| 00:00.08 | nny | suppose* |
| 00:00.41 | nny | so I am making a script that will populate calls based on the number of meetme rooms in range where participants=1 |
| 00:00.50 | nny | my call file generation script is bash |
| 00:00.54 | nny | and works pretty well right now |
| 00:01.03 | *** join/#asterisk WiretapWork (~wiretap@unaffiliated/wiretap) |
| 00:01.31 | nny | thinking AMI to pull the list of meetme rooms, grab the first one in the list, and populate the call file with the name of the room for use in my ocntext |
| 00:01.33 | nny | context* |
| 00:01.49 | nny | i understand bash isn't best for socket connections, gonna see what query will work in AMI for that with some sed magic |
| 00:02.37 | *** join/#asterisk gundy (~gabe@69.169.150.25.provo.static.broadweavenetworks.net) |
| 00:02.46 | WIMPy | From bash, 'asterisk -rx' might be easier. |
| 00:02.51 | WIMPy | No. Will. |
| 00:04.10 | carrar | Gettin' Jiggy wit It |
| 00:04.26 | nny | WIMPy: ooh, nice will try |
| 00:04.30 | nny | WIMPy: thanks |
| 00:04.34 | gundy | Hey all, been a while since I've done any Asterisk... just wondering, is Asterisk RealTime Architecture still the best way to get/use dynamic settings? |
| 00:04.36 | nny | why didn't I think of that hahaha |
| 00:08.05 | Bezzie | Hello all. I'm toying with asterisk-gui for a project (seeing if it fits my needs as it's more lightweight than freepbx), but I'm having issues getting asterisk's webserver to let me through. It's set to support static http etc, but giving me 404's and perm denied errors. Does asterisk's webserver log anywhere? Or can I direct it to log? I don't see it in the logs in /var/log/asterisk or when running |
| 00:08.07 | Bezzie | asterisk -rvvv |
| 00:12.11 | *** part/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 00:17.48 | *** join/#asterisk supkrs (~kaushal@14.97.194.232) |
| 00:17.50 | supkrs | Hi |
| 00:18.33 | supkrs | I am running libpri-1.4.11.5-1_centos5 |
| 00:18.53 | carrar | Thats HOT++ |
| 00:22.36 | supkrs | As i see http://downloads.asterisk.org/pub/telephony/libpri/releases/ |
| 00:22.50 | supkrs | can i have rpm version of |
| 00:22.53 | supkrs | LibPRI 1.4.12 |
| 00:22.57 | supkrs | for CentOS 5.6 |
| 00:23.57 | supkrs | since i have installed binary packages using https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages#AsteriskPackages-YUM%28CentOS%2FRedHat%29 |
| 00:25.01 | supkrs | I am having really burning issue |
| 00:25.16 | supkrs | My Entire asterisk system is down |
| 00:26.20 | supkrs | I dont see any updates available when i do yum list updates |
| 00:29.21 | ChannelZ | Is there a fixed bug or something in 1.4.12 that you think would solve all your problems? |
| 00:30.35 | supkrs | I see error 101 while i use 8 PRI Lines |
| 00:30.41 | ChannelZ | Bezzie: are you confusing Asterisk-GUI with Asterisk's built-in mini-HTTP server which is really for serving up some phone configs? |
| 00:30.48 | WIMPy | Ah, that one again. |
| 00:30.56 | ChannelZ | (still) |
| 00:31.00 | Bezzie | Well, asterisk gui uses the built in http server |
| 00:31.03 | supkrs | i dont see any error 101 when i use 4 PRI lines |
| 00:31.21 | Bezzie | it serves from the /static section enabled with enablestatic |
| 00:31.30 | WIMPy | supkrs: What does that mean? |
| 00:32.05 | Bezzie | Unfortunately I'm following the directions to the T, and before I go bug the asterisk-gui guys, I want to confirm where the built in http server is trying to pull the static files from :P |
| 00:32.06 | WIMPy | It breaks if you configure more than 4 lines? |
| 00:33.39 | supkrs | WIMPy: yes |
| 00:33.52 | WIMPy | Interesting. |
| 00:34.06 | supkrs | WIMPy: is there a way to know the version of libpri running ? |
| 00:34.16 | WIMPy | That's not something I would have tried. |
| 00:34.20 | supkrs | i did compiled now |
| 00:34.35 | WIMPy | 'pri show version' |
| 00:34.59 | WIMPy | You need to compile Asterisk after installing libpri. |
| 00:35.44 | supkrs | WIMPy: thats painful |
| 00:35.59 | supkrs | I am using binary version of asterisk |
| 00:36.20 | WIMPy | Never tried that. |
| 00:36.47 | supkrs | host0039*CLI> pri show version |
| 00:36.48 | supkrs | libpri version: 1.4.11.5 |
| 00:36.54 | hardwire | nny: I think the System I was doing was part of mixmonitors execute call. sigh. |
| 00:37.25 | supkrs | WIMPy: got it now |
| 00:37.41 | supkrs | I restarted asterisk server using /etc/init.d/asterisk restart |
| 00:37.52 | supkrs | pri show version |
| 00:37.52 | supkrs | libpri version: 1.4.12 |
| 00:37.54 | supkrs | :) |
| 00:38.24 | supkrs | rpm -qa | grep libpri |
| 00:38.25 | supkrs | libpri-1.4.11.5-1_centos5 |
| 00:38.35 | nny | hardwire: heh yeah, I used Trysystem for this, it's a phps cript to populate a database to be used in generating call files, idc what happens after it's executed, as long as it's executed |
| 00:38.37 | WIMPy | Does it work now? |
| 00:38.39 | supkrs | WIMPy: will that hurt ? |
| 00:38.41 | nny | also WIMPy you;re my hero |
| 00:38.52 | nny | asterisk -rx 'meetme list concise' 55125!1!0!00:00:12!1!0 |
| 00:38.59 | nny | the output from that command is perfect |
| 00:39.31 | WIMPy | supkrs: Err, what hurts? |
| 00:39.54 | supkrs | WIMPy: i mean the package version installed on the server is still libpri-1.4.11.5-1_centos5 |
| 00:40.03 | supkrs | rpm -qa | grep libpri |
| 00:40.24 | supkrs | Does that matter |
| 00:40.29 | WIMPy | Sure. Unless you uninstall it or build a rpm of the new version and install that. |
| 00:40.48 | supkrs | WIMPy: ok |
| 00:41.13 | WIMPy | You can have multiple versions of the same lib. But the next rpm update might create a symlink to an older version again. |
| 00:41.41 | supkrs | WIMPy: so i should believe pri show version and not rpm -qa | grep libpri ? |
| 00:41.54 | WIMPy | Definitely. |
| 00:42.03 | supkrs | WIMPy: great |
| 00:42.15 | supkrs | You made my day :) |
| 00:42.20 | WIMPy | rpm will only show you rpms. It doesn;t know about reality :-) |
| 00:42.28 | *** join/#asterisk Dougy (me@openvpn/community/user/dougy) |
| 00:42.45 | WIMPy | Only what it did itself. |
| 00:43.10 | supkrs | WIMPy: will there be a rpm version available for 1.4.12 ? |
| 00:43.20 | WIMPy | NFI |
| 00:43.27 | supkrs | I contacted CentOS and they redirected me here :( |
| 00:43.32 | Dougy | Hi all, have a question! I just got a new DID. I'm trying to figure out how to set it so that when you call the DID, it calls all 5 phone numbers at once then hangs up after X seconds. Is there any docs out there on how to properly do this? I cant seem to find..... |
| 00:43.38 | supkrs | WIMPy: NFI ? |
| 00:43.40 | WIMPy | There might already be one. But I have no idea, where to search. |
| 00:44.08 | WIMPy | No F***ing Idea |
| 00:44.13 | supkrs | :( |
| 00:44.35 | WIMPy | Dougy: You just need the help for the Dial() application. |
| 00:44.35 | supkrs | bows down to WIMPy |
| 00:44.45 | Dougy | WIMPy: hrm |
| 00:45.32 | Dougy | WIMPy: my issue is i guess.. if i want to set the # to go to extension 500.. how od i make it so i dont need to register a sip device actively to have the extension available? |
| 00:45.38 | Dougy | if that makes sense? |
| 00:45.43 | Dougy | 500 (Unspecified) D N 0 UNKNOWN |
| 00:46.16 | WIMPy | Dougy: No that doesn;t make too much sense. What are you trying to do? |
| 00:47.31 | Dougy | I'm trying to figure out the best way to do this.. here is how i formulated it.. haave the incoming call go to extension 500.. i define another extension per person.. then ahve 500 ring all those extensions at once? |
| 00:47.31 | Dougy | or am i way over complicating that |
| 00:47.58 | WIMPy | Yes. |
| 00:48.05 | riddlebox | Dougy, you have 1 number that you want to ring to multiple endpoints? |
| 00:48.27 | WIMPy | You only need one extension and put all device to ring inside a Dial(). |
| 00:48.42 | Dougy | riddlebox: yes. here is my plan. its one incoming DID that needs to call each of the listed numbers (6 cell phones) at once |
| 00:48.47 | Dougy | nobody will answer the number |
| 00:48.47 | Dougy | it is just an 'alert' |
| 00:49.05 | Dougy | so peoples cell phone will ring, then they will call a different number for more info (a landline) |
| 00:49.30 | *** join/#asterisk lemonadelegs (~lemonadel@75.97.198.153.res-cmts.sth.ptd.net) |
| 00:50.22 | WIMPy | Dougy: Just do a Dial(sip/provider/mobile1&sip/provider/mobile2&... |
| 00:50.38 | Dougy | ok so ill pastebin this |
| 00:51.38 | Dougy | WIMPy: http://paste2.org/p/1544146 |
| 00:51.38 | Dougy | that? |
| 00:52.37 | *** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf) |
| 00:52.38 | WIMPy | Use sip/peer/number instead of sip/number@peer. The later can be misinterpreted. |
| 00:53.22 | Dougy | so sip/vitel-outbound/number ? |
| 00:53.26 | WIMPy | yes |
| 00:54.37 | Dougy | ok that's wroking, now i have a caller ID issue i ned to fix |
| 00:54.37 | Dougy | lol |
| 00:54.59 | WIMPy | Set(CALLERID(num)=1234) |
| 00:56.36 | Dougy | yeah i got it :P thank you |
| 00:56.37 | Dougy | thanks guys! |
| 00:56.37 | Dougy | :) |
| 00:56.40 | *** part/#asterisk Dougy (me@openvpn/community/user/dougy) |
| 01:04.54 | ChannelZ | Using Vitelity, you'll probably have a DTMF problem you'll discover too... |
| 01:09.56 | Bezzie | just an fyi: figured my issue out. digium's asterisk 1.8 package for debian references /usr/share/asterisk/static-http instead of /var/lib/asterisk/static-http |
| 01:12.27 | nny | how reliable is AMD? |
| 01:22.35 | *** join/#asterisk Kumbang (~unknown@180.245.137.5) |
| 01:27.09 | *** join/#asterisk Kumbang (~kumbang@180.245.137.5) |
| 01:35.09 | *** join/#asterisk f2Knight (~ben@c-76-115-44-207.hsd1.or.comcast.net) |
| 01:36.32 | f2Knight | I am here to try and help if I can. |
| 01:36.56 | WIMPy | ~ask |
| 01:36.56 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 01:46.16 | ChannelZ | ~tell |
| 01:46.24 | ChannelZ | oh.. right.. |
| 01:48.50 | echinos | Are there people really here for the latter reason? |
| 01:48.52 | echinos | :/ |
| 01:50.17 | *** join/#asterisk nightrid3r (~kvirc@91.176.170.204) |
| 01:50.58 | *** join/#asterisk lemonadelegs (~lemonadel@75.97.198.153.res-cmts.sth.ptd.net) |
| 01:51.16 | echinos | So, I have an asterisk server on a hosted machine. I have an SPA-2102 here, and a linux firewall (iptables) here inbetween them. Difficulty rating? |
| 01:51.26 | *** join/#asterisk gladier (~gladier@pedobear.molested.me) |
| 01:51.55 | echinos | I've thought of having an asterisk go-between on the firewall, but that kinda defeats the purpose of having asterisk on the hosted machine |
| 01:52.02 | WIMPy | moderate |
| 01:52.36 | gladier | hey guys - i'm trying to hunt around for some creative commons on hold music for a day spa, everything i've looked at so far hasn't been fantastic. does anyone have any suggestions for sites to look at? |
| 01:52.37 | echinos | I've seen iptables has a SIP conntrack module, I'll try that first |
| 01:53.44 | WIMPy | echinos: yes, nf_conntrack_sip is good, but don't load nf_nat_sip. |
| 01:53.54 | echinos | ok, but why not? |
| 01:54.47 | WIMPy | echinos: It tries to modify SIP messages to work around the NAT issue. Better activate NAT support in the Asterisk peer. |
| 01:54.54 | ChannelZ | doesn't conntrack_sip handle the ports but nat_sip hack the actual sip messages? |
| 01:55.48 | nightrid3r | gladier : some clasic music maybe, |
| 01:55.49 | WIMPy | conntrack does what the name says. It tracks the connections. The nf_nat mangles the packets payloads. |
| 01:56.14 | echinos | ie you don't need to mangle if you are doing conntrack...? |
| 01:56.37 | WIMPy | No, you don;t need to mangle if you have Asterisk on the other end. |
| 01:56.50 | echinos | ah |
| 01:57.42 | WIMPy | I tried to load it and use nat=no, but without and nat=yes is more reliable. |
| 01:57.54 | echinos | cool |
| 01:58.00 | echinos | tyvm |
| 01:58.47 | WIMPy | Ot at least it was for me. |
| 01:58.58 | echinos | well, so long as it works well-ish |
| 01:59.17 | echinos | I tried a year or so ago, didn't get it going, and am revisiting now |
| 01:59.47 | echinos | I want a home voip line that will still answer/take messages if my internet etc. goes down |
| 02:00.13 | WIMPy | Doesn't your ITSP provide that? |
| 02:00.19 | echinos | and, do all the other cool things that are possible |
| 02:00.29 | echinos | Yeah, but the cool stuff... |
| 02:00.33 | WIMPy | Ok, tehy won't do that. |
| 02:00.48 | echinos | ring my cell, let me call out and record the call, etc |
| 02:01.00 | echinos | need I say monkies? |
| 02:04.45 | *** join/#asterisk coppice (~coppice@m121-202-62-134.smartone-vodafone.com) |
| 02:06.48 | WIMPy | f2Knight: So what's your hint of the day then? |
| 02:15.01 | WIMPy | What do you call the opposite role of a maintainer? |
| 02:15.25 | echinos | bug tester? |
| 02:15.41 | echinos | hacker? |
| 02:15.54 | *** join/#asterisk Dougy (me@openvpn/community/user/dougy) |
| 02:16.15 | f2Knight | WIMPy, umm.. Should I try to have a tip of the day? ... interesting suggestion |
| 02:16.45 | Dougy | How hard is it to set up something where a person can call in, record a message, press pound or something, then it will call 6 people at once, and then play the message back when they answer? |
| 02:17.38 | p3nguin | not hard |
| 02:19.42 | WIMPy | echinos: I was in the direction of client. But maybe more specific. |
| 02:20.46 | echinos | consumer? |
| 02:21.13 | WIMPy | Sounds like someone going wild in a supermarket. |
| 02:21.28 | echinos | umm... user? |
| 02:21.36 | echinos | Am I on a game show? ;) |
| 02:21.47 | WIMPy | Ja, maybe I have to stick with that. |
| 02:23.21 | echinos | dependants? :) |
| 02:24.38 | WIMPy | Seems to fit. does it sound unfriendly? |
| 02:25.00 | echinos | User doens't really sound unfriendly, depending on the context |
| 02:25.17 | echinos | clients is more professional |
| 02:25.49 | WIMPy | Han, dependant. That sounds unfriendly to me. |
| 02:25.57 | echinos | dependants is... comically truthful |
| 02:26.26 | echinos | I can see how you might not like the term users, it's kinda... distant? |
| 02:26.45 | WIMPy | And can mean round about anything. |
| 02:27.05 | echinos | consumer/customer is no good? |
| 02:27.54 | WIMPy | I wanted to make the hirachy clear. But without being unfriendly if possible. |
| 02:28.07 | echinos | how about end-users |
| 02:28.45 | echinos | user community? |
| 02:28.49 | WIMPy | There might be more than two levels. |
| 02:29.04 | WIMPy | So dependant seems technically perfect. |
| 02:31.19 | f2Knight | crash tester |
| 02:31.37 | *** join/#asterisk mindCrime (~chatzilla@cpe-076-182-089-009.nc.res.rr.com) |
| 02:31.53 | echinos | ...freeloader? |
| 02:32.23 | echinos | Maybe something like committed client |
| 02:32.59 | *** part/#asterisk gundy (~gabe@69.169.150.25.provo.static.broadweavenetworks.net) |
| 02:33.07 | echinos | as in 2 tiers, committed and uncommitted |
| 02:33.19 | echinos | is that kinda what you're getting at? |
| 02:34.33 | WIMPy | It's more like domain stuff where everyone has a maintainer. |
| 02:36.56 | echinos | liabilities? ;) |
| 02:37.10 | echinos | I'm out of ideas, I think |
| 02:37.32 | *** join/#asterisk disgrntld81 (~asdf@CPE-70-92-225-30.wi.res.rr.com) |
| 02:37.54 | WIMPy | I'll use dependant in the code. Maybe I find something else to output :-) |
| 02:38.13 | WIMPy | Or actually I did. |
| 02:38.29 | echinos | It's in the code? Use lusers. |
| 02:39.23 | WIMPy | It's ony to point upwards or downwards in the hirachy. It need not be at the bottom level :-) |
| 02:39.32 | echinos | ah :) |
| 02:45.32 | echinos | I really have to recompile the kernel to get sip conntrack? |
| 02:46.16 | WIMPy | echinos: If you don't have it... |
| 02:47.20 | f2Knight | curious has anyone started looking at asterisk 10? |
| 02:47.24 | f2Knight | just curious |
| 02:47.28 | echinos | oh well. |
| 02:48.15 | echinos | I was also thinking of creating a VPN or SSH tunnel or something to make them seem local to each other |
| 02:48.20 | WIMPy | I installed it on a hosted box, but it doesn't run, yet :-) |
| 02:48.40 | WIMPy | SSH is TCP only, so useless. |
| 02:49.09 | WIMPy | But something like openvpn will eliminate all NAT issues. |
| 02:49.12 | f2Knight | Curious . AGI scripts... they would execute as the 'asterisk' user. so possibly root right? |
| 02:49.21 | echinos | ah, yeah... oh well. |
| 02:49.37 | echinos | so long as the vpn is stable :) |
| 02:49.38 | WIMPy | f2Knight: yes |
| 02:49.50 | f2Knight | WIMPy, thought so thanks :-) |
| 02:50.04 | WIMPy | It shouldn;t be less stable than without. |
| 02:51.35 | echinos | sweet |
| 02:51.48 | *** join/#asterisk ariel_ (~chatzilla@pdpc/supporter/active/abatista) |
| 02:52.07 | echinos | that sounds like easie |
| 02:52.09 | echinos | r |
| 02:52.19 | WIMPy | The only trouble with VPNs is that tehy break TC/QoS/whatever you call it. |
| 02:53.01 | *** part/#asterisk ariel_ (~chatzilla@pdpc/supporter/active/abatista) |
| 02:53.13 | f2Knight | WIMPy, every use python for AGI scripts? |
| 02:53.29 | WIMPy | nope |
| 02:56.37 | f2Knight | umm.. okay. thanks.. |
| 03:03.37 | Dougy | 02:17 < p3nguin> not hard |
| 03:03.46 | Dougy | any docs on how to look into doing it |
| 03:04.11 | p3nguin | Not really. It's all dial plan. You have to develop dial plan logic yourself. |
| 03:04.37 | Dougy | rubs chin |
| 03:04.39 | Dougy | this is going to be hard |
| 03:04.39 | Dougy | ok |
| 03:05.14 | p3nguin | It shouldn't be difficult at all. Think it out. Draw it on paper if necessary. |
| 03:05.22 | f2Knight | Dougy, You will want to look at either the M option or A option to the dial statement. (depending on how you want to implement it) |
| 03:05.35 | Dougy | writes down |
| 03:05.49 | Dougy | p3nguin: im having enough trouble running a single extension pbx here |
| 03:05.53 | f2Knight | Dougy, Don't forget to answer the incoming call first, before you do a record. |
| 03:06.07 | p3nguin | You'll never be able to have a single extension in Asterisk. |
| 03:06.22 | p3nguin | A single extension in Asterisk won't get you anywhere. |
| 03:06.22 | Dougy | ok, sorry, a single phone pbx |
| 03:06.31 | p3nguin | Well that's different. |
| 03:06.37 | Dougy | yes indeed |
| 03:06.42 | Dougy | 2 extension i guess |
| 03:07.07 | p3nguin | If you only want to be able to call two phones, two extensions might be enough. |
| 03:07.16 | f2Knight | p3nguin, id we just use default as an extension with _X. we COULD have a single extension if all it did was play a file and hung up :) |
| 03:07.23 | p3nguin | Even with very few phones, I still end up with hundreds of extensions. |
| 03:07.29 | Dougy | my current setup is 1 DID, 1 IP phone |
| 03:07.39 | Dougy | real simple |
| 03:07.42 | Dougy | except not so simple lol |
| 03:08.58 | f2Knight | p3nguin, we were working on something the other day , figured out the Channel variable missing context thingy ChannelZ had the answer I was looking for, /n solved all the problems I was having |
| 03:09.07 | f2Knight | ChannelZ, thanks again btw. :-) |
| 03:09.21 | p3nguin | The local channel strikes again! |
| 03:10.32 | *** part/#asterisk gladier (~gladier@pedobear.molested.me) |
| 03:10.39 | f2Knight | p3nguin, yep it was a bear. but adding that worked. Took him like 3 min to notice it lol |
| 03:10.56 | f2Knight | that is one that goes in your hat of things learned the hardway |
| 03:23.28 | *** join/#asterisk navaismo (~navaismo@201.123.89.106) |
| 03:24.58 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 03:24.58 | navaismo | hello, can anyone help me? I can't see the incoming CID in the analog trunks using TDM410P(wctdm24xpp+ driver) |
| 03:25.03 | *** join/#asterisk batphone (~boo@unaffiliated/batphone) |
| 03:25.12 | batphone | so i think i have a couple of ip501s |
| 03:26.02 | batphone | but the sip.cfg i have doesn't seem to support them. they make no mention of "501", as they do of "4000", "301", and even "601" |
| 03:26.31 | *** join/#asterisk vinhdizzo (~vinh@pool-173-60-112-55.lsanca.fios.verizon.net) |
| 03:26.38 | batphone | i see this in my quest to increase the chassis speaker volume on my phone. the speakerphone is simply not loud enough. |
| 03:27.19 | batphone | wading through the XML, i find these under the "gains" tag: voice.gain.rx.analog.chassis="0" voice.gain.rx.analog.chassis.IP_300="0" |
| 03:28.10 | batphone | although I am certain an amount of permutation on the end of that object would find my phone, i was rather hoping that someone could point me to a sip.cfg that already has this taken care of. |
| 03:29.37 | batphone | voice.gain.rx.analog.chassis=.IP_500="0" voice.gain.rx.analog.chassis.IP_501="0" |
| 03:29.46 | p3nguin | If I have extensions 2, 20, and _2XXX in any given context along with a WaitExten() where a caller needs to enter an extension, pressing 2 creates a several second wait before running extension 2. Is there anyway to globally reduce the time that it waits to match more digits? |
| 03:30.01 | batphone | this is what i am looking for, but with all the requisite framework within the xml. |
| 03:30.19 | f2Knight | p3nguin, what version of asterisk? |
| 03:30.24 | p3nguin | 1.4.something |
| 03:30.35 | batphone | p3nguin: i would look at the phone's digit map |
| 03:30.36 | WIMPy | p3nguin: The art is to not require a timout at all. |
| 03:30.53 | WIMPy | Ugh. Overlapping extensions. Thats ugly. |
| 03:31.02 | batphone | inevitable though |
| 03:31.04 | f2Knight | WIMPy, I agree :-) but some times you know.. wait(1) |
| 03:31.06 | p3nguin | batphone: I'm not talking about phones, I'm talking about Asterisk. |
| 03:31.27 | batphone | p3nguin: i'm not talking about asterisk, i'm talking about phones ;P |
| 03:31.36 | p3nguin | That's not my problem. |
| 03:31.42 | f2Knight | p3nguin, I know they changed all the wait stuff.. there is like a wait digits and a wait exten and a wait |
| 03:31.50 | batphone | point taken. good luck. |
| 03:32.00 | WIMPy | batphone: And you are talking about sip.conf and things that certainly don't go in there. |
| 03:32.06 | p3nguin | It's not the Wait() or the WaitExten() that's giving me a problem. |
| 03:32.26 | batphone | p3nguin: digit maps are a programmatic function of the phone |
| 03:32.39 | p3nguin | It's the fact that I have overlaps in the extensions. |
| 03:32.47 | p3nguin | batphone: I'm STILL not talking about phones. |
| 03:32.48 | batphone | p3nguin: they determine how many digits to "wait" for before sending to the switch. |
| 03:32.52 | f2Knight | p3nguin, if i understand you correctly you have a long time out because Asterisk is waiting for the 'POSSIBILITY' of you entering something other then 2 such as 2555 |
| 03:32.55 | p3nguin | Still not. |
| 03:33.03 | batphone | ok what phone.. |
| 03:33.06 | p3nguin | f2knight: That's correct. |
| 03:33.13 | p3nguin | batphone: None. Not talking about phones here. |
| 03:33.17 | WIMPy | batphone: I'm petty sure, p3nguin knows that at least as well, as you do. |
| 03:33.37 | batphone | i am just being difficult. i will be quiet now. |
| 03:34.05 | navaismo | hello, can anyone help me? I can't see the incoming CID in the analog trunks using TDM410P(wctdm24xpp+ driver) with europe fsk signalling |
| 03:34.46 | WIMPy | p3nguin: TIMEOUT(digit)? |
| 03:35.02 | p3nguin | TIMEOUT() something, but I want it to be global rather than per channel. |
| 03:35.04 | f2Knight | WIMPy, just beat me to it |
| 03:35.14 | f2Knight | WIMPy, there is also an absolute timeout |
| 03:35.28 | WIMPy | absolute if for the entire call, not for dialing. |
| 03:35.37 | f2Knight | BUT the issue I think is that you will have users pressing 2000 and not having enough time. |
| 03:35.38 | WIMPy | is |
| 03:35.46 | p3nguin | I want the digit timeout, but I don't want to have to add it on every extension that has a waitexten on it. |
| 03:36.55 | p3nguin | There is surely some global setting for timeouts, isn't there? |
| 03:37.01 | f2Knight | umm docs say that WaitExten over riders timeout(response)... |
| 03:37.27 | f2Knight | Try Timeout(digit) = 3 |
| 03:37.43 | p3nguin | If I have a timeout of 1 and a waitexten of 5... |
| 03:38.05 | f2Knight | p3nguin, we will find out as soon as you run it :) |
| 03:38.12 | p3nguin | Once I press any key, it should only wait 1 second before executing that exten, right? |
| 03:38.26 | WIMPy | If we are lucky. |
| 03:38.33 | f2Knight | that is the assumption.. but that means your 2xxx will never be able to be entered |
| 03:38.43 | f2Knight | and why i suggested 3 seconds or maybe 2. |
| 03:38.46 | p3nguin | WaitExten(5) has nothing to do with the timeout of the digit acceptance. |
| 03:38.57 | f2Knight | Ideally dont use the 2 or dont use 2xxx |
| 03:39.07 | p3nguin | That's just how long you have to start pressing keys. |
| 03:39.36 | f2Knight | WaitExten is the wait for them to start. |
| 03:39.40 | p3nguin | right |
| 03:39.41 | WIMPy | p3nguin: What channel do you come from? dahdi? |
| 03:39.42 | p3nguin | That's what I said. |
| 03:39.53 | p3nguin | Nah, I'm just doing SIP. |
| 03:40.06 | f2Knight | the timeout(digit) is "When this timeout expires, |
| 03:40.06 | f2Knight | <PROTECTED> |
| 03:40.06 | f2Knight | <PROTECTED> |
| 03:40.17 | WIMPy | So it is not dialing, but IVR? |
| 03:40.27 | p3nguin | I really can't see any way to keep from having the overlaps. Consider if you want someone to be able to press 2 for something OR to dial 2001 to reach a phone... from the menu. |
| 03:41.12 | DrDigital | how can i see what ip last used an extension? |
| 03:41.20 | f2Knight | p3nguin, try the TIMEOUT(digit)=3 it SHOULD wait 3 seconds for the user to put all digits in. That would give them time to enter 2000 |
| 03:41.21 | WIMPy | Make them dial some extra digit to enter the extension? |
| 03:41.50 | f2Knight | p3nguin, it should also mean if they press 2 it will wait no more then 3 seconds to switch them. |
| 03:42.16 | p3nguin | TIMEOUT(digit) is most likely the fix for each channel as it is created. The question is now "How do I get a digit timeout to be a GLOBAL setting rather than having to add it to every extension that has a WaitExten() in it?" |
| 03:42.28 | f2Knight | p3nguin, also from a dialplan point of view.. unless you have 10 menu items on a single level.. moving a digit shold not be an issue. |
| 03:42.52 | p3nguin | moving a digit? |
| 03:43.30 | p3nguin | You're saying skip any menu number that is the beginning number of a more-lengthy extension number? |
| 03:43.32 | f2Knight | p3nguin, umm... put it in globals ? (nothing in docs say you can or cant) |
| 03:43.52 | f2Knight | p3nguin, i am saying in stead of having the caller press 2 have them press 5 |
| 03:44.04 | p3nguin | Your suggestion is that if the phones are 2XXX, don't use 2 in the menu? |
| 03:44.11 | f2Knight | p3nguin, right |
| 03:44.34 | WIMPy | That should help a lot. |
| 03:44.55 | f2Knight | p3nguin, ever notice that some systems that let users put extension in directly skip 2? lots of old phone system use 2xx as the users extensions |
| 03:45.14 | p3nguin | That's an interesting thought, but now I have to check to see if it only happens on single digits which have overlaps. I was thinking it happened on everything. |
| 03:45.46 | f2Knight | p3nguin, yea the only time is should be a problem is if you have 10 menu items on the fist level.. in which you better change your dialplan anyways because the callers going to get confused after about 4 options |
| 03:45.55 | WIMPy | If it happens on everything there must be more to your dialplan than you might think. |
| 03:46.03 | p3nguin | I was thinking if I hit 9 and didn't have anything else starting with 9 that it still had a 3-second pause. |
| 03:46.27 | p3nguin | I have quite large dial plans. |
| 03:46.36 | p3nguin | I don't have multi-level menus, though. |
| 03:46.40 | f2Knight | p3nguin, test the timeout(digit) and see if that works |
| 03:46.53 | p3nguin | I can set functions in the globals section? |
| 03:47.08 | f2Knight | p3nguin, not sure. |
| 03:47.12 | WIMPy | If it has a match and no other possible matches, it should start to execute that extension immediately. |
| 03:47.44 | p3nguin | That's what I would think, too, but I'm getting lots of dead air waiting on menu responses to execute. |
| 03:47.58 | WIMPy | I think I saw a global variable in some really old dialplan, but that might not be supported any more since it has become a function. |
| 03:48.24 | WIMPy | Do you include something you don't want to? |
| 03:48.49 | p3nguin | Not really. I only include what needs to be available from that side of a call. |
| 03:48.52 | f2Knight | WIMPy, makes a good point if there is an include... it will wait even longer |
| 03:49.13 | p3nguin | I have includes. |
| 03:49.20 | f2Knight | something like a _X. will kill you on waiting.. as would a _9X. |
| 03:49.38 | p3nguin | I'm glad you can now understand my frustration. |
| 03:49.38 | WIMPy | No. It won't wait longer. It will always wait the same time, but only if there are other possible matches. |
| 03:49.41 | f2Knight | p3nguin, you might need to clean up your dialplan (it happens) |
| 03:50.10 | f2Knight | WIMPy, that is what I was meaning, vs. an instant match |
| 03:50.12 | WIMPy | Yes, some wildcard can be a possible match. |
| 03:50.49 | f2Knight | p3nguin, what you CAN do ... for global is to set a variable, say DIGIT_TOUT=3 |
| 03:51.56 | f2Knight | nvm. still need to set it in each context. but you could at least use a global variable to set it for the whole system. |
| 03:52.05 | echinos | WIMPy: what about setting up a STUN server on my linux firewall? |
| 03:52.06 | f2Knight | exten => s,n,Set(TIMEOUT(digit)=${DIGIT_TOUT}) |
| 03:52.28 | p3nguin | If I'm going to do that, I might as well use the function directly with the value. |
| 03:52.42 | WIMPy | echinos: Don't do that. Just enable nat=yes on asterisk and nothing else. That will work best. |
| 03:52.47 | p3nguin | I'm trying to avoid that. |
| 03:53.00 | echinos | WIMPy: I still need the conntrack etc tho |
| 03:53.53 | WIMPy | echinos: No. You could open up lots of ports. conntrack just lets you use the RELATED match, which is a lot nicer, off course. |
| 03:54.06 | f2Knight | p3nguin, what text editor are you using? |
| 03:54.11 | p3nguin | vim |
| 03:54.40 | echinos | WIMPy: so basically set up a DMZ-type thing and just forward RTP/SIP ports to the SPA? |
| 03:54.47 | echinos | hm.. |
| 03:54.59 | f2Knight | p3nguin, not a vim liker but thats okay... personal prefs. heres an Idea... if you can grep out the lines, then you would know where you are calling them at right? |
| 03:55.08 | p3nguin | Don't use DMZ. You don't understand DMZ. |
| 03:55.34 | f2Knight | p3nguin, if you can do that you could use something like sed or maybe gawk to rewrite the line, doing that you would be able to handle all your entries |
| 03:55.35 | p3nguin | f2knight: I'm not sure what you're saying. |
| 03:55.43 | WIMPy | echinos: That's a possibility. But as you only want to talk to that server you can be rather specific about what kind of traffic to allow. |
| 03:55.45 | echinos | I'm just saying it would be kind of like a DMZ is all. |
| 03:56.14 | WIMPy | You don;t need sed if you use vi :-) |
| 03:56.36 | f2Knight | WIMPy, maybe not but I do not know vim. |
| 03:56.55 | f2Knight | WIMPy, or vi. I 'like' nano from my old pico/pine days |
| 03:57.10 | WIMPy | I used to know it a little. But I prefer others. |
| 03:57.34 | WIMPy | Actually I haven't found any editor on *X that I liked so far. |
| 03:58.27 | f2Knight | p3nguin, my suggestion is mostly that if you can locate the line (shouldnt be to heard) you can save that line, add a line before it then rewrite both lines back 'inline' and runn it on a whole file. |
| 03:58.38 | f2Knight | p3nguin, do this real quick.. |
| 03:59.31 | WIMPy | Unless you do evil tricks with priorities, like I sometimes do. |
| 03:59.41 | f2Knight | p3nguin, grep --color -i 'wait' extensions.conf.bak |nl |
| 04:00.37 | f2Knight | p3nguin, it will tell you at least HOW many times wait appears in your extensions. then you know if its going to take you 10 min to hand edit it or 2 hours to write a script. |
| 04:01.10 | WIMPy | Or one miute to use a decent editor. |
| 04:01.28 | f2Knight | WIMPy, I would think that the Wait, WaitExten should be pretty close to the top and not be numbered by hand or with any crazy +101's |
| 04:01.31 | DrDigital | how can i see what ip last used an extension? |
| 04:01.33 | p3nguin | I have no problems using vim. |
| 04:01.52 | f2Knight | p3nguin, I am currious how many of these waits you have to deal with |
| 04:02.11 | f2Knight | p3nguin, I am thinking that you could have just hand edited it all by now :-) |
| 04:02.23 | p3nguin | drdigital: IPs don't use extensions, so good luck with that. Well... unless you are talking SIP URIs, then maybe I would consider IPs using extensions. |
| 04:02.40 | f2Knight | p3nguin, and if your doing that add that line I pasted earlier for you so that you can set a global time out |
| 04:02.58 | p3nguin | I'm going to see if the function works as a global. |
| 04:03.17 | f2Knight | p3nguin, http://www.voip-info.org/wiki/view/Asterisk+dialplan+sample+-+quick+office+dialplan |
| 04:04.14 | f2Knight | p3nguin, https://wiki.asterisk.org/wiki/display/AST/Background+and+WaitExten+Applications from the asterisk wiki |
| 04:04.23 | DrDigital | i had an extension 00 and 123123 added to my system, i want to know what system connected to the extension |
| 04:04.31 | DrDigital | i didn't make them |
| 04:05.38 | p3nguin | It doesn't seem to work as a global. I still had to wait six seconds after I pressed 2 before it executed. |
| 04:06.18 | f2Knight | p3nguin, I know its a hard pill to swallow. the first link is a working dialplan example for exactly what your doing. the second is docs from asterisk wiki that says well the same thing. |
| 04:06.51 | f2Knight | DrDigital, if you had special 'extensions' added that means that your system has been compromised. |
| 04:07.57 | f2Knight | DrDigital, More exactly it is not 'asterisk' that was likely at fault. Your computer has more likely been rooted. and they used those credentials to just modify your asterisk configs and create your new extensions. |
| 04:08.53 | ChannelZ | I blame squirrels |
| 04:09.01 | ChannelZ | They're always up to no good |
| 04:09.03 | f2Knight | DrDigital, depending on how your system is setup, you may have a trace of them in your syslog, (but if they were half good) they would have cleared that. Unless you are sending syslogs to an external server you really wouldn't have a way of knowing. |
| 04:09.39 | WIMPy | Or just look who's talking to china right now. |
| 04:10.20 | f2Knight | DrDigital, Not easily anyways. I would also look at your /etc/passwd file to see if any users have an ID of 0, and if any unknown users were added, also checking your sudo settings |
| 04:10.40 | f2Knight | WIMPy, show channels :) |
| 04:11.16 | f2Knight | DrDigital, My guess is that there might be an 'extra' sip account or two as well. and that your default contexts allow outbound dialing.. |
| 04:11.22 | ChannelZ | Do you mean someone added sip peers called 00 and 123123 or added extensions to your dialplan? |
| 04:11.43 | f2Knight | ChannelZ, I think he means someone modified his dialplan |
| 04:12.40 | f2Knight | ChannelZ, either way, I would say he is compromised. even if its a sip account added. |
| 04:13.54 | f2Knight | DrDigital, STRONG recommendation. new server, restore clean configs and redeploy system. |
| 04:13.59 | ChannelZ | Yeah but there's a few different levels of severity depending on what this personal actually did |
| 04:14.42 | WIMPy | If he was able to edit either config file, the situation is definitely bad. |
| 04:14.47 | ChannelZ | Sure |
| 04:14.48 | f2Knight | ChannelZ, true but I think its safe to say someone edited the config files either way which requires shell access. |
| 04:14.56 | WIMPy | But maybe it was only a compromised database? |
| 04:15.02 | WIMPy | "only" |
| 04:15.18 | f2Knight | WIMPy, if his database was publicly exposed. what else was exposed |
| 04:15.20 | ChannelZ | Not necessarily. Maybe he has AMI running, badly configured, no firewall, and the person telnetted in and added extensions to the dialplan dynamically |
| 04:15.20 | Nugget | telnet is eeeeeeevil! |
| 04:15.49 | WIMPy | You can add extensions via AMI? |
| 04:15.52 | f2Knight | ChannelZ, does AMI ship by default enabled? |
| 04:16.11 | f2Knight | WIMPy, you can do a lot via AMI, but it is limited ot only a few connections at a time. |
| 04:16.16 | ChannelZ | I didn't scroll back that far but didn't see if DrDigital said his extensions.conf was actually edited. Or if it was realtime from a DB |
| 04:16.42 | WIMPy | He didn't tell us. |
| 04:16.51 | f2Knight | no he just said extensions |
| 04:17.09 | ChannelZ | right. So there's 100 different possibilities ranging from "oops you learned your lesson" to "wipe your hard drive and start over" |
| 04:17.43 | ChannelZ | In any case he seems quite busy scrambling to figure it out judging by the lack of replies :) |
| 04:18.57 | ChannelZ | (and yes you can pretty much issue any command via AMI, given sufficient permissions, that you can on the console.. like 'dialplan add extension') |
| 04:19.34 | f2Knight | ChannelZ, :-) agreed.. I don't think hes using a DB (like mysql or anything) only basing that on the fact that I think most people that know how to setup mysql would be also knowledgeable to make it only accessable by local host, or at least set firewall rules. (I am prob. wrong) |
| 04:19.46 | WIMPy | Oh right. There was that command. Never found any use for that :-) |
| 04:21.00 | f2Knight | Lesson to be learned: FIREWALLS ARE A MUST, LOGGING FAILURES, and use Fail2Ban guys at the VERY least. |
| 04:21.48 | WIMPy | And I never understood why someone might consider fail2ban a good thing. |
| 04:21.56 | f2Knight | DrDigital, hope all is well with your system and we have not made you rush to unplug the powercable. ... or that you have |
| 04:22.01 | p3nguin | Setting the digit timeout does reduce the wait time. I added it in the dial plan where the problem is occurring most often. |
| 04:22.45 | WIMPy | Maybe he already jumped out of the window? |
| 04:24.02 | f2Knight | WIMPy, Fail2ban is appealing because it can monitor for multiple kinds of attacks. SSH / APACHE Telnet, asterisk etc. It is configurable to look for things that are important to you, it can send email alerts, logging sms messages, and modify the firewall rules for you and its only one tool to run use and know. |
| 04:24.18 | f2Knight | p3nguin, I thought that would fix your problem. Glad it worked for you |
| 04:24.58 | f2Knight | WIMPy, I hope its a short fall if he did. Headlines: "Asterisk causes man to SPLAT" |
| 04:24.58 | p3nguin | I may have been a little over the edge with wanting to do it globally. |
| 04:25.05 | WIMPy | Yes, but the reaction times are bad. And it is likely to give a false sense of security. |
| 04:25.29 | f2Knight | p3nguin, you wanted a simple solution. sadly this was not a global option, and if you think about it.. maybe rightfully so. |
| 04:26.28 | f2Knight | WIMPy, I agree response times are based on polling though I have had the watcher updated and working where it gets triggers on every write. still, its better then nothing if someone does not know anybetter. |
| 04:26.37 | p3nguin | After I thought about it, I probably don't need to reduce the timeout for all places with a waitexten. Right now, it is only showing its ugly face in a couple spots... so I set the timeout in those two spots. |
| 04:26.58 | DrDigital | I've known its been compromised, i am in the process of rewiring the network and can't take the system down, they can just keep using it because we don't have anything else to replace it with till the network is in the new room we built with a 50K BTU a/c unit |
| 04:27.50 | DrDigital | they are getting in from freepbx exploit on elastix |
| 04:28.15 | DrDigital | I'm just not gonna worry about it, i should have the new systems up on friday |
| 04:28.18 | WIMPy | f2Knight: There are much faster possibilites. I use iptables logging to evaluate and block traffic. Yet I consider that to be too slow. |
| 04:28.27 | *** join/#asterisk gogasca (~gogasca@c-71-202-75-108.hsd1.ca.comcast.net) |
| 04:29.17 | WIMPy | A version where the iptables result is comming from userspace would be the real solution. |
| 04:30.12 | p3nguin | Good ol' Elastix. Good ol' FreePBX. |
| 04:30.29 | WIMPy | Or at least one that really helps. |
| 04:31.15 | f2Knight | WIMPy, from userspace? in what way do you mean? the config acually just executes iptables commands directly. I have a modified config I use to use UFW because I have other things that use UFW as well. This way it all 'flows' I also have modified to keep my info after a restart. |
| 04:31.52 | WIMPy | In a way where every incomming connection would go though userspace for approval. |
| 04:32.09 | f2Knight | DrDigital, I am sorry for your fail, but curious how you found out it was an exploit in that package? (This goes back to my belief that a PBX should NOT have a web GUI) |
| 04:32.26 | WIMPy | That's the only way to avoid the situation where an attacker just sends 2000 requests before your ban gets activated. |
| 04:32.38 | DrDigital | simple |
| 04:32.50 | DrDigital | everyone knows the login and password to freepbx on elastix |
| 04:32.56 | DrDigital | and theres no easy way to change it |
| 04:33.12 | f2Knight | DrDigital, um yes there is |
| 04:33.17 | WIMPy | ?!? |
| 04:33.29 | f2Knight | No offense if you did not change your default password... |
| 04:33.53 | f2Knight | you deseved to be hacked. thats not a bug thats a very bad admin mistake. and he should be fired. |
| 04:34.12 | DrDigital | http://nerdvittles.com/?p=737 |
| 04:34.29 | DrDigital | FIRED |
| 04:35.53 | f2Knight | So are you using PBXIF , Trixbox or Elastic? |
| 04:36.08 | f2Knight | Each has there own 'defaults' |
| 04:36.42 | DrDigital | no my defaults i changed |
| 04:37.17 | f2Knight | trixbox has a set of scripts called passwd-* including the amp, maint, and meetme's |
| 04:37.26 | f2Knight | trixbox 101. http://www.freepbx.org/forum/freepbx/installation/howto-change-admin-password |
| 04:37.35 | *** join/#asterisk The_REV (~The_REV@c-76-21-1-80.hsd1.ca.comcast.net) |
| 04:37.49 | f2Knight | they use htpassword to restrict access (sucsk I know) |
| 04:37.54 | p3nguin | PiaF is a pile of shit. |
| 04:37.57 | DrDigital | when i found out about asteriskuser:eLaStIx.asteriskuser.2oo7 my system was already hacked |
| 04:38.01 | echinos | They are defined in the parameters AMPDBUSER and AMPDBPASS in the /etc/amportal.conf file. |
| 04:38.29 | f2Knight | echinos, yes freepbx does store the copy in the DB but its also in the static config |
| 04:38.35 | echinos | gah |
| 04:38.43 | f2Knight | Really though, do you need a GUI ON the PBX? |
| 04:38.55 | WIMPy | And on the internet? |
| 04:39.01 | p3nguin | Some people aren't bright enough to do it any other way. |
| 04:39.09 | f2Knight | I have a set of python scripts I wrote to add users, and use AEL for dialplan logic |
| 04:39.14 | *** join/#asterisk mik3yy (~root@76.203.197.196) |
| 04:39.27 | f2Knight | p3nguin, true enough, |
| 04:39.47 | p3nguin | If I have to access something like that, I keep it off the internet and access it through an ssh tunnel. |
| 04:39.58 | f2Knight | WIMPy, good point!, (i confess I have a few clients that use Trixbox onsite) but you have to vpn in to get to the config |
| 04:40.03 | mik3yy | Can someone help me with getting my Cisco 2940 working with asterisk. I can dial out but my phone won't ring when dialling in |
| 04:40.17 | p3nguin | 7940? |
| 04:40.24 | mik3yy | yes.. |
| 04:40.26 | mik3yy | 7940 |
| 04:40.32 | f2Knight | thats for the rest of you guys. |
| 04:40.34 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-163.hsd1.nj.comcast.net) |
| 04:40.35 | gogasca | Sip or sccp? |
| 04:40.40 | mik3yy | SIP |
| 04:40.40 | p3nguin | Show me your dial plan and I'll tell you why it doesn't work. |
| 04:40.40 | f2Knight | when someone wants to buy a couple I have a box of them |
| 04:40.54 | p3nguin | Are they cheap? |
| 04:41.14 | f2Knight | p3nguin, private message me |
| 04:41.32 | gogasca | From where u r dialing? Pstn ir other ext ? |
| 04:41.58 | gogasca | Can u dial other ext? |
| 04:42.06 | mik3yy | p3nguin: do you want me to post on the channel? |
| 04:42.11 | p3nguin | ~pb |
| 04:42.11 | infobot | [~pb] 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.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 04:42.36 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
| 04:42.51 | p3nguin | pastebin it. |
| 04:44.40 | mik3yy | http://pastebin.com/Hs23D8nL |
| 04:45.56 | mik3yy | I think my issue might be that the phone is not registering |
| 04:46.13 | mik3yy | 101/101 (Unspecified) D 0 Unmonitored |
| 04:46.21 | p3nguin | What you have pasted is not asterisk dial plan. |
| 04:46.38 | p3nguin | A failure to register WILL prevent a call from going to the phone. |
| 04:46.52 | p3nguin | Asterisk does not know where to send the call if it isn't registered. |
| 04:49.44 | p3nguin | But if the phone provides the right auth, it can make calls. |
| 04:50.16 | mik3yy | http://pastebin.com/hJuswF7X |
| 04:51.05 | mik3yy | that is the extensions file |
| 04:51.45 | joako | I have some really strange error: translate.c:249 ast_translator_build_path: No translator path from unknown to g723. But I don't have anything with G723 in my system, much less my config files. |
| 04:52.42 | *** join/#asterisk thepinkster (~mrpink@adsl-99-103-239-171.dsl.emhril.sbcglobal.net) |
| 04:53.31 | *** part/#asterisk thepinkster (~mrpink@adsl-99-103-239-171.dsl.emhril.sbcglobal.net) |
| 04:54.19 | mik3yy | p3nguin: this is the error that asterisk shows on incoming call http://pastebin.com/1gmPWahw |
| 04:54.39 | mik3yy | WHen I connect with a softphone , it works. |
| 04:54.50 | p3nguin | You have a lot of broken lines in your extensions.conf. |
| 04:54.55 | p3nguin | _411? Just 411. |
| 04:55.47 | p3nguin | No spaces between extension, priority, and app. Not 411, 1, Dial()... just 411,1,Dial(). |
| 04:56.03 | mik3yy | I'm not sure.. I think I cut/paste from other people's config :( |
| 04:57.22 | joako | Is it possible to use ogg or even mp3 as a voicemail format? |
| 04:57.30 | WIMPy | no |
| 04:57.50 | WIMPy | The are read-only. |
| 04:57.52 | batphone | p3nguin: do you have any experience authenticating polycoms with dot1x and certs? |
| 04:58.09 | mik3yy | Maybe I should upgrade my phone firmware to the latest. I am using 7-1 ...Latest is 8-12 |
| 04:58.10 | p3nguin | Nope, sure don't. |
| 04:58.16 | p3nguin | I don't use Polycom at all. |
| 04:58.34 | p3nguin | If you want to upgrade, go to 8.11 not 8.12 |
| 04:58.39 | batphone | i just happen to have some, and i'll need to get some call managers and 7940s going like that. |
| 04:58.41 | p3nguin | 8.12 has a caller ID bug. |
| 04:58.48 | mik3yy | ok.. |
| 04:58.54 | batphone | thats why i am rebuilding my lab from years ago. |
| 04:58.54 | f2Knight | joako, along time ago I think mp3 was an options.. but no its not possible. You CAN save it as a wave and use something like LAME to re-encode it as an MP3 that you could then send by email to a user if you wanted. |
| 04:59.36 | joako | f2Knight, Great, but that doesn't work with voicemail to email |
| 04:59.49 | mik3yy | Can I go from 7-01 to 8-11 or do I need to upgrade to intermediate fws? |
| 05:00.25 | p3nguin | It should be direct. |
| 05:00.46 | *** join/#asterisk irroot (~irroot@41.52.28.231) |
| 05:00.49 | f2Knight | joako, nope it doesnt it means you have to NOT use the build in asterisk voicemail to email feature and instead use an external script to craft the email message for you, while its doing that it can do the mp3 encoding |
| 05:00.49 | p3nguin | Something as old as like 5.x would need an intermediate. |
| 05:01.12 | f2Knight | joako, I have been meaning to write one myself for my phone servers but have not got around to it yet. |
| 05:01.16 | joako | And what about ogg? I set format=ogg and the caller things the vm is recorded, but I get: -- Recording was 0 seconds long but needs to be at least 5 - abandoning (and a bunch more errors) |
| 05:01.24 | f2Knight | joako, mostly because I want pretty html emails. :) |
| 05:01.32 | irroot | morning |
| 05:01.42 | p3nguin | mik3yy: Did you set your phone to register? The SIP<MAC>.cnf and SIPDefault.cnf control if the phone registers or not. |
| 05:01.42 | joako | f2Knight, Why have to do all that if the support could be built in? |
| 05:02.08 | f2Knight | joako, because asterisk only support WAV, WAV49, and GSM for voicemail files |
| 05:02.46 | f2Knight | joako, and it doesnt support html email unless you hact the voicemail.c file, and then it breaks on an upgrade. |
| 05:03.06 | *** join/#asterisk freeman_u (~freeman@193.110.114.54) |
| 05:03.20 | mik3yy | i set register=1 |
| 05:03.21 | joako | Basically I want a simple way so I can send a voicemail to email and the user can play it in Gmail |
| 05:03.27 | joako | Gmail doesn't play wave or gsm |
| 05:04.03 | p3nguin | mik3yy proxy_register: 1 |
| 05:04.55 | p3nguin | Maybe that's why it doesn't work. Wrong syntax, wrong options... sure to fail. |
| 05:05.56 | f2Knight | joako, they play you just have to download them :) |
| 05:06.04 | joako | f2Knight, ulaw seems to work |
| 05:06.19 | joako | f2Knight, Actually I am told even if you download them they have issues playing on MacOS |
| 05:06.31 | p3nguin | I use wav49 for voicemail-to-email. It plays on all platforms I've encountered. |
| 05:06.36 | f2Knight | joako, well technically any supported audio file by asteirsk should work. but mp3 is not supported |
| 05:06.40 | WIMPy | Ok, two things not to use: gmail and Mac OS. |
| 05:06.51 | f2Knight | WIMPy, lol +1 that |
| 05:07.10 | f2Knight | But really I use gmail all the time. Its not so bad to click download |
| 05:07.20 | joako | f2Knight, Then why do I get nothing when I try format=ogg? |
| 05:07.38 | f2Knight | joako, because ogg is not a supported format |
| 05:07.40 | WIMPy | And you can probably tell your browser to play it instead of saving it. |
| 05:07.49 | joako | What do you mean by "supported format?" |
| 05:07.57 | f2Knight | WIMPy, on windows anyways you could tell it to open instead of save. |
| 05:08.00 | ChannelZ | oO |
| 05:08.05 | WIMPy | It is not a valid configuration option.; |
| 05:08.15 | mik3yy | p3nguin: proxy_register: 1 was already added |
| 05:08.41 | p3nguin | oic |
| 05:08.43 | f2Knight | joako, officially? format |
| 05:08.45 | f2Knight | The format setting selects audio file format(s) to use when storing voice mail messages. The value is a string defining the audio format(s) of the message file. The default format string is wav49|gsm|wav, meaning that Asterisk will save the voicemail message in all three supported formats. When emailing the attachment, however, it will send only the first of the formats defined here. When playing back (as with all file playback) Aster |
| 05:08.45 | f2Knight | isk will attempt to use the optimum format based on the codec used for the current channel, in order to provide the best sound quality and to reduce transcoding processing time. |
| 05:08.45 | f2Knight | wav49: In this format, the file size will be small, the quality good, and it's a good choice for sending voicemail messages in email. The file will have a .WAV extension, which all Windows users should have no problems with, and users on other platforms should also be able to easily play these sound files. |
| 05:08.46 | WIMPy | f2Knight: I think the OS has nothing to do with that. |
| 05:08.49 | f2Knight | gsm: Voicemail saved in this format will have about the same file size and same audio quality as wav49. It may be less well supported by client operating systems if sent to users in email, however. |
| 05:08.51 | f2Knight | wav: This is an uncompressed sound format with a .wav extension, so the file size is very large. Sound quality will be great, but you probably don't want to email it, and you must have adequate disk space. |
| 05:08.55 | f2Knight | g723sf: The sample voicemail.conf file has this option commented out. If you try to activate it, you will find that it doesn't work. |
| 05:08.56 | mik3yy | It works.. |
| 05:08.59 | f2Knight | If you're running Asterisk on Mac OS X, which has a case-preserving but case-insensitive file system, you can use .wav but you won't be able to use .WAV(wav49). Asterisk tries to pick the best file in the Playback command based on the encoding of your channel, and then looks for the file by file type in a specific order. In either case, asterisk will look for filename.wav, and if there is a filename.WAV, the file lookup will succeed. |
| 05:09.03 | f2Knight | Asterisk will then try to play the file as "wav", and choke on the format. This is especially noticeable in voicemail. If you have both wav49 and wav in the format= option, one will replace the other. You must remove wav49 from the format= option in voicemail.conf. |
| 05:09.04 | p3nguin | jebus |
| 05:09.07 | f2Knight | Opps sorry that was supposed to be a link |
| 05:09.14 | mik3yy | Upgraded the firmware and it worked :) |
| 05:09.20 | p3nguin | Great! |
| 05:09.57 | joako | f2Knight, I know all that. What I am saying is "ogg" is a "supported format" "in asterisk" |
| 05:09.58 | f2Knight | joako, the answer is that if you want MP3 email YOU have to process it. Asterisk won't do it for you and thats about all there is to it. |
| 05:10.01 | mik3yy | thanks for the tip about 8-12. |
| 05:10.11 | joako | I can do file convert whatever.ulaw whatever.ogg and it will generate (write) an ogg |
| 05:10.16 | f2Knight | joako, okay well keep trying with that |
| 05:10.38 | WIMPy | joako: But not by the voicemail module. |
| 05:10.40 | p3nguin | mik3yy: It's not a major bug, but it's still annoying to many people. It's a problem with spaces in the caller ID names. |
| 05:10.53 | WIMPy | joako: As I already stated, ogg and mp3 are read-only. |
| 05:10.53 | f2Knight | joako, we already told you what you have to do to use MP3's either you write a script to do it all for you or you work with what they provided. |
| 05:11.11 | joako | f2Knight, Do you know where I can get that script? |
| 05:11.25 | p3nguin | When I used SIP on my Cisco phones, I used 8.11 for as long as I used SIP on them. |
| 05:11.26 | f2Knight | joako, you can write it :-) |
| 05:11.37 | WIMPy | joako: It goes from your fingers to your keyboard. |
| 05:11.45 | joako | I am sure I am not the first person that tried to send voicemail as MP3 |
| 05:11.50 | f2Knight | joako, bash, python, c++ what ever you like I like python personally |
| 05:11.53 | joako | Thanks for your "help" anyways. |
| 05:12.02 | f2Knight | joako, but have not had time to get to writing one yet. |
| 05:12.06 | WIMPy | Maybe google knows more? |
| 05:12.34 | p3nguin | You could write an external script to accept the voicemail files as wavs and transcode them to something else. |
| 05:13.03 | f2Knight | joako, you will need to use the mailcmd command in your voicemail.conf to call an external program to do what ever you like with voicemail. |
| 05:13.28 | f2Knight | joako, as I said Pretty HTML emails, embeded flash players, what ever you want. |
| 05:13.53 | p3nguin | Ick, it's already past 1 AM and 6 is going to come early. |
| 05:14.43 | mik3yy | Is it difficult to set up voicemail? |
| 05:14.45 | f2Knight | p3nguin, told him thats what he has to do already.. but he like you didn;t want to listen to the advice ... you got smart and took it and moved on :) He will have to learn that the answer is simple enough. Write a shell script or python or what ever. |
| 05:14.56 | f2Knight | mik3yy, nope supper easy. |
| 05:15.19 | f2Knight | mik3yy, actually probably the eaisest thing in asterisk is the voicemail. |
| 05:15.21 | joako | How about the "along time ago I think mp3 was an options" where can I get that version of asterisk? Why were even useful valuable features REMOVED? |
| 05:16.05 | WIMPy | Because you have to pay a licence to encode mp3. |
| 05:16.18 | f2Knight | joako, long time ago.. like one of the very first versions I think , its not something to use as it is WAY out dated. it was removed (or never encluded) because it didnt work right or well |
| 05:16.31 | f2Knight | joako, and that too. ! |
| 05:17.24 | f2Knight | joako, the answer is simple, anything with voicemail incoming can be handled with mailcmd= it does what ever you want with the file to guess what .. EMAIL them |
| 05:17.28 | mik3yy | f2knight : Can you redirect me to a URL on how to set it up. |
| 05:18.16 | joako | But does that work 100% Say I switch to mysql or imap voicemail will it continue to work, or need modifications? |
| 05:18.30 | f2Knight | joako, if you WANT to reformat that you can do just that. if you want to make something look pretty you can do that too. The secreat is YOU need to learn how to either use shell scripts to use the tools on your computer or You need to use what's there. |
| 05:19.08 | joako | NO! I can't even use wav49 |
| 05:19.10 | joako | [Jul 27 01:21:45] WARNING[6719]: translate.c:249 ast_translator_build_path: No translator path from unknown to g723 |
| 05:19.11 | joako | [Jul 27 01:21:45] WARNING[6719]: file.c:201 ast_writestream: Unable to translate to format wav49, source format slin |
| 05:19.11 | joako | [Jul 27 01:21:45] WARNING[6719]: app.c:655 __ast_play_and_record: Error writing frame |
| 05:19.11 | joako | <PROTECTED> |
| 05:19.12 | f2Knight | joako, no it only works like once out of every 100000 voicemails. |
| 05:19.42 | joako | not sure why it even says g723 |
| 05:19.44 | f2Knight | mik3yy, to add voicemail? |
| 05:20.03 | f2Knight | joako, paste your voicemail.conf file |
| 05:20.06 | f2Knight | ~pb |
| 05:20.06 | infobot | [~pb] 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.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 05:20.10 | joako | if I set format=wav49 it hangs up! |
| 05:21.19 | f2Knight | mik3yy, [default] |
| 05:21.20 | f2Knight | 1234 => 1234,Some User,email@address.com,pager@address.com,saycid=yes|dialout=fromvm|callback=fromvm|review=yes|operator=yes |
| 05:21.24 | mik3yy | f2knight: yeap |
| 05:21.54 | f2Knight | thats the basics of it. |
| 05:22.17 | f2Knight | mik3yy, look in voicemail.conf it has all the examples and even a default mailbox already assigned. |
| 05:22.23 | batphone | -fxs |
| 05:22.37 | mik3yy | ok.. thanks .. I'll go look it up. Thanks |
| 05:23.06 | f2Knight | mik3yy, on my phone systems I uses the clients EXTEN as there voicemail number so I can do a simple Voicemail(${EXTEN}) in my dialplan.. |
| 05:23.22 | joako | http://paste2.org/p/1544616 |
| 05:23.31 | f2Knight | mik3yy, well technically I have a prefix to that but its the same thing. |
| 05:24.05 | joako | How can I even know what formats are loaded in Asterisk? I see no such command |
| 05:24.29 | f2Knight | joako, is this your first asterisk box? |
| 05:24.44 | f2Knight | joako, or are you administering someone elses system |
| 05:24.51 | joako | f2Knight, No. I've been using asterisk for many years, since the pre-1.0 days. |
| 05:25.04 | joako | I am adminstering my own system. It is behaving in unexpected ways. |
| 05:25.16 | f2Knight | WIMPy, ....??? |
| 05:25.38 | WIMPy | Here! |
| 05:25.47 | joako | format=wav works: no messages, warnings or errors. VM are properly stored |
| 05:26.04 | joako | format = wav|wav49 fails, warnings errors & nothing is written to disk |
| 05:26.24 | joako | format = ulaw works too |
| 05:26.58 | joako | And like I said I can't find a command that shows the possible audio file formats, just isn't there and can't recall ever seeing anything than the worthless show translations |
| 05:27.24 | WIMPy | 'core show translation' |
| 05:27.41 | f2Knight | core show translations |
| 05:27.43 | f2Knight | core show translation |
| 05:27.50 | f2Knight | ohh wimpy beatme to it. |
| 05:28.05 | f2Knight | you probably do not have the other codecs loaded. so for YOU they are not supported. |
| 05:29.15 | f2Knight | joako, if you pb the output we can see if you have the codecs |
| 05:29.27 | batphone | anyone know how to tell a Soundpoint IP 501 to upload it's configuration file? |
| 05:30.14 | f2Knight | joako, I believe your looking at adpcm for the wav's I could be wrong |
| 05:31.06 | f2Knight | joako, For the record I use wav on my production servers too |
| 05:31.15 | joako | Ok so there's a bug in format_wav_gsm.so it seems |
| 05:35.03 | f2Knight | format=wav , mailcmd=myemailscript, problem solved |
| 05:37.14 | joako | format_wav_gsm.so requires codec_gsm.so, but if codec_gsm.so is disabled format_wav_gsm.so still loads, and exhibits odd behavior when it's called |
| 05:38.31 | f2Knight | joako, that doesn't sound like a bug, that sounds like a dependency. |
| 05:38.52 | f2Knight | joako, and judging from the module name sounds like it is a reasonable one to expect. |
| 05:39.02 | joako | f2Knight, A module with unfufilled dependancies should not load |
| 05:39.17 | joako | format = sfsdf234 |
| 05:39.17 | joako | <PROTECTED> |
| 05:39.43 | f2Knight | joako, okay use that then. |
| 05:40.50 | joako | LOL, just illustrating that specifing an invalid codec causes it to fall back to wav |
| 05:41.47 | f2Knight | we have been telling you all night to use wav, |
| 05:41.59 | *** join/#asterisk e7e5 (~rudenko@188.134.2.33) |
| 05:42.20 | f2Knight | if you want to use something other then that then to use the build in documented function of how to convert the file to something else called mailcmd |
| 05:42.37 | f2Knight | if you don't want to do that. thats okay |
| 05:45.33 | joako | Ok, I found one that might work: http://www.fscker.ca/rc/wp-content/uploads/2011/06/sendmp3voicemail.txt |
| 05:45.51 | joako | I'll just need to test if it's compatible with IMAP & MySQL voicemail |
| 05:47.20 | *** join/#asterisk mintos (~mvaliyav@115.241.228.235) |
| 05:47.52 | f2Knight | oh yea thats the EXACT same script that is in the Asterisk voicemail documentation.. says to use the mailcmd function of asterisk |
| 05:49.04 | f2Knight | modify it up a little make pretty html email and add a flash based player and your good to go with any email server that supports html email. |
| 05:49.34 | f2Knight | http://www.voip-info.org/wiki/view/Asterisk+Voicemail |
| 05:49.34 | *** join/#asterisk hrolf (~hrolf@static-host202-61-49-9.link.net.pk) |
| 05:50.42 | f2Knight | make a few more tweaks to it and you can use an external SMTP mailserver and get rid of the extra services running on your server all together. |
| 05:52.06 | joako | If asterisk would support standard MP3 format that would be it, it would automatically play in gmail no flash or anything fancy needed |
| 05:55.18 | WIMPy | Nice nick. |
| 05:55.20 | f2Knight | if MP3 was a free audio format it might have, although I see support for m4a, before I see support for mp3, m4a is free already, based on mpeg4 audio stream and plays on every device out there computer, ipod, android, tablet and phone. and is the unencrypted (DRM-FREE) version of what that format callled... AAC? Apples "format"? |
| 05:56.48 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 06:02.02 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 06:02.15 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 06:02.23 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 06:07.05 | *** join/#asterisk vfabi (~fabi@host-static-93-116-255-140.moldtelecom.md) |
| 06:08.19 | f2Knight | Joako, has made me want to expiated my asterisk voicemailer script. 1) Use smtp services, 2) Pretty HTML emails 3) embeded audio. any other suggestions? |
| 06:09.02 | WIMPy | A callback link? |
| 06:10.24 | *** join/#asterisk hrolf (~hrolf@unaffiliated/hrolf) |
| 06:10.59 | *** join/#asterisk Ad-Hoc (~nimbus@194.30.241.105) |
| 06:14.19 | aiksa[LV] | a link to add to task list :)) |
| 06:14.21 | f2Knight | WIMPy, good idea... get a call that you have voicemail.. umm... would have to have an external config then to process that. |
| 06:14.25 | aiksa[LV] | deferred execution |
| 06:14.45 | f2Knight | aiksa[LV], like to add to Google Calendars? |
| 06:14.49 | aiksa[LV] | yup |
| 06:15.08 | WIMPy | was thinking of a link to call the person who left the message. |
| 06:15.10 | f2Knight | aiksa[LV], sounds good, but I think I am streaching my python skill on that one. |
| 06:15.24 | aiksa[LV] | or setup deferred call to you to remind later |
| 06:15.31 | aiksa[LV] | f2Knight: oh a python lover :) |
| 06:16.51 | f2Knight | aiksa[LV], i do like it.I came from windows where I did a lot of VBA scripting so Python felt 'natural' to transition too |
| 06:17.09 | aiksa[LV] | f2Knight: being reptilophile myself, python is a nice thing to interface to asterisk |
| 06:17.35 | f2Knight | aiksa[LV], the fact that it is cross platform, needs less code then other languages and is easy to understand even if you do not know the lanuguage is a bonus |
| 06:17.37 | aiksa[LV] | f2Knight: if you havent already - may i suggest StarPy and Twisted framework in general |
| 06:18.08 | WIMPy | If you like real BASIC, try brandy. But be warned: It won't do sockets. |
| 06:18.36 | aiksa[LV] | WIMPy: python aint basic |
| 06:18.38 | f2Knight | aiksa[LV], I am using pyst (and hate its ugly output on the console, i just remarked it out) but I do not understand twisted and didn't want to load a larger library |
| 06:19.05 | f2Knight | WIMPy, no fan of GOTO 10 statements |
| 06:19.28 | WIMPy | The 70s are over. |
| 06:19.46 | f2Knight | that being said RealBASIC is crossplatform |
| 06:19.47 | aiksa[LV] | f2Knight: take a good look at it. Async approcah might seem a bit odd at the first, but once you get a hold of it - its an unlimited power wating to be harnessed |
| 06:21.11 | f2Knight | aiksa[LV], After this currnent project I just might do that. But right now I needed to get something up quick and easy and didn't have time to learn another framework.. still trying to get an understanding of web2py (which I want to use to create a gui interface to asterisk realtime and a custom billing setup) |
| 06:21.52 | f2Knight | i do not like a2billing or astcc |
| 06:22.15 | f2Knight | I think they take over too much control of the system. |
| 06:23.10 | f2Knight | starPy might be good to learn for that as I want to have independent set of servers and have asterisk write to it that way. |
| 06:23.21 | f2Knight | aiksa[LV], are you on G+? |
| 06:24.48 | aiksa[LV] | f2Knight: if by G+ you meant El Goog's latest shot at social networking - yes. |
| 06:25.37 | f2Knight | aiksa[LV], if you want to add me I will put you in my astiersk circle and python group. gplus.to/tuxpowered |
| 06:25.58 | f2Knight | aiksa[LV], I think this time they might have got it right though. |
| 06:31.02 | *** join/#asterisk BuenGenio (~Gene@059148208218.ctinets.com) |
| 06:31.06 | BuenGenio | hello |
| 06:31.18 | BuenGenio | does Asterisk have some sort of a GUI/Web manager interface? |
| 06:31.40 | WIMPy | No, but you can get some. |
| 06:31.55 | BuenGenio | 3rd party, doesn't matter... |
| 06:32.02 | *** join/#asterisk cerberus_za (~coert@41-134-110-162.dsl.mweb.co.za) |
| 06:32.04 | BuenGenio | which ones are the better ones (more complete) out there? |
| 06:32.29 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 06:32.30 | schmidts | good morning |
| 06:32.38 | *** join/#asterisk timahvo1 (~rogue@41.215.1.35) |
| 06:40.24 | f2Knight | BuenGenio, AsteriskNOW, |
| 06:41.00 | *** join/#asterisk aiksa[LV] (~aiksaLV]@212.70.182.16) |
| 06:41.13 | f2Knight | i think is a good source as its made by the makers of digium so its good to work. freepbx is popular but I don't like alot of the way it works, its strongest feature is the plugins but thats about it to me |
| 06:43.20 | *** join/#asterisk singler (~singler@78-60-139-121.static.zebra.lt) |
| 06:43.59 | *** join/#asterisk sulex (~sulex@pdpc/supporter/professional/sulex) |
| 06:44.39 | BuenGenio | how much is Asterisk Now? |
| 06:45.33 | pabelanger | $0 |
| 06:46.29 | BuenGenio | oh, I was thinking about Swtichbox |
| 06:46.31 | BuenGenio | nm |
| 06:46.31 | BuenGenio | thanks |
| 06:47.57 | f2Knight | switchvox is a good solution if you need support |
| 06:48.19 | f2Knight | if you roll your own then AsteriskNOW is a good place to start. (better then free pbx IMO |
| 06:48.38 | f2Knight | you could also use Druid by Voiceroute, |
| 06:49.09 | WIMPy | Or Gemeinschaft. |
| 06:49.25 | WIMPy | But I don't know anythig about the GUIs. |
| 06:49.43 | BuenGenio | okay, I'm looking to just play around for now, get the hang of dialplan, and try to set up a simple office network |
| 06:50.10 | f2Knight | looks like voiceroute is DOA, I got a copy of the software excellent parser lets you keep your hand edited configs and all |
| 06:50.11 | BuenGenio | I can use any/most softphones, right? Polycom, Cisco, etc...? |
| 06:50.20 | WIMPy | In that case you probably don;t want a GUI. |
| 06:50.28 | f2Knight | AsteriskNow use the asterisk GUI |
| 06:51.00 | f2Knight | WIMPy, when someone learns better by seeing a change a gui can help. SO LONG as he looks at the configs lol |
| 06:51.35 | *** join/#asterisk irroot (~irroot@196-209-227-53.dynamic.isadsl.co.za) |
| 06:51.40 | BuenGenio | yeah, I'm pretty much a CLI freak, but I think GUI in this case is much better for giving a visual overview of Asterisk's different components |
| 06:51.42 | WIMPy | And uses a GUI that doesn't produce a hell of a dialplan, almos noone understands. |
| 06:54.27 | BuenGenio | well, I'm sure I'll get to get dirty to get things properly customised ... ) |
| 06:54.31 | schmidts | Wimpy there is a GUI which can do this? |
| 06:54.49 | WIMPy | Not sure. |
| 06:54.59 | schmidts | i know one but i have written it by myself :D |
| 06:55.14 | BuenGenio | so can I use most softphones? we have a couple of relatively old Polycom and Cisco units lying around |
| 06:55.45 | schmidts | BuenGenio you can use "every" SIP compatible Phone except of snoms, they are just evil |
| 06:55.59 | BuenGenio | ok |
| 06:56.25 | WIMPy | likes Snom. |
| 06:56.39 | BuenGenio | how is SNOM different from a SIP phone? |
| 06:56.42 | WIMPy | It's the least evil SIP hardware I've seen so far. |
| 06:57.57 | irroot | sitting at wimpy for breakfast |
| 06:58.06 | schmidts | BuenGenio as i said they are evil :D but if you have some polycom and ciscos they are fine |
| 06:58.44 | BuenGenio | and do any of you guys have experience with international PBX -> IP -> PBX with billing (minutes/credits) systems? |
| 06:58.49 | WIMPy | irroot: Breakfast? What's that? :-) |
| 06:59.20 | irroot | lol its the thing we do sometimes if we see big glowing ball in sky |
| 06:59.26 | BuenGenio | WIMPy, for me breakfast was the massive dinner from yesterday _ |
| 06:59.28 | BuenGenio | =) |
| 06:59.40 | BuenGenio | that didn't finish digesting overnight :P |
| 06:59.56 | WIMPy | Yes, that's a version I know :-) |
| 07:00.13 | schmidts | irroot you are from the south side of earth, maybe its different at yours :P |
| 07:00.38 | BuenGenio | the Big Yelly |
| 07:00.48 | BuenGenio | it's up and running over here... |
| 07:00.59 | BuenGenio | doing its nuclear thing |
| 07:01.09 | BuenGenio | TGFA/C |
| 07:01.26 | schmidts | havent seen it for two weeks, i guess its even warmer at irroots place even they have winter now :( |
| 07:01.38 | BuenGenio | schmidts, germany? |
| 07:01.49 | schmidts | austria :D |
| 07:01.58 | BuenGenio | heh, close |
| 07:02.09 | irroot | lol main highway closed for ice / snow we dont know how to deal with it ... |
| 07:02.25 | WIMPy | Sounds like Britain :-) |
| 07:02.47 | schmidts | irroot on the weekend we had snow down to 1000m :( |
| 07:03.28 | irroot | hehe well 5cm fell overnight and shutdown the transport network |
| 07:03.46 | irroot | its only 5c here today its freezing |
| 07:04.24 | schmidts | :D its the same here in the winter, only a few snow flakes and everybody goes crazy |
| 07:06.26 | irroot | the snow is still 200km from here |
| 07:07.24 | irroot | have fun today got someone wanting live demo before they have aggreed to installation or ability to put server on network |
| 07:07.51 | irroot | they assume "linux" is a program that can be run on the windows server |
| 07:09.07 | irroot | how is "Testing 10" going |
| 07:10.09 | irroot | anyone running SRTP with the "snom" 80/32 taglen patch ?? |
| 07:10.28 | WIMPy | What patch? |
| 07:11.19 | WIMPy | I haven't tried SRTP in a longer while. I didn't like the built in DOS support. |
| 07:11.37 | irroot | there is a patch to support 32bit auth tag len |
| 07:11.49 | irroot | asterisk only does 80 properly |
| 07:12.02 | WIMPy | Why do you want that? |
| 07:12.18 | irroot | https://reviewboard.asterisk.org/r/1173/ <- here is proper support for it |
| 07:12.31 | irroot | its not the key len that is 128 |
| 07:12.43 | WIMPy | What's wrong with 80 bit? |
| 07:12.45 | irroot | its the tag that is sent in sip header |
| 07:13.03 | *** join/#asterisk singler (~singler@88-119-250-190.static.zebra.lt) |
| 07:13.21 | irroot | nothing wrong with 80bit unless phone uses 32 like snom 6.X |
| 07:13.50 | WIMPy | Oh. Well, maybe you should upgrade the phone then? |
| 07:13.53 | irroot | if the phone asks for 32 it barfs |
| 07:14.05 | irroot | lol tottally |
| 07:14.25 | irroot | but the RFC gives leeway |
| 07:14.33 | WIMPy | I know, but for me that was one click in the phones web-i/f. |
| 07:15.04 | irroot | or proper script to provision it |
| 07:15.38 | WIMPy | That's waht I did after confirming it worked after the click :-) |
| 07:16.26 | irroot | but l-users are not clever like us |
| 07:16.36 | WIMPy | At least until Asterisk crashed. |
| 07:17.28 | irroot | asterisk never crashes :P |
| 07:17.40 | WIMPy | Rarely. |
| 07:17.48 | WIMPy | Usually it just locks up. |
| 07:17.54 | irroot | its "strong like a ox with large penis" <- common local saying |
| 07:18.21 | irroot | yeah unless you use single core :P no locks |
| 07:18.45 | WIMPy | Hmm. Haven't tried that. |
| 07:19.32 | irroot | may get some "blocking" like with timerfd |
| 07:22.31 | *** join/#asterisk wonderworld (~ww@port-92-201-100-159.dynamic.qsc.de) |
| 07:27.55 | schmidts | irroot even with a single core cpu you can have multiple thread which locks each other ,) |
| 07:29.07 | irroot | Start a new party no SMP no HT :P |
| 07:29.17 | *** join/#asterisk cyborg-one (1000@85-238-110-229.broadband.tenet.odessa.ua) |
| 07:30.03 | irroot | solve a great deal of the problem but eliminates 99% of the performance just like real politions |
| 07:30.27 | ChannelZ | We had that, it's called MacOS 9 |
| 07:30.59 | ChannelZ | see, he was running it |
| 07:31.05 | ChannelZ | him too |
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| 07:38.10 | *** join/#asterisk pdtpatr1ck (~pdtpdt@ip68-4-0-113.pv.oc.cox.net) |
| 07:38.33 | pdtpatr1ck | Question - has anyone gotten google voice to work and if so how? Each time i call it says this google person you are trying to reach is not available |
| 07:38.40 | pdtpatr1ck | and takes me to VM .. but then i can get the vm |
| 07:38.58 | pdtpatr1ck | asterisk cli shows no transaction going through |
| 07:39.51 | *** join/#asterisk tamiel (~tamiel@213.30.183.226) |
| 07:40.46 | irroot | just orderd a coke with no iceblocks dam things get stuck in my nose |
| 07:41.06 | irroot | pdtpatr1ck there was a recent protocol change |
| 07:41.29 | irroot | have you tried latest 1.8 SVN or 10b |
| 07:41.32 | pdtpatr1ck | can u please explain or point me to where i can read up.. |
| 07:41.36 | pdtpatr1ck | im using 1.8 right now |
| 07:41.43 | coppice | irroot: coke is usually pretty bad for your nose |
| 07:41.56 | irroot | think its the bubbles |
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| 07:42.55 | ChannelZ | pdtpatr1ck: If you're calling in with GV, you need to make your dialplan answer and then dial 1 to accept the call |
| 07:43.02 | ChannelZ | It's in the examples on the Wiki |
| 07:43.36 | ChannelZ | https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google |
| 07:44.23 | pdtpatr1ck | haha thats the one i followed |
| 07:44.52 | ChannelZ | You might need more delay |
| 07:45.09 | pdtpatr1ck | so change the wait to something like 10? |
| 07:45.23 | ChannelZ | or Google might not be reading the DTMF properly, I've seen people complain about that a lot |
| 07:46.18 | irroot | dont look at me :P oh wait i added dtmf support .... *duck* |
| 07:46.44 | pdtpatr1ck | asterisk doesn't even see the call |
| 07:46.47 | pdtpatr1ck | *sigh* |
| 07:46.47 | ChannelZ | I don't even remember my google voice number right now |
| 07:46.51 | pdtpatr1ck | haha |
| 07:46.58 | ChannelZ | oh well then you have an entirely different problem |
| 07:47.25 | pdtpatr1ck | btw for dialplan configuration .. i noticed they've been using context local all along |
| 07:47.40 | pdtpatr1ck | and then all of a sudden the context is defined in extensions.conf |
| 07:47.59 | pdtpatr1ck | is local just meaning everything else in the extensions.conf ? |
| 07:48.19 | ChannelZ | no it's just a sample |
| 07:49.13 | ChannelZ | They just don't show those sample extens in the 'local' context. You put them where you want and point your gtalk to that context accordingly |
| 07:50.11 | *** join/#asterisk singler (~singler@78-60-139-121.static.zebra.lt) |
| 07:54.29 | pdtpatr1ck | okay so here's my gtalk |
| 07:54.30 | pdtpatr1ck | http://pastebin.com/XQhn4HS0 |
| 07:55.00 | pdtpatr1ck | http://pastebin.com/505KUQzE ==> jabber.conf |
| 07:55.46 | pdtpatr1ck | extensions.conf => http://pastebin.com/iskXusRe |
| 07:56.31 | pdtpatr1ck | and then sip |
| 07:56.31 | pdtpatr1ck | http://pastebin.com/yk8unx85 |
| 07:56.41 | pdtpatr1ck | does something look wrong there to you? |
| 07:57.20 | ChannelZ | useasl=yes |
| 07:57.26 | ChannelZ | in jabber.conf |
| 07:58.23 | pdtpatr1ck | changed it |
| 07:58.25 | ChannelZ | and you might need allowguests=yes in your gtalk.conf for it to work |
| 07:58.36 | ChannelZ | (in [general]) |
| 07:58.40 | pdtpatr1ck | keeps getting google voice subscriber you have called is not available |
| 07:59.19 | ChannelZ | did you reload? Have you turned on jabber debug and see if it's yelling at you about something? |
| 07:59.46 | pdtpatr1ck | let me do that |
| 08:00.49 | pdtpatr1ck | jabber debug doesn't work for me |
| 08:00.51 | ChannelZ | well good luck, I really have to go to bed |
| 08:01.01 | ChannelZ | what do you mean it doesn't work |
| 08:01.08 | pdtpatr1ck | ptaylor*CLI> jabber debug |
| 08:01.08 | pdtpatr1ck | No such command 'jabber debug' (type 'core show help jabber debug' for other possible commands) |
| 08:01.17 | ChannelZ | jabber set debug on |
| 08:01.21 | irroot | cheers ChannelZ catch you on the flip side |
| 08:01.30 | pdtpatr1ck | ptaylor*CLI> jabber set debug on |
| 08:01.30 | pdtpatr1ck | No such command 'jabber set debug on' (type 'core show help jabber set' for other possible commands) |
| 08:01.30 | ChannelZ | (help jabber) |
| 08:01.38 | ChannelZ | Oh then that's your main problem |
| 08:01.42 | ChannelZ | module load res_jabber |
| 08:01.48 | ChannelZ | module load chan_gtalk |
| 08:02.05 | pdtpatr1ck | cannot be loaded |
| 08:02.12 | ChannelZ | There begins your quest |
| 08:02.17 | pdtpatr1ck | :) |
| 08:02.18 | pdtpatr1ck | thanks |
| 08:02.23 | pdtpatr1ck | enjoy the night |
| 08:02.25 | pdtpatr1ck | thanks for everything |
| 08:02.27 | ChannelZ | irroot et al: nighty |
| 08:04.44 | irroot | pdtpatr1ck it working ?? |
| 08:05.06 | irroot | im off to office in bit may be able to help then i have not used G in ages |
| 08:05.11 | irroot | about time i start |
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| 08:07.57 | pdtpatr1ck | no |
| 08:08.04 | pdtpatr1ck | modules are missing |
| 08:08.08 | pdtpatr1ck | trying to recompile |
| 08:08.51 | pdtpatr1ck | /usr/lib/asterisk is missing those two |
| 08:09.54 | *** join/#asterisk wasanzy (~emmanuel@196.201.43.55) |
| 08:10.14 | irroot | ok cool l8r |
| 08:10.43 | *** join/#asterisk Faustov (~fst@gentoo/user/faustov) |
| 08:10.50 | pdtpatr1ck | *sigh* .. this is kinda frustrating . Take care irroot |
| 08:11.16 | irroot | yeah once its built it is built ... build it and they will come :P |
| 08:12.39 | pdtpatr1ck | dang even the .deb packages crap out |
| 08:12.47 | pdtpatr1ck | must not like ubuntu or something |
| 08:13.24 | irroot | use the source luke |
| 08:13.58 | pdtpatr1ck | did but sudo make menuselect or make menuselect does nothing. |
| 08:14.10 | pdtpatr1ck | which is what the guide is saying to use to find out why the modules didn't load |
| 08:15.02 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
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| 08:15.35 | pdtpatr1ck | gonna try 1.6 and see what happens.. thats part of the repo |
| 08:16.03 | irroot | asterisk 1.6 ?? |
| 08:16.10 | irroot | try latest 1.8 |
| 08:16.48 | pdtpatr1ck | yeah thats part of the repo and apt-get installs it fine .. i tried 1.8 and broken dependencies all over the place. Maybe i might be able to get the modules from 1.6 and port it over to 1.8 ? |
| 08:16.57 | pdtpatr1ck | have you gotten 1.8 to work for ya? |
| 08:17.01 | pdtpatr1ck | what os u using ? |
| 08:18.17 | irroot | uses built from source linux |
| 08:18.26 | pdtpatr1ck | how does one get this window? |
| 08:18.27 | pdtpatr1ck | <PROTECTED> |
| 08:18.32 | pdtpatr1ck | https://wiki.asterisk.org/wiki/display/AST/Asterisk+Module+Support+States |
| 08:19.40 | irroot | ok need to get source run ./configure then "make menuconfig" |
| 08:19.48 | irroot | on way to office l8rz |
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| 09:22.05 | BuenGenio | what's the default AsteriskNOW username/password ) |
| 09:31.49 | BuenGenio | if I want to set up an internal office network - can I use localhost as the SIP trunk? |
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| 09:48.28 | irroot | Looking into the "security" requirements to go to astricon its a mess dont think im going |
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| 10:13.11 | *** mode/#asterisk [+o file] by ChanServ |
| 10:19.43 | schmidts | irroot what security requirements do you mean? |
| 10:19.47 | BuenGenio | do I need a trunk for an internal office PBX? |
| 10:21.59 | Gugge | BuenGenio: what is a trunk? |
| 10:22.02 | irroot | schmidts need to have a interview / intterigation ... then the scanners/body searches |
| 10:22.09 | irroot | can rather do withouit |
| 10:25.01 | BuenGenio | Gugge, are you new here or is it me :) |
| 10:25.55 | schmidts | irroot really why? cause you are from south africa? |
| 10:26.30 | Gugge | BuenGenio: im not new. |
| 10:26.45 | irroot | nah its for last 2 years they been rolling out "naked" body scanners and or invasive rubber glove searches |
| 10:27.22 | BuenGenio | by trunk in Asterisk context I understand the outside line through which Asterisk talks to the outside world.... |
| 10:27.39 | irroot | the Visa is 1000R +/- 100E [other charges apply above this] and have to have a interview at embassy before application is concidered |
| 10:28.35 | Gugge | BuenGenio: so a sip peer/friend? :) |
| 10:28.37 | irroot | BuenGenio yes and no ... . but the term "trunking" is refering to IAX2 also |
| 10:28.47 | BuenGenio | yes |
| 10:29.06 | Gugge | BuenGenio: if you want to receive calls from PSTN via sip, you need some provider |
| 10:29.10 | BuenGenio | my goal at this stage is to set up an office PBX - without an outside connection |
| 10:29.18 | BuenGenio | so an inter-desk set up |
| 10:29.26 | Gugge | If you dont need PSTN access, you dont need any outside connections |
| 10:30.58 | BuenGenio | so, I'm inside AsteriskNow now, and trying to add a user -> says no DialPlans defined, but when I try to add a DialPlan it says I don't have any calling rules defined, but it won't let me define calling rules without a "Trunk" |
| 10:31.11 | awk | Is there any asterisk ringtime that will not be supported, aka 7 seconds? |
| 10:31.25 | awk | Or can any value be accepted? |
| 10:31.42 | irroot | ~freepbx |
| 10:31.43 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 10:31.59 | irroot | BuenGenio maybe more help there |
| 10:32.43 | BuenGenio | not actually using freepbx |
| 10:32.48 | BuenGenio | just the Asterisk GUI |
| 10:33.13 | BuenGenio | btw, if you haven't noticed I'm a total n00b with * |
| 10:34.03 | kaii | you dont need an outside line for that |
| 10:36.02 | BuenGenio | When I click Save the dialogue doesn't disappear |
| 10:36.08 | BuenGenio | doesn't actually save |
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| 12:07.04 | kaii | hm .. is it possible to use variable variables in asterisk? lets say ${${DIALSTATUS}_action} for example? |
| 12:07.53 | kaii | sorry wrong question. i know this example does not work. the question is: has anybody got an idea how to do this with a workaround maybe? |
| 12:09.02 | WIMPy | If you think that wil look for something like ${BUSY_action}, that will work. |
| 12:10.11 | irroot | Wimpy maybe the HASH function ?? |
| 12:11.13 | *** join/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 12:11.51 | kaii | WIMPy: hm thanks.. i came up with this to my collegue and he said it doesnt work. i'll go check whats wrong.. |
| 12:12.12 | irroot | kaii core show function HASH |
| 12:12.50 | kaii | irroot: i looked at hash but i dont think that fits. if ${${varname}} doesnt work, i'll go try EVAL |
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| 12:26.24 | [gnubie] | waves |
| 12:29.04 | tuxxie | I have two asterisk servers I would like to allow one server to transfer calls to extension on the other server that start with 71xx. I have created a rule in my dial plan "exten => _720X,1,Dial(SIP/${EXTEN}@10.2.3.247,20,r)" . My issue is that the remote server is not sending my call to it;s local extension. What do i need to do on the server that is getting the transfer to sent it directly to the extension? |
| 12:29.18 | [gnubie] | i am still having a problem with no audio on my snom 300 when the externip's value is the public ip address of the firewall.. kindly check the diagram at http://imagebin.org/164663 for a better idea on my network |
| 12:30.01 | beek | tuxxie: The correct dialplan. |
| 12:31.12 | irroot | tuxxie need a guest account with correct context to allow calls in |
| 12:32.21 | tuxxie | irroot: I will need to configure sip peers between the serves? |
| 12:33.42 | irroot | tuxxie no you wont need too you can call like this |
| 12:34.13 | Dougy | sigh, could use f2knight to come back |
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| 12:36.25 | *** join/#asterisk Rico29 (~rico@sar-b123.olm.fr) |
| 12:36.27 | Rico29 | hi |
| 12:36.46 | Rico29 | just a little question : which version of FFA module should I use with 1.8 svn branch ? |
| 12:39.41 | leifmadsen | latest |
| 12:40.02 | Rico29 | ok, thanks |
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| 12:44.25 | tuxxie | irroot: I am not sure what you mean by setting up a guest account? |
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| 12:52.02 | irroot | tuxxie user |
| 12:53.03 | irroot | [guest] |
| 12:53.05 | irroot | type=user |
| 12:53.07 | irroot | context=ddi |
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| 13:35.06 | *** join/#asterisk rampage73 (~rampage73@mail.co.starke.in.us) |
| 13:37.36 | rampage73 | can i do a three way video conference using asterisk? or does anyone know of a open source project that would allow me to do this? I only have a total of 4 phones going on this system but I need to be able to do a 3 way video call in house without using a service on the internet. |
| 13:37.53 | chazzam | rampage73: try Asterisk 10 beta |
| 13:39.39 | rampage73 | chazzam, thanks for the response and forgive me for being a noob here but I thought asterisk was at version 1.8? |
| 13:39.48 | rampage73 | chazzam, ah nevermind i found it |
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| 13:43.18 | Chainsaw | rampage73: Big Blue Button can do it. |
| 13:43.32 | Chainsaw | rampage73: It uses Asterisk, among other things. |
| 13:45.49 | [gnubie] | can anyone explain to me about the externip parameter inside the sip.conf on asterisk v1.8.5? |
| 13:47.01 | [gnubie] | i mean, why i am getting a different result when assigning a value of a private ip address and on the public ip address? |
| 13:47.49 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 13:49.32 | [gnubie] | when i use a private ip address, the snom 300 hears nothing.. when using a public ip address, the x-lite hears nothing. kindly check the diagram at http://imagebin.org/164663 |
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| 13:51.59 | rampage73 | Chainsaw, big blue button? never heard of it thanks i will do some research |
| 13:52.27 | [gnubie] | the public ip address that is being used in the externip is the one assigned to the firewall/router.. but if i use the private ip address, it is the ip address of the asterisk server itself.. |
| 13:52.32 | Chainsaw | rampage73: It's an open source video conferencing solution mostly used in the academic field. |
| 13:53.34 | [gnubie] | my asterisk box comes only with a single ethernet port and a single ip address which is within the lan |
| 13:55.03 | [gnubie] | anyone? |
| 13:55.26 | rampage73 | chazzam, Chainsaw just in case I do not get the opportunity later thank you for your kindness and information. |
| 13:55.29 | *** part/#asterisk rampage73 (~rampage73@mail.co.starke.in.us) |
| 13:55.50 | anonymouz666 | [gnubie]: use externip AND localnet |
| 13:56.09 | [gnubie] | i actually use both |
| 13:56.27 | [gnubie] | localnet is set to the private network.. |
| 13:57.24 | [gnubie] | but, changing only the value of the externip with the private ip (it's own ip address) and the public ip address (firewall's public ip address) differs the result.. |
| 13:58.30 | [gnubie] | neither of the values solves my problem - the no audio problem |
| 13:58.56 | [gnubie] | on the firewall side, i set the dmz only to the asterisk box's private ip address |
| 14:08.22 | Hive | Hello everyone. I'm trying to make an attended transfer to an outside number, but when I press the "transfer" button on my polycom 500, a hangup is issued to all parties. |
| 14:08.37 | Hive | I have a debug log of the situation if anyone would care to take a look |
| 14:08.42 | leifmadsen | Hive: pastebin |
| 14:08.52 | leifmadsen | Hive: (this is not a volunteer to look at the issue though :)) |
| 14:09.34 | Hive | http://pastebin.com/qzP89mnf |
| 14:14.00 | *** join/#asterisk byronsmith (~smithers@84.12.253.146) |
| 14:15.14 | byronsmith | hey all, I see 'astertest' no longer exists as a website so I'm unable to download their Asterisk stress testing tool. Can anyone recommend a similar tool for stress testing? |
| 14:15.57 | byronsmith | more info here: http://www.asteriskguru.com/tutorials/astertest.html |
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| 14:24.52 | gork | Hello -- I have a couple of questions about SRTP that I cannot locate a 100% sure answer to in the docs or other material -- I have TLS/SRTP configured on 1.8.5 with polycom phones (transport=tls, encryption=yes in sip.conf and sec.srtp.enable="1" sec.srtp.offer="1" sec.srtp.require="1" in polycom phone config). First question: is there any way on the asterisk side outside of packet capture to verify that the RTP streams are actually e |
| 14:24.54 | *** join/#asterisk coppice (~chatzilla@116.92.35.40) |
| 14:25.22 | Naikrovek | verify that they actually are encrypted? |
| 14:25.26 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 14:25.43 | Naikrovek | show sip peers will have an (E) next to the last column if communications are encrypted, I believe |
| 14:25.58 | Naikrovek | sip show peers, rather |
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| 14:29.32 | gork | it does not appear to do that; no. sip show tcp is the only indication that TLS is being used.. sip show peer <peer> shows Encryption: Yes but what i was hoping is that sip show channel <something> would indicate it or something.. even rtp set debug on does not indicate if the packets are encrypted sadly |
| 14:30.09 | chazzam | is there a sip show channels verbose? |
| 14:30.27 | gork | nope |
| 14:30.32 | chazzam | I know there is a core show channels verbose, but I dunno that it would show encryption there |
| 14:31.21 | chazzam | Wouldn't you be able to tell from the SDP information whether it was establishing the connection using encryption? |
| 14:31.33 | gork | it's ok i can do a packet trace.. what im really hoping here is that setting sec.srtp.require on the polycom will simply not allow the phone to transmit or receive unencrypted RTP |
| 14:32.16 | Naikrovek | what are you talking about with these phones |
| 14:32.26 | Naikrovek | what's so vital that it be encrypted? |
| 14:32.28 | Naikrovek | curious |
| 14:34.35 | gork | Just business sensitive calls that are going over public networks -- more of a compliance matter actually |
| 14:34.37 | Naikrovek | just make sure the phones talk to each other once the RTP stream is up. If Asterisk is in the mix (such as if you're recording a call) then Asterisk will maybe decrypt the stream for a time |
| 14:34.43 | Naikrovek | fair enough |
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| 14:37.00 | gork | if i route it over VPN i lose the ability to advantage myself of any ability to shape/prioritize traffic on the edge or with the isp because they cant identify that it's sip or rtp or whatever .. so SIP/tls + SRTP actually provides an advantage over IPSEC tunnels or MPLS or any of that nonsense. |
| 14:37.45 | gork | the assumption of course is that setting sec.srtp.required=1 on the polycoms will simply not allow them to transmit or receive plain RTP but the docs dont say that so i was just hoping there was an easy way to verify it if * was in the media path |
| 14:40.15 | _Corey_ | gork: You can always sniff the traffic with wireshark or something to ease your mind |
| 14:40.42 | _Corey_ | just try to play back the media... it should sound like white noise |
| 14:43.08 | gork | _Corey_: yes thats what im gonna have to do; its just kind of a pain since I have to also have it sniff/decrypt the SIP session which is using TLS to get it to actually pick up the RTP packets as RTP |
| 14:43.36 | gork | it doesnt seem to recognize RTP without being able to decode the signalling from what i have seen |
| 14:43.46 | _Corey_ | gotcha |
| 14:46.21 | gork | second question is a problem i have noticed between * and polycom when using TLS; and im unsure whether or not it's a polycom bug or an asterisk bug: Polycom 3.3.1 when using TLS transport and SRTP will not respond to asterisk's request to reauthenticate on INVITE (since the tls session has already been authenticated when the phone registered, this actually seems reasonable) so i have to set insecure=invite in sip.conf |
| 14:46.56 | gork | first question is whether or not this is safe considering that it's using transport=tls, and second question is which end is misbehaving |
| 14:47.55 | Naikrovek | you could do it over the vpn and prioritize at the vpn endpoints. you needn't rely on the ISP for that |
| 14:53.58 | gork | Naikrovek: well i didnt say its impossible just that its an awful pain.. you ever try to get a cisco rotuer to set the IP Precedence bits on VPN traffic *after* its been encrypted? and then still residential isps and stuff (much as we hate them) do actually tend to classify and prioritize RTP so long as they can identify it (at least from what I am told) |
| 14:54.26 | vader-- | how is centOS 6 coming along? I am thinking about building a VM for a small office's freePBX, and was wondering if i should go with 5.6 or 6 |
| 14:54.51 | Naikrovek | it's released... beyond that i know little |
| 14:54.55 | Naikrovek | i'm sure it's fine |
| 14:54.59 | Naikrovek | check the list of known bugs |
| 14:55.11 | raden | Naikrovek, yooo bro ! |
| 14:55.24 | Naikrovek | also they ditched xen altogether (if you're using it as a vm host, that matters.) |
| 14:55.26 | Naikrovek | heya |
| 14:55.26 | pdtpatr1ck | Question .. im trying to get google voice to work. Here are my configs : http://paste.ubuntu.com/653151/ .. http://paste.ubuntu.com/653152/ ... http://paste.ubuntu.com/653154/ ...http://paste.ubuntu.com/653155/ |
| 14:55.27 | [gnubie] | anyone familiar on the use of the externip parameter here on the sip.conf and isolating problem on no audio? |
| 14:55.31 | raden | vader--, go with 5.6 more stable |
| 14:55.53 | pdtpatr1ck | the only thing i see on the cli is this: JABBER: pdtpatrick INCOMING: <presence from="pdtpatrick@gmail.com/f33b6b67" to="pdtpatrick@gmail.com"><show>xa</show><c hash="sha-1" node="http://telepathy.freedesktop.org/caps" ver="7U/rcOIc96YJUWwRYbSbF768gng=" ext="pmuc-v1 share-v1 voice-v1 video-v1" xmlns="http://jabber.org/protocol/caps"/><status/><x xmlns="vcard-temp:x:update"><photo/></x></presence> |
| 14:56.02 | beek | Hive: It's been a while but IIRC the only way I could get transfers to work properly was to ensure that I had "canreinvite=no" in sip.conf for each phone so that they HAD to go through Asterisk. |
| 14:59.24 | *** join/#asterisk d-_-b- (~d-_-b-@2607:f370:9999:dead:5ab0:35ff:fef7:6be3) |
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| 15:00.59 | Hive | Beek, thanks for the recommendation. I just tried canreinvite=no without any success. |
| 15:01.20 | beek | Hive: You did reload sip or restart Asterisk, right? |
| 15:01.21 | *** join/#asterisk bn-7bc (~bjarne-im@pdpc/supporter/active/bn-7bc) |
| 15:01.28 | Hive | I also tried "canreinvite=yes" and also "directrtpsetup=yes" which i guess is similar to canreinvite |
| 15:01.32 | Hive | yes, reloaded |
| 15:01.41 | Hive | cant restart :( users are online |
| 15:01.51 | beek | Okay... just making sure. |
| 15:01.58 | Hive | :) |
| 15:02.11 | beek | You can use "sip show peer" and one of your peers to ensure that it's picked up the parameter. |
| 15:02.11 | Hive | It's a transfer to an outside number which i feel like should not make a difference to asterisk |
| 15:02.19 | *** part/#asterisk [gnubie] (~marvin.pa@cm36.kappa8.maxonline.com.sg) |
| 15:02.22 | Hive | kk just a moment |
| 15:02.42 | _Corey_ | Hive: If you're on 1.8+ the option changed... you may need to do "directmedia=yes" |
| 15:02.45 | chazzam | didn't canreinvite become directmedia? |
| 15:02.51 | chazzam | err what _Corey_ said |
| 15:03.05 | Hive | ahh, this is good to know |
| 15:03.05 | beek | My example is from 1.4 & 1.6 |
| 15:03.21 | Hive | im on 1.8.4.4 (sorry for not making that clear) |
| 15:03.36 | _Corey_ | Yeah, I didn't read the changelog and had to discover that one the hard way... ;) |
| 15:03.55 | Hive | I just did sip show peer on my phone and it seems that DirectMedia is set to yes already |
| 15:04.04 | Hive | going to try turning that off... maybe something good will come of it heh |
| 15:04.12 | chazzam | pdtpatr1ck: they keep changing the format, make sure to try the latest release, or HEAD of SVN, and if it still doesn't work... then try to patch it? |
| 15:04.27 | _Corey_ | Hive: Yeah, I meant =no actually sorry |
| 15:05.29 | pdtpatr1ck | im using 1.8.5 |
| 15:06.08 | chazzam | pdtpatr1ck: then try HEAD of SVN, and if it still doesn't work... then try to patch it? |
| 15:06.18 | Hive | Hmmm |
| 15:06.27 | Hive | No luck |
| 15:06.38 | Hive | hangup is still being issued to all parties when i hit "transfer" button |
| 15:07.18 | pdtpatr1ck | chazzam, okay i will try that. Question - are you using that service? |
| 15:08.10 | chazzam | I was a while back, then I ran into the make menuselect issue and couldn't get any of hte channels compiled for a while |
| 15:08.17 | Hive | Thanks for trying to help though _Corey_ |
| 15:08.30 | chazzam | and now my box is turned off due to losing power, so ... I dunno where it stands anymore |
| 15:09.10 | pdtpatr1ck | yeah menuselect doesn't compile so i cannot get chan_gtalk and res_jabber |
| 15:09.16 | _Corey_ | Hive: I missed your original description of the problem, but it sounds like you have something else going on |
| 15:09.34 | pdtpatr1ck | so i had to use asterisk 1.6 which installs it but no incoming calls or anything |
| 15:09.39 | _Corey_ | directmedia should only help you with audio going somewhere strange |
| 15:09.49 | pdtpatr1ck | i think im just going to get myself a DID and forget this google voice stuff .. too much work |
| 15:10.06 | Hive | I'm trying to make an attended transfer to an outside number, but when I press the "transfer" button on my polycom 500, a hangup is issued to all parties. (Asterisk 1.8.4.4) Debug log is at http://pastebin.com/qzP89mnf |
| 15:10.11 | chazzam | yeah, apparently google keeps changing it which means code updates all the time =/ |
| 15:10.21 | Hive | Ahh ok |
| 15:10.30 | Hive | Yes I'm quite confused with this current problem |
| 15:11.32 | Hive | also the fact that the debug looks like hieroglyphics to me doesnt help hehe |
| 15:13.27 | pdtpatr1ck | chazzam, yeah that just sounds painful. Who wants to change their settings so often. Oh well back to my DiD option. On that note - do u have any sip service provider you've used and like that could be good for home users? |
| 15:13.52 | _Corey_ | Hive: I see a retransmit there.. wondering if maybe Asterisk is hanging up because of that |
| 15:14.57 | chazzam | pdtpatr1ck: I was trying to skip buying SIP service for a while longer by using google voice myself, so unfortunately, I don't have |
| 15:15.05 | _Corey_ | Hive: Are those IPs in your debug bogus or are you in africa? |
| 15:15.15 | Hive | hahaha they are bogus |
| 15:15.24 | Hive | I was worried about putting too much information online :x |
| 15:15.58 | pdtpatr1ck | chazzam .. yeah damn you google and your smart ways! BTW we use velocity at work and they've done a fine job. I haven't tried them for home use so we'll see in couple of days how that pans out and whether it is cheap enough to be worth it |
| 15:16.22 | Hive | I'm not sure what the deal with those retransmits are. or what they do really |
| 15:16.24 | Hive | googling now |
| 15:16.31 | *** join/#asterisk iratik (40c68559@gateway/web/freenode/ip.64.198.133.89) |
| 15:16.55 | *** join/#asterisk anonymouz666 (~anonymouz@187-28-37-118.poolip.RJO.embratel.net.br) |
| 15:16.56 | chazzam | pdtpatr1ck: ah ok, cool |
| 15:17.36 | anonymouz666 | irroot: ITSP that supports T.38 should RE-INVITE with T.38 stuff, after detect the fax tone at the far end ? |
| 15:18.06 | anonymouz666 | and before the dial to ITSP I should have FAXOPT with t38gateway = yes |
| 15:19.01 | iratik | I know this is not the place, but perhaps you guys might know where I should look. I'm looking for a place to sell a 4 year license for queuemetrics we won't be needing ? Is there something like a classifieds site for call centers? |
| 15:23.57 | Hive | Weirrrrrrd... Transfers to internal extensions work... |
| 15:25.23 | ChannelZ | pdtpatr1ck: remember you need to ./configure before make menuselect will do anything |
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| 15:27.34 | _Corey_ | Hive: I've seen things like the rpid updates in 1.8+ cause issues with some equipment, you may want to disable that stuff on your provider SIP peer |
| 15:28.30 | pdtpatr1ck | ChannelZ, i've done that .. configure works fine and then when u do menuselect it just does about 10 lines and says leaving directory. |
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| 15:36.10 | McBoing | MusiconHold(30) <--- I thought that played the musiconhold for default for 30 seconds but it just keeps on trucking |
| 15:36.55 | _Corey_ | it's MusicOnHold(class,duration) |
| 15:36.59 | McBoing | sorry nm, I see now MusicOnHold([class]) |
| 15:37.30 | McBoing | oh I knew there was a duration in there somewhere heh so would "MusiconHold(,30)" work? |
| 15:38.43 | McBoing | can always try, been having some trouble getting an interent stream going to my moh, but still digging around |
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| 15:42.01 | _Corey_ | I doubt it, but MusicOnHold(default,30) should work |
| 15:42.32 | McBoing | when a variable/class is declared like this "default =>quietmp3:/var/lib/asterisk/mohmp3/stream,http://ckqb.player.amri.ca/" do I call class "default" or is it the [name] heading above it that I call? |
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| 15:45.38 | fireman_biff | Hi, is it possible to use a call file to make an external call from my extension, using a specific trunk instead of the default one for that external number? |
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| 15:50.20 | Hive | _Corey_: looking into that now |
| 15:50.49 | Hive | _Corey_: you're talking about stuff in sip.conf yes? |
| 15:51.00 | _Corey_ | yes |
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| 15:58.15 | ChannelZ | pdtpatr1ck: you're probably missing ncurses or something |
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| 15:59.07 | Hive | _Corey_: Once again thanks for the direction. I'm kind of hesitant to mess with my provider peer durring business hours so i'll probably try that out later :) |
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| 16:05.12 | Hive | McBoing, http://www.voip-info.org/wiki/view/Asterisk+config+musiconhold.conf |
| 16:05.25 | Hive | See the part that says [default] |
| 16:05.31 | Hive | that is the class name |
| 16:06.00 | _Corey_ | Yeah, no problem. I'm not saying it's your fix but I've had some issues with RPID updates breaking stuff and completing the transfer would be when that'd happen, so it's worth a look |
| 16:07.42 | Hive | At this point I'm willing to try anything. The offices I provide for dont seem to do any external transfers which is probably why I haven't received any tickets about it yet hehe |
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| 16:33.53 | paulc | Caller A is in the Queue application talking to Agent B, who then redirects Caller A to a different queue via AMI. Straw poll: Would you expect caller A's h extension to fire in this scenario? |
| 16:34.27 | ChannelZ | not just from the redirect, no |
| 16:34.38 | ChannelZ | or do you mean when they really do finally hang up? |
| 16:34.56 | Bipul | p3nguin, ping |
| 16:37.21 | paulc | ChannelZ: I meant from a redirect. Only caveat here is I triggered the h extension doing a redirect from the console, the AMI bit is still pending. But either way, my expectation is that a redirect shouldn't trigger h extension on the original call - seem reasonable? |
| 16:44.41 | *** join/#asterisk wasanzy_ (~emmanuel@196.201.43.55) |
| 16:44.48 | wasanzy_ | hi guys |
| 16:45.11 | ChannelZ | paulc: I would think it wouldn't since that channel has never terminated, but I could be wrong |
| 16:45.33 | ChannelZ | aloha wasanzy_ |
| 16:45.50 | paulc | ChannelZ: Yeah, that was kind of my expectation. I'm going to do some more testing and see how it goes. The current behaviour is a bit of a pain for my application. |
| 16:46.42 | ChannelZ | Sure it wasn't the 'h' exten of the agent? |
| 16:46.47 | *** join/#asterisk n0tk (~n0tk@gateway/tor-sasl/n0tk) |
| 16:50.21 | paulc | ChannelZ: I'm pretty sure it was the original call.. like 98% sure.. don't have the box+phone in front of me right now |
| 16:50.52 | paulc | I could see it being the agent channel, that'd be understandable.. but it definitely looked like the original caller channel.. I'll test some more and see how it goes.. |
| 16:50.57 | paulc | thanks for the sanity check though :-) |
| 16:51.23 | wasanzy_ | am compiling Zaptel and dahdilinux and this is the error am getting: fatal error: linux/autoconf.h: no such file or directory, I searched google but couldn't find any solution |
| 16:52.02 | wasanzy_ | I don't know is this have to do with kernel version or some thing |
| 16:52.05 | irroot | wasanzy_ Zaptel is replaces with DAHDI |
| 16:52.10 | wasanzy_ | any idea? |
| 16:52.20 | *** join/#asterisk joelsolanki (~Joel@123.237.173.95) |
| 16:52.29 | wasanzy_ | so no need to use Zaptel |
| 16:52.46 | ChannelZ | install autoconf |
| 16:52.59 | wasanzy_ | ok, but still when am compiling the DAHDI, I get that error |
| 16:53.14 | wasanzy_ | I installed autoconf but still |
| 16:53.16 | irroot | yeah ^^^ ChannelZ |
| 16:53.51 | wasanzy_ | that was the first thing I did when I got the error but still |
| 16:53.51 | irroot | start with a clean base |
| 16:54.09 | wasanzy_ | make clean? |
| 16:54.13 | ChannelZ | hmmm.. |
| 16:54.31 | leifmadsen | or make distclean |
| 16:54.39 | wasanzy_ | ok |
| 16:54.44 | ChannelZ | oh |
| 16:54.45 | irroot | and have a clean linux dev |
| 16:54.51 | leifmadsen | make distclean && ./configure |
| 16:54.53 | ChannelZ | do you have the linux kernel headers installed? |
| 16:55.01 | ChannelZ | you need those for building drivers |
| 16:55.05 | leifmadsen | (and do they match the kernel you're running) |
| 16:55.19 | irroot | its autoconf.h from the linux tree ... snap ChannelZ |
| 16:55.25 | wasanzy_ | am running ubuntu 11.04 |
| 16:55.41 | irroot | wasanzy_ need the kernel dev package |
| 16:55.48 | ChannelZ | I don't actually have autoconf installed here so it must be coming from elsewhere - make sure you have the kernel headers |
| 16:56.09 | wasanzy_ | ok, let me install that and see |
| 16:56.44 | ChannelZ | look for "linux-headers-generic" if you're running the generic kernel for instance |
| 16:57.15 | wasanzy_ | ok |
| 16:59.58 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 17:10.04 | *** join/#asterisk tzafrir_laptop (~tzafrir@2001:41b8:203:dc11:21c:23ff:fe20:991f) |
| 17:15.56 | wasanzy_ | I got the lunix-header-generic install but still the same error |
| 17:18.53 | Qwell | What is the error, exactly? |
| 17:19.08 | Qwell | pastebin the entire thing |
| 17:19.17 | wasanzy_ | ok |
| 17:23.22 | irroot | wasanzy_ "ls /usr/include/linux" <- PB that too |
| 17:24.51 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 17:26.22 | wasanzy_ | http://pastebin/ncYc2h6c |
| 17:26.30 | wasanzy_ | that is the error |
| 17:26.59 | wasanzy_ | irroot: ok, I will check that as well |
| 17:30.36 | wasanzy_ | irroot: there is no autoconf.h file there |
| 17:31.07 | wasanzy_ | but when trying to reinstall it, it is telling me , is already the latest version |
| 17:31.44 | irroot | /usr/src/linux/include/linux/autoconf.h |
| 17:32.08 | irroot | try do a clean build now |
| 17:33.16 | *** join/#asterisk voipnet-tech (~voipnet-t@66.63.72.130) |
| 17:33.22 | irroot | recent 2.6 kernels changed things a bit /usr/src/linux/include/linux i used to ln to /usr/include/ |
| 17:33.25 | voipnet-tech | hello once again asterisk chat :) |
| 17:34.13 | Qwell | wasanzy_: ...you didn't put anything on that pastebin. |
| 17:35.25 | irroot | Qwell lol thought me browzer was fritzen |
| 17:35.47 | voipnet-tech | so.... what brings me here today you might ask... well... i'm having trouble with directrtpsetup... not sure if i'm doing something wrong perhaps. originally, my * box anchored all the RTP. Now that i've enabled directrtpsetup * is only sending the phone's IP in the SDP in one direction: ie: assymetrical RTP. server still anchors one direction of the audio. what might i do to start troubleshooting it? I have pcaps of the SIP messaging and can |
| 17:35.48 | McBoing | still trying to get streaming to work, now just trying to figure out if its my link that is the problem or the method I am using http://pastebin.com/KDY5LFSu, can someone take a quick peak? |
| 17:35.52 | ChannelZ | dpkg -S autoconf.h gives me linux-headers-2.6.32-33-generic: /usr/src/linux-headers-2.6.32-33-generic/include/linux/autoconf.h |
| 17:35.58 | ChannelZ | This on Ubuntu 10 but still... |
| 17:38.08 | wasanzy_ | http://pastebin.com/ncYC2h6c |
| 17:38.22 | wasanzy_ | sorry if I did typing mistake |
| 17:38.46 | wasanzy_ | irroot, sorry there is no path like that on the machine |
| 17:39.15 | irroot | wasanzy_ you have linux source package installed at all |
| 17:39.24 | irroot | recent 2.6 kernels changed things a bit /usr/src/linux/include/linux* i used to ln to /usr/include/ |
| 17:42.17 | wasanzy_ | I have this: /usr/src/linux-headers-2.6.35.22-generic/include/linux and /usr/src/linux-headers-2.6.35.22/include/linux but there is nothing autoconf under them |
| 17:42.39 | trollnet-tech | might there be anyone here gracious enough to share with me their experiences with getting IP phones to send direct RTP when able? |
| 17:43.04 | Hive | McBoing, did you download that mpg123 file? |
| 17:43.18 | Hive | http://pbxinaflash.com/forum/showthread.php?t=2666 looks like it addresses what you are trying to do |
| 17:43.26 | Hive | (kind of old though so some stuff might have changed) |
| 17:43.58 | wasanzy_ | trollnet-te:put this in your sip.conf directmedia=yes |
| 17:44.42 | irtroll | <PROTECTED> |
| 17:44.45 | ChannelZ | wasanzy_: http://forums.virtualbox.org/viewtopic.php?f=7&t=38584 |
| 17:45.31 | wasanzy_ | ok |
| 17:45.33 | ChannelZ | wasanzy_: do you have /include/generated ? |
| 17:45.54 | wasanzy_ | let me check |
| 17:47.18 | McBoing | Hive: yeah I have mpg123 all setup, I am pretty sure its my url |
| 17:47.31 | wasanzy_ | No I don't have |
| 17:47.51 | Hive | Strange, that url opens a stream for me |
| 17:48.17 | wasanzy_ | let me locate autoconf and know where it is hidden so I can ln |
| 17:49.32 | ChannelZ | you might have to run the mlocate job to update the DB |
| 17:49.39 | ChannelZ | else use find |
| 17:49.51 | ChannelZ | it seems like something whack is going on with Ubuntu 11 |
| 17:50.14 | wasanzy_ | ahhhh I found it here: /usr/src/linux-headers-2.6.35.22-generic/include/generated/autoconf.h |
| 17:50.47 | wasanzy_ | I do link it to the right path and see |
| 17:51.39 | McBoing | Hive: Asterisk plays the stream I have setup for you? or you mean your browser? |
| 17:52.21 | Hive | McBoing, my browser. The url for that stream opens the stream on my PC |
| 17:52.56 | McBoing | Hive: yeah of course that works for me, but when applying it to asterisk I see it attempts to run the class but its quiet, so most likely the url isnt working |
| 17:53.01 | *** join/#asterisk Deetz (~pauldittr@174.4.128.244) |
| 17:53.33 | Hive | McBoing, what does the line look like where you set your MoH? |
| 17:54.52 | *** join/#asterisk elfelvin (~elfelvin@87-194-69-88.bethere.co.uk) |
| 17:55.34 | McBoing | Hive: http://pastebin.com/KDY5LFSu |
| 17:55.57 | anonymouz666 | I am in trouble with this: t.30 machine -> Asterisk -> t.38 -> itsp -> PSTN fax machine |
| 17:57.10 | pdtpatrick | Do you guys know of any proxy servers i could use for sip communication.. AT&T is a pain in the butt. The one im currently using uses TLS and registers fine but when u call u cannot hear speech. |
| 17:58.34 | Hive | McBoing, i mean where you say Set(CHANNEL(musiconhold)=stream) or something like that |
| 17:58.39 | Hive | how are you applying the music on hold class |
| 17:59.24 | McBoing | Hive: oh I use exten => 4001,n,MusiconHold(stream,30) |
| 17:59.24 | *** join/#asterisk fullstop (~fullstop@173.210.91.4) |
| 17:59.33 | fullstop | Afternoon, all |
| 17:59.43 | Hive | do you know what version of asterisk you are running? |
| 18:00.02 | fullstop | Dumb question. Is there any point to installing dahdi if I have no dahdi hardware? |
| 18:00.41 | jaytee | fullstop, what version of *? |
| 18:00.45 | Hive | this is how I set my MOH class exten => blah,n,Set(CHANNEL(musicclass)=className) |
| 18:00.53 | fullstop | jaytee: 1.6.2.X |
| 18:01.06 | Hive | i see you are trying to only have it play for a certain time, but that is the only way I was able to make it work |
| 18:01.19 | Hive | Try that syntax out and see if it works |
| 18:02.40 | jaytee | fullstop, unless you need to use the MeetMe application you won't need dahdi for timing. |
| 18:03.05 | fullstop | jaytee: good to know. |
| 18:04.22 | McBoing | Hive: got a working stream now, it was the url, wow this is nice |
| 18:04.57 | fullstop | also, I tried updating from 1.6.2.11 to 1.6.2.19, and something in chan_sip has changed.. another sip peer was rejecting our calls with "SIP/2.0 481 Call/Transaction Does Not Exist" |
| 18:05.13 | fullstop | I rolled back to 1.6.2.11 and things were fine again. |
| 18:05.42 | Hive | McBoing, nice! strange that I couldn't get MusiconHold(class) to work for me when I set mine up a couple days ago haha. |
| 18:05.49 | fullstop | My concern is that I will again have the problem when we eventually move to 1.8.X (or 10.0 Ha!) |
| 18:06.13 | McBoing | Hive: no point, just use MusiconHold([class]), bam it works |
| 18:06.59 | Hive | no point what? |
| 18:10.30 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v000-096.mobile.uci.edu) |
| 18:11.56 | *** join/#asterisk KryptoKnight (~MDalby@cpc16-stkp7-2-0-cust163.10-2.cable.virginmedia.com) |
| 18:11.57 | mickecarlsson | If I want to use cdr_adaptive_odbc.conf instead of cdr_mysql.conf, do i also need cdr_odbc.conf? |
| 18:12.14 | leifmadsen | dont' think so |
| 18:12.17 | mickecarlsson | Or how do I match connection= |
| 18:12.18 | leifmadsen | just res_odbc |
| 18:12.23 | mickecarlsson | ok, looking |
| 18:12.40 | leifmadsen | cdr_adaptive_odbc --> res_odbc --> unixodbc |
| 18:12.47 | KryptoKnight | hey guys, I've had a trawl of google and can't seem to find any answers, maybe you can help? Is there any way to set rtp port ranges on a per peer basis? |
| 18:12.57 | leifmadsen | KryptoKnight: no |
| 18:12.59 | mickecarlsson | ok, so a match between connection= to section in res_odbc.conf |
| 18:13.03 | KryptoKnight | feck |
| 18:13.03 | leifmadsen | mickecarlsson: yes |
| 18:13.06 | KryptoKnight | :) |
| 18:13.11 | mickecarlsson | Thanks, batc to testing |
| 18:13.15 | mickecarlsson | back that is |
| 18:13.22 | ddickenson | are there advantages to using ODBC vs mysql for CDR? especailly if you're using mysql for other db related things like realtime and queue stats |
| 18:13.47 | mickecarlsson | cdr_mysql is deprectaed |
| 18:13.52 | mickecarlsson | deprecated that is |
| 18:15.08 | ddickenson | ahh, didn't know that. Have been working with older versions on most of my installs. Forgive my ignorance but is ODBC a database in and of itself or what? Basically what is the backend db for the ODBC route |
| 18:15.45 | mickecarlsson | http://en.wikipedia.org/wiki/ODBC |
| 18:15.51 | ddickenson | fair enough |
| 18:16.12 | mickecarlsson | :-) |
| 18:17.19 | ddickenson | so from what I'm seeing it is just the interface between the two, the database could still be a sql database |
| 18:17.28 | keith4 | is it possible to disable the "from <someone> at <timestamp>" that prefaces voicemail playback? |
| 18:17.49 | mickecarlsson | ddickenson: Yes, you can use whatever database you want that has odbc interface |
| 18:17.50 | wasanzy_ | ODBC I think is not a database but in can say is a driver for database connections |
| 18:18.31 | Qwell | keith4: Sure. Asterisk is open source. |
| 18:18.35 | ddickenson | mickecarlsson: thanks |
| 18:19.46 | keith4 | Qwell: without recompiling app_voicemail? ;-) |
| 18:19.49 | mickecarlsson | That seems to work: [2011-07-27 20:17:19] NOTICE[22662] res_odbc.c: res_odbc: Connected to testing [MySQL-asterisk] |
| 18:19.57 | mickecarlsson | [2011-07-27 20:17:19] VERBOSE[22662] cdr_adaptive_odbc.c: -- Found adaptive CDR table cdr@testing. |
| 18:20.11 | mickecarlsson | Ok, so now its time to make some calls |
| 18:23.23 | keith4 | Qwell: ah, finally found the "envelope" setting |
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| 18:32.31 | ddickenson | mickecarlsson: what would be best practice as far as cdr on a box that has realtime already on it, same db as realtime and it's own table or seperate db for cdr data? |
| 18:33.05 | mickecarlsson | Well, I would go for a separate db |
| 18:33.30 | mickecarlsson | You can move that to another server if you want without to much hazzle |
| 18:34.26 | mickecarlsson | In FreePBX we have db=asterisk for configuration and db=asteriskcdrdb for cdr |
| 18:34.50 | mickecarlsson | A lot of our users move cdr to another server |
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| 18:37.46 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 18:37.54 | fireman_biff | Hi, can anybody recommend an asterisk appliance for a 100 room hotel? |
| 18:37.59 | fireman_biff | analog phones |
| 18:38.20 | *** part/#asterisk fasta (1334259@xs8.xs4all.nl) |
| 18:38.21 | WIMPy | analog is evil. |
| 18:38.48 | WIMPy | icl. billing? That sounds like fun. |
| 18:38.48 | Jcook_5xData | Need some help. I have polycom 650 I am trying to load a separate 0000000000-directory.xml file per phone. I have them set as macaddress-directory.xml. for some reason it is not loading.anyone have an idea? |
| 18:38.49 | fireman_biff | WIMPy: i agree |
| 18:38.56 | fireman_biff | evil |
| 18:39.13 | fireman_biff | yes, including billing if possible |
| 18:42.41 | pdtpatrick | Question .. i'm seeing this.. has something seen this before? |
| 18:42.42 | pdtpatrick | http://pastebin.com/iVhQyKEH |
| 18:43.01 | Hive | Does anyone use sipstation as a SIP provider? |
| 18:43.25 | *** join/#asterisk adyn (~adyn@pdpc/supporter/active/adyn) |
| 18:47.18 | fireman_biff | if I'm using 4 or 5 channel banks to power analog phones, is there any need to get a proper asterisk appliance to work with the channel banks, or should a regular small server be fine? |
| 18:48.14 | WIMPy | Do you think some sort of appliance has more power that a standard PC? |
| 18:48.15 | fullstop | pdtpatr1ck: I'd turn on sip debugging. It looks like it is hanging up when you answer. |
| 18:48.47 | fullstop | what's up with the TIMEOUT call? |
| 18:49.07 | WIMPy | pdtpatrick: Who did that dialplan? It looks strange. Or wasn't that one call? |
| 18:49.30 | fireman_biff | WIMPy: well I'm looking at Rhino's appliance and it seems pretty powerful, but more importantly we already have fairly decent servers in stock (nothing too powerful though) |
| 18:50.33 | WIMPy | fireman_biff: To just make phone calls you get away with very little. You just need enough slots for the interfaces. |
| 18:50.39 | fullstop | I know that this is frowned upon, but, for running asterisk 1.6.2 in a VM, what timing module would you use? |
| 18:50.45 | fullstop | timerfd or pthread? |
| 18:51.11 | fireman_biff | WIMPy: thanks, how do channel banks connect to the server? pri cards? |
| 18:51.25 | fireman_biff | (T1) |
| 18:51.27 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 18:51.38 | WIMPy | That's the version I have heard of, yes. |
| 18:52.00 | fireman_biff | have you ever seen a pri card with more than 2 ports? |
| 18:52.17 | fireman_biff | nevermind |
| 18:52.18 | fireman_biff | found one |
| 18:52.24 | WIMPy | Yes, they are available with 1, 2, 4 or 8 ports. |
| 18:52.46 | fireman_biff | 8, cool didn't find that one yet, what brand? |
| 18:53.02 | WIMPy | Sangoma |
| 18:53.38 | fireman_biff | thanks |
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| 19:04.34 | bent_screwdriver | i have an issue where if user "A" answers a call then transfers to user "B" and user "B" doesn't answer, after the timeout the caller is dropped into user "A" voicemail. any ideas? Asterisk 1.6.2.6 - http://pastebin.com/BHEkmn4Y |
| 19:05.04 | *** join/#asterisk bamdad (~bamdad@86.109.36.175) |
| 19:05.16 | bamdad | Hi |
| 19:05.41 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 19:08.06 | bamdad | I want to run a script when i have incoming pstn call , i've edited the extention_custom.conf [from-custom-pstn] context but the event doesnt triger when I have incoming calls , any clue? |
| 19:08.07 | bent_screwdriver | also, the user is transferring with the phone transfer button (Poly ip650) |
| 19:08.28 | *** join/#asterisk nny (~SM@cpe-174-107-223-014.sc.res.rr.com) |
| 19:08.46 | pdtpatrick | I am getting this on asterisk CLI. Can someone please help -- http://pastebin.com/iVhQyKEH |
| 19:08.49 | bamdad | the script works with diffrent contexts , i have problem only with pstn calls |
| 19:10.36 | nny | need some walls to bounce off of. Chasing down a problem, the ultimate symptom is all the phones in the network end up being reported as <guest> like 10.0.0.109 (None) 29606be4-bfeaf7 0x0 (nothing) No Rx: SUBSCRIBE <guest>. it's like the sip stack is dying. I can't do a backtrace due to the volume ont he system, when it's enabed the load surpasses anything useful. I am having to do some seat of the pants diagnostic. What would cause the symptom |
| 19:10.50 | WIMPy | bamdad: #freepbx |
| 19:11.51 | nny | are there modules i can check on next time the system hits this state? It happens 3-4 times day, we are building a profile of what types of calls/sip peers etc are in use during the times when it breaks. |
| 19:12.53 | WIMPy | nny: Have you tried 'core show locks'? That might hold a clue. |
| 19:13.12 | WIMPy | If you have thread debugging enabled, that is. |
| 19:13.40 | nny | WIMPy: can't use the debug version, it kills the system load, and it's a pretty hefty server |
| 19:14.17 | nny | WIMPy: we're trying to compartmentalize the system over to another server, but in the meantime looking at possible causes. i assume that command only works with debugging enabled |
| 19:15.05 | WIMPy | There are different debugging options. I don;t think you need the don't optimize bit for that to work. |
| 19:16.05 | WIMPy | However the output might be less informative then, I guess. |
| 19:16.06 | *** join/#asterisk Aliosity (~aliosity@94-192-160-216.zone6.bethere.co.uk) |
| 19:16.36 | nny | WIMPy: i'll look into it, thanks |
| 19:16.40 | *** join/#asterisk tokozedg (~torne@92.241.87.150) |
| 19:19.04 | tokozedg | hello, trying to configure asterisk realtime with a2billing and got problem connecting to mysql. realtime mysql status show that asterisk has successfully connected, but when i reload config i get mysql_reconnect: MySQL RealTime: Unable to select database: general. Still Connected and realtime doesnot work also. Any advice? |
| 19:20.08 | nny | WIMPy: digium has a 1.8.5.0-debugthreads source folder on here. I am in the make menuconfig. Any advice on which debug aspects can be enabled without killing performance? |
| 19:20.53 | WIMPy | Huh? A different source? |
| 19:21.05 | nny | WIMPy: yeah it uses a different init script |
| 19:21.25 | nny | WIMPy: allows parralell testing of the debug version or switch back to normal |
| 19:21.40 | nny | think it uses other independent folders for asterisk, i can check /var/lib etc |
| 19:22.00 | WIMPy | Interesting. |
| 19:22.04 | nny | has DON'T OPTIMIZE and DEBUG THREADS |
| 19:22.14 | nny | maybe try it with DON'T OPTIMIZE unchecked but debug threads on? |
| 19:22.29 | WIMPy | That would be my idea. |
| 19:22.34 | nny | WIMPy: k thanks, will do |
| 19:25.23 | bamdad | Wimpy , can you help me on running an script when a pst-call occurs ? |
| 19:26.05 | WIMPy | bamdad: It looks like you're using FreePBX. |
| 19:26.23 | bamdad | Wimpy , i'm using elastix |
| 19:26.47 | bamdad | But i think it is just a web ui for the asterisk |
| 19:26.51 | bamdad | am i right ? |
| 19:26.56 | bamdad | im new to asterisk world |
| 19:27.31 | WIMPy | Not "just". That makes things pretty different. |
| 19:28.10 | bamdad | Wimpy , so the configuration file that i'm using is for the freePBX , not pure asterisk |
| 19:28.44 | bamdad | Wimpy, is there any ASTERISk way solution for running a shell script when we have pstn call? |
| 19:28.53 | WIMPy | It's pretty hard to tell what the GUIs configure. |
| 19:29.10 | WIMPy | Yes, but you need to find out, where to put it within the GUIs configs. |
| 19:30.05 | bamdad | if i use AsteriskNOw , instead of Elastix? |
| 19:30.36 | WIMPy | No idea. |
| 19:31.32 | bamdad | another question , is there anyway to open a web page in a browser when a sip client rings? |
| 19:31.38 | WIMPy | You won't get much help here when using a GUI. Use the channels for the GUIs |
| 19:31.55 | WIMPy | That depends on the client. |
| 19:32.23 | bamdad | Do you know any client that do this ? |
| 19:33.14 | WIMPy | I have definitely seen that, but I can;t remember on which client. There's a good chance it was Zoiper. |
| 19:33.35 | bamdad | Thank you Wimpy :) |
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| 19:51.34 | pdtpatrick | Question ... i keep getting Received incoming SIP connection from unknown peer. Any ideas what could be possible cause? |
| 19:52.12 | WIMPy | Someone trying to find an account on your system for free calls to china? |
| 19:53.44 | pdtpatrick | ha no |
| 19:53.55 | pdtpatrick | im using a SIP softphone for my iPhone |
| 19:54.39 | pdtpatrick | it registers fine but when i call in .. it just says unknown peer |
| 19:59.27 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 20:02.57 | nny | WIMPy: hmm FYI digium thinks it's DEBUG THREADS that causes the load, not DON'T OPTMIZE. I'll let you know which one works for knowledge's sake |
| 20:03.57 | *** join/#asterisk mhaddog_ (~mhaddog@adsl-184-32-47-9.mia.bellsouth.net) |
| 20:04.48 | Qwell | DONT_OPTIMIZE adds very little overhead. DEBUG_THREADS is far more. |
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| 20:13.38 | nny | Qwell: yeah thanks, that's what we found out.. |
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| 20:26.22 | WIMPy | bad luck |
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| 21:14.51 | *** join/#asterisk jonnyboy (jonnyboy@66-188-73-231.dhcp.athn.ga.charter.com) |
| 21:18.24 | Hive | Is there any channel variables that will let me know that two calls were involved in a transfer? |
| 21:18.31 | Hive | Kind of vague question :x |
| 21:18.38 | Hive | Are* |
| 21:18.59 | WIMPy | 2 calls, 1 transfer? |
| 21:19.05 | singler | Hive: use DumpChan to see what variables are available |
| 21:19.31 | Hive | WIMPy, I probably didnt say that right but i'm sure you can figure out what i meant ;-) |
| 21:19.39 | Hive | singler, ok will try that now |
| 21:20.00 | hardwire | data dip |
| 21:20.23 | WIMPy | nope |
| 21:20.43 | Hive | like, A calls B, B is doing attended transfer to C so they call C, then transfer A to C |
| 21:23.11 | WIMPy | That was a transfer, but I still have no clue, what you're trying to find out. |
| 21:29.26 | Hive | singler, thanks that was quite helpful |
| 21:29.43 | singler | np |
| 21:29.47 | Hive | WIMPy, i think i need to spend some more time with a pad of paper :\ |
| 21:30.16 | Hive | hard to discribe something when it's still cloudy in my own head hehe |
| 21:33.37 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 21:33.37 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 22:08.38 | *** join/#asterisk vfabi (~fabi@host-static-188-237-242-227.moldtelecom.md) |
| 22:08.57 | *** join/#asterisk BenC[UK] (~bcummins@cpc2-lock3-2-0-cust77.6-1.cable.virginmedia.com) |
| 22:09.19 | BenC[UK] | Hi Guys, I am having weird issues (probably something I've done) when trying use Dial() to call |
| 22:09.34 | BenC[UK] | I am using php to connect to asterisk manager and originate a call |
| 22:10.02 | BenC[UK] | it gets through to my context, and starts going through, but its not working anything like I expected |
| 22:10.59 | BenC[UK] | http://pastebin.com/sd6BdDm0 |
| 22:11.07 | BenC[UK] | is my php code for sending the originate, and my context |
| 22:11.37 | BenC[UK] | I have done clean asterisk install, and other than adding a couple of lines in sip.conf and extensions to add my trunk, everything else is standard |
| 23:40.04 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 23:40.04 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.0.0-beta1 (2011/07/22), 1.8.5.0 (2011/07/11), dahdi-linux 2.4.1.2 (2011/04/11), dahdi-tools 2.4.1 (2011/03/03), libpri 1.4.12 (2011/07/06) -=- Visit the new official Asterisk wiki: wiki.asterisk.org -=- http://xkcd.com/806/ |
| 23:40.43 | BenC[UK] | anone around? |
| 23:41.12 | WIMPy | You know that thing about asking, don't you? |
| 23:50.01 | *** join/#asterisk Cesare (~Adium@creati59.lnk.telstra.net) |
| 23:50.21 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 23:50.38 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 23:50.38 | Dougy | hahah WIMPy |
| 23:52.35 | BenC[UK] | yeah, sorry - I asked questions nearly tow hours ago... and no one has talked since |
| 23:53.48 | WIMPy | Ok, any console output? |
| 23:55.48 | BenC[UK] | http://pastebin.com/KKypgAae |
| 23:56.48 | BenC[UK] | Im seeing other things I am finding odd too |
| 23:56.54 | BenC[UK] | If I make it ring me.. |
| 23:56.59 | BenC[UK] | and I hang up, it keeps calling me |
| 23:57.11 | *** part/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 23:57.40 | WIMPy | I'm missing a destination in your originate. |
| 23:57.41 | nny | i am trying to run asterisk with DON'T OPTMIZE, any way to confirm it's in that state? |
| 23:59.18 | *** join/#asterisk reeveskevin (~chatzilla@ip68-4-190-9.oc.oc.cox.net) |