| 00:02.37 | *** join/#asterisk jakent (n=john@c-98-233-13-157.hsd1.va.comcast.net) |
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| 00:10.38 | p3nguin | I haven't seen a flake of snow all day. |
| 00:12.48 | jaytee | me neither, I've seen a few flakes but those were just people I work with |
| 00:24.13 | *** part/#asterisk pkecastillo (n=pirruar@190.113.141.122) |
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| 00:30.19 | sudhir492 | Hi All |
| 00:30.45 | sudhir492 | Can anyone tell me when to use OpenSER with asterisk? |
| 00:32.02 | voipmonk | When u have a crapload of deviceto keep track of |
| 00:32.07 | voipmonk | Or load balancing |
| 00:32.34 | voipmonk | Or fill in the blank |
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| 00:33.32 | sudhir492 | Can you elaborate about load balancing |
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| 01:49.22 | normaldotcom | hello, my asterisk server is able to ring extensions when an incoming number rings, but extensions cannot dial other extensions. When extensions dial anything, logs state 'call rejected because extension not found' |
| 01:50.21 | jaytee | normaldotcom, then your extensions are missing from your dialplan |
| 01:50.58 | jaytee | normaldotcom, are you running standard asterisk or asterisknow with freepbx |
| 01:51.07 | normaldotcom | standard asterisk |
| 01:51.28 | jaytee | pastebin your sip.conf and extensions.conf files masking passwords |
| 01:51.34 | jaytee | ~pb |
| 01:51.35 | infobot | [~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://bin.cakephp.org/ , or apt-get install pastebinit |
| 01:52.37 | normaldotcom | will do. I'm also using asterisk-gui, so things are a bit convoluted |
| 01:53.27 | jaytee | that's why then. |
| 01:53.45 | jaytee | asterisk-gui isn't supported in this channel |
| 01:54.06 | normaldotcom | ah, is there a more appropriate channel for discussing asterisk-gui related issues? |
| 01:54.29 | jaytee | and it uses the users.conf file instead of having extensions in extensions.conf |
| 01:54.46 | normaldotcom | heads over to #asterisk-gui, thanks |
| 01:55.50 | jaytee | normaldotcom, looks like the channel isn't there anymore |
| 01:56.03 | jaytee | most people are using freepbx nowadays that use a gui |
| 01:56.59 | normaldotcom | hmm, I'll look into that |
| 01:57.28 | jaytee | it will limit your flexibility. it trades power for ease of use |
| 02:01.47 | *** join/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 02:03.15 | [TK]D-Fender | Asterisk GUI = unsupported jsut about period |
| 02:03.52 | [TK]D-Fender | normaldotcom: pastebin your users.conf masking only passwords, along with your extensions.conf and CLI output with SIP DEBUG enabled |
| 02:13.19 | *** join/#asterisk hluesea (n=hulusika@88.247.127.66) |
| 02:16.10 | normaldotcom | [TK]D-Fender: http://dpaste.com/152412/ (users.conf) and http://dpaste.com/152414/ (extensions.conf), http://dpaste.com/152416/ (output) |
| 02:23.00 | [TK]D-Fender | normaldotcom: I said with SIP DEBUG : help sip |
| 02:28.25 | p3nguin | System() returns APPERROR, but that isn't a SYSTEMSTATUS reported in core show application System. Where should I look for info on how to analyze the problem? |
| 02:29.15 | ManxPower-work | p3nguin: The source code, unfortunately. |
| 02:33.50 | *** join/#asterisk aidinb (n=Aidin@64.134.224.200) |
| 02:34.00 | p3nguin | Should System() normally handle a pipe in the command? System(/bin/echo "Please see attachment" | /usr/bin/mutt -s "New Fax" -a /var/spool/asterisk/fax/${UNIQUEID}.tif -- fax-manager) |
| 02:34.02 | normaldotcom | {TK}D-Fender: http://dpaste.com/152420/ |
| 02:35.10 | *** join/#asterisk etnos (n=etnos@adsl-10-116-134.mia.bellsouth.net) |
| 02:48.05 | Katty | me and riddick are having a blast in the yard :P |
| 02:48.26 | [TK]D-Fender | normaldotcom: pastebin your sip.conf as well please |
| 02:48.52 | Katty | p3nguin: you're missin some mighty fine snow!! |
| 02:48.55 | Katty | goes back outside |
| 02:49.08 | p3nguin | Still haven't seen a single flake. |
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| 02:51.39 | normaldotcom | [TK]D-Fender: http://dpaste.com/152427/ |
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| 02:52.58 | [TK]D-Fender | normaldotcom: Looking for 1001 in 12.12.12.12,163.11.146.65 (domain 11.11.11.11) <--- somehow your sip config picked up "12.12.12.12,163.11.146.65" as the target context name. Check your settings, bu |
| 02:53.08 | [TK]D-Fender | normaldotcom: Something looks whacked |
| 02:54.22 | jaytee | http://www.listal.com/list/m-p-t-t-t |
| 02:55.10 | normaldotcom | i'm thinking that it might be better for me to switch to freepbx, as I don't have much experience with asterisk and I don't need many advanced features |
| 02:56.02 | Katty | okay, i'[m done. i can't feel my toes |
| 02:56.47 | Katty | p3nguin: enjoy the show? :P |
| 02:57.19 | *** join/#asterisk youngproguru (n=youngpro@cpe-76-180-188-78.buffalo.res.rr.com) |
| 02:57.24 | p3nguin | It wasn't bad. Not much happened, though. |
| 02:57.43 | Katty | what were you expecting? a parade? ;) |
| 02:57.56 | p3nguin | I had no expectations. |
| 02:58.14 | [TK]D-Fender | normaldotcom: Or you could learn the little it takes to do it yourself your way |
| 02:59.14 | *** join/#asterisk Cain` (n=Geek@unaffiliated/cain) |
| 03:00.17 | Katty | riddick wanted back outside |
| 03:01.19 | p3nguin | I was going to tell you that you forgot your dog. |
| 03:01.31 | Katty | lol he's stuck |
| 03:01.33 | Katty | brb |
| 03:03.12 | *** join/#asterisk dkirker-openmobl (n=dkirker@openmobl/ceo/dkirker) |
| 03:03.32 | Katty | too funny |
| 03:03.42 | Katty | at least he tells me when he's stuck |
| 03:05.24 | normaldotcom | [TK]D-Fender: thanks haha, just got it fixed, my domain setting was messed up |
| 03:07.39 | Katty | p3nguin: oh! i found a new bird today. |
| 03:07.40 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 03:07.47 | Katty | p3nguin: a chipping sparrow |
| 03:08.10 | p3nguin | The only one I saw was a cardinal. Everything else was in the rodent family. |
| 03:08.49 | Katty | a lot of the birds just stay down in the yard, picking up what the squirrels drop |
| 03:09.04 | Katty | i know it's difficult to actually determine breeds from the webcam |
| 03:09.24 | Katty | the only thing that gave it away was the bird was visiting the fruit baskets and the suet....something i've never seen the other sparrows do |
| 03:09.39 | Katty | so i got my binoculurs and had an even closer look....which revealed a red crown (= |
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| 03:48.39 | Katty | searches for life |
| 03:48.59 | Katty | pokes around |
| 03:49.13 | Katty | thinks |
| 03:49.19 | Katty | MANDRAKE IS THE BEST |
| 03:49.21 | Katty | waits. |
| 03:51.32 | ChannelZ | The best at what/? |
| 03:51.58 | [TK]D-Fender | ChannelZ: Summoning potions. |
| 03:52.07 | [TK]D-Fender | cackles |
| 03:52.47 | Katty | cheers |
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| 03:55.47 | *** part/#asterisk fozzmoo (n=fozz@209.41.95.5) |
| 03:55.53 | ManxPower-work | Cardinals are totally crazy birds. |
| 03:56.27 | ManxPower-work | I used to have one that would over and over again fly directly into the reflection in my window when the sun was at the right angle. Lasted for about an hour each day. |
| 03:56.40 | Katty | lol |
| 03:56.44 | Katty | that's hilarious |
| 03:56.49 | ChannelZ | I've had that, though I dunno if it was a cardinal |
| 03:57.01 | ManxPower-work | *thunk* A few mins latter *thunk* a few mins latter *thunk*. |
| 03:57.03 | ChannelZ | damn thing woke me up, banging into the window about once every 3 seconds |
| 03:57.18 | ChannelZ | I think he was horny and attacking himself as competition |
| 03:57.44 | Katty | LOL |
| 03:59.51 | Katty | can't say i've ever had that happen, but i know my parents have |
| 04:01.04 | ChannelZ | He flew into my bedroom window so hard once while I was asleep, it scared the shat out of me. Then a couple of days later he was at it again at a living room window |
| 04:01.20 | ChannelZ | Then that was it. I half wonder if he finally broke his neck on someone else's window |
| 04:01.26 | Katty | was there light coming from the opposite side? |
| 04:01.49 | ChannelZ | perpendicular to the window, yeah propably |
| 04:02.12 | Katty | i've read that migrating birds who don't know the landscape well can easily get confused if they see light coming from a certain direction |
| 04:02.44 | Katty | that would explain why it stopped |
| 04:02.49 | ChannelZ | although each window is on a different side of the house.. bedroom faces south, living room was west. The blinds were closed on both so I imagine it made it more mirror-like since there was nothing of any detail that could be seen through the window |
| 04:02.52 | Katty | if mister birdy was migrating |
| 04:03.03 | Katty | ah...hmm |
| 04:03.13 | Katty | who knows then |
| 04:03.24 | ChannelZ | yeah. Maybe he was just a retard |
| 04:03.35 | Katty | bird brains. |
| 04:03.39 | Katty | feather heads. |
| 04:03.40 | Katty | etc. |
| 04:03.51 | ChannelZ | bird flu |
| 04:08.55 | Katty | tweet fever? |
| 04:14.33 | *** join/#asterisk coppice (n=chatzill@234.157.17.210.dyn.pacific.net.hk) |
| 04:22.47 | ChannelZ | argh this is ridiculuous |
| 04:22.55 | ChannelZ | you can't turn off echo cancellation in Skype |
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| 04:27.19 | coppice | skype works with those sound card where echo cancellation cannot, so if you can't turn it off it must have some automatic turn off mechanism |
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| 04:34.09 | *** join/#asterisk bevins (n=bla@modemcable029.52-37-24.mc.videotron.ca) |
| 04:34.25 | bevins | . |
| 04:34.45 | *** part/#asterisk bevins (n=bla@modemcable029.52-37-24.mc.videotron.ca) |
| 04:35.21 | ChannelZ | eh? |
| 04:37.11 | ManxPower-work | "echo cancellation" usually referrers to echo caused on the far end analog loop and must be done at the PSTN/VoIP conversion point. "acoustic echo cancellation" is what we usually call the EC that's done for a speakerphone (or computer mic + speakers), that has to be done at the speaker+mic/VoIP conversion point. |
| 04:38.50 | ChannelZ | yeah. If I call my * via Skype (SFA on the * side) and do an Echo() and hum into the mic on my headset, skype sees the return audio as echo (which in this case it is) and tries to cancel it.. so it phases in and out |
| 04:40.06 | ChannelZ | Normally this isn't a big problem, but if both sides are on a headset and there is no "real" echo occurring, it will often get confused if you're speaking and someone speaks over you a bit, causing your skype to think their voice is yours echoing. |
| 04:40.11 | ManxPower-work | It should not see it as echo. The latency should be far too high. |
| 04:42.07 | ChannelZ | It doesn't much for short words, but if you go 'uhhhhhh' or something it does since it's a constant sound |
| 04:43.05 | ManxPower-work | sounds like half duplex, not EC |
| 04:43.15 | ChannelZ | Apparently there was an API call to shut it off but they disabled it. |
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| 04:50.04 | coppice | that's not echo cancellation. sound more like some kind of odd AGC action |
| 04:50.43 | ChannelZ | That you CAN turn off but no difference. |
| 04:51.16 | ChannelZ | If I just record my stream (leave myself voicemail for instance) it's fine |
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| 05:01.16 | *** join/#asterisk philm (n=philm@242-51.202-68.tampabay.res.rr.com) |
| 05:01.35 | philm | Hi |
| 05:01.47 | ChannelZ | oh hai |
| 05:02.24 | philm | I have a problem I'm working on |
| 05:03.08 | philm | I'm trying to setup a conference call with 3 user (admin, user 1, user2) |
| 05:04.05 | ChannelZ | is that the problem? |
| 05:04.40 | philm | I want the admin to talk to user1 and listen to user2, but not let user2 talk to user1 |
| 05:04.48 | philm | Any ideas? |
| 05:05.49 | ChannelZ | send user2 into the conference as listen only |
| 05:06.13 | ChannelZ | or does user2 get to talk to admin? |
| 05:06.26 | philm | User2 talks to the admin |
| 05:06.37 | ChannelZ | that's complicated |
| 05:06.46 | philm | yep |
| 05:07.55 | philm | It's like spy(listen) on user2, and talk to user1 from the admin's view |
| 05:08.33 | *** join/#asterisk corretico (n=laguilar@201.201.46.106) |
| 05:10.26 | ChannelZ | well I guess that would be the way if there's just 3 parties involved, user2 would ChanSpy on admin's channel with whisper mode.. at least I think |
| 05:12.28 | philm | So if the call is bridges, I can use chanspy to listen to the call, and whisper to only one end? |
| 05:12.47 | ChannelZ | yeah |
| 05:13.04 | philm | That might work. Thanks for you help. |
| 05:13.17 | ChannelZ | canreinvite=no |
| 05:13.17 | philm | Thanks for the help. |
| 05:13.42 | ChannelZ | good luck |
| 05:24.30 | ManxPower-work | Nashville has 6" - 8" of snow right now. Sucks to be them |
| 05:34.23 | coppice | ManxPower-work: its down to 25C here right now. winter is a time for suffering |
| 05:35.15 | ManxPower-work | coppice: practically freezing there in HK. 8-| |
| 05:35.56 | p3nguin | Okay, so still no idea why System() chokes. |
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| 06:33.03 | *** join/#asterisk benngard (n=benngard@90-230-92-67-no148.tbcn.telia.com) |
| 06:33.51 | mmlj4 | HK? neat |
| 06:36.13 | *** join/#asterisk lordmortis (n=lordmort@CPE-121-221-11-22.lns1.wel.bigpond.net.au) |
| 06:57.24 | ManxPower-work | waves to mmlj4 |
| 07:02.49 | mmlj4 | waves back |
| 07:02.56 | mmlj4 | HK as in Hong Kong? |
| 07:05.00 | benngard | morning folks |
| 07:11.04 | ChannelZ | hi and bye |
| 07:11.10 | *** join/#asterisk benngard (n=benngard@90-230-92-67-no148.tbcn.telia.com) |
| 07:11.14 | benngard | can someone plz go to one of our offices and answer my call plz ;) |
| 07:14.55 | *** join/#asterisk benngard (n=benngard@90-230-92-67-no148.tbcn.telia.com) |
| 07:17.09 | ChannelZ | Can you stay connected for more than 30 seconds? |
| 07:17.21 | benngard | i wish i could |
| 07:17.36 | benngard | blame telia (my adsl supplier) |
| 07:51.44 | *** join/#asterisk ttl- (n=patrick@d5153A420.access.telenet.be) |
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| 08:12.22 | compassiontara | hi everyone. I upgraded to ubuntu 8.04 and gdb says Program exited with code 01. (I don't have another asterisk running) |
| 08:13.07 | *** join/#asterisk rycar (n=rycar@adsl-76-244-169-60.dsl.bkfd14.sbcglobal.net) |
| 08:23.40 | benngard | is trying to figure out how "connected line info" i working... is there some readme/doc/howto? |
| 08:23.51 | benngard | is* |
| 08:24.52 | *** join/#asterisk Tim_Toady (n=moi@77.49.165.135.dsl.dyn.forthnet.gr) |
| 08:28.16 | ChannelZ | compassiontara: are you running packaged asterisk or did you build it yourself? |
| 08:28.37 | compassiontara | ChannelZ: the pacakged asterisk |
| 08:28.53 | benngard | setup: h323 phone - avaya - ooh323 - asterisk sip phone(s), h323 phone calls a sip phone, sip phone answer, display of h323 phone changes to "name of sip phone" in my case Magnus Benngard, but when sip phone calls another sip phone, no changes on the display... |
| 08:29.15 | ChannelZ | hmm not sure then |
| 08:29.26 | compassiontara | pbx.c:2968 ast_register_application: Already have an application 'VMAuthenticate' |
| 08:29.36 | compassiontara | that is the last line I get before it exits. |
| 08:34.31 | compassiontara | ChannelZ: yea, i don't know either obviously :( |
| 08:51.41 | *** join/#asterisk infobot (ibot@rikers.org) |
| 08:51.41 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.2.1 (2010/01/15), Asterisk 1.6.1.13 (2010/01/15), 1.6.0.21 (2010/01/15), 1.4.29 (2010/01/15), *-Addons 1.6.2.0 (2009/12/18), 1.6.1.2 (2009/12/02), 1.6.0.4 (2009/12/02), 1.4.10 (2009/12/02), dahdi-linux 2.2.0.2 (2009/07/23), dahdi-tools 2.2.0 (2009/06/24), Libpri 1.4.10.2 (2009/10/20) -=- Related channels |
| 08:52.35 | timahvo1 | Hi people |
| 08:53.18 | timahvo1 | I can't seem to get asterisk to ignore incoming calls |
| 08:54.16 | ChannelZ | trace cord to computer, pull cord |
| 08:54.36 | *** join/#asterisk jetlag (jetlag@pool-70-18-186-205.pskn.east.verizon.net) |
| 08:54.52 | ChannelZ | Works every time. Guaranteed. |
| 08:54.58 | compassiontara | asterisk -X 'stop now' would do it as well, but then it wouldn't be running asterisk either :P |
| 08:55.08 | *** join/#asterisk scardinal (~supreme@90.184.100.119) |
| 08:55.22 | timahvo1 | after I hit ignore (on a softphone) the call gets respawned and indicates as originating from asterisk |
| 08:56.01 | ChannelZ | Perhaps your dialplan is such that it occurs? What does the console say is happening? |
| 08:56.43 | *** join/#asterisk profxavier (~chatzilla@unaffiliated/neverblue) |
| 08:57.39 | timahvo1 | ChannelZ: lemmie pastebin it |
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| 08:59.28 | *** join/#asterisk Daviey (~Daviey@ubuntu/member/pdpc.gold.Daviey) |
| 08:59.31 | timahvo1 | ChannelZ: http://pastebin.ca/1771748 |
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| 09:03.08 | ChannelZ | plus that paste does not show enough (like what happens afterwards, where is it making the additional call) |
| 09:03.09 | timahvo1 | ok lemmie get irssi set up on the box and pastebin whatever you need to help me , will just be a sec |
| 09:03.52 | benngard | poor guy at our boras office, he has 033211118 internally we can just dial 118 to call him, 118118 in sweden is the number to dial when u wanna ask "who has a number" those who have mobiles with mex has to preped with a "*" so they dial *118118, np but for example friday night, then gonna call a cab, they are a little bit "drunk" so they dial 118118 without *, the poor boras guy has a mobile with mex so guess how many calls he get fr |
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| 09:11.40 | compassiontara | benngard: haha Poor guy! :( |
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| 09:13.25 | timahvo1 | brb |
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| 09:15.53 | m0t3jl | Hi, I was wandering, is it possible (or is it wise) to mangle with the outgoing caller's id? Now, I am not talking about the actual number, I am talking about the name that goes with it. I was thinking about appending our company's name to it. So if the user is internally called John Smith, I would convert his name to John Smith (Company Inc.). Is this just a stupid idea? Thanks a lot ;) |
| 09:15.58 | timahvo1 | ChannelZ, ok am setup so what do you need ? |
| 09:16.13 | ChannelZ | extensions.conf |
| 09:17.22 | ChannelZ | m0t3jl: are you on a PRI? you may not even have the capability to change the CID - there is also a length limit |
| 09:19.40 | m0t3jl | ChannelZ, well, to be honest, I don't even know that many about the technologies involved, I just thought it would look good if the receiving party saw our employees' name together with the company's name instead of just our SIP phone number ;). |
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| 09:21.00 | ChannelZ | well if you're on a PRI you might be able to change it yourself; If you're using an ITSP, they may or may not allow you to change it. And I think the max length is for the name is something piddly like 16 characters |
| 09:22.48 | m0t3jl | ChannelZ, oh... |
| 09:23.52 | m0t3jl | ChannelZ, pitty there are still such funny limitations like 16-character-long id in the 21st century :(... |
| 09:24.13 | ChannelZ | you just have to check with your provider. I suppose some might let you change the name but still enforce the number |
| 09:24.52 | timahvo1 | ChannelZ, http://pastebin.ca/1771763 |
| 09:25.00 | ChannelZ | Well in the analog world caller ID is transmitted before or between rings so there's a limited amount of time to transmit |
| 09:25.14 | m0t3jl | ChannelZ, the number is okay, I want the called party to see the number, I just though it would look better if the called party saw not only the user's name, but also our company ;) |
| 09:26.21 | m0t3jl | ChannelZ, it would be better to allow like 80-character-long name for SIP2SIP calls and trim it for SIP2Analogue line calls. |
| 09:27.29 | ChannelZ | a certain amount of that does happen |
| 09:29.36 | timahvo1 | ChannelZ, and when I make an incoming call from an outside line this is what I get even after I hit Ignore http://pastebin.ca/1771765 |
| 09:29.36 | m0t3jl | ChannelZ, yes, but still, pitty ;) |
| 09:29.52 | timahvo1 | ChannelZ, and it keeps ringing |
| 09:31.40 | ChannelZ | timahvo1: Oh. You never Answer() the call. |
| 09:32.24 | m0t3jl | :) |
| 09:32.28 | ChannelZ | As far as Asterisk is concerned, Hangup() does nothing because the channel was never picked up in the first place. So as far as the telco is concerned, the line is still ringing |
| 09:33.08 | timahvo1 | ChannelZ, ah ok |
| 09:33.17 | ChannelZ | I'm also not sure what you're doing with the wierd arguments to Hangup but that's unrelated |
| 09:33.59 | timahvo1 | ChannelZ, yeah I've removed the extra arguments , was an error on my part |
| 09:34.27 | m0t3jl | timahvo1, , I see a lot of examples when they sometimes use Answer, sometimes not. But in the Book it says that some commands do Answer by themselves, still they say it's a good habit to Answer anything first prior to doing something. |
| 09:35.09 | timahvo1 | <PROTECTED> |
| 09:35.26 | ChannelZ | Answer() |
| 09:36.08 | m0t3jl | timahvo1, some providers will terminate the call after some amount of seconds having not received the Answer even though you may be speaking on the line ;) |
| 09:36.26 | ChannelZ | though frankly the whole thing would be a little confusing to your caller.. they call, they hear the phone ring a couple of times, and then the call just terminates.. |
| 09:36.57 | m0t3jl | ChannelZ, +1 |
| 09:37.13 | timahvo1 | ideally what I would like is the user to see the caller ID then decide wheter to answer the call or ignore it |
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| 09:37.49 | timahvo1 | problem is whenver I hit ignore on the softphone the call just gets respawned indicating callerid =asterisk |
| 09:37.52 | ChannelZ | what happens if they ignore it? You really want to just hang up on the person calling? |
| 09:38.01 | timahvo1 | ChannelZ, yes |
| 09:38.17 | ChannelZ | ..well ok then.. |
| 09:39.10 | m0t3jl | :) |
| 09:39.24 | timahvo1 | ChannelZ, so the first thing to do would be Answer instead of s,1,Dial(SIP/400) ? |
| 09:39.36 | ChannelZ | yeah |
| 09:39.46 | timahvo1 | ok lemmie try that |
| 09:40.29 | m0t3jl | timahvo1, although many examples just Dial, no Answer.... |
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| 09:40.52 | ChannelZ | m0t3jl: yes but as I said, you can't hang up on someone if you haven't picked them up first. |
| 09:40.55 | m0t3jl | ChannelZ, is Dial one of the functions that Answers by themselves or not? |
| 09:41.33 | ChannelZ | No, not unless someone picks up the channel being Dialed, then it would bridge them and by proxy answer the first channel |
| 09:41.39 | m0t3jl | ChannelZ, no no, I am just saying to Answer always even though there are many howtos that do not Answer ;) |
| 09:42.29 | ChannelZ | That works assuming you're going to either answer the call or just let it ring forever until the person gives up |
| 09:42.29 | timahvo1 | ChannelZ, I follow your logic but think of it this way ....your cell phone starts ringing and you see the callerid and don't want to answer it ... |
| 09:42.33 | m0t3jl | ChannelZ, that's what I though, so when it is not possible to connect the caller with the called party, the caller's call would not be picked up? |
| 09:42.49 | timahvo1 | you can either let it ring or hit the call end button and terminate the call |
| 09:43.00 | ChannelZ | timahvo1: yeah.. I either let it ring until they give up or it goes to voicemail, or if I hit ignore the same thing happens |
| 09:43.10 | mattwj2002 | hey guys |
| 09:43.16 | ChannelZ | but it doesn't just *click* and drop the caller |
| 09:43.22 | mattwj2002 | I got my asterisk server running on a netbook |
| 09:43.36 | timahvo1 | ChannelZ, but can't I hit ignore and the line goes dead on them ? (rude I know) |
| 09:43.55 | m0t3jl | ChannelZ, I actually had one phone (was it Siemens?) that did not hang up when I hit ignore, but only stopped vibrations and any sound, so the caller would think I am just not answering ;) |
| 09:43.56 | ChannelZ | Yes, but you HAVE TO ANSWER() IN ORDER TO HANGUP() for the 5th time |
| 09:44.39 | ChannelZ | m0t3jl: that's how it usually works, or it times out and goes to voicemail. |
| 09:44.53 | m0t3jl | ChannelZ, it was a cell phone, actually, perhaps I should have mentioned that ;) |
| 09:45.02 | ChannelZ | same difference |
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| 09:45.05 | m0t3jl | ;) |
| 09:45.49 | timahvo1 | ChannelZ, is there any possible way to jump straight to ignore without answering first ? |
| 09:45.55 | m0t3jl | ChannelZ, sometimes I think it's not so rude to let the caller hang up eventually as to hang up by myself ;) |
| 09:46.03 | ChannelZ | timahvo1: When someone calls in, the telco is sending you an indication as such, but * hasn't done anything with that yet. It's just like your phone is ringing and ringing. |
| 09:47.13 | ChannelZ | You are then doing a Dial() to another channel. But * has not done anything to the calling channel yet. You then hit 'ignore' on your SIP phone which in your case goes on to a Hangup() in your dialplan. But again, * has not 'picked up the phone' yet, but regardless, your dialplan terminates. |
| 09:47.43 | ChannelZ | However the telco doesn't know anything about *, anything about your SIP phone, or that you hit ignore. As far as it is concerned, the phone is still ringing |
| 09:48.04 | ChannelZ | And that's exactly what is happening. * sees the channel ringing, and the whole cycle starts over. |
| 09:48.38 | ChannelZ | So you have three choices. |
| 09:48.42 | ChannelZ | 1. Live with it |
| 09:48.52 | ChannelZ | 2. Answer() first |
| 09:49.32 | ChannelZ | 3. Add an Answer(1000) as the first priority before doing a Hangup() |
| 09:50.31 | m0t3jl | I'd just go with Answer(),Dial(),GotoIf(STATUS is OK, 6),Playback(congestion),Congestion, Hangup() |
| 09:50.42 | m0t3jl | I am not sure about the GotoIf condition... |
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| 09:51.18 | timahvo1 | ChannelZ m0t3jl ok guys thanks for the help , got a better idea of whats going on now |
| 09:51.25 | ChannelZ | that whole Goto thing seems over-done and unnecessary |
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| 09:52.10 | ChannelZ | since it doesn't look like you want voicemail, a simple system would just be to Answer, Dial, and Hangup. Because when Dial comes back, it doesn't matter why. |
| 09:53.09 | m0t3jl | ChannelZ, I just thought it would be better for the caller not to hear the congestion sound from my Asterisk, when the call has successfully finished ;) |
| 09:53.19 | ChannelZ | So don't play it |
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| 09:53.57 | m0t3jl | ChannelZ, but then I wanted them to hear it when the Dial was unsuccessful (like the Ignore was hit) ;) |
| 09:54.40 | ChannelZ | So it rings, and then is suddenly busy? That seems no less confusing than just dumping the call |
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| 09:54.58 | ChannelZ | but hey, this is why you can program it to do whatever the hell you want it to do |
| 09:55.13 | ChannelZ | so have fun. I'm finally going to bed |
| 09:55.54 | timahvo1 | ChannelZ, think voicemail is the answer |
| 09:55.59 | compassiontara | ChannelZ: it was a config file, but I have NO idea which one.. :) |
| 09:56.17 | ChannelZ | why did you switch them all at once? |
| 09:56.29 | compassiontara | ChannelZ: I got it all working.. Thanks for your help !! |
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| 09:57.22 | m0t3jl | ChannelZ, perhaps you're right ;) |
| 10:00.36 | *** join/#asterisk Caplain (shayne@caplain.loves.boys.fbi.gov.silverelitez.org) |
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| 10:21.05 | benngard | yay, found i way to get clid working from avaya to asterisk |
| 10:22.21 | benngard | and connected call infoe works also from avaya to asterisk |
| 10:24.11 | benngard | has alot of configuration to do but its worth it! |
| 10:30.10 | *** join/#asterisk slima (slima@unaffiliated/slima) |
| 11:04.38 | benngard | but why cant i get caller id to work on my avaya phone, it is working on my siemens :( |
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| 12:56.24 | *** join/#asterisk Ast001 (~snezana@cable-94-189-232-243.dynamic.sbb.rs) |
| 12:59.24 | Ast001 | Hi, I want to be able to call ordinary phone with my xlite softphone through asterisk pbx and to be billed for talking but problem is that asterisk is writing in mysql cdr data everything from moment call starts (phone is ringing or is busy but billsec is counted) Any ideas what the problem might be ? |
| 13:00.46 | Ast001 | it seems dahdi emidetely answers the call before called party pick up a phone. |
| 13:07.05 | steliosk | I am having an issue with received faxes over pri. The setup is E1->asterisk 1.2.31->Linksys PAP->Fax. Both sides say that the fax was send/received ok, but i get a lot of "faint" lines |
| 13:07.08 | Ast001 | which leads to problem like calling party did not pick up or answer a call but asterisk wrote in cdr that call is answered by dahdi/1-1 and have billsec which should be 0 because I did not even talk to him because calling party did not answer |
| 13:08.11 | steliosk | playing with rxgain in pri alot, found a threshold after which increasing it more results in broken pages |
| 13:12.07 | *** join/#asterisk Akiraa (~Akiraaaa@79.112.24.192) |
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| 14:07.03 | jaytee | mornin [TK]D-Fender |
| 14:07.10 | [TK]D-Fender | y0 |
| 14:07.43 | jaytee | cold there? |
| 14:08.58 | [TK]D-Fender | jaytee: jaytee A balmy -1F |
| 14:09.14 | [TK]D-Fender | jaytee: You? |
| 14:09.21 | jaytee | makes it look like a heatwave here at 16F |
| 14:09.50 | [TK]D-Fender | would cry... if it weren't for the risk of icicles forming off his lashes |
| 14:10.02 | coppice | ooh, nasty. its down to 24C right now |
| 14:11.08 | jaytee | the older I get the closer to the equator I want to be |
| 14:11.25 | coppice | jaytee: bangalore is nice |
| 14:11.47 | [TK]D-Fender | jaytee: Maui.... |
| 14:11.48 | jaytee | although being disgustingly rich and being able to afford to follow summer around the world would be nicer |
| 14:12.23 | jaytee | basking in the Maui sun.....like Whitman, Price and Haddad. :-) |
| 14:15.31 | *** join/#asterisk chendy (~chatzilla@116.25.172.169) |
| 14:17.00 | jaytee | man, I'd love to get my hands around the neck of the bastard that wrote the Trojan.FakeAlert trojan. |
| 14:17.56 | jaytee | or the engineers at Symantec that maintain their worthless threatshield component |
| 14:18.54 | coppice | jaytee: venting on the bogus anti-malware makers makes a lot more sense than venting on the people who right malware |
| 14:19.09 | jaytee | true |
| 14:19.14 | *** join/#asterisk TimeRider (~steve@78.32.26.1) |
| 14:20.05 | coppice | Trojan.FakeAlert is your friend. without these nasty programs to highlight security issues the people writing the really evil non-obvious stuff would get away with it |
| 14:20.38 | jaytee | especially since Antimalware by Malwarebytes is supposedly written by a single developer, has a free version and manages to detect, quarantine and destroy most of the trojans I've come across compared to the high end corporate software like Symantec's that doesn't stop crap |
| 14:20.58 | jaytee | coppice, another good point |
| 14:25.07 | *** join/#asterisk Obeliks (obeliks@gentoo/contributor/Obeliks) |
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| 14:43.19 | Katty | morning |
| 14:45.20 | Katty | lots of visitors this morning |
| 14:45.23 | jaytee | morning |
| 14:46.19 | *** join/#asterisk voipmonk (~shido6@dsl-67-204-40-42.acanac.net) |
| 14:46.35 | Katty | hi monmk |
| 14:46.36 | Katty | monk |
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| 14:51.39 | *** mode/#asterisk [+o putnopvut] by ChanServ |
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| 15:04.42 | benngard | any using aastra rfp-32? |
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| 15:26.47 | jaytee | Internet Hitler's reaction to the iPad http://www.youtube.com/watch?v=9_EcybyLJS8&feature=player_embedded |
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| 15:44.55 | Mhaddog | hello |
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| 15:59.56 | p3nguin | Looks like a mess of cardinals on the feeders. |
| 16:00.20 | p3nguin | (and the ground) |
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| 16:08.07 | hackeron | hey, I'm seeing this in syslog: asterisk[2923]: rc_avpair_new: unknown attribute 1490026597 -- what does it mean? |
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| 16:19.09 | *** mode/#asterisk [+o ChanServ] by zelazny.freenode.net |
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| 16:53.42 | *** join/#asterisk tw56 (~tw56@94-195-1-113.zone9.bethere.co.uk) |
| 16:53.55 | tw56 | Hi can anyone help me with the following issue i'm having? http://pastebin.com/m3d1743bb |
| 16:53.55 | wjs | Hello, I was wondering if anyone could help me with a trunk I'm trying to setup |
| 16:54.55 | wjs | I'm not able to get the trunk to get into any context's except the default |
| 16:56.31 | wjs | it is only going to from-sip-external |
| 16:58.06 | p3nguin | Paste your entire sip.conf into a pastebin. |
| 16:59.44 | wjs | k |
| 17:00.21 | wjs | might be a bit tough bc of freepbx |
| 17:02.45 | p3nguin | ~freepbx |
| 17:02.46 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 17:03.29 | Akiraa | Is it possible to do this with asterisk: password individual phones; i.e. when the phone rings, the user is prompted for a p/w, and only then is the call routed |
| 17:04.52 | p3nguin | You want a user to have to enter a password on his ringing phone? |
| 17:05.05 | p3nguin | If you don't get the password right, I can't talk to you? |
| 17:05.16 | Akiraa | p3nguin: yes |
| 17:09.03 | [TK]D-Fender | Akiraa: Of course. |
| 17:09.10 | [TK]D-Fender | Akiraa: "core show application dial" |
| 17:09.14 | wjs | http://pastebin.com/m3aafe970 |
| 17:10.08 | wjs | If needed I'll hop over to freepbx |
| 17:11.09 | [TK]D-Fender | wjs: pastebin a failed call with SIP DEBUG enabled. |
| 17:14.10 | dlynes_laptop | Akiraa, yes, you can do whatever you would like |
| 17:17.27 | wjs | http://pastebin.com/d56e5c725 |
| 17:17.34 | wjs | let me know if thats useful |
| 17:17.44 | wjs | there are a few other trunks that are going up and down |
| 17:17.55 | [TK]D-Fender | wjs: -- Found no matching peer or user for '208.87.41.22:5060' |
| 17:18.09 | [TK]D-Fender | wjs: It isn't even matching your entry... which may also help to have a HOST line |
| 17:18.45 | [TK]D-Fender | wjs: host=sip.ringcentral.com:5060 <- shouldn't shove a port on this |
| 17:18.50 | [TK]D-Fender | (for the other) |
| 17:19.01 | wjs | host=sip.ringcentral.com |
| 17:19.02 | wjs | ok |
| 17:19.17 | [TK]D-Fender | wjs: [17327498416] <- should have a "username=" filed in all likelyhood as well |
| 17:19.56 | *** join/#asterisk mpe (~mpe@x1-6-00-1e-2a-2a-b3-a2.k758.webspeed.dk) |
| 17:20.24 | wjs | changes made, will test again |
| 17:26.39 | [TK]D-Fender | Sofa king coal deer... |
| 17:26.47 | [TK]D-Fender | shivers |
| 17:29.54 | wjs | i'm not making much progress. Thanks for the help, i'll poke around the freepbx channel |
| 17:32.03 | *** join/#asterisk jamesh1 (~jhenderso@xob.neospire.net) |
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| 18:50.55 | ChannelZ | yawns |
| 18:51.37 | *** join/#asterisk joat (~joat@ip70-174-79-200.hr.hr.cox.net) |
| 18:53.49 | *** join/#asterisk Linuturk (~linuturk@unaffiliated/linuturk) |
| 18:54.39 | [TK]D-Fender | tosses a peanut into ChannelZ's gaping maw |
| 18:54.43 | [TK]D-Fender | 3-Pointer! \o/ |
| 18:57.54 | ChannelZ | mmmmm peanutsssss |
| 19:06.57 | *** join/#asterisk Akiraa (~Akiraaaa@79.112.33.11) |
| 19:10.56 | *** join/#asterisk ManxPower-work (~EWieling@216.186.151.147) |
| 19:13.28 | ManxPower-work | ~answers |
| 19:13.29 | infobot | answers is, like, Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: : /path/to/src/asterisk/UPGRADE*.txt |
| 19:18.01 | *** join/#asterisk xpot-mobile (~xpot@70-91-210-233-BusName-Utah.hfc.comcastbusiness.net) |
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| 19:33.46 | *** join/#asterisk ruben23 (~AGENT@122.55.48.243) |
| 19:35.41 | ruben23 | hi guys is it in a bad shape running asterisk with load average of 12.49.. |
| 19:41.29 | *** join/#asterisk louben (~lou@212-70-216-131.ath.static.tee.gr) |
| 19:41.58 | carrar | ruben23, depends, got 32 processors? |
| 19:42.39 | ruben23 | <PROTECTED> |
| 19:43.05 | carrar | why do high? |
| 19:43.08 | carrar | do=so |
| 19:47.27 | ruben23 | carrar:im using AGi, conference and bulk calls |
| 19:47.42 | carrar | running anything else on the box? |
| 19:48.25 | carrar | Are you having audio problems? |
| 19:48.44 | *** join/#asterisk Torrieri (~Torrieri@nelug/crew/torrieri) |
| 19:50.16 | *** join/#asterisk mpe (~mpe@x1-6-00-1e-2a-2a-b3-a2.k758.webspeed.dk) |
| 19:50.20 | ruben23 | carrar:yes...im having...choppy sounds on calls |
| 19:50.44 | carrar | What process is using the most resources? |
| 19:51.01 | carrar | How many concurrent calls? |
| 19:51.10 | carrar | doing transcoding? |
| 19:51.11 | ruben23 | i think the dialer system install. |
| 19:51.26 | ruben23 | im using vicidial predicitive dialer |
| 19:51.28 | *** join/#asterisk QbY (~QbY@c-24-98-101-168.hsd1.ga.comcast.net) |
| 19:51.50 | QbY | if I mark my self as an admin entering a conference, why don't i get any of the menus--specifcally to mute all non-admins??? |
| 19:52.10 | ruben23 | so far my setup is i used softphnes with g729 codec then g729 codec alos on server. |
| 19:52.34 | carrar | So you have to transcode? |
| 19:53.24 | carrar | no hardware transocding modules? |
| 19:53.33 | ruben23 | <PROTECTED> |
| 19:53.53 | carrar | transcoding is gonna hit your system hard |
| 19:54.03 | ruben23 | does my setup transcode..? g729 on user end and g29 on server end aslo on voip end.. |
| 19:54.07 | carrar | if everycall has to do it |
| 19:54.44 | carrar | where is your dialtone coming from? |
| 19:54.46 | ruben23 | there are no other codec used so are there transcoding |
| 19:54.56 | ruben23 | form my asterisk server |
| 19:55.15 | carrar | asterisk server is the box with a load average of 12 righ? |
| 19:55.22 | ruben23 | yes |
| 19:55.31 | carrar | so where is your dialtone coming from |
| 19:55.35 | *** join/#asterisk fofware (~chatzilla@190.7.25.160) |
| 19:55.41 | carrar | aka provider |
| 19:55.53 | ruben23 | form my voip carrier.. |
| 19:56.02 | carrar | are they sending you g.729 |
| 19:56.07 | ruben23 | yes |
| 19:56.11 | carrar | are you sure |
| 19:56.18 | ruben23 | its agreed to used g729 codec |
| 19:56.24 | ruben23 | my channel are open for it |
| 19:56.27 | carrar | you'll need to check |
| 19:56.29 | ruben23 | yeah sure |
| 19:56.41 | carrar | how many concurrent calls |
| 19:57.28 | ruben23 | ill cehck now |
| 19:59.12 | ruben23 | around 60 calls |
| 19:59.18 | carrar | thats not that many |
| 19:59.26 | carrar | might also need to talk to vicidial |
| 19:59.44 | carrar | also maybe see if you can make your AGI better/smaller/faster or compiled |
| 19:59.57 | carrar | How fast of intels? |
| 20:00.20 | ruben23 | 2.4 Ghz. |
| 20:00.52 | carrar | did you vrify you are not trasncoding |
| 20:00.53 | ruben23 | yeah.. |
| 20:01.14 | ruben23 | i will verify it again now |
| 20:01.37 | carrar | also make sure any recordings you play back are also recording as 729 type audio files |
| 20:01.44 | carrar | otherwise you are transcoding to play wav files |
| 20:03.22 | ruben23 | <PROTECTED> |
| 20:03.30 | carrar | there you go |
| 20:03.34 | carrar | thats your killer |
| 20:03.47 | carrar | turn off recording and see what happens |
| 20:03.52 | carrar | as a test |
| 20:04.06 | ruben23 | i need the recordings |
| 20:04.11 | carrar | writting to disk and trasncoding |
| 20:04.27 | ruben23 | so typical my work around is |
| 20:04.34 | ruben23 | increase my hardware specs |
| 20:05.36 | carrar | Why do you have to use 729? |
| 20:06.10 | carrar | recording takes a big hit |
| 20:07.16 | carrar | definately think about going to SCSI 15k RPM drives |
| 20:07.16 | *** join/#asterisk benngard (~benngard@90-230-92-67-no148.tbcn.telia.com) |
| 20:07.28 | carrar | and more processors |
| 20:07.38 | ruben23 | im doing quad core |
| 20:07.48 | ruben23 | xeon quad core |
| 20:07.49 | carrar | dual quad |
| 20:08.06 | ruben23 | are there anyways ways to minimize the effect. |
| 20:08.19 | carrar | What are your IO stats in vmstat? |
| 20:08.32 | ruben23 | carrar: im doing quad core with 2 asterisk load balance. |
| 20:08.45 | carrar | vmstat 1 > binpaste that |
| 20:08.54 | carrar | 20 seconds worth |
| 20:08.57 | ruben23 | carrar: ok |
| 20:09.51 | [TK]D-Fender | Recording is fine.... transocding to do it LIVE is not. |
| 20:09.56 | [TK]D-Fender | ruben23: Change your format |
| 20:09.59 | ruben23 | http://pastebin.com/m8927729 |
| 20:11.30 | carrar | yeah I'd say increase cpu |
| 20:12.10 | carrar | If you can make your software faster |
| 20:12.26 | carrar | might also make sure you've stripped out all the debugging code to make it slightly faster |
| 20:14.09 | ruben23 | <PROTECTED> |
| 20:14.17 | carrar | and lots of cpu being used |
| 20:14.29 | carrar | disc isn't heavy |
| 20:14.46 | ruben23 | <PROTECTED> |
| 20:14.52 | *** join/#asterisk Tim_Toady (~moi@77.49.165.135.dsl.dyn.forthnet.gr) |
| 20:15.05 | carrar | well what makes cpu high is bulky code or code thats slow |
| 20:15.08 | carrar | but yeah |
| 20:15.17 | carrar | see if there are things that can speed up processing |
| 20:15.57 | carrar | more cpu may or maynot fix it |
| 20:16.14 | [TK]D-Fender | ruben23: You aren't showing us the calls.... |
| 20:17.35 | *** join/#asterisk Corydon76-dig (indigo@69.137.80.31) |
| 20:17.35 | *** mode/#asterisk [+o Corydon76-dig] by ChanServ |
| 20:18.27 | ruben23 | [TK]D-Fender:the call logs..? |
| 20:18.34 | ruben23 | ill paste it now |
| 20:21.27 | ruben23 | <PROTECTED> |
| 20:22.25 | ruben23 | http://pastebin.com/m3d8421ac |
| 20:25.05 | [TK]D-Fender | ruben23: [Jan 31 04:21:30] -- Executing [8309@default:2] Monitor("Local/58600051@default-6a45,1", "wav|JEN_20100130-152129_6318648111_TITAN") in new stack <- BAD |
| 20:25.29 | [TK]D-Fender | ruben23: Stop doing this live and your load will drop dramatically |
| 20:26.48 | *** join/#asterisk youngproguru (~youngprog@cpe-76-180-188-78.buffalo.res.rr.com) |
| 20:27.13 | ruben23 | [TK]D-Fender: i will test it, but my client requireds the recordings that the problem.. |
| 20:27.51 | *** join/#asterisk TSM2 (~the_softw@87-194-32-212.bethere.co.uk) |
| 20:29.37 | [TK]D-Fender | ruben23: What part of "DON'T TRANSCODE YOUR RECORDINGS" are you having difficulty with? |
| 20:30.40 | *** join/#asterisk jeffgus (~jeffgus@2002:ad33:b504::1) |
| 20:30.54 | Corydon76-dig | [TK]D-Fender: I'm not sure that transcoding is the problem |
| 20:32.11 | [TK]D-Fender | Corydon76-dig: 60 calls transcoding & recording? |
| 20:32.19 | [TK]D-Fender | Corydon76-dig: I'm sure its significant |
| 20:32.36 | Corydon76-dig | Yes, but the wav format is basically slin |
| 20:32.52 | [TK]D-Fender | Corydon76-dig: Still a G.739 decode on each |
| 20:32.57 | [TK]D-Fender | 729* |
| 20:33.06 | Corydon76-dig | Ah, yeah, that'll do it |
| 20:33.13 | [TK]D-Fender | Corydon76-dig: Far less load to do them individually after the call |
| 20:33.32 | Corydon76-dig | But the recording isn't a problem there. They still have to be transcoded in order to mix. |
| 20:34.15 | [TK]D-Fender | Corydon76-dig: Yes, but 60 at a time all live = bad. You could off-load the transcoding to another machine to "batch" them very easily |
| 20:34.29 | Corydon76-dig | Or use a hardware transcoder |
| 20:34.41 | [TK]D-Fender | Corydon76-dig: that would also help |
| 20:34.56 | Corydon76-dig | Not batch, no. They've gotta be live, in order to mix the sounds in the Meetme |
| 20:35.00 | [TK]D-Fender | Corydon76-dig: My way is probably easy for him to do without investment however ;) |
| 20:35.17 | [TK]D-Fender | Corydon76-dig: MeetMe? Didn't see mention of that... |
| 20:35.45 | Corydon76-dig | I'm seeing a ton of MeetMe in his log. |
| 20:35.48 | [TK]D-Fender | G729 + MeetMe = BAD :p |
| 20:35.57 | [TK]D-Fender | Corydon76-dig: Yeah, I missed the big print there :) |
| 20:36.13 | [TK]D-Fender | ruben23: Get a transcoding card... its worth it now :p |
| 20:36.42 | ruben23 | [TK]D-Fender: any particular brand and model..? |
| 20:36.46 | Corydon76-dig | Or frontend boxes to transcode the sound to ulaw prior to sending to the MeetMe machine |
| 20:37.07 | Corydon76-dig | ruben23: the Digium one is going to be the easiest to integrate into your setup |
| 20:37.11 | carrar | never did say why he needed 729 |
| 20:37.50 | Corydon76-dig | carrar: I'm not going to secondguess his relationship with his customers |
| 20:37.59 | carrar | I am :) |
| 20:38.02 | ruben23 | carrar: i got limited bandwidth only..running in 2Mbps |
| 20:38.19 | Corydon76-dig | Ouch |
| 20:38.30 | carrar | thats not good either |
| 20:38.50 | carrar | You might be hitting your upper bw limit |
| 20:38.54 | ruben23 | thats why i prefer using g729 |
| 20:39.38 | carrar | 31.2*60 |
| 20:39.38 | carrar | 1872.0 |
| 20:39.44 | carrar | thats almost 2 megs |
| 20:39.59 | carrar | assuming you have a true 2 meg circuit |
| 20:41.27 | carrar | but thats not the cause of your high cpu |
| 20:41.29 | *** join/#asterisk mattwj2002 (~Matt@wikisource/pdpc.active.mattwj2002) |
| 20:41.35 | carrar | that will be another issue |
| 20:41.36 | mattwj2002 | hi guys |
| 20:41.56 | mattwj2002 | well I got an asterisk server running on netbook |
| 20:42.33 | ruben23 | <PROTECTED> |
| 20:42.43 | mattwj2002 | it is pretty simple I have callwithus server hooked up to it right now and one softphone....not much to talk about.....eventually I want to hook up a Cisco 7960 to it |
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| 20:42.49 | ruben23 | thats my samlple of my traffic |
| 20:43.14 | carrar | You're graphs are always good aveages |
| 20:43.22 | carrar | but not always exact |
| 20:44.30 | carrar | if you are doing 60 concurrent calls you are reaching the top end of your BW |
| 20:45.04 | mattwj2002 | what do you have? |
| 20:45.07 | mattwj2002 | E1? |
| 20:45.11 | carrar | You have a few things to fix |
| 20:45.22 | carrar | <ruben23> carrar: i got limited bandwidth only..running in 2Mbps |
| 20:45.55 | ruben23 | carrar: ill be increasing to 4Mbps in time.. |
| 20:46.30 | [TK]D-Fender | ruben23: Ok, here's a far better solution : use one * as a border server and do all the transcoding on that and all the ACTUAL work on a second server behind it in ALAW |
| 20:47.29 | carrar | and get that transcoding card on that border server |
| 20:48.45 | ruben23 | [TK]D-Fender: this si not a type of load balance between to asterisk right..? |
| 20:50.50 | [TK]D-Fender | ruben23: Correct. Single border server transcodes all inbound calls and sends to server #2 as ALAW. Your local phones would all be switched to ALAW. Your #2 server can do the meetme & recording rather easily |
| 20:51.23 | [TK]D-Fender | carrar: Not really required... MeetMe was a super-killer. 60 transcodes on a border server isn't a big deal in SW |
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| 20:55.18 | ruben23 | <PROTECTED> |
| 20:56.21 | ruben23 | <PROTECTED> |
| 20:56.21 | [TK]D-Fender | ruben23: that is... if you have the #2 server available already... if it comes to new equipement, the transcoding card idea may well be worth it |
| 20:56.43 | [TK]D-Fender | ruben23: Small task actually... just a question of equiment. The work in setting up = negligable |
| 20:57.24 | ruben23 | [TK]D-Fender: i have two asterisk noe doing load balance, i may change the other one into that setup.. |
| 20:58.03 | *** join/#asterisk catojo (~catojo@189.24.24.35) |
| 20:58.15 | ruben23 | [TK]D-Fender: what they do now, is they both transcode and share load on the concurretn calls generated. |
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| 21:02.03 | [TK]D-Fender | ruben23: Right now if G.729 hits each one doing the MeetMe then each is still getting hit pretty hard. |
| 21:02.57 | ruben23 | [TK]D-Fender:yeah thats the situation now.. |
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| 21:18.24 | p3nguin | [tk]d-fender: To get the call from the border server to the main server, would you simply Dial() to the other server? |
| 21:21.09 | [TK]D-Fender | p3nguin: Just like any other call. |
| 21:22.09 | p3nguin | Just making sure, in case I ever want to try something like that. |
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| 21:33.35 | p3nguin | When is the approprite time to use Transfer() instead of Dial()? |
| 21:35.56 | [TK]D-Fender | p3nguin: When you are completely throwing the other call off of one, and onto another |
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| 21:39.39 | *** join/#asterisk jmcdowell (~nooe@173-112-87-255.pools.spcsdns.net) |
| 21:39.43 | jmcdowell | hello all |
| 21:40.23 | jmcdowell | can anyone tell me, please what a digitmap / dial plan needs to look like to dial 9 to get out, and have it append 1xxx-xxx-xxxx to the SIP provider? |
| 21:40.30 | jmcdowell | I am at a loss at this point. |
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| 21:46.33 | carrar | on the phone use 9,[2-9]xx[2-9]xxxxxx |
| 21:46.42 | carrar | on asterisk append the +1 |
| 21:48.15 | carrar | as well as remove the 9 |
| 21:48.28 | carrar | (if it comes through) |
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| 21:53.49 | jmcdowell | I have to dial 1 314 555 5555 to dial a number |
| 21:53.54 | jmcdowell | will that work? |
| 21:54.15 | jmcdowell | 9,[2-9]xxx[2-9]xxxxxx |
| 21:54.20 | jmcdowell | shouldn't it look like that? |
| 21:54.34 | carrar | no |
| 21:55.01 | carrar | [2-9]xxx would give you 4 diigits |
| 21:55.19 | carrar | first digit being 2-9 |
| 21:55.26 | jmcdowell | that has to be a good read out there on this |
| 21:55.31 | jmcdowell | all the stuff I have read is crap. |
| 21:55.34 | carrar | it |
| 21:55.39 | carrar | it's pretty standard |
| 21:55.41 | carrar | yeah |
| 21:55.52 | jmcdowell | standard or not, I don't have a grasp on it. |
| 21:56.07 | carrar | I think there is a good part in the polycom admin guide |
| 21:56.34 | jmcdowell | So, funny thing.. |
| 21:56.39 | jmcdowell | These are polycom phones |
| 21:56.47 | jmcdowell | and I will never recomend them to anyone agian. |
| 21:56.49 | jmcdowell | again |
| 21:57.28 | jmcdowell | When I try and dial out, it strips off the last 2 digits evertime, even though I erased their dial plan using provisioning. |
| 21:57.41 | carrar | polycom has probably the best phone out there |
| 21:57.44 | carrar | better learn them |
| 21:57.51 | jmcdowell | I love their quality |
| 21:57.57 | jmcdowell | hate their dial plan horse shit |
| 21:58.10 | carrar | You probably have it misconfigured |
| 21:58.24 | ManxPower-work | jmcdowell: Dude, it's not the phone. paste your dialplan |
| 21:58.26 | carrar | have you taken the time to read through the admin guide? |
| 21:58.34 | [TK]D-Fender | No, he DEFINITELY has it misconfigured |
| 21:58.38 | carrar | heh |
| 21:58.42 | QbY | other than meetmeadmin() is there anyway to mute the entire conference--except admins, specifically from a admin user already dialed in's phone? |
| 21:59.01 | ManxPower-work | I asked for the phone dialplan several times yesterday when he was having problems. |
| 21:59.06 | [TK]D-Fender | QbY: Looka t AMI |
| 21:59.29 | jmcdowell | It keeps putting the default plan back in no matter what I do. |
| 21:59.35 | jmcdowell | But here it is.. |
| 21:59.35 | QbY | [TK]D-Fender.. got a link? |
| 21:59.52 | jmcdowell | <PROTECTED> |
| 22:00.06 | ManxPower-work | jmcdowell: What file are you putting that in? |
| 22:00.15 | ManxPower-work | jmcdowell: are you using a GUI for Asterisk? |
| 22:00.45 | [TK]D-Fender | QbY: ... |
| 22:00.46 | jmcdowell | I am using the polycom xml provisioning files to program the phones. |
| 22:00.48 | [TK]D-Fender | ~book |
| 22:00.49 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 22:00.51 | jmcdowell | I can't get past the phones |
| 22:01.01 | ManxPower-work | jmcdowell: yes, but if you put that string in the wrong file it won't be used. |
| 22:01.11 | [TK]D-Fender | jmcdowell: If you ever touched teh web interface, that overrides provisioning |
| 22:01.17 | jmcdowell | BUT I am using FreePBX behind that, but i am not getting past the phone. |
| 22:01.19 | jmcdowell | so hang on |
| 22:01.31 | ManxPower-work | jmcdowell: What version of the Polycom Firmware are you using? |
| 22:01.35 | jmcdowell | I will will tell you what xml file I am putting the dial plan in. |
| 22:01.37 | jmcdowell | 2.2 |
| 22:01.38 | ManxPower-work | jmcdowell: do NOT configure the phone via FreePBX |
| 22:01.41 | [TK]D-Fender | QbY: Better docs in your tarball |
| 22:01.58 | jmcdowell | I can't configure the phone via freepbx that I knwo of. |
| 22:01.59 | ManxPower-work | jmcdowell: Wow, that's like older than dirt. What phone model are you using? |
| 22:02.07 | jmcdowell | 601 |
| 22:02.19 | [TK]D-Fender | ManxPower-work: not that bad |
| 22:02.26 | trentcreek | I installed asterisk from their yum repos, and now asterisk wants to load the modules from /usr/lib64/asterisk/modules, but they are all located in /usr/lib/asterisk/modules. Where is that changed? |
| 22:02.43 | jmcdowell | I am making the change in 0004f21a8803-phone.cfg |
| 22:02.57 | ManxPower-work | jmcdowell: that's the exact file you do not want to be making the changes in. |
| 22:03.07 | jmcdowell | Really.. |
| 22:03.08 | ManxPower-work | that's the one file that the phone will overwrite. |
| 22:03.09 | [TK]D-Fender | jmcdowell: DON'T. That is the manual override file, not provisioning basic |
| 22:03.23 | jmcdowell | Damnet! I read that doc five times |
| 22:03.30 | jmcdowell | I thought for sure I had it right. |
| 22:03.35 | ManxPower-work | jmcdowell: would you like some sample config files? |
| 22:03.47 | jmcdowell | I would DEEPLY appreciate that. |
| 22:04.00 | jmcdowell | I need to read that DOC again thoguh |
| 22:04.04 | ManxPower-work | The ones that I've been writing over the past couple of weeks. |
| 22:04.09 | carrar | should be sip{MAX}.cfg |
| 22:04.12 | ManxPower-work | jmcdowell: standby. |
| 22:04.14 | carrar | not phone{MAC}.cfg |
| 22:04.20 | carrar | err MAC |
| 22:04.21 | ManxPower-work | carrar: it should be whatever it's configured for. |
| 22:04.31 | ManxPower-work | all the config files are specified in 000000000.cfg |
| 22:04.32 | carrar | well, the defaults anyway |
| 22:04.57 | carrar | if didn't massivly change the defaults from polycom |
| 22:04.59 | jmcdowell | my 0000000.cfg is rather small |
| 22:05.34 | jmcdowell | It doesn't contain a-lot |
| 22:05.36 | [TK]D-Fender | <PROTECTED> |
| 22:06.38 | jmcdowell | I read and read and read, but can't seem to wrap my head around this polycom stuff |
| 22:06.58 | carrar | There is always cisco phones ;) |
| 22:07.01 | carrar | heh |
| 22:08.06 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-155-147.cablep.bezeqint.net) |
| 22:10.01 | ManxPower-work | jmcdowell: I'm almost ready to give you the URL |
| 22:10.17 | *** part/#asterisk mattwj2002 (~Matt@wikisource/pdpc.active.mattwj2002) |
| 22:10.32 | jmcdowell | So sip.cfg is the file with all the changes for all the phones. |
| 22:10.40 | jaytee | yes |
| 22:10.49 | ManxPower-work | NO! |
| 22:10.57 | jaytee | no? |
| 22:11.08 | ManxPower-work | sip.cfg should never be changed or it will make future firmware upgrades a total headache |
| 22:11.16 | jmcdowell | xpot"sip.cfg contains parameters which will apply |
| 22:11.17 | jmcdowell | to all phones in an installation such as the preferred voice codec." |
| 22:11.38 | jmcdowell | their documentation sucks |
| 22:11.44 | jaytee | i put overrides to any settings in sip.cfg in my local-settings.cfg file |
| 22:12.17 | jaytee | and that comes before my sip.cfg file in my 000000000000.cfg file in the files section so it gets applied first |
| 22:12.59 | jaytee | and manx is right, you should never make changes to sip.cfg |
| 22:13.47 | jmcdowell | i need a map of what happens when these things try and provision |
| 22:14.04 | jmcdowell | Their own documentation controdicts it's self. |
| 22:14.15 | jaytee | that's in the white paper or the SIP Admin guide, but since you think their documentation sucks that probably won't help |
| 22:15.12 | jmcdowell | So.. From their own documentation.. "Create a local phone file for each phone containing only the parameters that you want to change. For example, 0004f2000607_phone1.cfg" < -- Is that correct? |
| 22:15.26 | ManxPower-work | read the provisioning config file whitepaper |
| 22:15.34 | jmcdowell | That is what I am reading |
| 22:15.38 | jmcdowell | that is where that came from.; |
| 22:15.47 | ManxPower-work | jmcdowell: now look at the examples I just sent you. |
| 22:15.53 | jmcdowell | hang on |
| 22:16.22 | ManxPower-work | jmcdowell: you can't use them direct, they have options only in 3.x firmware |
| 22:17.05 | jmcdowell | OK |
| 22:17.21 | jmcdowell | But I can use the same concept, right |
| 22:17.28 | jmcdowell | the 0000000000.cfg never changes |
| 22:17.38 | ManxPower-work | but pay special attention to the 0000000.cfg. no, the 000000.cfg DOES change |
| 22:17.46 | jmcdowell | and then the <mac>-phone.cfg per phone "changes" |
| 22:18.04 | ManxPower-work | The 000000000.cfg is what tells the phone where the rest of it's config files to load. |
| 22:18.11 | jmcdowell | right |
| 22:18.21 | jmcdowell | but it doesn't have phone specific changes |
| 22:18.36 | jmcdowell | <mac>-phone.cfg per phone "changes" that I would like to make. |
| 22:18.38 | jmcdowell | Right? |
| 22:18.48 | ManxPower-work | you don't need MAC-phone.cfg anymore. |
| 22:18.59 | ManxPower-work | well, at least in reasonably modern bootroms. |
| 22:19.22 | jmcdowell | it's in your tar ball |
| 22:19.28 | jmcdowell | with minimal changes |
| 22:19.28 | *** join/#asterisk tw56 (~tw56@87-194-151-219.bethere.co.uk) |
| 22:19.30 | jmcdowell | but it's there. |
| 22:19.45 | ManxPower-work | from my config file: Phone loads the overide file, then loads the files on order from left to right with the left most file taking highest priority |
| 22:20.07 | tw56 | Hey people. Can anyone help me with an misdn issue i have with a quadbri card. For some reason the connection hangs when i try to call out and i can't figure out why |
| 22:20.12 | *** join/#asterisk riddlebox (~james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 22:20.17 | ManxPower-work | local-sip.cfg I use for things specific to that PBX. pbxmax-sip.cfg I use for stuff I want to change from the defaults for ALL PBXs |
| 22:21.09 | ManxPower-work | jmcdowell: you can name the config files almost anything you want. The phone knows what the file names are because you specify them in 0000000000.cfg |
| 22:23.38 | jmcdowell | right and it will not overwrite your <mac>-phone1.cfg |
| 22:23.40 | jmcdowell | right? |
| 22:27.45 | tw56 | does anyone have any experience with misdn? |
| 22:37.11 | *** part/#asterisk QbY (~QbY@c-24-98-101-168.hsd1.ga.comcast.net) |
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| 22:46.47 | tw56 | I have figured out the outgoing but for some reason incomming doesn't work |
| 22:46.51 | tw56 | i get the problem "received 1k Unhandled Bchannel Messages" |
| 22:46.55 | tw56 | on the port |
| 22:46.57 | tw56 | any ideas? |
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| 22:53.36 | trentcreek | I installed asterisk from their yum repos, and now asterisk wants to load the modules from /usr/lib64/asterisk/modules, but they are all located in /usr/lib/asterisk/modules. Where is that changed, or can I just copy them over? |
| 23:01.47 | *** join/#asterisk infobot (ibot@rikers.org) |
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| 23:04.44 | trentcreek | I installed asterisk from their yum repos, and now asterisk wants to load the modules from /usr/lib64/asterisk/modules, but they are all located in /usr/lib/asterisk/modules. Where is that changed, or can I just copy them over? |
| 23:06.04 | *** mode/#asterisk [+cr] by niven.freenode.net |
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| 23:06.18 | *** mode/#asterisk [+q CRCinAU!*@*] by niven.freenode.net |
| 23:06.18 | *** topic/#asterisk by niven.freenode.net -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.2.1 (2010/01/15), Asterisk 1.6.1.13 (2010/01/15), 1.6.0.21 (2010/01/15), 1.4.29 (2010/01/15), *-Addons 1.6.2.0 (2009/12/18), 1.6.1.2 (2009/12/02), 1.6.0.4 (2009/12/02), 1.4.10 (2009/12/02), dahdi-linux 2.2.0.2 (2009/07/23), dahdi-tools 2.2.0 (2009/06/24), Libpri 1.4.10.2 (2009/10/20) -=- Related channels |
| 23:06.35 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 23:06.41 | *** join/#asterisk ruben23 (~AGENT@122.55.48.243) |
| 23:07.02 | ruben23 | hi what happened..? |
| 23:07.49 | ChannelZ | network throwup |
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| 23:12.50 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 23:13.04 | ruben23 | wehre are the big guys..? |
| 23:16.19 | Corydon76-dig | Big guys? Fetish? |
| 23:16.22 | p3nguin | I guess that depends on whom you are referring to. |
| 23:18.01 | ruben23 | there you are guys..:-D |
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| 23:20.23 | ruben23 | asterisk experts i mean.. |
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| 23:20.47 | ChannelZ | whatever that is |
| 23:21.09 | insanet[1] | dispositionstring does not work with cdr_mysql? |
| 23:22.52 | Corydon76-dig | Nice. Ask a question, then don't stick around for the answer |
| 23:26.24 | ruben23 | http://blog.freenode.net/ |
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| 23:30.50 | insanet[1] | does dispositionstring=no in cdr_mysql is working with asterisk 1.4 ? or it was never intended to work with cdr_mysql ? |
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| 23:33.09 | p3nguin | Does A2Billing take over confs similar to how FreePBX does? |
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| 23:41.35 | dlynes_laptop | p3nguin, most billing softwares (not familiar with a2billing) usually require you to install a custom module as well |
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| 23:41.53 | dlynes_laptop | p3nguin, but taking over the configs is not usually something they do |
| 23:42.20 | p3nguin | I didn't notice any module, but I didn't specifically look for one, either. |
| 23:46.21 | dlynes_laptop | p3nguin, yeah...I just remember running into that with at least one billing system |
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| 23:46.43 | dlynes_laptop | p3nguin, i think it might have even been a2billing...but I tried a2billing about 3 years ago, when it wasn't terribly useful |
| 23:47.06 | dlynes_laptop | p3nguin, a lot can change in three years |
| 23:47.22 | p3nguin | I'm wanting to put it on my personal system, but I'm not wanting to break anything by doing so. |
| 23:47.55 | dlynes_laptop | p3nguin, only way to be 100% guaranteed of that, without a shadow of a doubt, is to write your own, then :) |
| 23:48.13 | p3nguin | I'm not a programmer, so that won't be happening. |
| 23:48.14 | dlynes_laptop | but that's generally not practical... |
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