| 00:01.31 | *** join/#asterisk op3r (n=Op3r@222.127.34.249) |
| 00:10.08 | asdx | guys, i have a asterisk server with teliax as a gateway, i see teliax has some "mailboxes" option, is this for voicemail? |
| 00:10.28 | asdx | i'm trying to configure voicemail |
| 00:12.43 | asdx | any ideas? |
| 00:19.43 | unpaidbill | what is the best way to originate several separate calls simultaneously to different phones, each of which need to be able to answer the call |
| 00:19.55 | unpaidbill | im trying to make nagios notify all of our people in certain events |
| 00:20.20 | unpaidbill | but when i originate multiple calls one after another, it queues each call until the one originated before it answers |
| 00:20.36 | unpaidbill | oooh nagios |
| 00:20.41 | unpaidbill | why do you look so good |
| 00:23.27 | unpaidbill | then i thought, ok instead of originating the calls directly i should use channel: Local/cell#@dialout which answers then uses a macro to dial |
| 00:23.46 | unpaidbill | but the second it answers, ti playes the nagios message, instead of waiting for the actual call to answer |
| 00:23.55 | kaii | unpaidbill: Dial(SIP/1&SIP/2&SIP/3&....) |
| 00:24.07 | unpaidbill | that will hang up all the lines when one answers though, wont it? |
| 00:24.17 | kaii | yes |
| 00:24.29 | unpaidbill | i need them all to be able to hear the message |
| 00:24.31 | kaii | ah ok |
| 00:24.33 | unpaidbill | :/ |
| 00:24.40 | unpaidbill | i need multiple separate calls originated |
| 00:24.45 | unpaidbill | damn |
| 00:25.05 | kaii | unpaidbill: i'd say: place a callfile for each staff in the asterisk spool folder and send the callee the right context/extension |
| 00:25.48 | kaii | what you need is a dialmachine |
| 00:25.55 | unpaidbill | i was hoping for something more elegant..asterisk and nagios are running on different servers |
| 00:25.56 | kaii | http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out |
| 00:26.43 | kaii | elegant is: give the nagios a user account with ssh-authorized-key on your asterisk machine and let it call a script via sudo, that places the callfiles |
| 00:26.55 | *** join/#asterisk s0lid (n=s0lid@124.106.200.252) |
| 00:27.06 | unpaidbill | meh |
| 00:27.08 | kaii | or whats the alternative: let the nagios use a sipclient to place the call itself? |
| 00:27.19 | unpaidbill | i'd rather not have to call more than one perl script that connects to 5038 on the asterisk server |
| 00:27.21 | unpaidbill | if possible |
| 00:27.38 | unpaidbill | that's the easiest and least expensive way i can think of.. using sip clients, sshing files |
| 00:27.43 | unpaidbill | i dont like it :/ |
| 00:27.52 | unpaidbill | im just going to have to figure this crap out |
| 00:28.31 | unpaidbill | although for a temporary fix ssh + preshared key will have to suffice :) |
| 00:28.34 | unpaidbill | thanks |
| 00:28.37 | kaii | you can also place calls using the manager interface... |
| 00:28.45 | unpaidbill | yeah that's what i am using now |
| 00:28.55 | kaii | and what doesnt work ? |
| 00:29.09 | unpaidbill | manager interface, but when i originate 5 calls, the first call is created, but the rest of them sit in queue till call #1 gets answered |
| 00:29.28 | unpaidbill | when call #1 is answered, call #2 is placed, and the rest are held in queue till call #2 is answered.. and so on |
| 00:29.40 | kaii | mh thats strange |
| 00:29.49 | kaii | how do you place your calls? |
| 00:30.04 | unpaidbill | the test setup i am using is.. two asterisk machines, the system making the calls to the phones has in sip.conf: |
| 00:30.12 | kaii | and what about: connect,place call#1,reconnect,place cal#2,reconnect,... |
| 00:30.28 | unpaidbill | im pretty sure it would do the same thing |
| 00:30.34 | unpaidbill | i dont know though |
| 00:30.52 | kaii | how do you actually place your calls in manager inface? |
| 00:31.02 | kaii | ACTION: Originate ? |
| 00:31.04 | unpaidbill | http://pastebin.com/m79ecaa20 |
| 00:31.12 | unpaidbill | that is my dialing server SIP configuration |
| 00:31.31 | kaii | i only want to see the manager call ^^ |
| 00:31.37 | *** join/#asterisk egypcio (n=korn@unaffiliated/egypcio) |
| 00:32.00 | kaii | the string you send to the manager interface |
| 00:32.10 | unpaidbill | http://pastebin.com/m670bdfa |
| 00:32.13 | kaii | and the context/extension it is pointing to |
| 00:32.16 | unpaidbill | that's the string through POE asterisk module |
| 00:32.46 | kaii | mh ok |
| 00:32.49 | unpaidbill | basically i pull the #s of people who should be notified from mysql, then i go through a while loop, originating the calls |
| 00:32.54 | kaii | i'd say try the reconnect thing |
| 00:33.15 | unpaidbill | there has to be another way... |
| 00:33.24 | unpaidbill | some way to make originate ignore the fact that call #1 hasnt been answered |
| 00:33.33 | kaii | whats the problem with reconnecting to the manager interface |
| 00:33.48 | unpaidbill | it's clunky |
| 00:34.03 | kaii | i think the manager interface is bound to success/notsuccess of the action placed |
| 00:34.03 | unpaidbill | doesnt make sense |
| 00:34.10 | unpaidbill | hrmm |
| 00:34.11 | kaii | so you have to wait until action has finished |
| 00:34.21 | unpaidbill | maybe i need to use threads then |
| 00:34.25 | kaii | that makes a lot of sense and now i leave you alone wishing a good night |
| 00:34.39 | unpaidbill | it makes no sense :) later |
| 00:34.40 | kaii | you need multiple connections yes, or maybe just a reconnect works |
| 00:35.09 | kaii | if you do not depend on the results of your call for further processing |
| 00:35.12 | unpaidbill | the thing is, if i originate thosoe 5 calls and i have Answer in my local dialplan |
| 00:35.16 | kaii | then you will really need threading |
| 00:35.17 | unpaidbill | it immediately sends off another call |
| 00:35.23 | kaii | or forking |
| 00:35.54 | kaii | yo pal. answer is the success message for originate .. got the point ? |
| 00:36.09 | kaii | somebody answered == call sucessfully originated. |
| 00:36.15 | *** join/#asterisk Kumbang (n=kumbang@167.205.24.67) |
| 00:36.15 | kaii | makes sense? |
| 00:36.21 | kaii | gn8. |
| 00:36.23 | unpaidbill | not at all |
| 00:36.31 | l0verb0y | whats the issue? |
| 00:36.35 | unpaidbill | i should be able to ignore its success or failure |
| 00:36.42 | unpaidbill | and not have queued calls, im sure there is a way to do it |
| 00:36.44 | unpaidbill | and i will find it. |
| 00:37.29 | l0verb0y | hmmm... what language are you using for agi? |
| 00:37.40 | kaii | just give the reconnect a try |
| 00:37.42 | kaii | AGI ? |
| 00:38.06 | unpaidbill | even if reconnecting works, i wont do it that way |
| 00:38.08 | kaii | l0verb0y: hes using the manager interface |
| 00:38.20 | l0verb0y | i had a situation where i had to pull numbers from a database, orginate calls, and report on the status |
| 00:38.22 | kaii | unpaidbill: you will implement threads.. fine. |
| 00:38.27 | outtolunc | learn to use async: true |
| 00:38.39 | unpaidbill | see fuck i knew there was an easy fix |
| 00:38.42 | outtolunc | and you will have to reconnect from time to time (120 or so) |
| 00:38.42 | unpaidbill | thank you outtolunc |
| 00:38.51 | l0verb0y | i used ruby-asterisk, created a thread, and sent the status at the end of the thread |
| 00:38.56 | kaii | this sounds like a solution ^^ |
| 00:39.17 | unpaidbill | it's right there at the top of the damn manager api page |
| 00:39.23 | outtolunc | nods |
| 00:39.28 | unpaidbill | somehow i missed it. :/ |
| 00:39.37 | unpaidbill | (it's the beer) |
| 00:39.40 | seanbright | some of the suggestions for ways to resolve it are a little scary, too. |
| 00:39.53 | seanbright | creating threads, multiple connections, blah blah |
| 00:40.11 | kaii | :D |
| 00:40.41 | kaii | seanbright: well, i didnt knew about the async option in the manager api |
| 00:40.49 | seanbright | Dial(SIP/1234&SIP/5678&SIP/ABCD) |
| 00:40.51 | seanbright | DONE! |
| 00:40.52 | seanbright | heh |
| 00:41.02 | unpaidbill | no |
| 00:41.06 | kaii | seanbright: thats what i first said, too. |
| 00:41.08 | seanbright | ... |
| 00:41.08 | kaii | :) |
| 00:41.10 | unpaidbill | you didnt read the problem |
| 00:41.13 | unpaidbill | shame on you! |
| 00:41.16 | seanbright | unpaidbill: yes, i did |
| 00:41.19 | unpaidbill | outtolunc is the man! |
| 00:41.23 | kaii | heh |
| 00:41.26 | seanbright | hands unpaidbill a sense of humor, free of charge. |
| 00:41.29 | unpaidbill | seanbright that will connect the call to ONE of those sip users |
| 00:41.42 | unpaidbill | i need all the phones to ring, and all to be able to answer |
| 00:41.43 | seanbright | zzzzzzzzzz.... |
| 00:41.45 | seanbright | hmm? |
| 00:41.53 | unpaidbill | anyway, i fixed it and just tested |
| 00:41.54 | unpaidbill | and it works |
| 00:41.55 | seanbright | me re-types |
| 00:41.56 | unpaidbill | so yay |
| 00:41.57 | seanbright | hands unpaidbill a sense of humor, free of charge. |
| 00:42.09 | unpaidbill | i have a sense of humor, it just wasnt funny |
| 00:42.13 | unpaidbill | sorry! |
| 00:42.14 | seanbright | ohhh |
| 00:42.18 | unpaidbill | haha |
| 00:42.23 | unpaidbill | ok im kidding dont beat me up |
| 00:42.27 | unpaidbill | i am a robot |
| 00:42.32 | unpaidbill | i have no sense of humor |
| 00:42.44 | kaii | go robot go! |
| 00:44.05 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-37690db043f0c4ca) |
| 00:44.20 | unpaidbill | aww man async is so sweet. |
| 00:44.57 | unpaidbill | i am going to take down a server tonight at 3am just so i can hear everyone say 'wtf i got woke up at 3am' tomorrow |
| 00:45.15 | seanbright | the sweetest part is that it's fully documented i think |
| 00:45.22 | unpaidbill | it is |
| 00:45.27 | unpaidbill | right at the top of the page |
| 00:45.30 | seanbright | yeah |
| 00:45.34 | seanbright | that's so sweet |
| 00:45.36 | unpaidbill | that's what i get for drinking beer and playing. |
| 00:45.52 | asdx | it is possible to have mailbox=100 in a user in sip.conf and use VoiceMailMain() ? |
| 00:46.02 | asdx | so it ask them for their user/pass |
| 00:46.16 | seanbright | asdx: give it a shot and report back your results |
| 00:46.28 | seanbright | we'll all wait here |
| 00:46.31 | asdx | seanbright: i tried it... no luck |
| 00:46.39 | seanbright | so then it must not be possible |
| 00:46.47 | asdx | shit |
| 00:46.51 | seanbright | heh |
| 00:47.31 | seanbright | asdx: mailbox=100 is just for voicemail polling |
| 00:47.39 | seanbright | asdx: how are you calling VoiceMailMain |
| 00:47.50 | asdx | 700,1,VoiceMailMain() |
| 00:47.57 | asdx | in my [internal] context |
| 00:48.10 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-671d96e59c89da2e) |
| 00:48.28 | seanbright | asdx: and what happens when you dial 700? |
| 00:48.56 | asdx | it says "mailbox... password" |
| 00:49.02 | frieze | with polycom <MAC ADDRESS>.cfg files does it matter if the files are named with upper or lower case? |
| 00:49.14 | *** join/#asterisk Strom_C (n=strom@208.127.172.112) |
| 00:49.17 | seanbright | frieze: make them lowercase |
| 00:49.26 | seanbright | frieze: the polycom looks for them as lowercase |
| 00:49.38 | asdx | seanbright: but when i press 100/1234 it says "login incorrect" |
| 00:50.02 | seanbright | asdx: and you've configured mailbox 100 in voicemail.conf? |
| 00:50.08 | asdx | yes |
| 00:50.23 | seanbright | pb it |
| 00:50.28 | seanbright | your voicemail.conf |
| 00:50.32 | Strom_C | asdx: is VoicemailMain() looking in the correct voicemail context? |
| 00:50.42 | seanbright | asdx: Strom_C will help |
| 00:50.49 | asdx | thx |
| 00:50.57 | asdx | let me pastebin what i have |
| 00:53.38 | asdx | Strom: i have something like this... http://pastie.org/220105 |
| 00:53.49 | seanbright | "something like" |
| 00:54.18 | asdx | well i have that but is not the whole config |
| 00:54.32 | Strom | pastebin the CLI output of the failed voicemail attempt at verbosity 10 |
| 00:54.39 | asdx | ok |
| 00:56.00 | asdx | http://pastebin.com/mba4e8e1 |
| 00:56.40 | seanbright | hmmmm |
| 00:56.48 | seanbright | 'voicemail reload'? |
| 00:56.57 | Strom | asdx: tried "reload app_voicemail.so"? |
| 00:57.07 | asdx | let me try |
| 00:58.03 | asdx | "you have no messages" |
| 00:58.03 | asdx | :D |
| 00:58.04 | asdx | thx |
| 00:58.19 | asdx | voicemail needs a reload option |
| 00:58.27 | seanbright | it has one in trunk |
| 00:58.34 | asdx | cool |
| 00:59.03 | asdx | 1.4.21 here |
| 00:59.07 | asdx | can't wait to switch to 1.6 |
| 01:22.22 | frieze | stupid newbie question probably answered in the docs somewhere: any reason I should use both an extensions.conf and an extensions.ael file? or does make samples create both just for completeness? |
| 01:23.38 | _ShrikE | frieze: you only need to use one or the other. |
| 01:23.50 | frieze | thanks |
| 01:24.10 | _ShrikE | just two different approaches at writing dialplans. |
| 01:37.49 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 01:41.13 | *** join/#asterisk Mavvie (n=edwin@ppp121-44-116-180.lns10.syd6.internode.on.net) |
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| 01:46.11 | jeev | has begun the long and grueling task of redoing his deleted asterisk plans |
| 01:47.11 | jaytee | you deleted your dialplans? |
| 01:47.40 | l0verb0y | i hope you backed them up! |
| 01:48.02 | jaytee | I have all my config files tarballed and emailed in a nightly cron job. |
| 01:49.24 | codefreeze-lap | jaytee: wouldn't hurt to put all your config files under cvs or svn, just like source code; makes it easy to see the diffs, updates, etc. |
| 01:49.50 | codefreeze-lap | you could even use git (no db, etc) |
| 01:49.59 | jeev | i deleted * |
| 01:50.01 | jeev | i meant config. |
| 01:50.19 | jeev | jaytee, why dont you include me in the nightly email... |
| 01:50.44 | jeev | i dunno what i had done to the server, i brought it up from the datacenter (i havne't really gone into production yet), i accidently newfs'd it, couldn't recover.. even though i'm capable of recovering ufs2. |
| 01:52.04 | jaytee | you running OpenBSD or freeBSD? |
| 01:53.26 | *** join/#asterisk amitry (n=amitry@pool-96-241-63-179.washdc.fios.verizon.net) |
| 01:57.07 | jeev | free |
| 01:59.15 | amitry | hello, I am interested in deploying asterisk using a redfone TDMoE box, anyone have recommendations on a distro with a gui? |
| 02:00.34 | jaytee | CentOS 5 with the GUI |
| 02:00.43 | jaytee | or * with a GUI? |
| 02:01.13 | jeev | jaytee, my config wasnt that great anyway |
| 02:01.29 | jeev | i just need to set up like 20-25 lines |
| 02:01.33 | jeev | a few DID's |
| 02:01.40 | jeev | i dont even know how to do DID's anyway |
| 02:01.56 | jeev | my biggest issue is finding decent hardware |
| 02:02.14 | amitry | jaytee: * with a gui |
| 02:02.20 | amitry | no gui needed for centos |
| 02:03.15 | jeev | why dont you just get asteriskNOW |
| 02:03.52 | amitry | to get the redfone installed with the package manager in rpath (conary) looks like a royal pain |
| 02:05.08 | amitry | installing the redfone with yum is pretty straightforward |
| 02:09.27 | jaytee | there's a couple of howtos out there for just installing the * gui on CentOS but I haven't tried one yet. |
| 02:10.06 | amitry | do you prefer that to freepbx? |
| 02:10.15 | amitry | i got the redfone going in elastix |
| 02:14.36 | *** join/#asterisk WilliamK (n=noc@static-71-170-144-28.dllstx.fios.verizon.net) |
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| 02:17.21 | *** join/#asterisk Mavvie (n=edwin@119.11.47.148) |
| 02:18.45 | axisys | i have asterisk running on my laptop on port 5061 and ekiga softphone is up.. this is my sip.conf file looks like this http://pastebin.com/f5ff20ee3.. can any one help how do the test with ekiga? I tried sip:192.168.1.136 (ip of my laptop where asterisk is running) and nothing happends |
| 02:27.31 | *** join/#asterisk simprix (n=simprix@c-24-11-30-113.hsd1.mi.comcast.net) |
| 02:27.58 | Strom | axisys: you dial a number |
| 02:28.07 | Strom | i.e. an extension defined in extensions.conf |
| 02:29.03 | *** join/#asterisk spokra (n=spokra@74-61-42-127.sea.clearwire-dns.net) |
| 02:30.40 | *** part/#asterisk spokra (n=spokra@74-61-42-127.sea.clearwire-dns.net) |
| 02:31.28 | jeev | how could i check that i'm cnonected to my wholesaler? i forgot :/ |
| 02:31.30 | jeev | sip .. |
| 02:33.41 | Strom | "sip show registry" will show whether you've successfully registered |
| 02:33.50 | Strom | but you only "connect" when you place a call |
| 02:34.37 | jeev | i understand |
| 02:34.44 | jeev | because when i callt he number, i just hear the busy signal |
| 02:34.53 | Strom | busy signal or reorder tone? |
| 02:34.55 | jeev | i know that my wholesaler switched me over to the test server.. with ip only |
| 02:34.57 | jeev | busy signal |
| 02:35.01 | Strom | reorder is twice as fast as busy |
| 02:35.03 | jeev | so maybe i gotta have him change the ip |
| 02:35.05 | jeev | ahh let me see |
| 02:35.14 | Strom | busy signal only happens when the actual dialed station is busy |
| 02:35.15 | jeev | i definitely dont see anything come in via debug |
| 02:35.32 | jeev | yea i dunn what it is, i haen't heard a busy signal in a while then :D |
| 02:35.38 | jeev | either way, there is nothing coming into asterisk |
| 02:37.08 | jeev | damn, Visual Voiceplan Pro is confusing as hell heh |
| 02:37.28 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 02:37.36 | *** join/#asterisk simprix (n=simprix@c-24-11-30-113.hsd1.mi.comcast.net) |
| 02:38.46 | jeev | i dont have an incoming line for my dialplan anyway |
| 02:49.40 | axisys | Strom: thnx |
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| 03:13.04 | nemski | anyone here setup asterisk on solaris SPARC? |
| 03:13.28 | jeev | i can't believe i lost all my dialplans, jesus christ |
| 03:13.31 | jeev | i'm not gonna be able to set this up again lol |
| 03:15.12 | nemski | shoulda backed it up |
| 03:15.53 | [TK]D-Fender | Jesus saves |
| 03:17.00 | jeev | it's ok, Fendur promised me to email his dialplan |
| 03:19.09 | [TK]D-Fender | jeev: [default] exten => _!,1,Hangup ; GTFO! |
| 03:19.12 | [TK]D-Fender | jeev: there :p |
| 03:19.46 | jeev | ah |
| 03:19.47 | jeev | IT WORKS! |
| 03:20.38 | jeev | no but seriously Fender, what's the best way without this headache of a dialing plan to verify that my SIP provider is still working |
| 03:21.09 | DigitalIrony | jeev: heh try and connect to them with a softphone like xlite |
| 03:21.33 | [TK]D-Fender | jeev: Without a dialing you have nothing. |
| 03:21.38 | jeev | DigitalIrony, the softphone connects fine to asterisk |
| 03:21.41 | jeev | yea, blah |
| 03:21.44 | jeev | someone shoot me |
| 03:22.03 | [TK]D-Fender | looks around for his Barrett Light-50 |
| 03:22.03 | jeev | [Jun 22 20:18:07] NOTICE[46097]: chan_sip.c:14035 handle_request_invite: Call from '300' to extension 'xxxxxxxxx' rejected because extension not found. |
| 03:22.10 | jeev | has his Glock 26 right here.. |
| 03:22.13 | jeev | somewhere at least |
| 03:22.14 | jeev | lool |
| 03:22.16 | [TK]D-Fender | jeev: then go MAKE IT |
| 03:22.20 | DigitalIrony | jeev: no connect to your sip provider. with xlite to test. |
| 03:22.31 | jeev | ah |
| 03:22.35 | *** join/#asterisk coreyf52 (i=coreyf52@63.227.121.164) |
| 03:22.42 | jeev | i dont think it's set up like that |
| 03:22.47 | jeev | or it should be, hold up let me try |
| 03:23.34 | jeev | forbeeden! |
| 03:23.41 | jeev | he moved me and didnt leave the old one active |
| 03:23.42 | jeev | bah humbug |
| 03:26.36 | *** part/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 03:30.19 | *** join/#asterisk phy (n=phy@funky.monkey.org) |
| 03:31.22 | phy | quick question: is there a way redirect log output to the manager? e.g. so clients connected through the AMI interface can see Warnings |
| 03:31.56 | phy | i don't see a way through the logger.conf sample |
| 03:33.54 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 03:36.48 | coreyf52 | does anyone know where i can find a list of external programs used by asterisk? for example Monitor uses soxmix to combine in/out channels after the call.. |
| 03:39.18 | *** join/#asterisk ZX81 (n=matt@202.49.106.158) |
| 03:39.48 | asdx | cool, i just added voicemails |
| 03:39.50 | asdx | xD |
| 03:40.06 | asdx | for my 40 user |
| 03:40.08 | asdx | users |
| 03:41.57 | asdx | they just need to replace their xlite with voip phones now xD |
| 03:44.23 | *** join/#asterisk tengulre (n=tengulre@125.71.208.16) |
| 03:44.25 | tengulre | hi,all |
| 03:44.41 | asdx | hi |
| 03:45.57 | tengulre | how to using asterisk building video conference! |
| 03:46.00 | *** join/#asterisk mvanbaak (n=michiel@vanbaak.xs4all.nl) |
| 03:47.13 | gitguy | lol |
| 03:47.27 | gitguy | ~book |
| 03:47.28 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 03:47.50 | *** join/#asterisk gitguy (n=diego@adsl-142-230.click.com.py) |
| 03:47.58 | tengulre | gitguy: I m point video conference ,not voice conference! |
| 03:48.13 | gitguy | ok |
| 03:48.27 | gitguy | is clueless about video :/ |
| 03:51.50 | *** join/#asterisk gitguy (n=diego@adsl-142-230.click.com.py) |
| 03:53.24 | jeev | gitguy |
| 03:53.38 | jeev | so youre good at voice, eh |
| 03:54.38 | gitguy | i know a bit of asterisk now |
| 03:54.39 | gitguy | lol |
| 03:54.46 | gitguy | [TK]D-Fender helped me |
| 03:55.26 | gitguy | the book is cool too |
| 03:55.28 | jeev | so you wanna help me |
| 03:55.28 | jeev | ;D |
| 03:55.37 | gitguy | sure |
| 03:59.25 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 04:00.13 | [TK]D-Fender | s/hep/doitforhimbecauseheslazyandcluelessaverydangerouscombinationindeed |
| 04:00.16 | [TK]D-Fender | hlp* |
| 04:00.31 | gitguy | lol |
| 04:02.25 | jeev | fender |
| 04:02.30 | jeev | you forgot to say |
| 04:02.33 | jeev | include bastard |
| 04:05.44 | [TK]D-Fender | jeev: Sorry, don't know your mom well enough to know your pedigree ;) |
| 04:06.15 | [TK]D-Fender | jeev: Though you go hop around and whine like a Jack Russell ^%@#$ing terrier :P |
| 04:17.43 | gitguy | i discovered most things in my own too xD |
| 04:17.54 | gitguy | it was a bit of headache but well worth it |
| 04:18.31 | gitguy | now if my dtmf could work well in zoiper would be cool |
| 04:18.52 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.136) |
| 04:18.58 | gitguy | rfc2833compensate=yes helped a bit |
| 04:19.25 | gitguy | exten => 123,1,Read(NUMBER,,9) |
| 04:19.26 | gitguy | exten => 123,n,NoOp(${NUMBER}) |
| 04:19.31 | gitguy | but that still shows some missing digits |
| 04:20.02 | gitguy | damn i need something to drink... brb |
| 04:22.50 | jeev | [09:05pm] <[TK]D-Fender> jeev: Sorry, don't know your mom well enough to know your pedigree ;) |
| 04:22.52 | jeev | not cool! |
| 04:24.02 | [TK]D-Fender | jeev: You said I forgot to call you a bastard. As I have no idea of her marital status at the time I can't be accused of "forgetting" to call you something I don't know that you are :) |
| 04:24.32 | jeev | ok |
| 04:24.35 | jeev | my dad is lazy |
| 04:24.45 | jeev | if my mom didn't have a succesful business.. |
| 04:24.46 | jeev | and I |
| 04:24.48 | jeev | he'd be homeless |
| 04:25.01 | jeev | the guy listens to music more than people who get paid to listen to music |
| 04:27.16 | gitguy | heh |
| 04:31.15 | jeev | Fender |
| 04:33.14 | *** join/#asterisk denon (n=denon@tooth.decay.org) |
| 04:33.14 | *** mode/#asterisk [+o denon] by ChanServ |
| 04:43.02 | *** join/#asterisk Nuitari (n=Nuitari@cybernet.nuitari.net) |
| 04:43.50 | Nuitari | Hi |
| 04:44.33 | Nuitari | Is there anything that changed between 1.4.19.1 and 1.4.21 that would cause realtime to behave differently when peers/users where "lost" by doing a "sip reload" |
| 04:45.03 | Nuitari | it used to be able to find peers who previously registered automatically, but now it doesn't do that anymore and until the peer sends another register it is considered offline |
| 04:45.13 | jeev | if i have a single DID, how do i make that answer? |
| 04:46.22 | gitguy | jeev: with a extension... like, exten => thedidnumber,1,Answer() |
| 04:46.29 | jeev | i've tried :) |
| 04:46.29 | gitguy | in the correct context |
| 04:46.33 | gitguy | what happened |
| 04:47.12 | jeev | nothing |
| 04:47.15 | jeev | still rejecting the extension |
| 04:47.20 | jeev | it's calling the 10 digit number an extension |
| 04:47.41 | gitguy | hm |
| 04:48.21 | gitguy | is your extension in [default] ? |
| 04:48.22 | gitguy | or? |
| 04:49.06 | jeev | yea |
| 04:51.33 | Nuitari | did you do dialplan reload? |
| 04:52.42 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 04:53.45 | jeev | yes |
| 04:54.10 | [TK]D-Fender | jeev: pastebin is your friend... |
| 04:54.29 | jeev | http://pastie.org/private/caajoow1tgnungt1pdhnsa |
| 04:54.51 | jeev | pastie for life |
| 04:54.56 | gitguy | lol |
| 04:55.16 | Nuitari | jeev: who is your DID provider? |
| 04:55.22 | jeev | [Jun 22 21:52:08] NOTICE[46495]: chan_sip.c:14035 handle_request_invite: Call from 'myusernameattheprovider' to extension '3101234567' rejected because extension not found. |
| 04:55.26 | *** part/#asterisk bancus (i=treed@static-71-160-206-211.lsanca.dsl-w.verizon.net) |
| 04:55.29 | jeev | lugosoft |
| 04:55.53 | Nuitari | what is the context that you set for the sip peer? |
| 04:55.59 | [TK]D-Fender | jeev: and now SIP debug infor for the incoming call attempt... |
| 04:56.17 | jeev | incoming_calls |
| 04:56.20 | jeev | should it be default ? |
| 04:56.29 | [TK]D-Fender | jeev: ... CALL PLEASE... |
| 04:56.42 | gitguy | lol |
| 04:57.05 | jeev | huh |
| 04:57.06 | jblack | skips the obligatory joke |
| 04:57.08 | gitguy | what is lugosoft, lugo is the name of my new president |
| 04:57.11 | jeev | wnts to shoot himself |
| 04:57.26 | jeev | Fender, i enabled debugging.. |
| 04:57.44 | jeev | Looking for 3101234567 in incoming_calls (domain 75.x) |
| 04:57.48 | jeev | so it has to be default |
| 04:57.57 | [TK]D-Fender | jeev: "thats nice". Now wheres the pastebin of the full call attempt's SIP DEBUG? |
| 04:58.08 | jeev | woo hoo |
| 04:58.11 | jeev | hahah |
| 04:58.12 | jeev | tt monkeys |
| 04:58.20 | jeev | thanks, i'd like to thank the academy |
| 04:58.30 | gitguy | jeev: does it works now? |
| 04:58.31 | jeev | i couldn't have done it without gitguy's help.. all the help fender has given me throughout the months |
| 04:58.34 | [TK]D-Fender | jeev: Well you've just shown what contexdt its hitting, so that exten you made is not going to get called. |
| 04:58.42 | jeev | it worked fendereno :D |
| 04:59.15 | jblack | jeev: Yeah. He rocks |
| 05:00.52 | jeev | Fender, would it freak you out to say that my business does DNA reconsturction |
| 05:00.53 | jeev | ? |
| 05:00.56 | jeev | reconstruction |
| 05:01.47 | [TK]D-Fender | jeev: I firmly believe that many should never be allowed to breed... |
| 05:02.56 | jeev | me too. |
| 05:03.15 | *** join/#asterisk xenonex (n=xenonex@89.218.232.28) |
| 05:06.57 | [TK]D-Fender | ok, checkout time, later all. |
| 05:11.28 | *** join/#asterisk quazzmarsh (n=quazzmar@62.8.93.2) |
| 05:17.45 | gitguy | later |
| 05:18.27 | jeev | Nuitari, how would i set the outgoing caller id? my provider says i'm getting a generic number cause i haven't set my number. |
| 05:18.57 | Nuitari | you can set it using the caller id functions in Asterisk inside the dialplan |
| 05:19.01 | Nuitari | it depends on the version that you have |
| 05:19.35 | jeev | yea, i used to do it, then i lost my config.. |
| 05:19.41 | *** join/#asterisk echelon (n=echelon@ool-182cc7a4.dyn.optonline.net) |
| 05:19.42 | echelon | hi |
| 05:19.49 | gitguy | hi |
| 05:19.53 | Nuitari | jeev: what's your version? |
| 05:20.04 | echelon | what should i do if i have a softphone and asterisk running on the same machine? |
| 05:20.28 | jeev | 1.4.21 |
| 05:20.37 | echelon | wouldn't there be a port conflict? |
| 05:20.42 | jeev | Nuitari, i pasted it to you |
| 05:21.31 | echelon | since the sip user agent port and asterisk both listen on 5060? |
| 05:22.23 | echelon | whose port should i change? |
| 05:22.34 | gitguy | echelon: i tried that a while ago and worked... in my case i just had to make sip work behind nat |
| 05:22.50 | echelon | gitguy: the sip client? |
| 05:23.08 | gitguy | yes |
| 05:23.11 | gitguy | nat=yes |
| 05:23.12 | gitguy | in both |
| 05:23.15 | gitguy | and worked |
| 05:23.32 | gitguy | because my server was behind a firewall |
| 05:23.36 | gitguy | but worked fine |
| 05:23.45 | echelon | hmm.. also, i'm setting it up on a vpn |
| 05:23.58 | gitguy | that should work |
| 05:25.37 | echelon | also, have you tried the meetme conference bridge? |
| 05:25.48 | gitguy | yes |
| 05:26.02 | gitguy | works fine too |
| 05:26.30 | gitguy | but i had some problems on a xen environment with zaptel/ztdummy... although they are working on soft-based conferencing for 1.6 now |
| 05:26.37 | gitguy | which is very cool |
| 05:26.44 | gitguy | http://www.russellbryant.net/blog/index.php/2008/06/16/asterisk-16-now-with-a-new-timing-api/ |
| 05:27.02 | gitguy | but meetme() works fine for me |
| 05:27.16 | gitguy | on real hardware |
| 05:27.24 | echelon | i'm new to all this |
| 05:27.24 | gitguy | with zaptel/ztdummy |
| 05:27.32 | gitguy | ok |
| 05:27.49 | gitguy | asterisk is cool, it's fun |
| 05:27.53 | echelon | i just need a software-based voip solution so my friends can talk over vpn in conference |
| 05:28.02 | gitguy | use meetme |
| 05:28.17 | echelon | that's part of asterisk right? or is it a plugin? |
| 05:28.25 | gitguy | it's part of it |
| 05:28.30 | echelon | great |
| 05:29.12 | *** join/#asterisk LeddyHM (i=leddy@da-club.with.my.beerandcondoms.com) |
| 05:30.33 | *** join/#asterisk grEvenX (n=even@ap39pb.ip.ssc.net) |
| 05:31.03 | echelon | you don't know how many times i've built asterisk, then tried to configure and failed because i didn't know what all the features were for |
| 05:33.02 | unpaidbill | shit yeah asterisk!!! |
| 05:33.12 | unpaidbill | hey gitguy what kind of xen environment were you using |
| 05:33.23 | unpaidbill | i want to try running 4 asterisk servers on a quad core c2d with xen |
| 05:33.53 | unpaidbill | in the hopes that i can pump out close to 4x the amount of power |
| 05:34.18 | unpaidbill | were you doing this for large installs? |
| 05:37.30 | echelon | how do i configure asterisk/sip so anyone can sign in with any name without being registered? |
| 05:38.27 | echelon | it's only for a few friends over vpn, so additional authentication is an inconvenience |
| 05:38.50 | *** join/#asterisk PepOSX (n=angeldav@200.93.28.168) |
| 05:40.30 | gitguy | unpaidbill: amazon ec2 |
| 05:43.22 | *** join/#asterisk PepOSX (n=angeldav@200.93.28.168) |
| 05:46.15 | echelon | is libpri needed for meetme? |
| 05:48.41 | *** join/#asterisk newbie1 (n=banik@58.145.184.1) |
| 05:48.52 | gitguy | echelon: you have to compile zaptel though |
| 05:48.57 | gitguy | echelon: and then asterisk |
| 05:49.04 | gitguy | echelon: for getting meetme |
| 05:49.16 | echelon | i didn't do that :/ |
| 05:49.16 | gitguy | echelon: and modprobe ztdummy |
| 05:49.20 | gitguy | ok |
| 05:49.26 | gitguy | then you probably have that setup already |
| 05:49.29 | gitguy | if you have meetme |
| 05:49.40 | echelon | how do i know if i have meetme? |
| 05:50.23 | echelon | gitguy: ? |
| 05:51.38 | gitguy | module show like app_meetme.so |
| 05:51.40 | gitguy | do that |
| 05:51.49 | echelon | huh? |
| 05:51.59 | gitguy | in asterisk -r |
| 05:52.03 | gitguy | in the asterisk cli |
| 05:52.21 | gitguy | if you see some output it means you have it |
| 05:52.29 | gitguy | Module Description Use Count |
| 05:52.32 | gitguy | app_meetme.so MeetMe conference bridge 0 |
| 05:52.32 | gitguy | some output like that |
| 05:52.35 | gitguy | 1 modules loaded |
| 05:53.22 | echelon | no :( |
| 05:53.29 | *** part/#asterisk Nuitari (n=Nuitari@cybernet.nuitari.net) |
| 05:53.40 | echelon | but isn't zaptel for hardware support? |
| 05:54.03 | gitguy | yes but zaptel has ztdummy.ko kernel module |
| 05:54.11 | gitguy | which provides timing for meetme |
| 05:54.19 | echelon | but i wouldn't need the actual hardware, right? |
| 05:54.19 | gitguy | you don't need hardware for meetme |
| 05:54.22 | gitguy | but ztdummy.ko |
| 05:54.24 | gitguy | for timing |
| 05:54.31 | echelon | oh ok |
| 05:54.43 | gitguy | it's a dependency for meetme |
| 05:55.06 | gitguy | with asterisk 1.6.1 you wont need zaptel for conferencing |
| 05:55.07 | gitguy | i think |
| 05:55.19 | gitguy | or with further 1.6 versions |
| 05:55.32 | gitguy | app_confbridge... i'm waiting for you |
| 05:55.41 | echelon | :\ |
| 05:55.56 | echelon | i found this earlier.. http://sourceforge.net/projects/appconference/ |
| 05:56.02 | gitguy | yeah |
| 05:56.04 | gitguy | you can use that too |
| 05:56.08 | gitguy | app_conference |
| 05:56.10 | echelon | oh cool |
| 05:56.12 | gitguy | that doesn't require zaptel |
| 05:56.20 | echelon | how do i use it? |
| 05:56.47 | gitguy | compile it, load the module and put Conference() in your dialplan |
| 05:56.57 | echelon | dialplan? |
| 05:57.05 | JT | ~thebook |
| 05:57.07 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 05:57.12 | gitguy | i tried that before but it lacked some features that i needed (notification sounds) etc, and i went back to meetme() |
| 05:57.41 | gitguy | echelon: read the book ;) |
| 06:00.41 | *** join/#asterisk MrNaz (n=naz@ppp59-167-142-211.lns3.mel6.internode.on.net) |
| 06:05.45 | *** join/#asterisk magic_hat (n=geoffdou@h-68-164-228-112.chcgilgm.dynamic.covad.net) |
| 06:07.14 | magic_hat | hi everyone. I have an xlite behind a router/firewall, and I'm trying to connect with an * box that is not behind a firewall. Registration is timing out, and I'm seeing no indication from the * client window that the xlite is actually hitting asterisk. Suggestions on how to proceed? |
| 06:07.31 | tengulre | echelon: do u successful using appconference module in asterisk? |
| 06:10.44 | jblack | magic_hat: I'd look at the nat issues. You'll need to make sure port 5060 works both ways, and the rtp ports too |
| 06:10.55 | echelon | i'm trying it now |
| 06:12.12 | magic_hat | jblack: yeah, i have port forwarding turned on at the router -- 5060-5063 UDP and also 10000-20000 and 69 |
| 06:15.47 | jeev | anyone run zaptel in bsd? |
| 06:16.32 | jql | I've wanted to... never taken the time to try |
| 06:17.14 | gitguy | ~sipnat |
| 06:17.14 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 06:17.20 | gitguy | magic_hat: you might want to read that |
| 06:17.24 | jeev | hmm |
| 06:17.51 | gitguy | magic_hat: add nat=yes in your user |
| 06:18.56 | magic_hat | cool....reading now |
| 06:19.35 | gitguy | magic_hat: adding nat=yes in your user section would be enough i think |
| 06:20.24 | magic_hat | Okay, I'm trying now... |
| 06:23.01 | *** join/#asterisk remibemol (n=remibemo@lev92-1-82-67-255-63.fbx.proxad.net) |
| 06:24.54 | *** join/#asterisk af_ (n=getsmart@88-149-230-105.dynamic.ngi.it) |
| 06:25.46 | jeev | damn |
| 06:25.50 | jeev | meetme is sending me to a voicemail system |
| 06:25.50 | jeev | heh |
| 06:26.26 | magic_hat | if my xlite is behind a NAT and my asterisk is outside of it, do I need to use stun? |
| 06:26.51 | jeev | what the hell is stun |
| 06:26.53 | jeev | i used it fine |
| 06:26.55 | jeev | xlite behind NAT |
| 06:27.14 | jeev | what's your * server info |
| 06:27.14 | jeev | let me try |
| 06:27.46 | magic_hat | jeev... server's not set up for production yet, I don't wanna go giving everyone the coordinates. |
| 06:27.57 | jeev | :) |
| 06:27.58 | jeev | hm |
| 06:28.05 | jeev | i had it working man, easily |
| 06:28.08 | jeev | i dont have the config anymore |
| 06:28.13 | jeev | infact, i'm behind NAT right now |
| 06:28.19 | jeev | and i'm connected just fine to the outside network * |
| 06:28.36 | magic_hat | huh. yeah, it's not even touching the * box, according to the * client log... |
| 06:29.11 | jeev | is your nat functioning properly? |
| 06:29.34 | magic_hat | jeev: not positive, but I imagine so. |
| 06:29.58 | mchou | magic_hat: as long as client expects SIP, you best turn on STUN |
| 06:30.42 | mchou | magic_hat: what NAT are you using? |
| 06:31.06 | mchou | magic_hat: i.e. is it some sort of linux firewall? |
| 06:31.12 | magic_hat | mchou: My crappy 2Wire router. |
| 06:31.27 | *** join/#asterisk creativx (n=creadure@226.62-97-205.bkkb.no) |
| 06:31.32 | mchou | magic_hat: proprietary firmware? |
| 06:31.40 | magic_hat | i have 5060-5063 and 10000-20000 forwarded. Do i need anything else for the stun traffic? |
| 06:31.41 | echelon | cool, i got an echo test to work :D |
| 06:32.16 | magic_hat | mchou: I dunno... but I've had an xlite working on here before. |
| 06:32.19 | mchou | magic_hat: the whole point of STUN is so you dont have to set up ANY port forwarding |
| 06:32.27 | *** join/#asterisk creativx (n=creadure@226.62-97-205.bkkb.no) |
| 06:32.28 | magic_hat | duh. lol. |
| 06:32.31 | magic_hat | okay. |
| 06:34.34 | magic_hat | i had it working w/ this router with port forwarding, not stun, before. |
| 06:34.36 | *** join/#asterisk tengulre11 (n=tengulre@125.71.208.16) |
| 06:34.59 | *** join/#asterisk icenicola (n=pavlovni@83.244.78.241) |
| 06:35.10 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-ecfe6f6321600b29) |
| 06:35.16 | mchou | magic_hat: it's up to you which philosophy to go with. both work |
| 06:35.16 | icenicola | i have two servers |
| 06:35.33 | *** join/#asterisk Hackbanger (n=hackbang@213.209.114.6) |
| 06:35.39 | icenicola | one running asterisk and one runing a voipswitch. i monitor rtp packets using wireshark |
| 06:36.14 | mchou | magic_hat: STUN works better if you have multiple SIP clients behing your NAT (as an example) |
| 06:36.23 | icenicola | between those two servers. there is delay like 16 seconds when using g729 and very low delay using g711 |
| 06:36.29 | icenicola | is that normal? |
| 06:36.34 | magic_hat | shrug. gonna try restarting the router. lol |
| 06:37.15 | mchou | 16 second delay is certainly not normal |
| 06:37.15 | icenicola | the delay is only from the asterisk -> voipswitch |
| 06:37.22 | icenicola | i know |
| 06:37.35 | icenicola | voipswitch -> asterisk is ok |
| 06:37.42 | mchou | so why are you asking? lol |
| 06:37.45 | JT | all audio is delayed by 16 seconds, or there is 16 seconds of silence? |
| 06:38.10 | icenicola | changing codecs in sip.conf the delay between asterisk -> voipswitch was reduced to 500ms |
| 06:38.46 | icenicola | i want to ask if the g729 has a problem |
| 06:39.01 | icenicola | should i re-install it? |
| 06:39.14 | JT | icenicola: all audio is delayed by 16 seconds, or there is 16 seconds of silence? |
| 06:39.35 | jblack | icenicola: Not to ask the obvious, but did you buy the license for g729? |
| 06:39.58 | icenicola | i don't use it for encoding |
| 06:40.06 | icenicola | so i didn't buy a license |
| 06:40.11 | JT | icenicola: can you answer my question? |
| 06:40.32 | icenicola | in wireshark, monitoring only RTP i have a maxumim Delta 16 seconds |
| 06:41.01 | JT | yes but what does the call sound like? |
| 06:41.08 | icenicola | i have other values like 4, 5, 6 seconds |
| 06:41.14 | icenicola | very bad |
| 06:41.23 | icenicola | the sound was very bad |
| 06:41.27 | mchou | lol |
| 06:41.33 | icenicola | i could hear them very well but they can't here me |
| 06:41.34 | JT | very bad, that's quite some dexription |
| 06:41.41 | JT | description |
| 06:41.49 | icenicola | bad quality!! |
| 06:41.57 | JT | be more specific. |
| 06:42.04 | mchou | lol |
| 06:42.17 | mchou | who's on first? |
| 06:42.21 | icenicola | when i call through the system, i could hear everythin on the other side |
| 06:42.27 | icenicola | but they can't hear me at all\ |
| 06:42.52 | jblack | that's nat |
| 06:42.53 | icenicola | they can hear only one word out of 20 |
| 06:43.00 | jblack | or a firewall. |
| 06:43.03 | jblack | They can hear any words? |
| 06:43.23 | icenicola | i tested it it with a disabled firewall, same problem |
| 06:43.28 | JT | where do the calls come from originally? |
| 06:43.49 | icenicola | sip device registered through asterisk |
| 06:45.29 | icenicola | when i changed the preference of the codecs in sip.conf the delay was reduced |
| 06:46.20 | icenicola | using ulaw for example, the max delay i had was 500ms |
| 06:47.17 | icenicola | does g729 codec apply delays when i speak and they can't hear me? |
| 06:47.27 | icenicola | though i can hear everythin |
| 06:55.01 | *** join/#asterisk magic_hat (n=geoffdou@h-68-164-7-249.chcgilgm.dynamic.covad.net) |
| 06:57.08 | magic_hat | okay, still having issues. When I do telnet "my asterisk ip" 5060 from my client machine, I get connection refused. But then I'm seeing stuff in my * logs like SIP read from 63.211.239.2:5060, which means my * box has a connection open on 5060, right? |
| 06:57.09 | Nugget | telnet is eeeeeeevil! |
| 06:57.39 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 06:57.58 | *** join/#asterisk pa (n=pa@unaffiliated/pa) |
| 06:58.43 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
| 07:01.38 | *** join/#asterisk steliosk (n=Stelios@athedsl-277899.home.otenet.gr) |
| 07:05.22 | echelon | tengulre11: i got app_conference to work :) |
| 07:05.25 | echelon | :D |
| 07:08.52 | *** join/#asterisk rcy (n=rcy@66.183.58.28) |
| 07:08.53 | jeev | ~moh |
| 07:08.54 | jbot | it has been said that moh is Music On Hold. Good information about how to set it up in the various possible ways can be found at http://www.voip-info.org/wiki/index.php?page=Asterisk+config+musiconhold.conf, or originally from http://www.freeplaymusic.com |
| 07:09.53 | magic_hat | okay, does earthlink block 5060? I can't seem to get out on it no matter what. |
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| 07:11.13 | jeev | hmm |
| 07:11.20 | jeev | that'd be messed up |
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| 07:12.30 | echelon | magic_hat: get a sip broker? |
| 07:12.40 | echelon | sign up with one, i mean |
| 07:13.19 | echelon | can you use music on hold with conference? |
| 07:13.38 | echelon | when one person is by themself |
| 07:13.47 | echelon | sipphone.com does this |
| 07:14.48 | magic_hat | what is a sip broker? |
| 07:16.22 | echelon | it acts like a proxy |
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| 07:35.13 | mvanbaak | magic_hat: make sure you enabled the manager, and that you have permissions to access it |
| 07:35.32 | mvanbaak | magic_hat: see manager.conf |
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| 07:38.32 | nomad_cz | Hi. Is it possible with Asterisk during one to one call invite another person ? |
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| 07:52.13 | icenicola | when i run "sip show peers" in asterisk CLI i get all the peers but with status unmonitored |
| 07:52.35 | icenicola | is there a certain configuration that i have to setup in order to monitor? |
| 07:53.55 | mort_gib | Use the hint |
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| 07:54.21 | icenicola | what do u mean? |
| 07:56.25 | mort_gib | Are you starting any extension with something like exten => 100,hint,SIP/100 (if your SIP user andm extension is 100) |
| 07:59.11 | icenicola | i don't use hint |
| 07:59.11 | mvanbaak | mort_gib: that's not for the status in sip show peers |
| 07:59.30 | mvanbaak | icenicola: in your sip.conf, add 'qualify=yes' for every peer |
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| 08:00.08 | icenicola | ok, thank u |
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| 08:16.05 | nomad_cz | <PROTECTED> |
| 08:17.55 | jblack | yeah, it's called three way calling. :) |
| 08:18.09 | jblack | hit the flash button on your phone, the new number, say hi to them, then flash again |
| 08:19.55 | nomad_cz | jblack: So it is enabled by default , right ? |
| 08:20.26 | jblack | Usually. |
| 08:20.42 | nomad_cz | jblack: Ok. I'll try ;) |
| 08:20.45 | jblack | Either your ATA, or your phones should be able to do it without needing something special from * |
| 08:28.32 | synthetiq | anyjingle people here? was wondering what would be the best sip <-> jingle gateway to use (freeswitch,ast,ser) in the case of you dont want to have the xmpp user to register on the gateway |
| 08:39.55 | jblack | bleh. theonion.com is gay today. |
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| 08:45.25 | yangvnc | Is it posible to set a time-delayed forwarding of the call to extension, for example all calls after 1630 ring a desired extension? |
| 08:45.46 | yangvnc | and where could i read about it? |
| 08:46.10 | jblack | I'll give you a fragment that does it |
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| 08:46.15 | jblack | it's not hard |
| 08:46.53 | icenicola | when runing sip show peers |
| 08:47.01 | mvanbaak | yangvnc: you have to use 'GotoIfTime()' |
| 08:47.09 | icenicola | i have a status column, saying OK and a number in ms |
| 08:47.11 | yangvnc | mvanbaak: thanks |
| 08:47.16 | icenicola | what is this number? |
| 08:47.26 | jblack | yangvnc: http://pastebin.com/m3004b21b |
| 08:47.51 | icenicola | is it rtt or one way? |
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| 08:48.14 | mvanbaak | icenicola: rtt |
| 08:48.18 | yangvnc | jblack: thank you too !!! |
| 08:49.18 | jblack | feel free to paypal me at jblack@merconline.com . :) |
| 08:53.19 | yangvnc | jblack: a little something for a beer :) |
| 08:53.23 | icenicola | mvanbaak: thank u |
| 08:54.09 | jblack | Heh. Thanks |
| 08:54.27 | jblack | Yay! I own a pound! |
| 08:55.16 | jblack | That's worth something like, 250,000 dollars US. |
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| 09:13.26 | yangvnc | jblack: 1:2 :) |
| 09:14.02 | TheH | is now dCap certified |
| 09:14.10 | yangvnc | dcap ? |
| 09:14.21 | TheH | digium certified asterisk thingy |
| 09:14.26 | yangvnc | cool |
| 09:14.41 | yangvnc | Where did you take the exam? |
| 09:14.47 | TheH | a'dam |
| 09:15.06 | yangvnc | You had to show your skills on asteriskNOW platform? |
| 09:15.18 | TheH | nope plain old asterisk |
| 09:15.22 | TheH | 1.4.19 |
| 09:15.22 | yangvnc | nice |
| 09:15.48 | TheH | practium exam was setting up a server+asterisk+pri+analogue+sip peers+dialplans |
| 09:15.58 | TheH | and then the theory was 120 questions |
| 09:16.11 | mvanbaak | cool. congrats |
| 09:16.16 | TheH | thanks |
| 09:16.17 | mvanbaak | I should get my dCap as well |
| 09:16.25 | TheH | it aint that difficult |
| 09:16.58 | mvanbaak | ok. maybe this summer during my holidays |
| 09:17.23 | TheH | yeah although they are not often in the netherlands, normally its the uk where you can take exams |
| 09:17.31 | TheH | Clustering with asterisk ! » http://www.youtube.com/watch?v=eUochCGC9Ag&feature=related |
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| 09:47.56 | Tebi | how to change TYPE OF NO : .010.... = national number to TYPE OF NO : .100.... = subscriber number ? |
| 09:48.07 | Tebi | PRI |
| 09:48.27 | TheH | vi /etc/zaptel.conf |
| 09:48.55 | Tebi | is´t termtype? |
| 09:50.00 | TheH | sorry its /etc/asterisk/zapata.conf |
| 09:50.03 | TheH | ;prilocaldialplan=national |
| 09:50.03 | TheH | ; |
| 09:50.03 | TheH | ; PRI callerid prefixes based on the given TON/NPI (dialplan) |
| 09:50.03 | TheH | ; This is especially needed for euroisdn E1-PRIs |
| 09:50.03 | TheH | ; |
| 09:50.05 | TheH | ; sample 1 for Germany |
| 09:50.07 | TheH | ;internationalprefix = 00 |
| 09:50.09 | TheH | ;nationalprefix = 0 |
| 09:50.11 | TheH | ;localprefix = 0711 |
| 09:50.13 | TheH | ;privateprefix = 07115678 |
| 09:50.15 | TheH | ;unknownprefix = |
| 09:50.27 | JT | ~pb |
| 09:50.28 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 09:50.47 | Tebi | ;-) thx |
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| 10:53.16 | Dr-Linux|home | what's the difference between cdr-csv and cdr-custom? |
| 10:58.27 | DarKnesS_WolF | mmm why everytime i reload asterisk i lose the parking extension and i have to reload res_features.so manualy ?? |
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| 11:16.59 | yangvnc | Ehem...I was looking for asterisk voicemail files, are they also available in another language than english? |
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| 11:34.06 | datacompboy | Hi all! How to ignore CLIR presentation on incoming calls? |
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| 11:34.31 | datacompboy | I know, that telephone provider send CLI always, even if it should be hidden (CLIR=1) |
| 11:34.46 | datacompboy | but i got "CID withheld" in *. |
| 11:34.55 | datacompboy | zaptel, card TE400 |
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| 11:50.18 | icenicola | where i can set global file where a sip device can retreive info regarding codecs in asterisk? |
| 11:50.39 | icenicola | so, no matter what r the configurations in the device are, they will be ignored |
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| 11:52.19 | TheH | icenicola » sip.conf |
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| 11:54.14 | icenicola | TheH: so if i have in the device for example codec g711 |
| 11:54.41 | icenicola | and as a preference in the sip.conf g729, g729 will be applied? |
| 11:55.08 | TheH | ok you want to configure all the devices to use g729 then in your sip.conf [general] context you set disallow=all ,allow=g729 |
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| 11:55.50 | TheH | if you want to spec it for indivual devices you set it up with in the context of the device so [myphone] allow=g729 |
| 11:58.04 | icenicola | ok great |
| 11:58.20 | icenicola | what if i have 500 users with g729 |
| 11:58.26 | icenicola | and 500 g711? |
| 12:04.27 | TheH | well you can configure variables for it or even use different files to separate them. Also there is some scripting possiblities that you can say if range is 0-500 use g711 if range is 501-1000 use g729 |
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| 12:26.15 | wonderworld | hey, we have an asterisk box running on an E1-line everything worked well for a week or so. today the machine froze completely. no ssh-login, no local console. the only way to fix it was to remove the power cord. the logs didn't say anything suspicious, maybe the kernel had no time to write something. we are using debian/stable with the debian asterisk-packages for stable (1.2.something). might this crash be related to asterisk or shouldn't |
| 12:26.15 | wonderworld | it crash at all? |
| 12:27.48 | wonderworld | as there are no other services running on the box atm and debian/stable never crashed for me in years i am really puzzled and don't know where to look for the error |
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| 12:31.35 | roadies | anyone who maybe working the Asterisk Appliance Device know where to get the latest software for the AAD? It doesnt seem to be apparent on the Asterisk website |
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| 12:43.39 | x86 | roadies: call Digium's support |
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| 12:46.38 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 12:46.59 | roadies | X86, I just looked at the AOL page... sadly the box I have (bought a year ago) is dead .... *sign* |
| 12:47.15 | roadies | oops thats EOL webpage |
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| 12:58.33 | hsv-al | hello all |
| 13:01.32 | x86 | roadies: you didn't buy more than a year's worth of support for it? |
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| 13:02.38 | x86 | interesting |
| 13:02.49 | x86 | would think they would still support it after only a year old... |
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| 13:10.24 | Takapa | is it possible to trap 302 redirects in the dialplan somehow? e.g. if someone cfwd's their phone, i'd like to change the callerid, for instance |
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| 13:19.41 | railsmunky | Howdy. Can anyone help me with queues. I have it so after 2 mins callers drop to voicemail but there's a requirement for if anymore than 2 people are in the queue. I need to drop the call |
| 13:20.51 | railsmunky | No sorry hang on :) |
| 13:21.02 | railsmunky | I need to drop anymore than 2 calls directly to voicemail |
| 13:21.06 | *** join/#asterisk sakajawebe (n=chazz@216.207.245.1) |
| 13:21.26 | railsmunky | my queue is set to only allow 2 people - so anymore are currently being dropped |
| 13:22.01 | lmadsen | I'm not sure how you would really do that... |
| 13:23.57 | creativx | railsmunky: queue returns an error when trying to join a full queue |
| 13:24.20 | creativx | no wait.. that was joinempty |
| 13:24.21 | creativx | my bad! |
| 13:24.27 | creativx | or wait! haha |
| 13:24.34 | creativx | <PROTECTED> |
| 13:24.35 | creativx | <PROTECTED> |
| 13:24.35 | creativx | <PROTECTED> |
| 13:24.38 | creativx | FULL should do the trick |
| 13:25.11 | *** join/#asterisk bootc_ (n=bootc@78-33-44-114.no-dns-yet.enta.net) |
| 13:30.52 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 13:32.45 | *** join/#asterisk fish-bulb (n=cstewart@216.207.245.1) |
| 13:38.48 | *** join/#asterisk elguero (n=elguero@ns1.nashuacs.com) |
| 13:39.28 | railsmunky | creativx: cool i'll have a looksee at that - thanks |
| 13:45.13 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 13:45.32 | *** join/#asterisk Overclock_Orange (n=tty0@unaffiliated/overclockorange/x-000001) |
| 13:46.18 | Overclock_Orange | Hello, I need a spanish support channel, know you a spanish channel? |
| 13:46.41 | anonymouz666 | maybe #asterisk-es |
| 13:47.21 | anonymouz666 | but if you know a little bit of english (just like me) it's better to ask in here |
| 13:47.54 | Overclock_Orange | thanks anonymouz666 |
| 13:47.55 | Overclock_Orange | :) |
| 13:48.33 | *** join/#asterisk ido- (n=wtf@212.199.189.65) |
| 13:49.08 | Overclock_Orange | anonymouz666: but my english is bad, :P |
| 13:49.31 | ido- | hello. I have to pots lines, however when connecting the fxo to one of them, asterisk does not hangup when i hangup |
| 13:50.01 | TheH | did you but a hangup in your dialplan ? |
| 13:50.06 | TheH | but=put |
| 13:50.22 | TheH | so Exten,n,Hangup() |
| 13:50.47 | ido- | i think so |
| 13:50.56 | ido- | when using the other pots line with the same exact setup it works fine |
| 13:51.11 | TheH | mmm |
| 13:52.38 | ido- | hrm. |
| 13:52.48 | ido- | i get a busy 486 error when trying to make a call |
| 13:52.51 | lmadsen | Overclock_Orange: your english seems decent thus far |
| 13:52.56 | lmadsen | Overclock_Orange: I've seen much worse :) |
| 13:53.02 | Overclock_Orange | lmadsen: :) |
| 13:53.03 | ido- | actually, also with the other (working) pots line, it sometimes happens. |
| 13:53.38 | TheH | well you need to provide some CLI debug info when you make the outgoing call and also your extensions.conf part which is used |
| 13:54.03 | colulu | Hi, I am getting a E1 line for my asterisk and I want to use a TDM card, do I need any software to handle SS7 singalling? |
| 13:54.16 | TheH | nope |
| 13:54.23 | *** join/#asterisk xenonex (n=xenonex@89.218.234.49) |
| 13:54.54 | Overclock_Orange | lmadsen: thanks :), i study english in my university but, in my country only speak spanish, then my english is bad :P |
| 13:55.07 | colulu | TheH: why? |
| 13:55.15 | ido- | TheH, what debugging level ? |
| 13:55.19 | colulu | TheH: is it hangled in zaptel already? |
| 13:56.12 | TheH | colulu: you can configure ss7 handling in the configs |
| 13:56.17 | TheH | ido: set verbose 7 |
| 13:56.41 | TheH | ido: dial BLABLA@your trunj |
| 13:56.59 | *** join/#asterisk wonderworld (n=ww@ip-62-143-31-126.hsi.ish.de) |
| 13:57.06 | colulu | TheH: which configs? |
| 13:57.18 | colulu | TheH: is it the ziptel config? |
| 13:58.09 | TheH | colu: vi /etc/asterisk/zapata.conf signalling = ss7 |
| 13:59.05 | TheH | colulu : Google is a good resources for full examples by the way » http://lists.digium.com/pipermail/asterisk-ss7/2006-October/000554.html |
| 13:59.55 | bootc_ | colulu: unless you know you need to use SS7, you probably don't |
| 14:00.09 | bootc | colulu: I assume you're using ISDN? |
| 14:00.25 | colulu | yes |
| 14:00.31 | bootc | colulu: in Europe? |
| 14:01.10 | colulu | bootc: in Hong Kong |
| 14:01.41 | bootc | not 100% sure about the details in Honk Kong but what you use is ISDN signalling, and on an e1 that's pri_net signalling |
| 14:02.03 | bootc | you'd need to check the exact details with your telco |
| 14:02.09 | TheH | mmm is that not NI-1 in hong kong ? |
| 14:02.40 | colulu | bootc: i want to understand how SS7 is handled with using TDM card. |
| 14:03.23 | bootc | but do you know that you're actually using SS7? |
| 14:03.51 | colulu | yes, i asked. |
| 14:04.24 | colulu | so, zaptata does do the SS7-to-SIP signalling conversion, right? |
| 14:04.45 | *** join/#asterisk guilherme-jorge (n=guilherm@200.213.210.2) |
| 14:04.54 | *** join/#asterisk DarylVoip (n=daryl@75.147.121.177) |
| 14:05.25 | bootc | colulu: Zaptel does the Asterisk to PSTN conversion |
| 14:05.36 | bootc | be that SS7, ISDN PRI, BRI, Analogue, etc... |
| 14:06.23 | colulu | is it true that Zaptel provides MTP3 and above? |
| 14:07.10 | bootc | no idea, I've never used SS7 |
| 14:07.51 | bootc | I have 2x 30 channel ISDN PRI trunks |
| 14:08.04 | Overclock_Orange | Thanks to all |
| 14:08.06 | Overclock_Orange | good bye :) |
| 14:08.08 | *** part/#asterisk Overclock_Orange (n=tty0@unaffiliated/overclockorange/x-000001) |
| 14:09.14 | colulu | bootc: what hardware are you using? |
| 14:09.49 | bootc | colulu: Sangoma A104's at the moment, pushing for an upgrade to A104d's |
| 14:11.41 | colulu | bootc: other than pricing, what other advantage does Sangoma has over a physical gateway? |
| 14:12.14 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 14:12.23 | *** join/#asterisk spokra (n=spokra@host093-179-178.sea0.speakeasy.net) |
| 14:13.53 | bootc | colulu: we needed an all-in-one PBX system, so the servers needed to have the cards built-in |
| 14:14.17 | bootc | we also thought it would be better for us than ISDN to Ethernet (TDMoE) bridges |
| 14:16.15 | *** join/#asterisk anthm (n=anthm@68-30-36-193.area4.spcsdns.net) |
| 14:20.58 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 14:23.18 | *** join/#asterisk friedrich| (i=friedric@trem-servers.com) |
| 14:23.47 | *** join/#asterisk mindCrime (n=chatzill@216.27.62.2) |
| 14:27.05 | *** join/#asterisk RoyK (n=roy@fw.fortel.no) |
| 14:28.42 | *** join/#asterisk Horvath (n=Horvath@bas1-toronto26-1279685211.dsl.bell.ca) |
| 14:32.37 | colulu | does anyone know where I can get some help on SS7? |
| 14:33.07 | Horvath | I need to convert some incoming faxes to PDF. I'm less than 1ms away from my SIP trunk and PSTN on LAN so no latency / jitter issues is there a way to do fax to pdf conversion with asterisk reliably or should I just look for an appliance/service? rxfax? haylafax? |
| 14:35.13 | errr | Horvath: Id use hylafax |
| 14:35.47 | Horvath | errr, is it going to be reliable tho? |
| 14:35.57 | errr | Horvath: hylafax+iaxmodem has been working great for us for 18 months now |
| 14:36.07 | [TK]D-Fender | Horvath: the moment VoIP comes into the picture, you risk trouble. |
| 14:36.12 | errr | we have a very large fax volume |
| 14:36.35 | Horvath | [TK]D-Fender, yea I understand thats why I have avoided doing fax over ip... cant be avoided in this instance |
| 14:37.00 | Horvath | errr, do you run your hylafax on a different box or same as your asterisk? |
| 14:37.18 | [TK]D-Fender | Horvath: then even over LAN you risk losing faxes. but go for it. Start with SpanDSP + rxfax, and go from there |
| 14:37.26 | errr | Horvath: we dont us it with voip except for the iaxmodem part, the rest is though our pri. hylafax is running on our asterisk server |
| 14:37.33 | *** join/#asterisk ManxPower (n=manxpowe@168.sub-70-222-30.myvzw.com) |
| 14:37.50 | Horvath | Well it would be over lan out to PRI |
| 14:38.36 | [TK]D-Fender | Horvath: LAN itself can be a problem. 1 delayed or lost packet is 20ms and risks failure |
| 14:38.52 | x86 | [TK]D-Fender: iaxmodem + hylafax is WAY more reliable |
| 14:39.19 | [TK]D-Fender | x86: ok, fine, sure |
| 14:39.44 | Horvath | [TK]D-Fender, unless your lan is flooded or you have a bad cable somewhere I cant see why you would have dropped packets |
| 14:39.53 | x86 | [TK]D-Fender: still though, you're right, it's not as reliable as a fax machine with an analog line connected to it ;) |
| 14:41.44 | *** join/#asterisk gaetronik (n=gaetan@eficaz2.manquehue.net) |
| 14:41.48 | gaetronik | Hi |
| 14:42.07 | gaetronik | i'm stuck with a problem with the AgentLogin command |
| 14:42.32 | gaetronik | i've to test in the dialplan when the comunication is over |
| 14:42.41 | gaetronik | ie when the agent hangup |
| 14:42.55 | [TK]D-Fender | gaetronik: You don't with that command. |
| 14:43.11 | *** join/#asterisk murdock_ut (n=chatzill@mail.kimballequipment.com) |
| 14:43.18 | [TK]D-Fender | gaetronik: Look at AMI messages for call completion or trace queue-log entries. |
| 14:43.23 | gaetronik | [TK]D-Fender, how coul i know when the agent go away |
| 14:43.27 | [TK]D-Fender | gaetronik: But there will be no other way |
| 14:43.33 | gaetronik | ok it's what o don't want to do |
| 14:43.49 | gaetronik | shit |
| 14:43.53 | gaetronik | i'm screwed |
| 14:44.38 | gaetronik | it's the same with AgentCallBack login ? |
| 14:45.06 | *** join/#asterisk Whitor (n=Whitor@rrcs-24-97-4-150.nys.biz.rr.com) |
| 14:45.34 | ManxPower | [TK]D-Fender: chan_local would not be helpful? |
| 14:45.38 | [TK]D-Fender | gaetronik: No, agentcallbacklogin uses the dialplan. |
| 14:45.53 | [TK]D-Fender | ManxPower: not with agentlogin. |
| 14:45.58 | ManxPower | ah |
| 14:46.03 | *** join/#asterisk moy (n=moyhu@189.169.71.85) |
| 14:46.17 | [TK]D-Fender | gaetronik: So yes, if you use callback, or anything dialplan based, then you can track it there. |
| 14:46.17 | gaetronik | [TK]D-Fender, but i will return after the login |
| 14:46.43 | gaetronik | not when the agent go to home |
| 14:46.56 | gaetronik | ManxPower, i tried |
| 14:47.02 | gaetronik | th e chan local way |
| 14:47.31 | [TK]D-Fender | gaetronik: You can have the local channel method kick him out if he doesn't answer. |
| 14:48.13 | Whitor | Hi. I'm having trouble linking Flash Operator Panel and AsteriskNow ... FOP doesn't seem to be pulling any extension / trunk information from the server... All I see are the pre-setup buttons |
| 14:48.20 | Whitor | or any that I create manually |
| 14:49.01 | Whitor | I know that this is not precicely on-topic... but I figgured someone in her emight have a clue where to start looking |
| 14:49.22 | Whitor | I've configured the Op-server.cfg file appropriately ... I think |
| 14:49.38 | [TK]D-Fender | whiteclearly not. |
| 14:50.02 | [TK]D-Fender | Whitor: its all AMI, so its not dialplan thats off. |
| 14:50.03 | gaetronik | [TK]D-Fender, i will write a bunch of ugly line to catch ami event |
| 14:50.37 | [TK]D-Fender | gaetronik: well you can do it through dialing out to the agent on a local channel and catching the timeout. |
| 14:50.40 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 14:51.21 | gaetronik | [TK]D-Fender, but it will annoy the agent if it's on line, no? |
| 14:52.28 | gaetronik | when agentCall bcack return |
| 14:52.52 | gaetronik | at the end of the session or after the login? |
| 14:53.08 | ManxPower | Whitor: I suggest asking in the right place, #asterisk-now |
| 14:53.17 | ManxPower | or #asterisknow, I don't recall |
| 14:53.39 | Whitor | ManxPower, yes. Thank you ! |
| 14:53.59 | Whitor | I didn'te realize that existed... (supose I should have checked) |
| 14:54.05 | Whitor | Thanks [TK]D-Fender . too |
| 14:54.27 | ManxPower | Whitor: It says in the /topic |
| 14:54.37 | Whitor | ok. now I feel really dumb |
| 14:54.44 | Whitor | thx again |
| 14:54.50 | Whitor | :) |
| 14:54.54 | [TK]D-Fender | gaetronik: Once the call is in dialplan you can check if they're on the phone and not pass the call, or if they aren't, pass the call and if no answer, log them out, etc |
| 14:57.27 | wonderworld | our asterisk machine froze today after running perfectly for about a week. might this be related to asterisk or is it more likely that the hardware failed? |
| 14:58.16 | jets | explain froze |
| 14:58.31 | *** join/#asterisk jjshoe (n=jjshoe@72.37.252.50) |
| 14:58.39 | jets | the whole system froze and was unpingable, unmanageable or the asterisk quit 'responding'? |
| 14:59.00 | wonderworld | no console, no remote ssh, console screen 1 filled with garbage. only way to fix it was to plug the power cord |
| 14:59.43 | wonderworld | debain stable server with asterisk packages from debian stable |
| 14:59.49 | [TK]D-Fender | wonderworld: Based on your description its impossible to tell. |
| 15:00.18 | wonderworld | ok. i checked the logs but there was nothing. likely the kernel / asterisk had no time to write to the logs |
| 15:01.16 | wonderworld | *does* this actually happen sometimes with asterisk or would it be way more likely that the hardware failed? |
| 15:01.20 | *** join/#asterisk andrewn (n=andrew@76-191-151-229.dsl.dynamic.sonic.net) |
| 15:02.00 | wonderworld | debian has a quiet old version in it's repos. i was wondering if that might be another possibilty for the error. it's 1.2.13 |
| 15:02.33 | [TK]D-Fender | wonderworld: in a word : ANCIENT |
| 15:02.36 | Whitor | Is there a reccomended distro of linux to use for asterisk ? |
| 15:03.06 | wonderworld | ok. so i will install the latest 1.4.x and see if it happens again. |
| 15:03.18 | *** join/#asterisk Deeewayne (n=Deeewayn@216.207.245.1) |
| 15:03.18 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 15:04.43 | [TK]D-Fender | Whitor: As has been answered countless times : The one you can best administer and provide *'s dependencies with |
| 15:05.13 | Whitor | [TK]D-Fender, cool... I'm new here... so I havn't seen countless times yet. :) |
| 15:05.22 | *** join/#asterisk caio1982_ (i=caio1982@CAcert-br/caio1982) |
| 15:06.04 | ManxPower | George Carlin died 8-((( |
| 15:06.31 | hsv-al | and the winds of time move on |
| 15:06.37 | hsv-al | so too manxpower, we all meet the same fate |
| 15:06.42 | *** join/#asterisk CunningPike (n=arodgers@204.239.8.157) |
| 15:15.33 | *** join/#asterisk r0land (n=roland@193.227.191.91) |
| 15:15.35 | r0land | hellow all |
| 15:15.45 | r0land | anyone has experience with sipura 3102! |
| 15:16.14 | r0land | everything is setup perfectly asterisk's side.. yet on sipua's side.. if i call out! theres at least a 4 seconds delay before getting the dial tone! |
| 15:16.19 | r0land | wht might be the cause of tht! |
| 15:17.52 | [TK]D-Fender | r0land: which dial tone? |
| 15:18.24 | r0land | [TK]D-Fender any dialtone! if i wanna call a pstn line or a vonage..! theres no dial tone except after 4 seconds for pstn, and 8 seconds for vonage sip |
| 15:19.06 | railsmunky | Does anyone have any ideas if 0141 numbers in the UK need to present differently over an ISDN? |
| 15:19.15 | [TK]D-Fender | r0land: please describe your layout with some sense of ORDER to it. you have crap plugged into OTHER crap and we can't tell where the failure might be with it so far |
| 15:19.27 | ManxPower | r0land: There is the dialtone generated by the ATA, then there is the dialtone generated by the PSTN |
| 15:19.30 | railsmunky | Everything is working over my PRi apart from 0141's which are coming up that BT can't recognise the number |
| 15:19.40 | r0land | ManxPower ah ok |
| 15:19.49 | ManxPower | IF you have a PBX connected to it, then you have the dialtone from the legacy PBX. |
| 15:19.54 | r0land | ManxPower is there a way to shorten out this delay! |
| 15:20.04 | [TK]D-Fender | r0land: ...... |
| 15:20.13 | [TK]D-Fender | r0land: Your ATA provides dialtone, not PSTN <- |
| 15:20.14 | ManxPower | r0land: once we figure out what dialtione is delayed we might be able to |
| 15:20.17 | r0land | ManxPower the crap issue got explained to me by ManxPower so thanks :) |
| 15:20.19 | [TK]D-Fender | r0land: decsribe the LAYOUT. |
| 15:20.39 | r0land | ManxPower well the thing is, if i call from my softphone i get an immediate dialtone! |
| 15:20.52 | ManxPower | r0land: you should never get a dialtone from a softphone |
| 15:21.01 | r0land | ManxPower yet if i call from a phone device it works |
| 15:21.10 | [TK]D-Fender | r0land: and your softphone is faking ITS OWN DIALTONE IMMEDIATELY. |
| 15:21.10 | ManxPower | r0land: but until we see your layout we really can't help you |
| 15:21.22 | r0land | ManxPower when i dial 01, i get to my PSTN |
| 15:21.24 | ManxPower | Analog phone -> Sipura 3120 -> PSTN line |
| 15:21.34 | r0land | so if i dial 01, on softphone i *immediatly* get the pstn's dial tone |
| 15:21.39 | ManxPower | or maybe Analog phone -> SIPura 3120 -> Asterisk -> SIP/ITSP |
| 15:21.55 | r0land | analog phone/softphone - > asterisk - sipura- pstn |
| 15:22.01 | ManxPower | or maybe analog phone -> SIPURA -> channel bank -> T-1card |
| 15:22.01 | [TK]D-Fender | r0land: describe how stuff is connected and configured or you are wasting our time. |
| 15:22.26 | ManxPower | r0land: try dialing the full number |
| 15:22.31 | r0land | ManxPower i did |
| 15:22.36 | r0land | ManxPower it takes 17 seconds |
| 15:22.44 | r0land | ManxPower if i just dial 01 to get to the pstn.. it takes 4 |
| 15:22.52 | r0land | thts on analog.. |
| 15:22.57 | [TK]D-Fender | r0land: pastebin a complete call. |
| 15:23.00 | r0land | though on softphone i have no prob whtsoever |
| 15:23.04 | r0land | [TK]D-Fender ok |
| 15:23.06 | [TK]D-Fender | r0land: your #'s mean nothing. |
| 15:23.18 | r0land | yet the thing is |
| 15:23.21 | r0land | im sure its not from asterisk's side |
| 15:23.23 | [TK]D-Fender | r0land: STOP! |
| 15:23.36 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-79-183-173-78.red.bezeqint.net) |
| 15:23.36 | r0land | since softphone works.. yet my analog is pluged directly to sipura |
| 15:23.37 | [TK]D-Fender | r0land: your continued description is becoming even MORE useless. |
| 15:23.51 | r0land | hehe u never cease to amaze me [TK]D-Fender |
| 15:23.55 | r0land | thank god for ManxPower |
| 15:24.07 | [TK]D-Fender | r0land: and the Sipura has *2* analog and we can never tell WHICH ONE you're talking about. You are making a huge mess out of your description. |
| 15:24.17 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 15:24.38 | [TK]D-Fender | r0land: Show the call. |
| 15:24.59 | r0land | this is my setup: softphones-> asterisk->sipura<- (1) analog phone (2) PSTN line |
| 15:25.16 | [TK]D-Fender | r0land: ... show the CALL <-- |
| 15:25.21 | *** join/#asterisk wonderworld (n=ww@ip-62-143-31-126.hsi.ish.de) |
| 15:25.41 | r0land | [TK]D-Fender wont do u any good, since its the same log tht comes out calling from a softphone |
| 15:25.42 | r0land | identical |
| 15:25.51 | r0land | so unless ud simply like to check it out |
| 15:25.54 | r0land | theres no use for it |
| 15:25.56 | r0land | its IDENTICAL |
| 15:25.57 | ManxPower | r0land: A critical thing to know is the relationship between the timing of getting the dialtone and the timing of stuff on the CLI |
| 15:26.00 | r0land | i checked it and double checked it |
| 15:26.00 | [TK]D-Fender | r0land: I don't care if its "identical", I want to SEE IT. |
| 15:26.14 | [TK]D-Fender | r0land: and maybe your eyes aren't working so well. |
| 15:26.32 | r0land | [TK]D-Fender i used araxis :) identical.. |
| 15:26.37 | ManxPower | r0land: you don't know even close to enough stuff to know what is important and what is not important. Stop trying to second guess the people trying to help you or they will stop trying to help you |
| 15:26.50 | [TK]D-Fender | "But I did everything right,, I checked!" <- typically BULLSHIT |
| 15:26.59 | r0land | ManxPower hmm u got a point |
| 15:27.02 | [TK]D-Fender | r0land: You aren't giving us anything sueful. |
| 15:27.08 | r0land | just a sec lemme pastebin |
| 15:27.08 | [TK]D-Fender | useful* |
| 15:28.16 | *** join/#asterisk LuisTorres (n=chatzill@bl6-196-27.dsl.telepac.pt) |
| 15:29.34 | *** join/#asterisk grEvenX (n=even@pc107-130.ktv.no) |
| 15:30.31 | railsmunky | anyone any ideas?> |
| 15:31.46 | ManxPower | railsmunky: just the standard stuff, finding the vaule of HANGUPCAUSE after the Dial() fails, that standard sort of stuff |
| 15:32.02 | ManxPower | railsmunky: pastebin of failed call, etc |
| 15:32.18 | railsmunky | http://pastebin.com/m49c0115a |
| 15:32.43 | railsmunky | thing is - it doesn't fail - Gets through to bt but then BT give me the "number not recognised" malarky |
| 15:32.45 | ManxPower | railsmunky: that is not a failed call |
| 15:32.51 | railsmunky | i know :) |
| 15:32.53 | *** join/#asterisk svenna_ (n=svenna@p548D0341.dip0.t-ipconnect.de) |
| 15:33.01 | ManxPower | railsmunky: turn off the inband setting in /etc/asterisk/zapata.conf |
| 15:33.09 | r0land | ManxPower [TK]D-Fender http://pastebin.ca/1053323 |
| 15:33.11 | [TK]D-Fender | railsmunky: Dial("SIP/900-c40ca2e0", "Zap/g0/1412433600") in new stack <--- doesn't look like 0141 to me... perhaps you trimed off too many digits? |
| 15:33.30 | [TK]D-Fender | railsmunky: no "0" <- |
| 15:33.40 | railsmunky | [TK]D-Fender on BT you send everything without the 0 |
| 15:33.53 | [TK]D-Fender | railsmunky: if you say so... |
| 15:33.59 | railsmunky | everything else has been working cool for the past month |
| 15:34.08 | railsmunky | if you send the 0 BT kicks off |
| 15:34.12 | ManxPower | railsmunky: never try to guess what the telco wants. |
| 15:34.23 | colulu | hi, is it possible to plug 4 8-port sangoma cards into one system? Can a dual xeon supports that kind of load? |
| 15:34.38 | ManxPower | railsmunky: but any future troubleshooting of the PRI will be much harder until you remove the inband setting |
| 15:34.42 | [TK]D-Fender | colulu: Highly unrecommended. |
| 15:34.54 | [TK]D-Fender | r0land: Now what are we looking at? |
| 15:35.02 | railsmunky | ManxPower okedoke - i'll give that a go |
| 15:35.20 | r0land | [TK]D-Fender i called out from an analog phone to 200 (pstn extension) i then got a dialtone (from pstn) and called out |
| 15:35.34 | colulu | TK: how many 8-ports card do you think can be supported in a dual-Xeon? What if I switch to a qual-core PC? |
| 15:35.51 | [TK]D-Fender | r0land: don't say "analog phone", tell us what its PLUGGED ITO that lets it talk to *. |
| 15:36.09 | [TK]D-Fender | r0land: And why would you be waiting for a 2nd tone from your SPA to dial out? |
| 15:36.26 | [TK]D-Fender | colulu: its not the CPU, its the bus. with interrupt hits |
| 15:36.38 | r0land | analog phone connected to pstn, called 200 which is in extensions.conf directing asterisk to the PSTN line thts setup in sipura. |
| 15:36.44 | [TK]D-Fender | colulu: What are you doing with so many PRI's? |
| 15:36.58 | [TK]D-Fender | r0land: if an analog phone is connected to the PSTN, then * is NOT INVOLVED. |
| 15:37.05 | r0land | its not |
| 15:37.20 | railsmunky | this is my current zapata.conf http://pastebin.com/m568dd7c4 |
| 15:37.23 | r0land | analog phone is connected to a port in sipura and tht port has an extensions assigned to it *200* |
| 15:37.23 | [TK]D-Fender | r0land: You are not making sense. |
| 15:37.39 | *** join/#asterisk MrNaz (n=naz@ppp59-167-142-211.lns3.mel6.internode.on.net) |
| 15:37.44 | railsmunky | so just ad in an priindication= outofband |
| 15:37.48 | railsmunky | ? |
| 15:37.52 | [TK]D-Fender | r0land: You jsut said analog phone was connected to the PSTN, and now you say its conencted to the SPA's FXS port. MAKE UP YOUR MIND |
| 15:37.59 | ManxPower | railsmunky: it should default to that if not set |
| 15:38.08 | railsmunky | default to outofband? |
| 15:38.15 | ManxPower | r0land: accuracy is very, very important |
| 15:38.21 | ManxPower | railsmunky: yup. |
| 15:38.27 | ManxPower | inband on PRI is an evil hack |
| 15:38.33 | [TK]D-Fender | r0land: So again, when dialing out the FXO port on your SPA, why aren't you giving it the full number to dial? |
| 15:39.01 | ManxPower | r0land: the problem is in the dialplan on your SIPura |
| 15:39.11 | r0land | analog phone has an extensions of 200 tht extensions comes from the setup on the port of sipura. on the other hand PSTN is connected to FXO poirt in the same sipura with extensions 100.. so wheni dial *01* from my analog phone, asterisks direct me to extensions 100 (thts how i configured it in extensions.conf) and tht way i get a dial tone coming from the pstn line |
| 15:39.16 | railsmunky | ManxPower: okedoke so is there some BT docs etc... on what they want (rather than guessing :)) |
| 15:39.30 | colulu | how many E1 can a PC handles? |
| 15:39.34 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 15:39.40 | [TK]D-Fender | r0land: well you wait on your FXS port because you didn't configure its dialplan better. |
| 15:39.46 | ManxPower | railsmunky: it all depends on how they set it up, but you almost never want inband on PRI. It basically makes the PRI act like an analog port. |
| 15:39.52 | r0land | [TK]D-Fender i have 8 lines, ive assigned 8 codes 01,02,04,... i have 2 steps dialing procedure |
| 15:40.06 | railsmunky | ManxPower: fair enough - but that's not the problem here |
| 15:40.06 | [TK]D-Fender | r0land: and you shouldn't be waiting for dialtone on your FXO anyways, you should be passing it the # to dial, not waiting for a 2nd tone. |
| 15:40.14 | ManxPower | r0land: your problem is that your dialplan is badly designed |
| 15:40.15 | r0land | hmm |
| 15:40.20 | r0land | ManxPower sipura's ? |
| 15:40.31 | [TK]D-Fender | r0land: Again, you TELL the SPA what to dial, you don't wait for IT to give you a tone. |
| 15:40.38 | [TK]D-Fender | r0land: YES, the Sipura dialplan. |
| 15:40.39 | ManxPower | r0land: since Asterisk is not involved, it would have to be the SIPura |
| 15:40.45 | r0land | ok |
| 15:40.47 | r0land | thanks for the help |
| 15:40.50 | r0land | ill figure it out :) |
| 15:40.57 | [TK]D-Fender | r0land: * doesn't give you dialtone with any of that equipment |
| 15:41.04 | r0land | i get it.. but pstn does |
| 15:41.07 | r0land | whic his taking forever |
| 15:41.15 | r0land | unlike when using softphone |
| 15:41.20 | [TK]D-Fender | r0land: And thats the FXS ports fault <- |
| 15:41.28 | r0land | ya i got it now |
| 15:41.30 | r0land | ill figureit otu |
| 15:41.32 | r0land | thanks :) |
| 15:41.36 | [TK]D-Fender | r0land: because you didn't configure its dialplan to hand it faster. |
| 15:41.57 | ManxPower | r0land: NO! IT should NOT be giving you a dialtone. You pick up the phone, the ATA gives you the dialtone, you dial, then the SIPura dials out the PSTN port without giving you audio until it's done dialing, so you should never hear a 2nd dialtone |
| 15:42.14 | ManxPower | you dial the COMPLETE number. |
| 15:42.18 | [TK]D-Fender | ManxPower: he's not passing the #. |
| 15:42.32 | ManxPower | [TK]D-Fender: I know and that's the entire problem |
| 15:42.37 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 15:42.50 | r0land | well i want it to b a 2 stage dialing procedure |
| 15:42.55 | [TK]D-Fender | ManxPower: half of it. His problem is that he's not happy with his FXS dialplan delay |
| 15:42.59 | [TK]D-Fender | r0land: WHY? |
| 15:43.01 | r0land | i dont want ppl to dial right away.. asi want them to speciofy from which line they want to call.. |
| 15:43.27 | [TK]D-Fender | r0land: 01987654321 <- there, take the prefix and dial out that SPA <- |
| 15:43.31 | r0land | land line phones are to b dialed from a specific line, mobile numbers are to b called from a specific line, international.. from a specific line |
| 15:43.38 | [TK]D-Fender | r0land: you make people wait and get a 2nd tone for nothing. |
| 15:43.54 | r0land | it helps them know if the line is busy on our end or the recievers |
| 15:44.01 | ManxPower | r0land: you have a bad design. I can't help you further. |
| 15:44.04 | [TK]D-Fender | r0land: if its going to fail, its going to fail, but your way they have to wait for dial timeout only to be disappointed later. |
| 15:44.18 | r0land | lemme try it ur way |
| 15:44.21 | r0land | just a sec |
| 15:44.23 | [TK]D-Fender | r0land: the SPA will respond BUSY EITHER WAY! |
| 15:50.38 | *** part/#asterisk goobsoft (n=chad@cpe-24-167-97-202.satx.res.rr.com) |
| 15:55.12 | [TK]D-Fender | ManxPower: *urg* |
| 15:58.17 | *** join/#asterisk egypcio (n=korn@unaffiliated/egypcio) |
| 16:02.00 | jetlagmk2 | Anyone know if there phones that will show me the status of other phones? As simple as "in use/not in use" is fine. |
| 16:03.27 | [TK]D-Fender | jetlagmk2: Yes, many phones support "presence" which will tell you if someone is on the phone, usually by a lit indicator |
| 16:03.41 | [TK]D-Fender | jetlagmk2: how many indicators do you need? |
| 16:04.10 | Rem| | when i do a 'zap show channel X' on channel X which has a fax call should echo cancel be turned off? |
| 16:05.24 | *** join/#asterisk exvito (n=exvito@mail.colours.pt) |
| 16:05.44 | [TK]D-Fender | Rem|: it should |
| 16:06.47 | jetlagmk2 | I would need 5 indicators right now. |
| 16:08.03 | exvito | hi all... two questions: any source on the info regarding limits in callgroup/pickupgroup ? (a quick grep of the source revealed nothing...) |
| 16:08.09 | [TK]D-Fender | jetlagmk2: That'd take a phone like a Polycom 6XX series, or Aastra 57i |
| 16:08.46 | [TK]D-Fender | jetlagmk2: for "instant view". Polycom 501 would do useing the micro-browser, but thats lagged,a nd less elegent. |
| 16:09.08 | [TK]D-Fender | exvito: limit of 63 IIRC. Go check the WIKI, its listed there. |
| 16:09.14 | Rem| | [TK]D-Fender: what if you have HW echo card? |
| 16:09.21 | [TK]D-Fender | exvito: And keep in mind its dated, and may have been expanded |
| 16:09.32 | [TK]D-Fender | Rem|: the same. You can't EC a fax call or it will fail. |
| 16:09.56 | exvito | [TK]D-Fender: ...I checked it, but I was wondering exactly that... how dated is that info ? Thanks anyway. |
| 16:10.09 | Rem| | [TK]D-Fender: I am not seeing the echo turning off when I receive a fax call which I think that is why they are failing |
| 16:10.25 | [TK]D-Fender | Rem|: make sure you set "faxdetect=both" |
| 16:10.41 | *** join/#asterisk implicit (n=implicit@dhcp-x223-230.mobile.uci.edu) |
| 16:11.42 | Rem| | [TK]D-Fender: thanks |
| 16:11.49 | Rem| | i'll try that |
| 16:14.48 | colulu | TK: If I want to use sangoma to receive T38 fax, will it work? |
| 16:14.49 | *** part/#asterisk exothermc (n=miles@74.85.89.146) |
| 16:15.03 | hardwire | Anybody a zaptel hacker here? |
| 16:15.17 | [TK]D-Fender | colulu: No.. because T.38 has nothing to do with PSTN <- |
| 16:15.23 | colulu | TK: i actually don't need that many E1 for now, I just want to know the possibility for future expansion using sangoma |
| 16:15.26 | exvito | (fyi: found it... max group is in fact 63 as of 1.4.21 @main/channel.c) |
| 16:15.36 | [TK]D-Fender | colulu: How many do you need? |
| 16:15.53 | [TK]D-Fender | exvito: There you go |
| 16:15.54 | colulu | TK: just one of those 8-port card |
| 16:16.12 | colulu | TK: so in the future if I want to expand, I would need a new PC, then? |
| 16:16.16 | Rem| | [TK]D-Fender: I thought faxdetect is if you want to pass it to the fax exten |
| 16:16.24 | *** part/#asterisk exvito (n=exvito@mail.colours.pt) |
| 16:16.50 | colulu | TK: so does it mean that if I use sangoma card, I won't be able to do any T38 faxing? or is there a solution? |
| 16:17.16 | [TK]D-Fender | Rem|: thats the "fax" standard exten. Not sure if this is separate, but it can't hurt... |
| 16:17.32 | [TK]D-Fender | colulu: it means that youa re talking about apples and oranges. |
| 16:17.42 | [TK]D-Fender | colulu: they have nothing to do with each other. |
| 16:18.08 | [TK]D-Fender | colulu: Want to know how many MPG I get on my bicycle? |
| 16:18.27 | colulu | TK: how many? |
| 16:19.00 | colulu | TK: so the fact that whether I can receive/send T38 fax has nothing to do with what TDM hardware I use, right? |
| 16:19.23 | Rem| | [TK]D-Fender: no, that option is if you want to send that fax to the fax exten |
| 16:19.39 | colulu | TK: the reason why I asked that is because all the hardware gateway providers have t38 in their spec, but sangoma does not. |
| 16:19.59 | [TK]D-Fender | Rem|: Ok, wasn't sure if the code was implicit or not. Other than that, Zaptel itself should detect fax/modem tones and disable EC accordingly. |
| 16:20.25 | bootc | colulu: T.38 is Fax over SIP, AFACU |
| 16:20.27 | [TK]D-Fender | colulu: T.38 is for sending faxes over VOIP <_ So far E1 has nothing to do with VoIP. |
| 16:20.44 | bootc | you wo |
| 16:21.05 | bootc | colulu: you would need Asterisk to turn T.38 back into noise and pass it back over the PSTN |
| 16:22.00 | bootc | colulu: because you can't reliably send fax over SIP/VoIP without demodulating the fax signal into bits and bytes then remodulating it to send it back down the PSTN |
| 16:22.41 | *** join/#asterisk ngvoice (n=linxrout@58.186.250.2) |
| 16:23.35 | ngvoice | ;;; |
| 16:23.37 | ngvoice | hello |
| 16:25.09 | colulu | bootc: so the hardware gateway would automatically do the analog-to-digital conversion whereas if I use a sangoma card, I would need Asterisk to do the conversion, is that correct? |
| 16:25.22 | bootc | colulu: that's right |
| 16:25.38 | bootc | it's only really important if you plan on putting lots of fax through your system |
| 16:26.21 | bootc | colulu: you would need a device on the fax end of things to do the conversion too, changing the fax into t.38 before sending it down ethernet |
| 16:26.27 | brad_mssw | colulu: sounds like you're looking for T.38 termination |
| 16:26.48 | brad_mssw | colulu: there was a patch set for asterisk a while back by coppice from what I remember |
| 16:27.09 | [TK]D-Fender | brad_mssw: Not yet he isn't |
| 16:27.13 | colulu | brad_mssw: i think callweaver does that too... |
| 16:27.21 | bootc | it's CPU intensive stuff, basically acting as a modem in software |
| 16:27.28 | [TK]D-Fender | brad_mssw: Don't project. So far he's mixing things in statements that don't add up |
| 16:27.35 | brad_mssw | [TK]D-Fender: pretty sure he did (we're talking like 1.5 years ago) |
| 16:27.47 | [TK]D-Fender | brad_mssw: ... YEARS? |
| 16:28.12 | [TK]D-Fender | brad_mssw: colulu just asked if a SANGOMA CARD supports T.28 MINUTES ago. |
| 16:28.16 | [TK]D-Fender | T.38 |
| 16:28.34 | brad_mssw | doesn't have anything to do with the card |
| 16:28.39 | [TK]D-Fender | brad_mssw: Exactly. |
| 16:28.55 | *** join/#asterisk SteveTotaro (n=Administ@pool-151-196-237-15.balt.east.verizon.net) |
| 16:28.57 | [TK]D-Fender | colulu: T.38 has nothing to do with being able to recieve faxes over E1. |
| 16:29.13 | brad_mssw | I'm almost positive the T.38 termination support was taken directly from coppice's patches for asterisk |
| 16:29.19 | colulu | TK: I was a bit confused before as the the difference between Sangoma and other hardware gateway in terms of fax support. |
| 16:29.26 | brad_mssw | (for callweaver that is) |
| 16:29.47 | [TK]D-Fender | colulu: PCI solutions don't do T.28. T.38 is for VoIP. |
| 16:29.55 | [TK]D-Fender | T.38* (damn typos) |
| 16:29.58 | colulu | TK: now i know it is only for the conversion from analog to digital |
| 16:30.01 | bootc | colulu: Sangoma's aren't hardware gateways, that's the main difference |
| 16:30.07 | bootc | they don't do SIP<=>PSTN |
| 16:30.22 | [TK]D-Fender | colulu: No, not from analog to digital, from audio to VOIP <- |
| 16:30.33 | bootc | they do Asterisk<=>Zaptel<=>PSTN |
| 16:30.51 | [TK]D-Fender | Where audio = TDM |
| 16:30.51 | bootc | though of course you can change the whole Asterisk/Zaptel side of things to something else completely, but this is #asterisk |
| 16:31.12 | *** join/#asterisk l0verb0y (n=l0verb0y@119.111.96.120) |
| 16:31.26 | bootc | T.38 only really applies to SIP |
| 16:31.34 | bootc | anyway, time to head home, see ya chaps |
| 16:31.49 | colulu | I get it now. Thanks you all. |
| 16:32.02 | *** join/#asterisk wonderworld (n=ww@ip-62-143-31-126.hsi.ish.de) |
| 16:32.10 | *** join/#asterisk dogmeat (n=Bob@unaffiliated/dogmeat) |
| 16:32.15 | jeev | Fender |
| 16:32.21 | colulu | TK: regarding performance, is a PCI card comparable with a hardware gateway? |
| 16:32.30 | jeev | you use this ? http://5sn.net/asterisk/rawplayer/ or heard of anyone using |
| 16:33.26 | [TK]D-Fender | jeev: Don't use, others do. |
| 16:34.06 | [TK]D-Fender | colulu: Depends on your idea of "comparable". What devcie I would recommend depends on specific requirements. |
| 16:34.14 | jeev | is it normal if it's still cat'ing the file after it says stopped ? |
| 16:34.15 | *** join/#asterisk grantm (n=grant@66.29.180.194.static.utahbroadband.com) |
| 16:34.27 | jeev | -- Stopped music on hold on SIP/... |
| 16:34.38 | jeev | and it's still running rawplayer with the file, and a cat blah.raw process |
| 16:34.55 | [TK]D-Fender | jeev: Not sure if its kept as a continuous stream and tapped into only when needed. |
| 16:34.59 | *** join/#asterisk Nuitari (n=Nuitari@cybernet.nuitari.net) |
| 16:35.39 | jeev | i guess i've got to test it more because i got a line of noise y esterday after the file was done, not sure if it was from me uploading another file in the middle |
| 16:35.51 | jeev | but i dont think it ends.. the source is rather small and all it does is cat it, it never ends i gues |
| 16:37.40 | colulu | TK: if I am on a high traffic environment and require stability, what would you recommand? |
| 16:38.50 | [TK]D-Fender | colulu: That offers no details for your requirement. |
| 16:42.02 | *** join/#asterisk wolfson (n=hehe@75.145.3.89) |
| 16:44.02 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.193) |
| 16:46.01 | hardwire | Anybody fingered out a way to administratively "down" a configured zaptel span? |
| 16:46.22 | hardwire | stop the cards from responding at all to any plugged in equipment |
| 16:47.33 | [TK]D-Fender | hardwire: yes, there's this little connector in the backplate of the card and you..... :) |
| 16:47.38 | [TK]D-Fender | *yank* |
| 16:48.46 | hardwire | [TK]D-Fender: Know any robots that can do that? |
| 16:49.09 | [TK]D-Fender | hardwire: Not specifically... but I'm sure I could rig something up :) |
| 16:49.23 | *** join/#asterisk Cyon (n=cyon@fxp1.dmz1.cro1.bestweb.net) |
| 16:49.47 | [TK]D-Fender | hardwire: Doesn't need to be a robot. a quick relay would do. |
| 16:50.10 | hardwire | or, a driver that deactivated the card from reacting or processing anything. |
| 16:50.14 | *** join/#asterisk sakajawebe (n=chazz@216.207.245.1) |
| 16:50.51 | hardwire | I don't know how much of the T1 signaling is handled without approval of the kernel. |
| 16:51.25 | Qwell | hardwire: There is support being added for service maintenance messages, which would do basically that |
| 16:51.33 | Qwell | take down a chan/span |
| 16:51.43 | hardwire | Qwell: but leave it configured? |
| 16:52.05 | hardwire | so asterisk will try to use a zap channel and it would get "congested" ? |
| 16:52.08 | Qwell | Deeewayne: is your branch in a state that it can be tested? |
| 16:52.20 | hardwire | Qwell: support from sangoma or digium? |
| 16:52.29 | Qwell | it's in libpri for the most part |
| 16:52.45 | Qwell | http://bugs.digium.com/view.php?id=3450 |
| 16:53.08 | hardwire | Qwell: I briefly looked at the kernel modules and all the MAINT related stuff had to do with loopbacks. |
| 16:53.32 | jeev | Fender, i got pretty far last night.. just reading and stuff |
| 16:53.46 | hardwire | I'd love to actively test out any libpri changes, I can't afford multiple isdnguard :) |
| 16:56.16 | Deeewayne | Qwell: the branch is definitely in a state for feedback but its not yet complete |
| 16:56.26 | Qwell | hardwire: there you have it |
| 16:56.53 | hardwire | Qwell: It doesn't seem like this change will modify how the card behaves when it's plugged in as PRI_CPE |
| 16:56.57 | Deeewayne | hardwire: please test and provide traces if you can |
| 16:57.27 | hardwire | like, I'm thinking you couldn't have both rx and tx shared between two cards in an HA situation. |
| 16:58.23 | hardwire | maybe it will, and I'm just not in the know enough (which is more than likely) |
| 16:58.34 | Deeewayne | hardwire: Asterisk will stop taking calls on a span if you use 'pri service disable channel <chan>' on all the b-chan's in said span |
| 16:58.48 | Qwell | Deeewayne: can you disable span too? |
| 16:59.03 | Qwell | I guess turning off a D wouldn't make sense, since you couldn't turn it back on |
| 16:59.31 | Deeewayne | I haven't been able to confirm disabling an entire span, but I have confirmed by disabling individual b-chans |
| 16:59.48 | hardwire | I'm not too interested in maintaining a true sync between HA systems and a PRI provider. an ISDNguard would work fine. |
| 17:00.16 | hardwire | so thats why I'm curious if tdm cards can simply set themselves to red and do nothing until otherwise instructed. |
| 17:00.28 | hardwire | allowing another host to go green sharing the same conductors. |
| 17:01.19 | Deeewayne | hardwire: that is not the purpose of the issue3450 branches |
| 17:01.21 | *** join/#asterisk hsv-al (n=hsval@66.0.46.210) |
| 17:01.32 | hardwire | Deeewayne: thank you for establishing that :) |
| 17:01.57 | *** part/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 17:02.03 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 17:08.15 | *** join/#asterisk Strom_C (n=strom@208.127.172.112) |
| 17:15.23 | Nuitari | I have a weird problem with IAX and Asterisk 1.4.21 |
| 17:15.47 | Nuitari | sometimes the call just drops and then asterisk 1.4.21 hangs indefinitely when I do a "iax2 show channels" |
| 17:15.56 | Nuitari | the peer is an Asterisk 1.4.19.1 installation |
| 17:19.50 | *** join/#asterisk Cabel (n=abel408@64.128.120.92) |
| 17:21.33 | Cabel | Can anyone give me a list of supported phones for asterisknow? |
| 17:23.08 | hsv-al | cabel as soon as I can find a 100% stable , and native sip, and/or iax soft client |
| 17:23.10 | hsv-al | for the 3G iphone |
| 17:23.13 | hsv-al | im getting an iphone |
| 17:25.42 | Cabel | well I'm talking about business phones. Like polycom ect |
| 17:27.14 | Cabel | nevermind, I found my answer |
| 17:27.32 | [TK]D-Fender | Cabel: Yes, basically every phone anyone's used with * before. |
| 17:27.47 | [TK]D-Fender | Cabel: (insert 100 company names/models here" |
| 17:27.50 | Cabel | thanks! |
| 17:28.09 | [TK]D-Fender | Cabel: Wher we'd recommend is another matter. Where are you located? |
| 17:28.13 | [TK]D-Fender | what* |
| 17:28.21 | Cabel | NY |
| 17:28.32 | *** join/#asterisk trapa (n=2264581@207.230.238.94) |
| 17:29.01 | trapa | I'm having some trouble using AGI, is anyone here particularly fluent in agi scripting |
| 17:29.03 | [TK]D-Fender | Cabel: Polycom is probably the best choice for you then. www.telephonydepot.com |
| 17:29.25 | [TK]D-Fender | trapa: Pastebin your code & problem and we'll see if someone has your specific answer. |
| 17:29.27 | Cabel | great, thanks a lot! |
| 17:30.03 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 17:30.47 | *** join/#asterisk gr0mit (n=tim@81.187.67.134) |
| 17:31.08 | trapa | Fender: Working on pastbin-ing |
| 17:31.51 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 17:34.08 | trapa | So both my extensions.conf section , and the test php script are pasted there. THe problem is that we can't make the script seem to do anything. Asterisk says it's executing and returing 0 but nothing seems to be happening on the script front. |
| 17:34.09 | trapa | http://pastebin.com/d5c260c8d |
| 17:34.54 | [TK]D-Fender | trapa: LOL |
| 17:35.06 | [TK]D-Fender | trapa: look at lines 31 & 32 |
| 17:36.09 | *** part/#asterisk Dandre (n=daniel@82.233.173.121) |
| 17:36.24 | trapa | Okay ... Unfortunatly i don't see the error .. |
| 17:36.32 | [TK]D-Fender | trapa: And next time also include the CLI output of your attempt with AGI debug enabled |
| 17:36.43 | trapa | we also tried calling it AGI() and agi, |
| 17:36.44 | [TK]D-Fender | trapa: Does 500 look like 502 to you? |
| 17:37.01 | trapa | Oh .. bugger ... thats NOT in the actual extensions tho |
| 17:37.06 | [TK]D-Fender | trapa: Yuo don't seem to have a 502 priority 1 |
| 17:37.20 | trapa | I was hand typing it because it's not the same machine and it was easier than copying the file over hte network |
| 17:37.21 | [TK]D-Fender | trapa: Show us real stuff or you'll be wasting our time. |
| 17:37.39 | trapa | No looking at the file it is 502 on all of them. |
| 17:37.45 | [TK]D-Fender | trapa: copy/paste <- it ain't Raw-Cat Science |
| 17:38.17 | [TK]D-Fender | trapa: So now include the other bits, as well as "ls -la" for your file including the FULL path, etc |
| 17:38.30 | trapa | Sure. |
| 17:39.25 | *** join/#asterisk angom (n=angom@201.170.65.143) |
| 17:39.25 | trapa | You metnioned agi debug enabled .... is that the "set debug 10" command? |
| 17:39.36 | trapa | or is there a special agi debug command? |
| 17:39.41 | seanbright | agi debug |
| 17:40.25 | trapa | thx |
| 17:41.20 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 17:42.37 | *** part/#asterisk Nuitari (n=Nuitari@cybernet.nuitari.net) |
| 17:43.21 | trapa | http://pastebin.com/d2cb19c5a |
| 17:43.23 | *** join/#asterisk drehlecom (i=ircbnc@unaffiliated/drehlecom) |
| 17:47.02 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 17:47.26 | seanbright | trapa: you need to consume the stuff from stdin before you can send the commands to stdout |
| 17:48.36 | *** join/#asterisk pikachu2000 (n=pikachu2@196-209-198-124-rrdg-esr-2.dynamic.isadsl.co.za) |
| 17:48.47 | trapa | I'm not sure i'm understanding .. ... Sorry. Can you explain in a bit more detail? |
| 17:48.55 | seanbright | http://asterisk.name/asterisk/0596009623/asterisk-chp-9-sect-3.html |
| 17:49.14 | seanbright | the part that starts "This section of code ensures that we have open file handles for..." |
| 17:49.24 | jeev | ahh, time to reconfigure the polycom :( |
| 17:49.33 | seanbright | grab that code, and put it into your AGI script right above the call to fwrite |
| 17:49.48 | trapa | Okay .. lemme go re-read and try stuff ... and try again. |
| 17:50.00 | seanbright | its as simple as copy and pasting the code i just referenced |
| 17:50.52 | trapa | It can't be because we did copy and paste that code ... and it didn't ever do anything .. so then we reduced the code to just this, just to see if we could make it stream one thing ... Clearly we missed somthing .. But our first test was copying the sample code from the good book. |
| 17:51.37 | seanbright | trapa: give this a shot... |
| 17:51.41 | seanbright | cd agi-bin/ |
| 17:51.51 | seanbright | ./plain_test.php |
| 17:51.56 | seanbright | ^D |
| 17:52.08 | seanbright | (where ^D is control-D) |
| 17:52.15 | trapa | okay. |
| 17:52.54 | trapa | Didn't even stop ... |
| 17:52.59 | trapa | asterisk:/usr/share/asterisk# cd agi-bin |
| 17:52.59 | trapa | asterisk:/usr/share/asterisk/agi-bin# ./plain_test.php |
| 17:52.59 | trapa | asterisk:/usr/share/asterisk/agi-bin# logout |
| 17:52.59 | trapa | trapa@asterisk:~$ |
| 17:53.13 | seanbright | so it's a script error |
| 17:53.40 | seanbright | oh... do you have a ?> at the end of your script? |
| 17:54.52 | trapa | No. |
| 17:54.56 | seanbright | :) |
| 17:54.58 | trapa | Hrmm ... intersting |
| 17:55.04 | seanbright | bets a $1 |
| 17:56.05 | seanbright | add a ?> to the end and give it another whirle |
| 17:56.10 | seanbright | s/whirle/whirl/ |
| 17:56.19 | *** join/#asterisk msetim (n=msetim@200.195.161.164) |
| 17:57.35 | trapa | nodnods. Yeah workin on it. |
| 17:57.53 | trapa | Why ^D btw? That seems to be a shell "logout comand" |
| 17:57.57 | *** join/#asterisk implicit_ (n=implicit@dhcp-x223-230.mobile.uci.edu) |
| 17:57.58 | *** join/#asterisk maagic (i=maagic@fsck.fi) |
| 17:57.59 | seanbright | trapa: its EOF |
| 17:58.19 | seanbright | trapa: and that was the wrong thing to tell you, since you aren't reading anyway, so the script wouldn't block |
| 17:59.18 | *** join/#asterisk DarKnesS_WolF (n=sherif@unaffiliated/sherif) |
| 18:01.05 | trapa | No effect when adding the ?> .,.. we're workin on comsuming the STDIN |
| 18:01.26 | seanbright | what happens if you run it from the command line after adding the ?>? |
| 18:02.16 | seanbright | sigh... |
| 18:02.26 | seanbright | why does copying and pasting take so long? |
| 18:02.32 | seanbright | goes back to work |
| 18:03.54 | *** join/#asterisk smach (n=smach@207.35.173.122) |
| 18:03.59 | *** join/#asterisk Rico29 (n=Rico@nor75-29-88-188-75-101.fbx.proxad.net) |
| 18:04.32 | trapa | sorry was in another window |
| 18:04.44 | trapa | It resulted in the exact same as before |
| 18:04.45 | smach | hi guys, I'm wondering why my cisco 7960 doesn't subscribe to MWI ? anyone played with that ? |
| 18:05.18 | trapa | asterisk:/usr/share/asterisk/agi-bin# ./plain_test.php |
| 18:05.18 | trapa | asterisk:/usr/share/asterisk/agi-bin# |
| 18:09.22 | [TK]D-Fender | smach: pastebin your sip peer masking only passwords |
| 18:09.38 | [TK]D-Fender | trapa: add a raw "echo" in there. |
| 18:09.45 | [TK]D-Fender | trapa: as a sanity check |
| 18:10.30 | trapa | okay |
| 18:10.40 | seanbright | and comment out the "error_reporting(0);" b.s. |
| 18:11.20 | trapa | Interesting ... No echo result :) |
| 18:11.43 | seanbright | did you comment out the error_reporting line? |
| 18:12.00 | trapa | Yes i have now. |
| 18:12.06 | trapa | But still no echo result. |
| 18:12.13 | seanbright | wow |
| 18:12.36 | trapa | So .. Hrm .. |
| 18:12.45 | seanbright | try this |
| 18:12.57 | seanbright | http://pastebin.com/d41cb6b7c |
| 18:13.00 | trapa | A hello world on either side of these three lines .. Only the first hello world shows. |
| 18:13.02 | trapa | # create file handles if needed |
| 18:13.02 | trapa | define('STDIN', fopen('php://stdin', 'r')); |
| 18:13.03 | trapa | define('STDOUT', fopen('php://stdout', 'w')); |
| 18:13.03 | trapa | define('STDERR', fopen('php://stderr', 'w')); |
| 18:13.03 | smach | [TK]D-Fender: isn't something that has to do with the set. that starts by sending a subscribe to the server ? |
| 18:13.30 | *** join/#asterisk mikealeonetti (n=mikel@static-72-68-153-122.nycmny.fios.verizon.net) |
| 18:13.54 | mikealeonetti | is a TDM2400P something that is for use with T1 phone lines? |
| 18:13.59 | [TK]D-Fender | trapa: See you removed the SAFETY checks in seanbright's link that prevents you from defining something that is already defined. |
| 18:14.02 | mikealeonetti | or something like it. |
| 18:14.13 | [TK]D-Fender | trapa: and I'm betting at least one of those does already. |
| 18:14.14 | seanbright | [TK]D-Fender: nice catch |
| 18:14.46 | [TK]D-Fender | ~[TK]D-Fender |
| 18:14.47 | jbot | [TK]D-Fender is the Zen Master of the blatantly obvious, and #asterisk 's resident Google-proxy. |
| 18:14.49 | [TK]D-Fender | ^^^ |
| 18:15.02 | [TK]D-Fender | mikealeonetti: No. TDM2400 is for analog lines. |
| 18:15.39 | mikealeonetti | [TK]D-Fender: hrm, is there a way to connect hardware wise to T1 lines or do I have to go to analog first? |
| 18:15.53 | trapa | OKay we'll add the safty checks back in ... Seanbright: That script works just fine. |
| 18:16.09 | seanbright | trapa: good. add the safety checks as suggested and report back the results. |
| 18:16.15 | [TK]D-Fender | mikealeonetti: Buy a proper card. Digium and Sangoma both have an extensive lineup |
| 18:16.32 | *** join/#asterisk anarcat (n=anarcat@pdpc/supporter/active/anarcat) |
| 18:17.04 | anarcat | any good suggestions for an asterisk-supported ATA (FXS) out there? |
| 18:17.08 | *** join/#asterisk Whitor (n=Whitor@rrcs-24-97-4-150.nys.biz.rr.com) |
| 18:18.01 | mikealeonetti | [TK]D-Fender: what might they be called? |
| 18:18.28 | [TK]D-Fender | anarcat: Linksys PAP2 on a budget, 2102 if you can spare the extra 10$, Mediatrix-2000 series if you need something high-end |
| 18:18.51 | [TK]D-Fender | mikealeonetti: www.digium.com <- if you can't find them in 30 seconds or less, then something is very wrong. |
| 18:19.39 | anarcat | [TK]D-Fender: thanks |
| 18:20.33 | *** join/#asterisk SteveTotaro (n=Administ@pool-151-196-237-15.balt.east.verizon.net) |
| 18:21.02 | mikealeonetti | think something is wrong :D |
| 18:21.11 | seanbright | jeebus |
| 18:21.33 | trapa | joeseanbright: Okay so adding those checks got us further .. Now we need to insert a EOF (^D) and now when i do it says: STREAM FILE northernly "" |
| 18:21.39 | seanbright | mikealeonetti: http://www.digium.com/en/products/digital/ |
| 18:21.53 | seanbright | trapa: perfect. try it through asterisk again. |
| 18:22.51 | *** join/#asterisk snapple42 (n=snapple4@h216-18-80-132.gtconnect.net) |
| 18:23.00 | mikealeonetti | seanbright: thanks |
| 18:23.38 | seanbright | mikealeonetti: no problem... you'll need someone else to pay for them, ship them, install them, and configure them, as [TK]D-Fender and i are unavailable. |
| 18:23.59 | trapa | Okay ... in the agi debug ... we're now getting a added line of : AGI Rx << STREAM FILE northerly "" but still not hearing her saying "northernly" |
| 18:23.59 | mikealeonetti | seanbright: that's unfortunately 'cause it sounds like I'd have a lot of questions |
| 18:24.11 | [TK]D-Fender | mikealeonetti: My rates are very accessible :) |
| 18:24.12 | seanbright | mikealeonetti: well you're in the right place. |
| 18:24.14 | seanbright | heh |
| 18:24.47 | mikealeonetti | so basically I plug my T1 device into here and I can set up separate lines through the config? |
| 18:25.03 | [TK]D-Fender | mikealeonetti: Separate lines? |
| 18:25.09 | mikealeonetti | phone numbers |
| 18:25.22 | seanbright | mikealeonetti: something like that, yes. |
| 18:25.27 | [TK]D-Fender | indeed |
| 18:25.32 | mikealeonetti | oh, what am I getting wrong? |
| 18:25.39 | seanbright | mikealeonetti: just some terminology |
| 18:25.47 | mikealeonetti | where? |
| 18:25.58 | [TK]D-Fender | mikealeonetti: T1 carries multiple channels. Not entirely appropriat to call those "lines" |
| 18:26.05 | seanbright | ^^^ |
| 18:26.23 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 18:26.54 | [TK]D-Fender | mikealeonetti: In PRI, DIDs are sent to your PRI circuit over the next available channel. DID's are "numbers". Your PRI would just be a channel interface for your telco to send you calls which target DID you pay for through. |
| 18:27.14 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 18:27.24 | [TK]D-Fender | mikealeonetti: In CAS mode, things will be a bit different. |
| 18:28.03 | x86 | CAS mode always uses the same channels for the different DIDs |
| 18:28.12 | x86 | or group of channels, if that's your case |
| 18:28.19 | x86 | (inbound hunt, etc) |
| 18:28.37 | mikealeonetti | <3 |
| 18:29.44 | [TK]D-Fender | mikealeonetti: x86 typically anyways... you CAN get DTMF based DID on CAS, etc (analog even), but it is rare |
| 18:30.08 | trapa | http://pastebin.com/d1180f3d so i'm getting agi rx << STREAM FILE .. but no streaming is happening. |
| 18:30.14 | [TK]D-Fender | mikealeonetti: Suffice it to say you DO want PRI if at all possible |
| 18:30.45 | mikealeonetti | [TK]D-Fender: is that something I request of the company? |
| 18:30.55 | *** part/#asterisk anarcat (n=anarcat@pdpc/supporter/active/anarcat) |
| 18:30.58 | [TK]D-Fender | trapa: And you are not clearing the respone to your call either |
| 18:31.04 | [TK]D-Fender | mikealeonetti: Yes. |
| 18:31.08 | mikealeonetti | oh word |
| 18:31.19 | [TK]D-Fender | mikealeonetti: You would be well served to give this a read : |
| 18:31.21 | [TK]D-Fender | ~101 |
| 18:31.21 | jbot | from memory, 101 is Telephony 101, which is a good read if you're unfamiliar with traditional TDM telephony. You can download it at http://www.stromcarlson.com/docs/basics/NTtelephony101.pdf |
| 18:31.41 | trapa | How do you mean clearing the response from your call? Should it be returining 1? Or somthing? |
| 18:31.48 | seanbright | ... |
| 18:31.50 | mikealeonetti | [TK]D-Fender: thanks |
| 18:31.58 | [TK]D-Fender | trapa: Go read the chapter on AGI all over again |
| 18:32.00 | mikealeonetti | I wish I didn't hate reading |
| 18:32.12 | [TK]D-Fender | mikealeonetti: Thats what consultants are for :) |
| 18:32.29 | mikealeonetti | everything costs all this money |
| 18:32.29 | seanbright | trapa: ls -al /var/lib/asterisk/sounds/northerly* |
| 18:34.38 | trapa | asterisk:/usr/share/asterisk/agi-bin# ls -al /usr/share/asterisk/sounds/northerly* |
| 18:34.38 | trapa | -rw-r--r-- 1 root root 15650 2008-06-20 17:13 /usr/share/asterisk/sounds/northerly.sln |
| 18:34.51 | seanbright | /usr/share... pfft |
| 18:34.53 | seanbright | heh |
| 18:35.10 | jeev | Fender! |
| 18:35.10 | trapa | Mine aren't in /var/lib/asterisk/sounds/ .... BUt doing a exten=> Playback(file) still works .. should i move them? Will they not work with agi calls from where they are? |
| 18:35.12 | ManxPower | trapa: sounds like you are using a package and did not compil from source. |
| 18:35.21 | seanbright | trapa: no, don't move them. |
| 18:35.31 | seanbright | trapa: freepbx? trixbox? |
| 18:35.35 | trapa | manxPower: Yeah .. I'm using debian... apt-get install baby .... But is this a issue? |
| 18:35.45 | trapa | apt-get install asterisk .... |
| 18:35.52 | seanbright | damn |
| 18:35.55 | *** join/#asterisk lanning (n=lanning@ip67-152-85-190.z85-152-67.customer.algx.net) |
| 18:35.58 | ManxPower | trapa: yes, as we don't where the packages put important files |
| 18:35.59 | seanbright | was hoping to just ~trixbox you |
| 18:36.52 | *** join/#asterisk SteveTotaro (n=Administ@pool-151-196-237-15.balt.east.verizon.net) |
| 18:37.00 | trapa | Sorry i can't, I can't dedicate this machine to just asterisk (Tho it's beefy enough to be used for asterisk and web pages) .. because the agi script we're trying to eventually get to making will add caller id info to a website amonst other things ... So a asterisk package wasn't for us ... |
| 18:37.10 | seanbright | trapa: after your fflush, add this: |
| 18:37.11 | seanbright | fgets(STDIN,4096) |
| 18:37.14 | seanbright | err |
| 18:37.15 | seanbright | fgets(STDIN,4096); |
| 18:37.16 | trapa | manxPower: Well ... maybe i can find importan files. |
| 18:37.35 | adeel | trapa, you should be able to list the files in a package with apt-get |
| 18:37.49 | seanbright | we aren't concerned about file locations right now |
| 18:37.53 | [TK]D-Fender | trapa: just look at asterisk.conf |
| 18:38.19 | seanbright | Playback(foo) is working, so /usr/share it is |
| 18:38.36 | trapa | seanbright: Should i execute that through asterisk and watch it in console agi debug, because at the command line it dind't return anything. |
| 18:38.41 | [TK]D-Fender | trapa: And do follow seanbright's advise. Additionally, you don't seem to be the most competant at PHP. Perhaps you should use php-agi wrapper for this |
| 18:38.47 | seanbright | trapa: yes, through asterisk |
| 18:38.53 | seanbright | trapa: and pastebin the output |
| 18:39.37 | trapa | fender: I'm the network admin, while i'm not super at php, the bigger issue is that i'm not the prorgammer .. the programmer isn't a system admin or asterisk administrator .... so i'm trying to do his job too all the while trying to help him do his attempts at fixing it ... |
| 18:39.59 | seanbright | trapa: can't you just point him at the AGI documentation and be done with it? |
| 18:40.43 | trapa | seanbright: I did ... he has the book, he's said he's read chapert nine several times .. |
| 18:41.02 | seanbright | trapa: might be time to look for new developers... but that is neither here nor there. |
| 18:41.10 | seanbright | trapa: add the fgets, and re-run |
| 18:41.21 | trapa | seanbright: also .. it's not like i can accuse him of being a iditot or not reading it .. I'm sorta not in a position of power here ... Working on pasbin fgets now. |
| 18:41.22 | [TK]D-Fender | trapa: Your cuurrent approach doesn't seem to be working too well for you. Yuo should rethink it. |
| 18:41.52 | seanbright | it's called the "pay 1 guy, but have a half dozen other people do his job for free" approach |
| 18:42.25 | seanbright | actually, that should just be the #asterisk motto |
| 18:42.45 | seanbright | i really should be pulling in 7 figures a year |
| 18:43.19 | adeel | honestly though, how would they ever know that asterisk wasn't installed from a package? i doubt they go around and monitor the files....as long as things work, i doubt they'll care |
| 18:44.05 | [TK]D-Fender | seanbright: You do... you need to work on moving the decimal point however ;) |
| 18:44.16 | trapa | http://pastebin.com/d27b5c2ee |
| 18:44.19 | seanbright | [TK]D-Fender: too true |
| 18:44.38 | trapa | That made it work .. .. when it had fgets .. it streamed the northerly file |
| 18:44.45 | seanbright | bows |
| 18:45.49 | trapa | why did that make it work |
| 18:45.53 | seanbright | trapa: [TK]D-Fender has a point, though. there are a couple PHP based libraries for AGI already built. your dev should look into them instead of reinventing the wheel. |
| 18:46.13 | ManxPower | trapa: the various AGI libraries should handle all the flushing for you |
| 18:47.20 | [TK]D-Fender | trapa: Use the library's procedures for all of this. You have enough issues, let it take care of the nitty-gritty so you can get down to work. |
| 18:47.51 | seanbright | and when all else fails, hire [TK]D-Fender |
| 18:47.59 | seanbright | and pay me a finder's fee |
| 18:48.06 | seanbright | i'm kinda like his pimp |
| 18:48.11 | ManxPower | isn't that the Fender Fee? |
| 18:48.26 | seanbright | very nice |
| 18:48.38 | ManxPower | seanbright: to be a pimp, don't people actually have to pay? |
| 18:49.04 | ManxPower | I seem to get about 1 in 500 people that I help sending me money |
| 18:49.10 | hsv-al | lulz |
| 18:49.28 | seanbright | ManxPower: i've only helped a half dozen people, so it'll be a while before it pays off. |
| 18:49.40 | hsv-al | manxpower, just study xhtml/php/mysql for a month |
| 18:49.47 | seanbright | [TK]D-Fender on the other hand should be rich |
| 18:49.52 | hsv-al | and make an asterisk consulting website , and charge people $$ for remote webex work :) |
| 18:49.58 | trapa | well without getting too much into things ... basically person A makes a software package .. the software package was built and intevnted on machine A .. in some foregin country ..... Now company wants copies of porgram A ... but program A isn't anywhere near ready to release and perosn A refuses to just give copy of said program .. he gives source code .. but of course it doesn't all work .. so now person B and myself re tasksed to making program |
| 18:50.27 | [TK]D-Fender | trapa: You've got a point buried deep in there somewhere, right? |
| 18:50.52 | seanbright | trapa: all we are saying - your current way isn't too efficient - find a better way. |
| 18:51.09 | hsv-al | what the |
| 18:51.17 | trapa | will work on it. |
| 18:51.27 | hsv-al | person A of the person b, of the person c, semantics of the a to the b raised to the power c |
| 18:51.34 | [TK]D-Fender | trapa: Your programmer isn't programming, and you aren't a programmer. This is nto a good combination. |
| 18:51.41 | trapa | But i don't get to make all the real decisions on how this crap works .. I'm just suppose to make what we have work |
| 18:51.48 | seanbright | without getting too involved... train A leaves moscow at 6:00 GMT heading west... |
| 18:52.04 | hsv-al | trapa, read: Four hour workweed by timothy ferris |
| 18:52.10 | hsv-al | and outsource all the work to VA's |
| 18:52.12 | hsv-al | virtual assistants |
| 18:52.18 | hsv-al | workweek |
| 19:00.52 | *** join/#asterisk quaqo (n=quaqo@83-103-40-166.ip.fastwebnet.it) |
| 19:02.33 | ManxPower | I liked "workweed" better. |
| 19:03.28 | outtolunc | likes the weedend better |
| 19:05.13 | mikealeonetti | that's a good alias |
| 19:05.18 | mikealeonetti | "workweed" |
| 19:05.20 | bkruse | ManxPower: Hello. |
| 19:06.07 | *** join/#asterisk robl^ (n=robl@pdpc/supporter/monthlybyte/robl) |
| 19:06.12 | robl^ | ~book |
| 19:06.13 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 19:07.18 | *** join/#asterisk gitguy (n=diego@adsl-142-230.click.com.py) |
| 19:08.21 | adeel | is there any downside to reducing the range of RTP ports usable? |
| 19:08.29 | adeel | by * that is |
| 19:10.45 | *** join/#asterisk troubled_ (n=troubled@pdpc/supporter/sustaining/troubled) |
| 19:18.35 | Cabel | I have asterisknow and I need some sort of call accounting utility to keep track of user calls. What is my best solution? |
| 19:18.45 | ManxPower | Cabel: #asteriskNOW |
| 19:19.04 | Cabel | well it doesn't have to be related to asterisknow |
| 19:19.04 | ManxPower | adeel: yes. You need two RTP ports per call. |
| 19:19.26 | ManxPower | Cabel: No, it does not, but we can't help you install it or integrate it with AsteriskNOW |
| 19:19.47 | ManxPower | Heck, we can't even recommend one that will work with AsteriskNOW |
| 19:20.17 | *** part/#asterisk mikealeonetti (n=mikel@static-72-68-153-122.nycmny.fios.verizon.net) |
| 19:20.33 | Cabel | well it doesn't have to install or integrate with asteriskNOW does it? |
| 19:21.00 | ManxPower | Cabel: Yes it does. |
| 19:21.02 | adeel | ManxPower, yeah i'm aware of the 2 ports per call...but since i'm not expect 5,000 calls at once, i don't think i need to open the 10000 ports defined in rtp.conf |
| 19:21.18 | ManxPower | adeel: clrrect |
| 19:21.21 | ManxPower | correct too. |
| 19:21.44 | adeel | ManxPower, i just wanted to make sure that if set the port range to 40 ports, that things wouldn't break |
| 19:21.45 | ManxPower | Cabel: for example, maybe AsteriskNOW uses AGIs or chan_local, does the accounting package support this CDRs? |
| 19:21.59 | Cabel | ManxPower, So I wouldn't be able to use Asterisk CDR? http://www.voip-info.org/wiki/view/Asterisk+CDR+Areski+GUI |
| 19:22.18 | Cabel | ok |
| 19:22.35 | ManxPower | Cabel: as we don't use AsteriskNOW here, how the hell would we know? |
| 19:22.52 | [TK]D-Fender | Cabel: Sure, that program will probably work, go try it |
| 19:23.09 | Cabel | thanks |
| 19:24.24 | bkruse | Cabel: #asterisk-gui |
| 19:24.27 | bkruse | Cabel: #asterisknow |
| 19:24.47 | bkruse | ManxPower: I am not sure if that is the asterisk gui anyways... :/ |
| 19:25.40 | anonymouz666 | how many * boxes for a solution with 10 E1s? How do you would distribute that? |
| 19:26.20 | bkruse | 10 e1s? |
| 19:26.30 | bkruse | You could do that with one beefy box....Ive done it before |
| 19:26.33 | jeev | Fender, this thing is capable of .. lets say, A calls in, wants to talk to B, we put A on hold, call B, conference them together.. |
| 19:26.44 | anonymouz666 | with one? |
| 19:26.48 | anonymouz666 | no no no. |
| 19:26.48 | ManxPower | anonymouz666: you KNOW BETTER than to ask that question! |
| 19:27.11 | lmadsen | jeev: that is usually a functionality of the phone |
| 19:27.14 | ManxPower | anonymouz666: You KNOW there are so many factors involved that asking that question just makes you look stupid. You've been here for MONTHS. |
| 19:27.23 | jeev | sup leif ericcson! |
| 19:27.29 | jeev | yea, polycom 330.. it has the conference button ;D |
| 19:27.29 | lmadsen | zup zup |
| 19:27.38 | jeev | looks like i got a lot of work ahead of me |
| 19:27.44 | bkruse | anonymouz666: you can do it with one |
| 19:27.56 | bkruse | keyword CAN |
| 19:27.57 | jeev | leif@asterisk# tar -zcvf asterisk.tgz /etc/asterisk |
| 19:27.59 | jeev | jacked your conf |
| 19:28.13 | *** join/#asterisk makkksimal (n=makkksim@e177214232.adsl.alicedsl.de) |
| 19:28.14 | ManxPower | bkruse: 10 T-1s with call recording and G729 on one box? |
| 19:28.24 | lmadsen | jeev: sweet, that local config sucks :) |
| 19:28.31 | jeev | bastard |
| 19:28.44 | jeev | how much time did it take into perfecting your favorite config? |
| 19:28.50 | bkruse | ManxPower: 12 t1s going to g729 sip termination.... |
| 19:29.07 | lmadsen | ummm... I don't have a favourite config I guess... took me 5 years of Asterisk full time to get to where I am if that helps any |
| 19:29.18 | jeev | hmm |
| 19:29.21 | bkruse | ManxPower: Again, plan for the worst, it IS possible. Heck I am 100% sure you can setup a 2950 in that config to do it... |
| 19:29.33 | jeev | i hope you've made at least a million bucks in that 5 years. |
| 19:29.44 | outtolunc | haha |
| 19:29.55 | lmadsen | lol... you've no idea how cheap people really are then :) |
| 19:30.10 | jeev | i do.. i was about to offer you $8/hr to help me with the config |
| 19:30.11 | jeev | but i guess now i can't! |
| 19:30.16 | jeev | hahaha i'm kidding |
| 19:30.29 | ManxPower | bkruse: my point is that 1) he did not provide enough information to receive an accurate answer 2) he's been here for months and months and unless he's a total idiot he already knows that scaling asterisk involved a lot of things to consider. |
| 19:30.40 | lmadsen | jeev: move the decimal place to the right one spot, then double it |
| 19:30.47 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
| 19:30.47 | ManxPower | jeev: $8/hr? That's insulting |
| 19:30.56 | lmadsen | minimum wage is $10/hr here |
| 19:30.58 | jeev | haha i'm just messin with him |
| 19:31.06 | *** kick/#asterisk [jeev!n=Leif@asterisk/documenteur-extraordinaire/blitzrage] by lmadsen (just messing with ya!) |
| 19:31.06 | *** join/#asterisk jeev (i=jeev@naptime.net) |
| 19:31.14 | jeev | bastage |
| 19:31.40 | lmadsen | twiddles his thumbs while waiting for ABE ISO to upload to clients server... |
| 19:31.43 | outtolunc | aim in.. unlock <G> |
| 19:31.53 | lmadsen | box 1! |
| 19:32.10 | jeev | lmadsen, i'm waiting for the girl to test it and see if she has any static |
| 19:32.17 | lmadsen | ok |
| 19:34.24 | *** join/#asterisk makkksimal (n=makkksim@e177214232.adsl.alicedsl.de) |
| 19:35.30 | *** join/#asterisk arcanine (n=saxon_m2@202.138.162.67) |
| 19:39.33 | jeev | lmadsen: i'm curious, what do you usually charge in total, you can't do it by the hour.. do you? to set up like a 10-20 person office |
| 19:39.49 | lmadsen | per hour |
| 19:39.56 | jeev | really |
| 19:40.06 | jeev | avg of how many hours |
| 19:40.08 | lmadsen | yes, I'm a consultant, it's always per hour unless someone pays a large retainer |
| 19:40.14 | jeev | ah cool |
| 19:40.19 | lmadsen | depends... I don't tend to build 10-20 person offices |
| 19:40.25 | jeev | what's the biggest you've done |
| 19:40.30 | Qwell | lmadsen: what about 0-9? |
| 19:40.33 | jeev | lol |
| 19:40.35 | lmadsen | Qwell: 1 hr! |
| 19:40.41 | ManxPower | jeev: you mean "what is the smallest you've done" |
| 19:40.44 | lmadsen | depends on the features required actually |
| 19:40.47 | jeev | what's the biggest office you've done |
| 19:40.50 | jeev | why smallest! |
| 19:40.57 | jeev | the smallest is his home probably |
| 19:41.00 | lmadsen | I've build a clustered, DB driven system for an ITSP |
| 19:41.10 | lmadsen | s/build/built |
| 19:41.38 | jeev | hmm |
| 19:41.45 | lmadsen | currently working on a multi-site, distributed, high availability, DB driven system |
| 19:41.50 | lmadsen | for a call centre |
| 19:41.57 | lmadsen | I do a lot of queue stuff now |
| 19:42.01 | Qwell | ie; his house |
| 19:42.03 | jeev | what do you use to make sure it's redundant.. for the DID |
| 19:42.04 | jeev | hahaa |
| 19:42.17 | lmadsen | jeev: asterisk tools such as DUNDi and func_odbc |
| 19:42.29 | jeev | so would that mean you're your own DID provider? |
| 19:42.33 | ManxPower | jeev: Because if you want him to install a 5 person system and he wants to use Polycoms, a Cisco switch, PoE, Asterisk and a PRI you would want to reconsider hiring him. |
| 19:42.41 | lmadsen | jeev: it means a client of mine is a DID provider |
| 19:42.48 | jeev | haha manx |
| 19:42.59 | lmadsen | I don't do small projects anymore |
| 19:43.03 | jeev | ok so |
| 19:43.07 | ManxPower | jeev: That's how *I* would build a 2 person system or a 200 person system. |
| 19:43.07 | Qwell | ManxPower: Cisco switch would be the killer. |
| 19:43.17 | jeev | i dont like the thought of PoE |
| 19:43.37 | ManxPower | jeev: Not I. It just adds to the cost and is not always needed. |
| 19:43.37 | unpaidbill | poe is awesome |
| 19:43.41 | jeev | lmadsen: so if i get my provider who does everything and i connect to them via my asterisk system, what could i do to ensure that if they drop, i dont. |
| 19:43.53 | ManxPower | jeev: you can't. |
| 19:43.57 | jeev | poe = rewire everything.. and i aint doin that! |
| 19:43.59 | unpaidbill | what isnt to like about it.. easier to handle power outages |
| 19:44.06 | Qwell | jeev: umm, no |
| 19:44.07 | unpaidbill | no power adaptors taking up space |
| 19:44.12 | Qwell | poe uses cat5e |
| 19:44.13 | jeev | true unpaid |
| 19:44.14 | ManxPower | jeev: PoE means NOT REWIRING EVERYTHING! |
| 19:44.27 | jeev | hmm |
| 19:44.33 | jeev | but i can't have regular data go over PoE could i? |
| 19:44.33 | unpaidbill | it means switching cabling on your patch panel! :( |
| 19:44.37 | jeev | yea |
| 19:44.39 | lmadsen | I'm going to be installing a small PoE switch at my new condo so I can get rid of half of my cables |
| 19:44.39 | Qwell | unpaidbill: no it doesn't |
| 19:44.40 | unpaidbill | yes you can jeev |
| 19:44.56 | jeev | i didn't set up the network here so it'd be a mess to do it |
| 19:44.57 | unpaidbill | qwell well you ahve to move the rj45 from your normal switch to the poe switch |
| 19:44.59 | ManxPower | jeev: the POINT of PoE is to run power and data in the same network cable. |
| 19:45.01 | unpaidbill | i did at least |
| 19:45.07 | jeev | yea i know that manx.. |
| 19:45.12 | unpaidbill | otherwise it would have been on the non poe switch! |
| 19:45.28 | jeev | i do hate the thought of having a power adapter for each phone.. |
| 19:45.57 | ManxPower | I like a compromise solution, some phones on PoE, some not. |
| 19:46.05 | unpaidbill | jeev i have mine set up like this... vlan30 = data, vlan50 = voice crap.. the cisco phones have a switch port that the computers plug into, which go on the default 30 vlan, everything works through one cable at the desk |
| 19:46.11 | Qwell | poe is nice - when we had power issues here recently, our phones were still working, since the switch was on the generator/ups |
| 19:46.23 | unpaidbill | and qos handles the network traffic so i always have sufficient bw for voice |
| 19:46.43 | jeev | hmmmm |
| 19:46.47 | jeev | i do dual wan |
| 19:46.52 | unpaidbill | also, my poe switch is plugged into a ups with the phone system, so everything stays up during a power outage |
| 19:47.05 | unpaidbill | rather than having a ups at every desk for each phone |
| 19:47.08 | unpaidbill | which is insanity |
| 19:47.11 | Qwell | ups really isn't enough.. |
| 19:47.20 | unpaidbill | i have only 25 phones |
| 19:47.23 | Qwell | there would be a fairly significant draw from the phones |
| 19:47.36 | jeev | how much could the phones draw? |
| 19:47.42 | unpaidbill | i can get 20m of uptime |
| 19:47.45 | Qwell | a few watts adds up |
| 19:47.52 | ManxPower | what do most phones draw? 7 watts? 14 watts? |
| 19:47.55 | jeev | if you dedicate a 180 watt UPS? |
| 19:47.56 | jeev | heh |
| 19:47.57 | unpaidbill | 20W 7960 |
| 19:47.58 | Qwell | ManxPower: something like that |
| 19:48.02 | unpaidbill | something crazy |
| 19:48.12 | Qwell | but, lets say 14W * 25 phones |
| 19:48.14 | jeev | ok so what do you guys suggest, i run PoE at the new office? |
| 19:48.25 | Qwell | that's 350W... |
| 19:48.26 | unpaidbill | oh srory, 15.4W |
| 19:48.28 | jeev | i think the company that was in there has cat5 everywhere, so it doesn't even go through the walls |
| 19:48.29 | ManxPower | jeev: No, install it, don't run it. |
| 19:48.31 | unpaidbill | per phone, 25 phones |
| 19:48.38 | ManxPower | running it would be silly. |
| 19:48.48 | jeev | so i could pretty much plug the data portion into the phone |
| 19:48.53 | jeev | the phone into the jack.. |
| 19:48.57 | jeev | and the PoE wouldn't ruin the pc? |
| 19:49.00 | ManxPower | unplug cat 5 cable from switch, plug it into the PoE all done. |
| 19:49.00 | unpaidbill | no |
| 19:49.07 | unpaidbill | jeev you can plug a pc into a poe port, it wont blow it up |
| 19:49.16 | ManxPower | jeev: Why don't you stop asking questions and start going and reading a few things. |
| 19:49.20 | unpaidbill | i wouldnt recommend it, but i have done it a few times accidentally without problems. |
| 19:49.25 | Qwell | unless you patch it to 110v |
| 19:49.27 | jeev | thge polycom 330 has the lan port and then a pc port.. |
| 19:49.29 | Qwell | then it might blow up |
| 19:49.31 | ManxPower | unpaidbill: that does not apply to the really cheap dumb PoE injectoes |
| 19:49.44 | unpaidbill | well im using a fastiron poe switch, it seems to detect things OK |
| 19:49.52 | unpaidbill | i dont know about cheap dumb things :) |
| 19:50.08 | jeev | so the polycom passes the power through it's lan port to the PC if it were being used? |
| 19:50.18 | unpaidbill | assuming it has a lan port, yes |
| 19:50.20 | ManxPower | unpaidbill: the dumb ones just put power on the correct pair(s), they don't do any PoE detection to see if the device can handle it. |
| 19:50.23 | Qwell | jeev: no |
| 19:50.26 | jeev | heh |
| 19:50.28 | unpaidbill | manx yea that would be bad |
| 19:50.35 | ManxPower | that said, the pairs with power on them should not be connected inside an ethernet device either |
| 19:50.40 | Qwell | it's a switch. there's no passthrough there |
| 19:50.49 | jeev | so it wont pass the power, cool |
| 19:50.55 | jeev | so it's safe to do, i guess i should get me a PoE switch? |
| 19:50.56 | Qwell | no, that would be dumb |
| 19:50.59 | unpaidbill | oh pass power, i read that as data somehow wtf |
| 19:51.28 | unpaidbill | i have an old cisco poe switch i might be willing to sell... |
| 19:51.34 | unpaidbill | 24 port! |
| 19:51.40 | ManxPower | We use PoE injectors (smart, not dumb), as the switch we have standardized on does not support PoE |
| 19:52.28 | unpaidbill | how does that work manx, is it a patch panel type thing, or a separate device on each run? |
| 19:52.54 | jeev | i dont get it, one person says it's stupid, one person says it's good |
| 19:52.55 | ManxPower | unpaid, it looks like a switch, two rows of ports, one row goes to the devices, the other to the switch |
| 19:52.59 | unpaidbill | ahh, nice |
| 19:53.05 | Qwell | jeev: no, I was saying passing power through would be stupid |
| 19:53.09 | jeev | oh |
| 19:53.14 | jeev | yea it would |
| 19:53.20 | jeev | so it's 100% safe and a good method to use |
| 19:53.21 | jeev | ? |
| 19:53.50 | unpaidbill | nah, all these companies just roll dice and hope they dont get sued for blowing people/pcs up |
| 19:53.51 | jeev | the only issue here is, i dont want to mangle the data and voice together with the dual wan system.. |
| 19:53.54 | ManxPower | jeev: The question is NOT is it safe, does it work? The question is will you pay the extra money for it? |
| 19:54.10 | jeev | yea,l why wouldn't i pay 400-500 bux for a PoE switch |
| 19:54.11 | unpaidbill | jeev that's what vlans are for |
| 19:54.22 | jeev | i realize that, but wont it cause NAT issues? |
| 19:54.24 | ManxPower | jeev: then you will not get PoE |
| 19:54.30 | jeev | i guess not, i'll just use pfsense then |
| 19:54.36 | unpaidbill | it wont cause any issues if you configure it correctly |
| 19:54.36 | jeev | why wont i get PoE ? |
| 19:54.42 | ManxPower | jeev: because you are too cheap. |
| 19:54.44 | jeev | ye i got a better understanding.. |
| 19:54.47 | jeev | who said i'm too cheap? |
| 19:54.49 | *** join/#asterisk gardo (n=gardo@121.97.195.215) |
| 19:54.49 | jeev | how do you assume this |
| 19:54.55 | unpaidbill | 48 port POE is gonna run you 3000 |
| 19:55.01 | ManxPower | I'm sure prices have come down, but we normally budget about $100 per PoE poer. |
| 19:55.03 | unpaidbill | maybe less if you have some kind of deal |
| 19:55.08 | jeev | hmm |
| 19:55.27 | ManxPower | so $2,400 for 24 ports. |
| 19:55.42 | jeev | what brand |
| 19:55.46 | ManxPower | Heck, even at $50/port (which you might be able to find) still adds up. |
| 19:56.03 | unpaidbill | i spent 3200 after shipping for my 4802-poe (48 port) |
| 19:56.07 | unpaidbill | not too bad |
| 19:56.20 | jeev | you're talking managed.. |
| 19:56.24 | unpaidbill | yes |
| 19:56.39 | unpaidbill | i dont think they make unmanaged poe switches? unless they're total crap |
| 19:56.41 | ManxPower | jeev: you can't do vlans on unmanaged switches |
| 19:56.42 | jeev | why would i need a managed switch |
| 19:56.47 | jeev | i'm not gonna do vlan's |
| 19:56.51 | jeev | i'll do source based routing. |
| 19:57.00 | jeev | source to destination |
| 19:57.12 | jeev | i'll set DHCP for the mac addresses for a specific block, and have it route in and out the proper interface |
| 19:57.54 | jeev | is that not a smart way |
| 19:57.55 | jeev | ? |
| 19:57.55 | ManxPower | jeev: http://www.voipsupply.com/index.php?cPath=271 PoE proces |
| 19:58.12 | *** join/#asterisk makkksimal (n=makkksim@e177214232.adsl.alicedsl.de) |
| 19:59.01 | ManxPower | I don't run voice over internet |
| 19:59.21 | jeev | ahh |
| 19:59.23 | jeev | hmm |
| 19:59.30 | jeev | what do you do? maybe i could get a t1 in there |
| 19:59.45 | ManxPower | For PSTN we use PRI |
| 20:00.19 | ManxPower | jeev: How much is a T-1/PRI where you are? |
| 20:00.47 | jeev | t1 will cost probably 250/m |
| 20:01.19 | ManxPower | I would guess at least twice that |
| 20:01.23 | jeev | I think. |
| 20:01.27 | jeev | well i could get one from speakeasy i'm sure |
| 20:01.36 | jeev | you dont believe in using a cable and/or dsl? |
| 20:01.36 | ManxPower | Best of luck. |
| 20:01.52 | jeev | i've pretty much had full uptime on both for the past 2-3 years |
| 20:01.55 | jeev | since i do dual wan.. |
| 20:02.30 | bkruse | ManxPower: Your right. "Can I run the fast asterisk with lots of calls? What do I need?". I gotcha, good point |
| 20:03.14 | jeev | huh |
| 20:03.46 | Strom | jeev: PRI in los angeles is relatively cheap |
| 20:04.01 | Strom | engineer for stability |
| 20:04.10 | jeev | hmm |
| 20:04.35 | Strom | no one cares that you're saving $70 a month when the DSL goes down at peak hour |
| 20:04.44 | jeev | true |
| 20:04.54 | jeev | i get the PRI through att pretty mcuh eh since they're the provider |
| 20:05.05 | Strom | get bids from AT&T and some of the CLECs too |
| 20:05.13 | jeev | i need additional hardware in the system, no ? |
| 20:05.18 | Strom | T1 card, yes |
| 20:05.43 | jeev | what would be the price difference though, i mean ATT is charging like $1200/month right now for the 20 lines |
| 20:05.47 | jeev | and they wont budge |
| 20:06.04 | unpaidbill | haha holy shit |
| 20:06.05 | Strom | last client i ordered a PRI for was something like $350/month from AT&T |
| 20:06.19 | jeev | but what's that include? i dont really get how it works |
| 20:06.45 | unpaidbill | i could get a full t1 for voice and a full t1 for data for less than you pay for 20 lines :O |
| 20:06.48 | Strom | well, your pricing will depend on call traffic, expansion estimates, etc |
| 20:07.09 | Strom | presumably you know how to run the Erlang formulae to determine all the metrics you need to feed to AT&T for maximum savings... |
| 20:07.19 | unpaidbill | you're in LA jeev? |
| 20:07.22 | jeev | yea unpaid |
| 20:07.25 | Strom | as am I |
| 20:09.46 | unpaidbill | man they're realling beating your wallet up charging 1200 bucks for 20 lines :/ |
| 20:10.08 | jeev | with verizon at the other office |
| 20:10.11 | jeev | it's another 400-450 bux |
| 20:10.15 | jeev | and we're combining it |
| 20:10.22 | unpaidbill | ah |
| 20:10.45 | jeev | i figured and calculated, with SIP.. i'll be paying like $125/month :) |
| 20:11.24 | unpaidbill | sounds good as long as the internet doesnt go down! |
| 20:11.28 | jeev | yea |
| 20:11.30 | jeev | either way |
| 20:11.36 | jeev | if the internet is down, the business is halted |
| 20:12.17 | jeev | i think the best thing to do here is to get my cable and DSL, work on getting a t1 in there.. |
| 20:12.20 | jeev | hope to god my SIP provider doesn't die |
| 20:12.40 | Qwell | or <random switch in Texas> doesn't die |
| 20:12.41 | unpaidbill | who are you using for voice service? |
| 20:12.53 | jeev | not texas! ;D haha |
| 20:12.57 | jeev | yea, stupid randomness |
| 20:12.59 | jeev | theroute could suck |
| 20:13.19 | jeev | i've taken all that into consideration, the good thing is that there is basically no incoming calls OTHER than personal.. mostly outgoing. and they dont complain when ATT is tapping the lines outside |
| 20:14.15 | unpaidbill | well in atts defense, the govt told them to.... |
| 20:14.20 | unpaidbill | just kidding, they're bastards. |
| 20:14.41 | jeev | yea man they're a rip |
| 20:14.43 | jeev | i'll be back, thanks guys |
| 20:15.56 | unpaidbill | i must say, POE::Component::Client::Asterisk::Manager is pretty bitchin. |
| 20:16.19 | Qwell | POE::Component? |
| 20:17.07 | *** join/#asterisk gaetronik (n=gaetan@eficaz2.manquehue.net) |
| 20:17.08 | unpaidbill | yeah.. poe stands for..portable something something.. |
| 20:17.27 | unpaidbill | even driven asterisk manager client! |
| 20:17.30 | Qwell | POE - portable multitasking and networking framework for Perl |
| 20:17.36 | unpaidbill | yea that one. |
| 20:18.03 | unpaidbill | it's kind of nice, i havent found a better perl module for the manager interface |
| 20:18.17 | Qwell | Net::Telnet |
| 20:18.22 | unpaidbill | heh |
| 20:18.52 | unpaidbill | i used that too |
| 20:22.25 | *** join/#asterisk makkksimal (n=makkksim@e177214232.adsl.alicedsl.de) |
| 20:22.51 | *** join/#asterisk maurot (n=usuario@host144.201-252-138.telecom.net.ar) |
| 20:22.59 | maurot | i people |
| 20:23.11 | unpaidbill | i people too |
| 20:23.14 | maurot | i have a problem when recorder th speech |
| 20:23.23 | maurot | i listening defasing |
| 20:23.30 | maurot | defase |
| 20:23.42 | maurot | the audio |
| 20:24.14 | Strom | holy god, man |
| 20:24.30 | Strom | please try and assemble your thoughts into cogent sentences |
| 20:25.42 | jblack | <yoda>defasing the audio, bad it is!</yoda> |
| 20:26.01 | *** join/#asterisk FinboySlick (n=FinboySl@207.134.8.4) |
| 20:26.46 | fogo | maurot: dephasing you mean? As in out of sync or what? |
| 20:26.59 | echelon | do both clients have to have the same audio codecs to talk when doing direct ip to ip? |
| 20:27.52 | FinboySlick | 'lo gang. I'm trying to diagnose a pick-up problem and would like to validate this simple config with you. http://www.pastebin.ca/1054162 My problem is that channel 2 doesn't seem to notice any ringing. Otherwise it works fine. |
| 20:28.32 | maurot | fogo: yeap out of sync |
| 20:28.37 | [TK]D-Fender | echelon: Not when going through *. * will transcode from one to the other |
| 20:28.40 | unpaidbill | echelon: if you dont want asterisk to be transcoding for the duration of the call they need to be able to use the same codec |
| 20:29.13 | fogo | maurot: is this during a chanspy monitoring, or a mixmonitor recording or what? |
| 20:29.45 | jaytee | anyone have a link to a good howto on setting up BLF keys for extensions on a GXP-2000 and * 1.4 and using call pickup? |
| 20:30.27 | echelon | [TK]D-Fender: how do i increase the voice quality when it goes through *? |
| 20:31.06 | [TK]D-Fender | echelon: There is no such concept as "increase the quality". |
| 20:31.17 | echelon | ok.. increase the bitrate |
| 20:31.35 | echelon | what codec does asterisk use? |
| 20:31.36 | FinboySlick | echelon: Well, logic would suggest that you cannot get any more quality/information out as you have coming in. |
| 20:31.37 | maurot | fogo: i tried with monitor and soxmix and with mixmonitor |
| 20:31.40 | [TK]D-Fender | echelon: The averaged quality of the call will be lowest codec used between each. |
| 20:31.49 | [TK]D-Fender | echelon: * supports MANY codec. |
| 20:31.58 | fogo | maurot: what version of * are you running? |
| 20:32.22 | [TK]D-Fender | echelon: Ttypically G.711 (ulaw/alaw) is the highest common denominatory (PSTN uses it) |
| 20:32.23 | gitguy | "core show codecs" |
| 20:32.30 | maurot | fogo: asteirsk is 1.4.21-rc1 |
| 20:33.00 | maurot | fogo: the call is all right but the recorder is out of sync |
| 20:34.03 | echelon | i'm using speex |
| 20:34.08 | Qwell | demonanitory? |
| 20:34.17 | Qwell | denomanitory* |
| 20:34.19 | [TK]D-Fender | echelon: Don't then. Speez is pretty low on the quality scale |
| 20:34.23 | Strom | i don't think that's a word :) |
| 20:34.26 | Qwell | [TK]D-Fender: I...think you made that up |
| 20:34.28 | fogo | maurot: hmm... my best guess is http://bugs.digium.com/view.php?id=12837 or http://bugs.digium.com/view.php?id=12296 |
| 20:34.29 | [TK]D-Fender | Denominator* |
| 20:34.48 | Qwell | I can't even say that word |
| 20:34.59 | [TK]D-Fender | Qwell: hukt on fonix werkt 4 u! |
| 20:35.11 | fogo | maurot: I've run into that on earlier versions, so I don't know |
| 20:35.13 | Strom | cocks in tonics? |
| 20:35.23 | jaytee | lol |
| 20:35.23 | Qwell | note to self: keep drinks away from Strom |
| 20:35.27 | Strom | strange colonics? |
| 20:35.37 | Strom | plague bubonics? |
| 20:35.37 | maurot | fogo: maybe the hard? |
| 20:35.47 | maurot | fogo: i have a inte lceleron 2gb 256 ram |
| 20:35.59 | maurot | fogo: 2gh |
| 20:36.06 | [TK]D-Fender | Plate tectonics? |
| 20:36.09 | [TK]D-Fender | shakes |
| 20:36.20 | jaytee | rattles |
| 20:36.25 | [TK]D-Fender | rolls |
| 20:36.56 | FinboySlick | ducks and covers. |
| 20:37.00 | fogo | maurot: well, if you've gor recordings falling out of sync, it's not generally the hardware (to my experience), just a bug in the software (which has been happening in earlier versions) |
| 20:38.17 | echelon | [TK]D-Fender: how about PCMA? |
| 20:38.40 | [TK]D-Fender | echelon: I've already answered that, specifically. |
| 20:38.47 | [TK]D-Fender | echelon: PCMA=alaw |
| 20:39.24 | echelon | oh |
| 20:40.44 | maurot | fogo: ok thanks |
| 20:40.50 | maurot | fogo: what version recommend me? |
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| 20:43.35 | *** mode/#asterisk [+o russellb] by ChanServ |
| 20:45.31 | fogo | maurot: heh. I've been plagued with out of sync issues since 1.4.18. You can get it working 100% (with a patch) on 1.4.19.1, but I didn't find that to be a great build. So far, 1.4.20.1 seems to be working well, despite some problems on chanspy |
| 20:45.48 | *** part/#asterisk echelon (n=echelon@ool-182cc7a4.dyn.optonline.net) |
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| 20:48.20 | hsv-al | http://youtube.com/watch?v=FnTK9R5dDp8 |
| 20:49.14 | unpaidbill | nice, cop schlong! |
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| 20:55.19 | maurot | fogo: ok but i have the 1.4.21 you recommend mu downgrade tu 1.4.20.1 ? |
| 20:58.06 | maurot | fogo: what about 1.6 ? |
| 20:59.10 | *** join/#asterisk Whitor (n=Whitor@rrcs-24-97-4-150.nys.biz.rr.com) |
| 20:59.18 | Whitor | Hi... I just installed asterisk... I also installed gastman ... I havn't setup any usernames or passwords for asterisk yet... but when I run gastman, I get a dialog box asking for such... Is there a default Username and password for admining Asterisk? My system u and p does not seem to work. |
| 20:59.49 | Whitor | - not using asterisknow anymore :) |
| 21:00.10 | Whitor | Am using the repository fed Ubuntu version atm |
| 21:00.12 | *** join/#asterisk implicit (n=implicit@ip68-5-93-237.oc.oc.cox.net) |
| 21:00.32 | unpaidbill | http://www.voip-info.org/tiki-index.php?page=Asterisk%20gastman look under 'set up asterisk' |
| 21:00.50 | Whitor | unpaidbill, tyvm |
| 21:02.20 | [TK]D-Fender | Whitor: it uses AMI so you need to set up an account for it under manager.conf |
| 21:03.03 | FinboySlick | Hmmm, I thought that was a zaptel issue, but seems my asterisk stops answering completely and won't even answer to a kill signal after a little while. |
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| 21:06.08 | Whitor | [TK]D-Fender, ty too ! |
| 21:07.07 | Whitor | well... its whistle time here... cheers and thanks to everyone who put up with my nubbish Q's |
| 21:10.49 | *** join/#asterisk CunningPike (n=arodgers@204.239.8.157) |
| 21:11.35 | gitguy | i have users in sip.conf with different codecs for each |
| 21:12.27 | gitguy | that would increase the performance in transcoding but shouldnt create problems right? |
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| 21:18.23 | gitguy | i use different codecs because some users have crappiest internet connections than others, etc |
| 21:18.48 | gitguy | then for those who have shitty ones i use gsm/speex |
| 21:19.02 | gitguy | and ulaw/alaw for those who have good ones |
| 21:20.06 | gitguy | it shouldnt be a problem right? |
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| 21:38.42 | jeev | is there a store that sells the polycom 330's at a good price and finances? |
| 21:39.10 | Strom | telephonydepot.com |
| 21:39.14 | outtolunc | sure, they call it visa/mastercard <G> |
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| 21:39.42 | jeev | har har har! |
| 21:39.45 | jeev | go back to lunch |
| 21:39.51 | outtolunc | chuckles |
| 21:40.12 | jeev | starts kicking people |
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| 21:56.36 | *** join/#asterisk echelon (n=echelon@ool-182cc7a4.dyn.optonline.net) |
| 21:57.15 | echelon | hi, i need some help.. i just set up a conference server |
| 21:57.29 | echelon | myself and someone else are on it |
| 21:57.35 | echelon | but a third person isn't able to join |
| 21:58.12 | echelon | they can't call the conference extension or even the echo test extension |
| 21:59.06 | jsmith | echelon: Have you looked at the Asterisk CLI to see if it tells you why they can't join? |
| 21:59.21 | echelon | how do i do that? |
| 21:59.44 | jsmith | "asterisk -vvvr" from a Linux prompt |
| 21:59.50 | echelon | k |
| 22:00.27 | echelon | ok.. i'm in |
| 22:00.50 | echelon | [Jun 23 18:07:46] NOTICE[25389]: chan_sip.c:5335 process_sdp: No compatible codecs, not accepting this offer! |
| 22:01.04 | Qwell | echelon: g729? |
| 22:01.10 | echelon | huh? |
| 22:01.32 | echelon | speex |
| 22:04.15 | echelon | ok, they're using this "Express Talk" |
| 22:05.56 | echelon | softphone |
| 22:06.05 | gitguy | i have a question... i have like 50 users, some of them use codecs like gsm/speex for low bandwidth and some others use ulaw/alaw for better bandwidth, would this cause any kind of problems? or will just increase the transcoding load in the server? |
| 22:06.59 | jsmith | gitguy: Shouldn't cause any problems per se... but yes, it will increase the transcoding load in the server. |
| 22:07.15 | gitguy | jsmith: ok, thanks |
| 22:07.27 | jsmith | gitguy: No worries... |
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| 22:09.30 | echelon | anybody know of a good SIP softphone for windows? |
| 22:09.41 | echelon | that works with asterisk |
| 22:09.44 | gitguy | X-Lite |
| 22:09.51 | gitguy | Zoiper |
| 22:09.56 | echelon | everyone keeps saying that.. where do i get it? |
| 22:10.03 | gitguy | google |
| 22:10.32 | gitguy | http://www.counterpath.com/ => X-Lite |
| 22:10.43 | gitguy | http://www.zoiper.com/ => Zoiper |
| 22:10.46 | echelon | they have two versions |
| 22:11.00 | gitguy | if you put "X-Lite" or "Zoiper" in the url bar in Firefox, it will redirect to their sites automatically... |
| 22:11.47 | gitguy | i don't know |
| 22:11.56 | echelon | [Jun 23 18:09:51] WARNING[25389]: chan_sip.c:1938 retrans_pkt: Maximum retries exceeded on transmission d03471b9-de3f-dd11-8d8a-0018deabee54@epsilon for seqno 1 (Critical Response) |
| 22:12.01 | echelon | what does that mean? |
| 22:12.08 | gaetronik | and for linux |
| 22:12.19 | gaetronik | I'm looking for a softphone decent |
| 22:12.26 | gaetronik | like X-Lite |
| 22:12.43 | gaetronik | which existed for linux |
| 22:13.39 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 22:14.07 | gitguy | X-Lite exist for Linux |
| 22:14.20 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 22:14.23 | pcrane | I/ve used zoiper on linux |
| 22:14.39 | jsmith | echelon: That means that you keep sending out a SIP message, but Asterisk never gets a response from the other end |
| 22:14.56 | gaetronik | gitguy, still exist? |
| 22:15.06 | echelon | so.. what's causing that? |
| 22:15.10 | gaetronik | i don't find any hint on the counterpath website |
| 22:15.14 | gitguy | gaetronik: yes, afaik |
| 22:16.43 | gitguy | gaetronik: http://storage.counterpath.com/downloads/X-Lite_Install.tar.gz |
| 22:16.54 | gitguy | gaetronik: that's X-Lite for Linux |
| 22:17.07 | echelon | what's the difference between X-Lite and X-Lite Demo? |
| 22:17.13 | echelon | i thought X-Lite was already free? |
| 22:17.37 | echelon | nvm |
| 22:17.49 | gaetronik | gitguy, i looked for it a couple of days ago i didn't find it |
| 22:18.04 | anonymouz666 | inside FUNCTIONS you shoud use ${var} or var? |
| 22:18.10 | gitguy | gaetronik: it's on their website |
| 22:18.22 | gaetronik | yes but very old |
| 22:18.36 | gitguy | gaetronik: when you download the software it asks you the platform |
| 22:18.49 | gaetronik | i was blind or drunk so |
| 22:19.33 | gaetronik | thanks a lot |
| 22:20.53 | gitguy | np |
| 22:20.56 | echelon | ok.. now my friend is able to connect to asterisk, but he doesn't get directed to the conference |
| 22:21.05 | echelon | instead he gets "demo-congrats" |
| 22:21.49 | gitguy | add a extension with MeetMe() in the proper context |
| 22:21.57 | echelon | huh? |
| 22:22.12 | echelon | i'm using Conference() |
| 22:22.21 | gitguy | ok, or that |
| 22:22.42 | echelon | gitguy: but my other friend and i were able to get into the conference just fine |
| 22:22.44 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177584412.dsl.bell.ca) |
| 22:22.55 | echelon | using 4321@53.36.37.1 |
| 22:23.02 | echelon | sip:4321@53.36.37.1 |
| 22:23.35 | gitguy | what is the context of your 3 users... |
| 22:23.46 | gitguy | context= |
| 22:24.00 | echelon | they're not registered users on asterisk |
| 22:24.13 | gitguy | hm |
| 22:24.23 | gitguy | inbound? |
| 22:24.27 | echelon | yeah |
| 22:24.47 | gitguy | in which context is your conference() ? |
| 22:24.51 | gitguy | [default] ? |
| 22:25.20 | gitguy | the conference extension |
| 22:25.21 | echelon | i dunno |
| 22:25.46 | echelon | exten => 4321,1,Conference(servername,N,1) |
| 22:26.09 | gitguy | look in which context it is |
| 22:26.38 | echelon | how? |
| 22:27.04 | gitguy | extensions are grouped by context, like: |
| 22:27.07 | gitguy | [context] |
| 22:27.09 | gitguy | exten => ... |
| 22:27.22 | gitguy | look in which section/context is your extension |
| 22:27.38 | gitguy | by opening the extensions.conf in a text editor and looking at it |
| 22:27.56 | echelon | ok |
| 22:28.16 | echelon | it's in [default] |
| 22:29.00 | echelon | should uncomment include => demo? |
| 22:29.02 | gitguy | when you dial in... where does the incoming calls looks at, at [default] right? |
| 22:29.17 | echelon | i guess so, since it works for me |
| 22:29.36 | gitguy | odd |
| 22:29.55 | echelon | so should uncommenting "include => demo" work? |
| 22:30.09 | gitguy | try it |
| 22:30.34 | gitguy | include => demo just includes the extensions of that context in your default context |
| 22:31.13 | echelon | how about exten => s,n(restart),BackGround(demo-congrats) ; ? |
| 22:31.22 | gitguy | make sure you don't have another 4321 extension in demo context... |
| 22:31.23 | gitguy | or something |
| 22:31.36 | gitguy | hm |
| 22:31.54 | echelon | nope, no other 4321 |
| 22:32.30 | gitguy | weird |
| 22:34.19 | gitguy | did you do dialplan reload after you do the changes? |
| 22:34.50 | echelon | dialplan reload? |
| 22:34.57 | gitguy | in the command line |
| 22:34.58 | echelon | i just restarted asterisk |
| 22:35.00 | gitguy | asterisk cli |
| 22:35.02 | gitguy | ok |
| 22:35.53 | echelon | weird.. now echotest won't work |
| 22:36.16 | gitguy | you commented include => demo? |
| 22:36.21 | echelon | yeah |
| 22:36.29 | gitguy | that's why |
| 22:36.37 | echelon | ok |
| 22:38.52 | echelon | [Jun 23 18:45:48] NOTICE[27861]: chan_sip.c:13669 handle_request_invite: Call from '' to extension '' rejected because extension not found. |
| 22:39.04 | echelon | that's the error i get in CLI |
| 22:39.19 | echelon | and he gets "no route to destination" |
| 22:39.19 | gitguy | when your friend calls? |
| 22:39.23 | echelon | yeah |
| 22:39.27 | echelon | but when i call, it's fine |
| 22:39.42 | gitguy | you probably have to configure your DID number properly |
| 22:39.49 | echelon | DID? |
| 22:40.34 | gitguy | Direct Inward Dialing |
| 22:40.39 | echelon | could it be the zoinker client? |
| 22:40.49 | echelon | because i'm using linphone, and my other friend was using sjphone |
| 22:41.12 | gitguy | weren't you using normal phones? |
| 22:41.14 | gitguy | i'm confused now |
| 22:41.17 | echelon | no |
| 22:41.19 | echelon | softphones |
| 22:41.32 | gitguy | 18:24 < gitguy> what is the context of your 3 users... |
| 22:41.33 | gitguy | 18:24 < gitguy> context= |
| 22:41.33 | gitguy | 18:25 < echelon> they're not registered users on asterisk |
| 22:41.44 | echelon | yeah |
| 22:41.48 | gitguy | how come you are using softphones but you don't have registered users? |
| 22:42.01 | echelon | we're just directly dialing the conference room |
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| 22:43.55 | gitguy | echelon: are the three users softphones? |
| 22:44.34 | echelon | yeah, all of 'em |
| 22:44.42 | [TK]D-Fender | echelon: they aren't dialing an extension first of all and there is no match for what they are dialing (blank) in the contexdt against which you receive the call |
| 22:45.39 | echelon | we all called sip:4321@IP |
| 22:45.54 | echelon | and it worked for linphone and sjphone |
| 22:45.58 | [TK]D-Fender | echelon:Call from â to extension â rejected because extension not found. <--- means exactly what it says. they are dialing '' (nothing) and you don't have an exten in the context (DIALPLAN - extensions.conf) that processed that call. |
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| 22:47.14 | gitguy | echelon: in other words, configure asterisk properly |
| 22:47.14 | echelon | ok.. how do you enter sip addresses in x-lite? |
| 22:48.43 | [TK]D-Fender | echelon: almost nobody allows unauthed SIP calls like that. As for how, it might be that it wasn't made to let you work like that either. |
| 22:49.03 | echelon | [TK]D-Fender: we're on VPN, so it's not important |
| 22:49.22 | [TK]D-Fender | echelon: then it should also be no problem to set them up properly. |
| 22:49.33 | [TK]D-Fender | echelon: go make sip.conf account entries for them |
| 22:50.37 | echelon | nevermind, we got it to work |
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| 23:18.06 | frieze | is the digium svn not working? |
| 23:18.51 | frieze | svn co http://svn.digium.com/svn/asterisk-addons/trunk asterisk-addons does not appear to work. have i caught the stupids again? |
| 23:18.53 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 23:20.32 | frieze | nevermind. I did have the stupids. reboot = firewall rules reload = no svn |
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| 23:45.28 | ManxPower | You'd think that in 2008, they could managed to keep the power on. |
| 23:45.34 | ManxPower | stupid power company |
| 23:48.31 | ManxPower | 3 multi-hour power outages since Memorial day -- with no storms to explain it. |
| 23:48.57 | ManxPower | Does anyone have suggestions for propane fueled generators? |