to have a "target number" and are thus sent to a single extension of some sort. This would depends on the specific device you are using. |
| 20:10.24 | MajesticFudgie | They just put in my home number from my provider and it goes into the FXS/FXO port |
| 20:10.33 | [TK]D-Fender | DAHDI-based analog cards send to "s" |
| 20:10.40 | [TK]D-Fender | in the context you ttold them to use |
| 20:10.50 | MajesticFudgie | hmkay |
| 20:10.51 | *** join/#asterisk Arkanos-Morph (~Arkanos-M@61.12.72.138) |
| 20:11.12 | [TK]D-Fender | To an FXO device there is no "number" it is simply "ringing" just like any dumb phone would see. It doesn't know WHY it is ringing. |
| 20:11.16 | [TK]D-Fender | it simply "is" |
| 20:11.18 | MajesticFudgie | yeah |
| 20:11.25 | [TK]D-Fender | And it will react accordingly |
| 20:11.49 | MajesticFudgie | http://paste.ubuntu.com/16195591/ I assume It'd do nothing with this dialplan? |
| 20:11.56 | [TK]D-Fender | Ringing > call sent into the dialplan at whattever Extension it is designed to use |
| 20:12.17 | [TK]D-Fender | Depends what that devices sends into the dialplan |
| 20:12.18 | [TK]D-Fender | ^^^ |
| 20:12.21 | MajesticFudgie | ah |
| 20:12.26 | [TK]D-Fender | and what context |
| 20:12.38 | MajesticFudgie | So I could setup an extension for the device to call to? |
| 20:13.03 | [TK]D-Fender | you WOULD set up one to match where the call would be sent... because you do want to actuall match and process it... |
| 20:13.21 | MajesticFudgie | okay, no idea how to |
| 20:14.22 | [TK]D-Fender | Again it depends on th what the device sends |
| 20:14.31 | MajesticFudgie | I bet I'm frustrating you plenty lol |
| 20:14.42 | [TK]D-Fender | you have to be SPECIFIC about exactly what device you will even be using as your interface to the FXO |
| 20:15.05 | *** join/#asterisk KerioMorgan (~Adium@gw-us.kerio.com) |
| 20:15.10 | MajesticFudgie | Would I configure the device in asterisk or does the device have a tool to configure it? |
| 20:15.29 | [TK]D-Fender | Devices don't magically configure asterisk for you |
| 20:15.39 | MajesticFudgie | lol |
| 20:15.41 | [TK]D-Fender | they are each responsible for themselves |
| 20:15.43 | [TK]D-Fender | that phone line is going to be plugged into a VERY specific piece of hardware. |
| 20:15.55 | [TK]D-Fender | time for you to NAME NAMES |
| 20:16.13 | [TK]D-Fender | "I;m going to use product XYZ for my FXO interface" |
| 20:16.16 | MajesticFudgie | yeah |
| 20:16.28 | [TK]D-Fender | then you can start getting specific answers as to "how" |
| 20:16.54 | MajesticFudgie | yeah |
| 20:17.25 | [TK]D-Fender | There are PC cards that DAHDI supports. There are VoIP gateway devices that take in the line and act similar to a phone and call into your server., etc |
| 20:17.32 | [TK]D-Fender | All are different |
| 20:17.40 | MajesticFudgie | yeah |
| 20:17.48 | MajesticFudgie | I were looking at something like this |
| 20:17.48 | MajesticFudgie | http://www.ebay.co.uk/itm/1-Port-FXS-FXO-Card-tdm110p-Analog-Card-Elastix-Freepbx-Asterisk-Card-x100p-/151780321510?hash=item2356cfe8e6:g:JXoAAOSwAYtWMIxr |
| 20:17.57 | MajesticFudgie | I only need 1 phone line in |
| 20:18.04 | MajesticFudgie | As its a home office setup |
| 20:18.40 | [TK]D-Fender | that is a cheap chinese card that is a hybrid of several more mainstream cards. |
| 20:19.08 | MajesticFudgie | Is there an affordable one with 1 port for incoming landline? |
| 20:19.12 | [TK]D-Fender | It might work, but if you have issues, odds are that you'll have a lot more trouble getting any kind of manufacturer support for it |
| 20:19.29 | *** join/#asterisk CeBe (~CeBe@p4FC1952E.dip0.t-ipconnect.de) |
| 20:20.38 | [TK]D-Fender | most commonly affordable option is something like the "SPA-3102", "SPA-232D", etc |
| 20:22.05 | MajesticFudgie | So something like this? http://www.ebay.co.uk/itm/Multi-Line-DECT-ATA-/231903874022?hash=item35fe8c4be6:g:uI0AAOSwOfxXBC15 |
| 20:22.32 | [TK]D-Fender | possibly |
| 20:22.44 | [TK]D-Fender | I'm not 100% sure you can use the FXO port completely independly on that |
| 20:22.49 | MajesticFudgie | ah |
| 20:23.02 | [TK]D-Fender | You should confirm wiht the MFG on the model. |
| 20:23.06 | [TK]D-Fender | reading the manual... |
| 20:23.11 | MajesticFudgie | I assume a device like that would essentially turn the incoming landline into a VOIP Peer? |
| 20:23.16 | [TK]D-Fender | the SPA-3102 I do know specifically works |
| 20:23.21 | MajesticFudgie | and I'd configure that device to call a specific extension? |
| 20:23.27 | [TK]D-Fender | yes, that is what I had said previously |
| 20:23.31 | MajesticFudgie | kk |
| 20:23.35 | MajesticFudgie | I'm slowly making sense of it all lol |
| 20:24.10 | [TK]D-Fender | So you don't need any special card phycially in the server thus allowing it to be phycially anywhere relative to your server as long as it can reach it via networking somehow |
| 20:24.26 | MajesticFudgie | yeah |
| 20:24.37 | MajesticFudgie | I could put it next to my router then |
| 20:24.43 | [TK]D-Fender | sure |
| 20:24.59 | MajesticFudgie | http://www.ebay.co.uk/itm/Brand-New-Unlocked-Linksys-SPA-3000-VoIP-FXS-FXO-PSTN-SPA3000-Phone-Adapter/190864921830?_trksid=p2047675.c100005.m1851&_trkparms=aid%3D222007%26algo%3DSIC.MBE%26ao%3D1%26asc%3D36382%26meid%3D1606867f098f4bf4b6656dae83e36be2%26pid%3D100005%26rk%3D1%26rkt%3D6%26mehot%3Dpp%26sd%3D141972817227 |
| 20:25.03 | MajesticFudgie | This what you're referring to? |
| 20:25.04 | [TK]D-Fender | Or put on in a foreign counrty to let you use a landline there to call out into that area as a "local" call. |
| 20:26.00 | MajesticFudgie | If I wanted to make an outside call then, I assume I call the devices peer id (insert appropriate terminology)? |
| 20:26.09 | MajesticFudgie | How would I pass the outside lines number |
| 20:27.25 | [TK]D-Fender | Dial(SIP/peername/numbertodial[,timeout][,options]) |
| 20:28.14 | MajesticFudgie | Ah |
| 20:28.46 | [TK]D-Fender | SIP is SIP, so this is also the format you'd use to call using an ITSP. |
| 20:28.51 | MajesticFudgie | yeah |
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| 20:30.02 | MajesticFudgie | Does asterisk support calling a list of extensions at once until one picks up? |
| 20:30.35 | [TK]D-Fender | Again a reminder on the use of the word "extension" |
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| 20:30.56 | MajesticFudgie | or peers |
| 20:30.56 | MajesticFudgie | idk |
| 20:31.00 | [TK]D-Fender | And Dial() can call multiple devices at once as well as : |
| 20:31.02 | [TK]D-Fender | "core show application queue" + queues.conf |
| 20:31.09 | MajesticFudgie | ah |
| 20:31.20 | [TK]D-Fender | Dial is a single call and any looping, etc is up to you making more dialplan code. |
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| 20:31.43 | [TK]D-Fender | Queue() is for call-center style queuei-ing |
| 20:32.19 | MajesticFudgie | yeah |
| 20:32.26 | MajesticFudgie | I'd not bother with that |
| 20:32.51 | [TK]D-Fender | if you just want to ring "a bunch" in one big shot, thena single Dial() is more than enough. |
| 20:33.54 | MajesticFudgie | So Dial(SIP/abc&SIP/xyz,20) |
| 20:33.59 | MajesticFudgie | What flags am I looking at |
| 20:34.17 | [TK]D-Fender | Depends what options you want on there |
| 20:34.27 | [TK]D-Fender | Check them out and see what you have a reason to care about. |
| 20:34.39 | [TK]D-Fender | I can think that "c" might be worthwhile... |
| 20:35.00 | MajesticFudgie | yeah |
| 20:35.04 | MajesticFudgie | I'm looking at it now |
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| 20:35.52 | MajesticFudgie | I think I'll have to return at a later date when I've got the actual hardware for it lol |
| 20:36.45 | [TK]D-Fender | You can still test option just dialing between any kind of device including multiple soft-phones |
| 20:36.55 | [TK]D-Fender | just to get a feel.Every call is jsut a call. |
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| 20:37.20 | MajesticFudgie | true |
| 20:37.44 | MajesticFudgie | If a person dials my house through the device, would it show as the devices caller id? |
| 20:37.53 | MajesticFudgie | Or would it forward the external number |
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| 20:38.57 | [TK]D-Fender | Every kind of device I've seen can send the CID of the calls taht comes into it |
| 20:39.04 | MajesticFudgie | ah okie |
| 20:40.45 | [TK]D-Fender | So unlike your phone peers your one for a gateway device you'd buy would NOT have a set callerid in the peer, but rather allow whatever the peer sends to be taken as-is |
| 20:41.30 | [TK]D-Fender | And on that note.. check-out time here. |
| 20:41.39 | MajesticFudgie | okie |
| 20:41.44 | MajesticFudgie | Many thanks for you help :P |
| 20:41.51 | MajesticFudgie | Helped me understand a lot more lol |
| 20:41.55 | [TK]D-Fender | hope that gets you a bit clearer in your understanding |
| 20:41.56 | [TK]D-Fender | heads home.... |
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| 21:20.42 | jiremek | Hello everybody |
| 21:21.05 | jiremek | I need some help about multiple account with same provider |
| 21:22.44 | jiremek | How to route calls from same provider base on different account with pjsip ? |
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| 21:29.54 | V5_12R\X | jiremek use different endpoints, one for each account. |
| 21:31.26 | V5_12R\X | Hello all, I'm using 13.8.2 on Debian Jessie (digital ocean vps) and compiling using the ./configure --with-pjproject-bundled option and I am getting a failure when building pjsua, notably on the python bindings. |
| 21:31.54 | V5_12R\X | Any word on what python interpreter/libraries that I need to install to get that to work? |
| 21:46.31 | V5_12R\X | OK so it also fails on 13.9.0-rc1 with the --enable-dev-mode configure flag. |
| 21:46.49 | V5_12R\X | jiremek Did you get it figured out? |
| 21:47.23 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 21:51.30 | jiremek | I need some help about multiple account with same provider |
| 21:51.38 | jiremek | How to route calls from same provider base on different account with pjsip ? |
| 21:53.04 | [TK]D-Fender | What does "route calls" actually mean here? |
| 21:53.13 | [TK]D-Fender | "route" is not a word to * |
| 21:53.53 | V5_12R\X | jiremek when you call the Dial application you will enter a dial string as one of it's parameters. That dial string will be user@endpoint, which in your case will be user@endpoint1 and/or user@endpoint2 where endpointN=accountN. |
| 21:55.58 | jiremek | sorry my english is poor, i will try to explain as clear of possible |
| 21:56.07 | jiremek | I Have To account from same provider |
| 21:57.03 | jiremek | All call from the number A should go to user A in context "int" |
| 21:57.04 | [TK]D-Fender | "to"? |
| 21:57.09 | [TK]D-Fender | do you mean *2*? |
| 21:57.22 | scv | i think the real question is whether he's using freepbx :) |
| 21:57.45 | [TK]D-Fender | Your wording is not getting clearer |
| 21:57.54 | [TK]D-Fender | calls go TO a number. |
| 21:57.59 | [TK]D-Fender | Calls come FROM places. |
| 21:58.02 | jiremek | all call from number B should go to user 2 . |
| 21:58.07 | [TK]D-Fender | Calls come in targeting numbers |
| 21:58.14 | [TK]D-Fender | be CLEAR about the direction |
| 21:58.23 | jiremek | with chan sip all is ok. |
| 21:58.24 | [TK]D-Fender | What "number B"? |
| 21:58.31 | jiremek | But now in pjsip |
| 21:58.32 | [TK]D-Fender | is that the phone number the caller DIALED? |
| 21:58.37 | jiremek | it's break |
| 21:58.40 | [TK]D-Fender | or are you talking about THEIR phone number? |
| 21:58.54 | [TK]D-Fender | there are *2* bumbers on calls. |
| 21:59.04 | [TK]D-Fender | the CALLING number, and the CALLED number |
| 21:59.09 | [TK]D-Fender | please get CLEAR fast. |
| 21:59.21 | jiremek | There are 2 public number but not sames usage |
| 21:59.33 | jiremek | all inbound for line 1 go to an user |
| 21:59.38 | [TK]D-Fender | Ok, so these are YOUR 2 numbers that someone might be calling? |
| 21:59.47 | jiremek | yes |
| 22:00.00 | [TK]D-Fender | ok, so let's look at the call |
| 22:01.20 | jiremek | the identify point is same but all calls goes to one user only |
| 22:01.41 | jiremek | or sometimes the 2 users are called |
| 22:01.44 | V5_12R\X | jiremek can you pastebin your working SIP configuration perhaps? |
| 22:01.50 | jiremek | yes |
| 22:01.51 | jiremek | sorry |
| 22:01.58 | V5_12R\X | jiremek no worries |
| 22:02.09 | [TK]D-Fender | forget the configuration |
| 22:02.14 | [TK]D-Fender | LOOK AT THE CALL |
| 22:02.29 | jiremek | I think I found the answer |
| 22:02.32 | V5_12R\X | jiremek I would like to see the configuration if you don't mind, thanks |
| 22:02.53 | jiremek | the match in identify should user user and not the ip of provider |
| 22:03.13 | V5_12R\X | jiremek correct |
| 22:04.02 | jiremek | is it possible to combine user and ip ? |
| 22:04.12 | V5_12R\X | jiremek the endpoint addressing schema (if you will) is rather different in PJSIP. |
| 22:04.46 | V5_12R\X | jiremek If I understand the question, you can use a full SIP uri there, such as sip:user@hostname:5060 |
| 22:05.02 | V5_12R\X | jiremek IIRC that is, check the docs for yourself. |
| 22:05.30 | jiremek | I checked but badly |
| 22:05.57 | jiremek | Ok, the match in identify section allow the full uri ? |
| 22:06.11 | jiremek | My current working conf for one of public number http://pastebin.com/raw/cuWCZ5gG |
| 22:07.46 | jiremek | The 2nd public number http://pastebin.com/raw/WQS0mVuq |
| 22:08.52 | V5_12R\X | jiremek I am using a url in mine, |
| 22:09.06 | V5_12R\X | jiremek match=sip.flowroute.com |
| 22:09.29 | [TK]D-Fender | auth_type=USERB |
| 22:09.34 | [TK]D-Fender | USRB is NOT an authtype |
| 22:09.46 | [TK]D-Fender | You should notice the obvious differences from your first account |
| 22:10.07 | jiremek | Sorry it's userpassword |
| 22:10.22 | jiremek | I maked a mistake |
| 22:11.02 | V5_12R\X | jiremek I'm sure that you can use the entire URL |
| 22:11.07 | V5_12R\X | jiremek "will identify inbound traffic by pulling the user from the "From:" SIP header in the packet" |
| 22:11.17 | V5_12R\X | jiremek https://wiki.asterisk.org/wiki/display/AST/PJSIP+Configuration+Sections+and+Relationships |
| 22:11.22 | jiremek | I will try it. |
| 22:11.55 | jiremek | no that's work but same username with another provider should make some trouble |
| 22:12.05 | jiremek | Thank you for you help |
| 22:12.21 | V5_12R\X | jiremek in the example they use an IP address, I used a DNS record. PSIP tends to accept sip:user@hostname:5060 format though |
| 22:12.29 | V5_12R\X | jiremek no worries good luck |
| 22:12.38 | V5_12R\X | * PJSIP |
| 22:12.48 | jiremek | I've got an other trouble but need first more search |
| 22:14.01 | V5_12R\X | jiremek yeah the auth_type is wrong, I don't actually believe that you need it (I didnt for SIP registration) |
| 22:14.51 | jiremek | I need it for outbound calls |
| 22:19.32 | V5_12R\X | jiremek are you sure? Do you require an md5 password on this or just user/pass auth? If it's just user/pass, thats the default so just remove that line. |
| 22:20.54 | [TK]D-Fender | I fail to see a problem at all |
| 22:21.01 | jiremek | It's effectively an user / pass . so this additionnal instruction isn't need |
| 22:21.30 | [TK]D-Fender | I have seen precisely 0 failures, and one config bit with what look like clear mistakes |
| 22:21.54 | jiremek | Thanks [TK]D-Fender, I first add the provider IP in match of identify section but know Now Used the user so that work |
| 22:22.09 | jiremek | My last trouble is only for one of users. |
| 22:22.33 | jiremek | her need to make hold after answering a call for have sound |
| 22:23.03 | jiremek | Probably du to nat... Need ot search more first |
| 22:24.25 | V5_12R\X | jiremek try adding "ice_support=yes" to the user's endpoint record and reloading pjsip |
| 22:27.13 | jiremek | That not work http://pastebin.com/raw/JxnHwPdr |
| 22:28.37 | V5_12R\X | jiremek remove "force_rport=yes", |
| 22:29.22 | jiremek | It's only ice or force_rport could be the error ? |
| 22:29.31 | V5_12R\X | probably force_rport |
| 22:29.44 | V5_12R\X | but I'll need more information if it doesnt |
| 22:33.16 | jiremek | Will try it in nextly |
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| 22:42.15 | jiremek | V5_12R\X that's worked |
| 22:42.46 | V5_12R\X | jiremek cool! |
| 22:43.10 | jiremek | Yeah, great |
| 22:43.14 | jiremek | thanks a lot |
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| 22:43.32 | V5_12R\X | jiremek no worries |
| 22:43.36 | jiremek | be happy |
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