| 00:06.08 | *** join/#asterisk coppice (~chatzilla@m121-202-84-127.smartone-vodafone.com) |
| 00:07.05 | *** join/#asterisk manunic (~manu_loca@pool-74-102-58-60.nwrknj.fios.verizon.net) |
| 00:07.21 | *** join/#asterisk retentiveboy (~pdugas@69.169.199.82) |
| 00:11.01 | *** join/#asterisk pabelanger-lap (~pabelange@CPE0013f7abc09a-CM0013f7abc096.cpe.net.cable.rogers.com) |
| 00:11.26 | manunic | I am using digium Asterisk Appliance 50. My problem is that Users(Extensions) can not call each other. Outgoing calls work, Incoming calls to any extensions work. Please advice on where to look |
| 00:16.43 | *** join/#asterisk mindCrime (~chatzilla@cpe-075-189-213-049.nc.res.rr.com) |
| 00:19.28 | manunic | I am seeing this activity online |
| 00:19.36 | manunic | (i mean on asterisk cli) |
| 00:20.23 | manunic | <--- Transmitting (NAT) to 192.168.1.105:5060 ---> |
| 00:20.23 | manunic | SIP/2.0 489 Bad event |
| 00:20.23 | manunic | v: SIP/2.0/UDP 192.168.1.105:5060;branch=z9hG4bK-79781fb2;received=192.168.1.105 |
| 00:20.23 | manunic | f: "105" <sip:105@192.168.1.225>;tag=d53a1bd75db675cao0 |
| 00:20.23 | manunic | t: <sip:192.168.1.225>;tag=as4cfdc1b8 |
| 00:20.24 | manunic | i: d9ceeafb-463150f6@192.168.1.105 |
| 00:20.24 | manunic | CSeq: 72 NOTIFY |
| 00:20.25 | manunic | Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY |
| 00:20.25 | manunic | k: replaces |
| 00:20.26 | manunic | l: 0 |
| 00:20.26 | manunic | <------------> |
| 00:20.37 | manunic | What is a "bad event"? |
| 00:20.47 | manunic | My phones are on 1.105 |
| 00:25.58 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 00:25.58 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 00:42.04 | *** join/#asterisk retentiveboy (~pdugas@69.169.199.82) |
| 00:43.06 | pabelanger-lap | manunic: Don't spam the channel, use PB |
| 00:43.37 | pabelanger-lap | manunic: 2nd, disable NAT on your phone, if you are on the same network |
| 00:43.43 | p3nguin | s/spam/flood/ |
| 00:44.44 | manunic | pabelanger-lap: thank you, also what is PB |
| 00:45.05 | pabelanger-lap | ~pb |
| 00:45.06 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 00:46.52 | *** join/#asterisk Alton35 (~alton@69.45.116.128) |
| 00:52.53 | *** join/#asterisk mattwj2002 (~Matt@wikisource/pdpc.active.mattwj2002) |
| 00:53.02 | mattwj2002 | hi guys |
| 00:53.46 | mattwj2002 | is there a website where you can do a mass search for a specific number through different providers? |
| 00:53.55 | mattwj2002 | for DIDs |
| 00:56.09 | p3nguin | That would be kinda neat. |
| 00:56.43 | *** join/#asterisk bandu (~coyotama@pool-71-164-233-190.dllstx.fios.verizon.net) |
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| 00:57.24 | mattwj2002 | indeed |
| 00:57.27 | mattwj2002 | :) |
| 00:57.28 | *** join/#asterisk dlynes (~dlynes@bas6-hamilton14-1176462822.dsl.bell.ca) |
| 01:13.07 | mattwj2002 | hey I have a question |
| 01:13.25 | mattwj2002 | for a particular town |
| 01:13.54 | mattwj2002 | is it possible that all of their voip numbers are different than all of their landlines? |
| 01:14.00 | mattwj2002 | *numbers |
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| 01:38.48 | *** join/#asterisk raden (~J@66-168-15-100.dhcp.stpt.wi.charter.com) |
| 01:44.09 | raden | morning katty :) |
| 01:44.15 | raden | except its night time dear |
| 01:46.16 | *** join/#asterisk Kumbang (~kumbang@rusnas.paume.itb.ac.id) |
| 01:53.55 | raden | I have a Syba 6 port serial card and lspci is showing it but i do not believe its setting to serial ports http://pastebin.com/9xjt5PC4 |
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| 02:01.37 | p3nguin | mattwj2002: no |
| 02:02.07 | p3nguin | mattwj2002: Well, possible, but totally not probable. |
| 02:02.55 | p3nguin | There would have to be some serious restrictions by a single phone company, and we know that isn't the case. |
| 02:03.56 | mattwj2002 | so p3nguin |
| 02:04.20 | mattwj2002 | I can't get a number from a local telephone company but I can port it? that doesn't make sense? |
| 02:05.17 | p3nguin | If you get a phone number from a phone company, you can port it out to another company. The company could be a VoIP provider. |
| 02:05.38 | *** join/#asterisk xuser (~xuser@unaffiliated/xuser) |
| 02:05.50 | mattwj2002 | but how do I get a phone number from a company if I don't live in that area? |
| 02:06.18 | p3nguin | Look in the pool of numbers from VoIP service providers. |
| 02:06.26 | mattwj2002 | I have no luck |
| 02:06.36 | p3nguin | Call some providers and ask them for specific area code and prefix. |
| 02:06.42 | p3nguin | What number are you looking for? |
| 02:06.52 | mattwj2002 | 715532XXXX |
| 02:07.24 | mattwj2002 | centurylink is the provider |
| 02:07.46 | p3nguin | Ladysmith, WI? |
| 02:07.51 | mattwj2002 | correct |
| 02:08.06 | raden | p3nguin, what up |
| 02:08.44 | raden | mattwj2002, what kinda number you need ? |
| 02:08.54 | raden | mattwj2002, you in wisconsin ? |
| 02:08.59 | raden | central i take it |
| 02:09.04 | mattwj2002 | actually I mean in MN |
| 02:09.17 | mattwj2002 | I want a SIP number |
| 02:09.17 | p3nguin | mattwj2002: VoIP.ms has Ladysmith numbers on backorder. Call them and ask what prefix they can get for you. |
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| 02:09.26 | mattwj2002 | okay |
| 02:09.29 | raden | mattwj2002, lemee see what i have for ladysmith |
| 02:09.35 | mattwj2002 | sounds good |
| 02:09.43 | mattwj2002 | a lot of providers have 715 403 |
| 02:09.44 | raden | p3nguin, i can tell him right now I have direct access |
| 02:09.54 | p3nguin | That's ever faster. |
| 02:10.03 | mattwj2002 | I have a specific reason I want 715 532 |
| 02:10.09 | raden | just waiting for portal to load |
| 02:11.33 | raden | looking |
| 02:11.43 | mattwj2002 | thanks raden |
| 02:11.44 | mattwj2002 | :) |
| 02:11.53 | p3nguin | Flowroute has some 715, but no 532. |
| 02:12.09 | raden | I can get tons of 715 numbers |
| 02:12.11 | raden | im 715 |
| 02:12.34 | raden | i have 1,000's available |
| 02:12.36 | p3nguin | Only prefixes I see in 715 are 254, 318, 942, 953 |
| 02:12.38 | raden | just getting 532 |
| 02:12.48 | raden | 942 vonage |
| 02:12.56 | raden | suprised a person can get it |
| 02:13.05 | *** join/#asterisk Mhaddog_Mac (~anonymous@adsl-233-199-204.mia.bellsouth.net) |
| 02:13.17 | p3nguin | two available. |
| 02:13.19 | raden | 350 |
| 02:13.28 | raden | 257 |
| 02:13.29 | mattwj2002 | what you can get a 5320? |
| 02:13.32 | mattwj2002 | oops |
| 02:13.36 | mattwj2002 | *532 |
| 02:14.07 | raden | 861 |
| 02:14.16 | raden | p3nguin, there lots available :) |
| 02:14.23 | raden | mattwj2002, looking |
| 02:14.31 | mattwj2002 | okay |
| 02:14.35 | mattwj2002 | :) |
| 02:14.56 | p3nguin | How do you get direct access to the pool listings? |
| 02:15.03 | raden | you need a 532 specific ? |
| 02:15.21 | mattwj2002 | any 715532XXXX will work |
| 02:15.41 | raden | p3nguin, Im a carrier |
| 02:15.48 | mattwj2002 | :O |
| 02:15.50 | mattwj2002 | :D |
| 02:16.10 | *** join/#asterisk aidinb (~Aidin@32.174.36.183) |
| 02:16.30 | mattwj2002 | easy to remember would be nice |
| 02:16.35 | raden | p3nguin, but i dont have the database app done to search by anything more than area code |
| 02:16.36 | mattwj2002 | but I am not picky |
| 02:16.37 | *** part/#asterisk rodrigonmap (~rodrigo@189-015-108-100.xd-dynamic.ctbcnetsuper.com.br) |
| 02:17.04 | raden | if you want 532 you will have to contact CTC communications in ladysmith and see if they will port the number out to another provider ( doubtful ) |
| 02:17.17 | mattwj2002 | bummer :( |
| 02:17.26 | raden | specific reason for 532 ? |
| 02:17.36 | mattwj2002 | yeah real simple |
| 02:17.51 | raden | ??? |
| 02:17.54 | mattwj2002 | my dad lives there |
| 02:18.05 | raden | so you want something local for 532 ? |
| 02:18.05 | mattwj2002 | he was in the hospital recently |
| 02:18.12 | raden | you use asterisk ? |
| 02:18.23 | mattwj2002 | and you could only dial out for 715532XXXX |
| 02:18.26 | raden | p3nguin, would you like access ? |
| 02:18.30 | mattwj2002 | no other local numbers! |
| 02:18.39 | raden | what about 800 # ? |
| 02:18.43 | mattwj2002 | oh |
| 02:18.50 | mattwj2002 | I hadn't thought about that! |
| 02:18.51 | mattwj2002 | :D |
| 02:18.57 | raden | *duh* |
| 02:19.00 | raden | 1.95 mo |
| 02:19.04 | raden | it cheap |
| 02:19.34 | raden | then u call him back save coin :) |
| 02:19.43 | mattwj2002 | :) |
| 02:19.44 | raden | or just do what i have at office |
| 02:20.00 | raden | when sales reps call in on 800 # system knows it and hands up and calls there caller id back |
| 02:20.21 | mattwj2002 | nice |
| 02:20.23 | mattwj2002 | :) |
| 02:21.23 | raden | p3nguin, u got a moment to help me get my 6 port PCI card working ? |
| 02:22.07 | mattwj2002 | thanks raden for your efforts |
| 02:22.11 | mattwj2002 | thanks everyone else too |
| 02:22.14 | raden | no prob |
| 02:22.15 | mattwj2002 | I am off to bed |
| 02:22.17 | mattwj2002 | :D |
| 02:22.20 | raden | get a 800 # |
| 02:22.23 | raden | there handy |
| 02:22.26 | mattwj2002 | will do |
| 02:22.33 | mattwj2002 | bye! |
| 02:22.34 | mattwj2002 | :) |
| 02:22.39 | *** part/#asterisk mattwj2002 (~Matt@wikisource/pdpc.active.mattwj2002) |
| 02:40.41 | *** join/#asterisk Majost (~majost@f00kie-1-pt.tunnel.tserv8.dal1.ipv6.he.net) |
| 02:42.24 | Majost | For the Dial() command, is there a value for indefinate timeout, other than specifing one? Eg: Dial(SIP/100,*,X) |
| 02:44.22 | Majost | I want the extention to ring until the caller hangs up (or google voice transfers to voicemail in this case) -- but I also want to be able to use call recording on my asterisk box... I got an error doing Dial(SIP/100,X) "app_dial.c:1847 dial_exec_full: Invalid timeout specified: 'X'. Setting timeout to infinite" |
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| 03:02.30 | dongs | how do i add + to extension? |
| 03:12.07 | dongs | i guess i just do. |
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| 04:04.22 | seanjohn | is the parameter rfc2833compensate or which is it? |
| 04:08.56 | seanjohn | if anyone is using sip, would they mind testing to see if they can register on sip.sheltoncomputers.com? |
| 04:09.04 | seanjohn | pm me for credentials |
| 04:10.53 | *** join/#asterisk Mish- (~mish@194.28.69.111.dynamic.snap.net.nz) |
| 04:11.09 | Mish- | I am about to build a home PBX, should I use 1.4 or 1.6? |
| 04:11.35 | *** join/#asterisk _mwoodj_ (~mwoodj@pdpc/sponsor/digium/hyper-eye) |
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| 04:23.59 | seanjohn | 1.4 is the mature version |
| 04:24.11 | *** join/#asterisk raden (~J@66-168-15-100.dhcp.stpt.wi.charter.com) |
| 04:24.15 | seanjohn | 1.6 works just as well but, depending on what you want to do, it may not work |
| 04:24.48 | seanjohn | use of 3rd party applications or etc |
| 04:25.03 | raden | seanjohn, i never have any issues but then again using a stable branch |
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| 04:27.35 | Alton35 | I'd use the 1.6 since there are a few bells and whistles. It's a danger to use the latest version of asterisk, but you'd think you could find something in 1.6 that would make you happy. |
| 04:58.43 | seanjohn | alton, there are 3 1.6 version |
| 04:59.18 | seanjohn | the 1.6.2 is NOT suggested, the 1.6.1 is possible production, and 1.6.0 is your 1.6 "fix" |
| 05:00.02 | seanjohn | I use 1.4 as I have no need for any new features of 1.6 |
| 05:02.58 | Alton35 | yes, I am aware of the differences, I was just making a general observation for the fellow |
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| 05:06.40 | *** join/#asterisk pinoyskull (~pinoyskul@122.55.80.205) |
| 05:09.00 | Mish- | I have asterisk installed, if I go to the web gui I get the "Asterisk Configuration Panel" but no matter what password/username I use it shows a red "Loading" in the top right corner then does nothing, just sits at the login screen. |
| 05:11.26 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 05:13.59 | kaldemar | Mish-: what exactly do you have installed? asterisk doesn't have a web gui. |
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| 05:16.54 | Mish- | package under suse called "asterisk-gui" |
| 05:18.19 | Mish- | "/var/lib/asterisk/static-http" |
| 05:18.53 | Mish- | "Asterisk-GUI - an Asterisk configuration interface" - "Copyright (C) 2008, Digium, Inc." |
| 05:21.16 | kaldemar | Mish-: i strongly encourage you not to use that. afaik, it is not maintained anymore. |
| 05:22.14 | DogBoy | not to mention gui is for sissies |
| 05:22.30 | DogBoy | a layer of bs |
| 05:22.43 | kaldemar | if you want a GUI, ask here for instructions: #freepbx |
| 05:24.15 | *** join/#asterisk soman (~somnath@118.102.130.6) |
| 05:24.43 | DogBoy | weee, here we go, gonna see if I can get it to work on an nslu2 running debian |
| 05:24.46 | Mish- | Right, understood. |
| 05:28.25 | DogBoy | debian is so weird sometimes |
| 05:28.35 | DogBoy | what do I need with gcc when installing asterisk |
| 05:29.06 | kaldemar | g++ |
| 05:29.41 | DogBoy | Unpacking gcc (from .../gcc_4%3a4.3.2-2_arm.deb) ... |
| 05:32.43 | DogBoy | and g++ but the question is the same |
| 05:33.40 | DogBoy | hmm, already it's not gonna work |
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| 06:47.36 | sawgood | What is a 'good' approach for using Asterisk to support a 'presence' system |
| 06:47.50 | sawgood | clients want more than just a DND option on the phone |
| 06:49.22 | sawgood | clients want to go to their SIP end points, and set a presence option (like out of office, at lunch, in a meeting, etc) |
| 06:49.47 | sawgood | depening on their presence choice, the dialplan would be changed (out to cell phone, etc.) |
| 06:51.30 | carrar | make it a menu where they set options that are stored in a db with routing also in a db |
| 06:51.45 | carrar | or a XML interface on the phone |
| 06:52.26 | sawgood | XML interface on a phone is a cool choice (some phones have progammable XML soft keys) ... very interesting approach |
| 06:52.35 | carrar | xml being the better choice |
| 06:52.40 | *** join/#asterisk hehol (~hehol@buero-gw.dortmund.loca.net) |
| 06:54.24 | sawgood | I know the Grandstream GXP2010 (known as their key system phone) has 4 or so programmable XML softkeys |
| 06:54.32 | sawgood | I will call them tomorrow to see how I can do this maybe |
| 06:55.44 | carrar | or get better phones |
| 06:59.11 | Alton35 | hah |
| 06:59.18 | Alton35 | polycom? cheap but reliable |
| 07:01.31 | *** join/#asterisk waschtl (~waschtl@3ed8a58a.d.d9tcloud.de) |
| 07:01.40 | *** join/#asterisk elliot98 (~elliot@unaffiliated/elliot98) |
| 07:02.13 | elliot98 | is there a channel variable to get the ip address of the incoming sip packet? |
| 07:03.09 | carrar | SIPCHANINFO(peerip) or SIPCHANINFO(recvip) |
| 07:03.53 | *** join/#asterisk stix (~stix@exchange2003.corporate.billetkontoret.dk) |
| 07:03.57 | elliot98 | thanks! |
| 07:04.12 | elliot98 | basically, for some reason the server stopped authenticating by ip address... |
| 07:04.21 | elliot98 | it's throwing everything into the default context |
| 07:04.31 | carrar | fix it |
| 07:04.39 | elliot98 | why does that happen? |
| 07:05.02 | carrar | user error probably |
| 07:05.06 | sawgood | Is this approach 'common' Asterisk box --- SIP trunk ----- SBC ---- IAX trunk --- ITSP |
| 07:05.21 | *** join/#asterisk lowlevel (~Stuart@lowlevel.ca) |
| 07:05.40 | elliot98 | nothing changed when the issue started |
| 07:06.05 | elliot98 | if I set the host in sip.conf for a client, it should be enough for ip authentication? |
| 07:06.07 | sawgood | Or even this ... Asterisk box ---- G.711 SIP trunk ---- SBC ---- G.729 SIP trunk --- ITSP --- PSTN |
| 07:12.56 | elliot98 | what is more appropriate peerip or recvip? |
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| 07:31.44 | elliot98 | what is the fromuser and fromdomain? |
| 07:33.36 | *** join/#asterisk [sr] (~Unknowned@pa8-84-91-197-8.netvisao.pt) |
| 07:33.38 | [sr] | howdy |
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| 07:34.11 | [sr] | is there a way to tell dahdi to NOT configure ISDN cards's, when i have both, POST & ISDN? |
| 07:38.28 | WIMPy | Load the other drivers first? Blacklist the dahdi parts you don't want? |
| 07:39.33 | [sr] | hi WIMPy :) |
| 07:39.42 | [sr] | hum... it's possible to blacklist cards on dahdi side? |
| 07:39.56 | [sr] | my ideia is to use LCR for ISDN, DAHDI for POST |
| 07:39.59 | WIMPy | How is yorfax doing? |
| 07:40.11 | WIMPy | modules.d |
| 07:40.21 | WIMPy | Or just don't build them. |
| 07:40.35 | [sr] | hum have to search about it |
| 07:40.40 | [sr] | about the fax, havent done yet |
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| 07:46.16 | [sr] | WIMPy: when you say modules.d, the system one, or dahdi has one too? |
| 07:46.33 | WIMPy | system |
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| 07:46.55 | [sr] | hum |
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| 07:51.22 | [sr] | im thinking |
| 07:52.10 | [sr] | the module to blacklist is the module name for dahdi, for LCR, it wont use it, since it's gonna use isdn stuff on kernel |
| 08:03.02 | tzafrir_laptop | [sr], certainly not "dahdi". Blacklisting a high-level module will not do you any good |
| 08:03.21 | tzafrir_laptop | You man want to blacklist wcb4xxp |
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| 08:22.53 | [sr] | tzafrir_laptop: that, only the dahdi module |
| 08:23.21 | tzafrir_laptop | [sr], blacklisting dahdi does nothing |
| 08:23.39 | tzafrir_laptop | blacklisting a module tells modprobe to ignore all of its aliases |
| 08:23.46 | [sr] | tzafrir_laptop: the dahdi module, the specific card module |
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| 09:10.47 | skatz | hi all, I need some help on how to set my CDR(billsec). I have an IVR which brgide the caller to an external number and I need to track the time from when the destination answers and the final hangup. Does anyone know how I can make this? |
| 09:12.08 | WIMPy | I guess ForkCDR would be your friend. Probably with M(). |
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| 09:13.22 | skatz | WIMPy, is there a chance to get it without creating a new CDR entry? |
| 09:16.06 | WIMPy | Ugh, can't remember exactely, but there might be something to reset the time. But it might be easier to just skip the unwanted CDRs. |
| 09:16.44 | skatz | WIMPy, ok many thanks for your help |
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| 09:40.32 | elliot98 | if I want all calls coming from a certain IP address to be associated with a certain user, I should set the insecure parameter? |
| 09:40.49 | vlt | Hello. Is there a way to "redirect" a call that is currently ringing on one extension to another one from the CLI? |
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| 09:41.05 | geemee | Hi there. |
| 09:41.20 | geemee | Curious if someone could assist in troubleshooting setting up a sip trunk. |
| 09:41.37 | geemee | Newish to astersik, ISDN's are fine but possibly moving to pureVOIP |
| 09:41.54 | WIMPy | vlt: Channel redirect |
| 09:43.02 | vlt | WIMPy: Thanks |
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| 09:43.48 | vlt | WIMPy: no such command |
| 09:44.20 | WIMPy | Time for an upgrade? |
| 09:45.33 | vlt | WIMPy: Ah ok, it's available on my 1.6 test setup but not on Debian's 1.4 ... |
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| 10:15.27 | elliot98 | Do I need to add the register statement in sip.conf if the server will only be using the remote trunk for placing calls, but not receiving them? |
| 10:15.58 | WIMPy | no |
| 10:19.57 | xheliox | rubs his eyes |
| 10:20.04 | xheliox | nothing suckier than a carrier problem at 4am |
| 10:35.56 | Dovid | xheliox: What's with back up routes and welcome to my life |
| 10:39.17 | xheliox | Dovid: Yeah, it's suppose to fail to another circuit or SIP trunk, and it's not. They're down down. Took me 10 minutes to even get on the phone with them -- all circuits busy to their call center. |
| 10:40.46 | *** join/#asterisk KrisW (~kris@cpc4-swin14-2-0-cust29.know.cable.virginmedia.com) |
| 10:42.43 | KrisW | So I upgraded my asterisk box from Ubuntu 9.10 to 10.04, and due to the kernel change, I had to rebuild asterisk / dahdi etc. Asterisk is running fine, but I keep getting "wcfxo: error" when starting dahdi, and as a result am unable to pick up PSTN calls, does anyone have any suggestions as to what I could try? |
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| 10:46.47 | KrisW | Here is what I am getting when (re)starting dahdi: http://pastie.org/1035741 |
| 10:48.00 | KrisW | Though, lsmod reports that it is loaded: http://pastie.org/1035742 |
| 10:55.07 | Dovid | does anyone know if in features .conf I can have more than one code for disconnect ? |
| 10:55.34 | Dovid | xheliox: Always fun. |
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| 11:19.49 | tuxx- | hey guys, is it possible to let dahdi pick a random channel for outgoing calls? Instead of always picking the highest number channel available? |
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| 11:38.49 | VEc | Hi guys, I'm having a odd problem where the CDR is not being written to when a user is put in the queue and then the queue times out and continues on the dial plan, please, see http://pastebin.ca/1896606 |
| 11:43.35 | tzafrir_laptop | tuxx-, not random, but use r/R instead of g/G |
| 11:43.57 | tuxx- | tnx! |
| 11:44.04 | tzafrir_laptop | This uses (Reversed?) round-robin order |
| 11:44.11 | tuxx- | gonna look it up |
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| 12:03.39 | Dovid | morning TK |
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| 12:11.14 | ariel_ | Morning |
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| 12:37.09 | geemee | Hi Again, Can anyone assits with my previous request? Looking for troubleshooting advice for setting up siprunk |
| 12:37.13 | geemee | sorry sip trunk |
| 12:38.42 | pabelanger | ~ask |
| 12:38.43 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 12:38.46 | pabelanger | geemee: ^^^ |
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| 12:44.27 | geemee | I have a virtual machine running latest Trixbox CE. I have 3 extensions. 2 physical and 1 softphone which operate correctly. Ports are open on firewall and can all communicate / register. I have been told SIP trunk is setup but ISP only have given me an ip address of their server, no secret or username. I have entered the trunk and routes but can make external calls. Getting All circuits are busy. Can SIP work without username / secret? I personally |
| 12:44.28 | geemee | think ISP giving false info. |
| 12:45.27 | [TK]D-Fender | geemee: that message i meaningless. |
| 12:45.36 | [TK]D-Fender | geemee: PASTEBIN a failed call with SIP DEBUG enabled. |
| 12:45.40 | [TK]D-Fender | ~pb |
| 12:45.40 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 12:45.42 | [TK]D-Fender | ^^^ |
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| 12:52.27 | geemee | http://pastebin.com/GrAZ1C0A |
| 12:52.45 | geemee | hopefully this should be enough info. Getting circuit-busy straight away |
| 12:55.59 | [TK]D-Fender | geemee: That isn't the ENTIRE CALL, nor do I see any hint of SIP DEBUG being enabled |
| 12:56.21 | [TK]D-Fender | geemee: Try again |
| 12:56.37 | geemee | D-Fender: looking into it now. |
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| 13:42.53 | snapple42 | morning all.. anyone here use dahdi_dynamic_loc? |
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| 13:45.16 | *** join/#asterisk Tagor (~none@s55928c6d.adsl.wanadoo.nl) |
| 13:46.10 | Tagor | Someone got in my Asterisk box. I just changed the password, put him in the firewall and reloaded asterisk. But like one minute later someone else logged in. Any idea how they get in? Cause I changed the password to something very difficult |
| 13:46.29 | Tagor | I'm running 1.4.21.2 |
| 13:47.23 | Tagor | Actually the server is pretty well secured. Looks like there some leak in Asterisk cause they didn't get access to other parts of the server. My asterisk recorded all calls and they didn't change any configs |
| 13:47.36 | *** join/#asterisk hugorebelo (~hugo@200-171-132-124.completo.com.br) |
| 13:48.15 | pabelanger | Tagor: Exploit in your OS? IE: ssh, telnet |
| 13:48.15 | Nugget | telnet is eeeeeeevil! |
| 13:48.34 | pabelanger | Tagor: Upgrade your OS |
| 13:48.54 | Tagor | pabelanger: no telnet access, SSH only allows access to 1 ip |
| 13:49.07 | Tagor | pabelanger: OS is upgraded to the latest stable version |
| 13:49.08 | pabelanger | Tagor: physical access? |
| 13:49.21 | fauxalliance | Tagor, use OpenBSD... |
| 13:49.46 | Tagor | pabelanger: no physical access. Looks like they only got access to that certain extension on Asterisk |
| 13:49.48 | snapple42 | Tagor: rootkit... check out http://www.rootkit.nl/projects/rootkit_hunter.html |
| 13:50.58 | Tagor | snapple42: no rootkits, it checks automatically every day and I just did a manual check too |
| 13:51.52 | Tagor | This *ss is still logged in. I just changed my config so when he calls out I get him on the line |
| 13:51.56 | [TK]D-Fender | Tagor: Block out their IP range. Pick stronger passwords. use fail2ban, etc |
| 13:52.53 | fauxalliance | unplug the network cable until you figure it out even... |
| 13:53.03 | Tagor | [TK]D-Fender: well that would be an option, but I prefer to know how he got in. I picked a password like fk34gGf5Gd92grr4 .. so that shouldn't be easy to bruteforce |
| 13:53.20 | fauxalliance | Tagor, again, unplug, preserve state and start diggin' |
| 13:53.38 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 13:54.04 | Tagor | fauxalliance: why unplug? installing another OS isn't really a solution... |
| 13:55.02 | fauxalliance | well, there is a hacker attached to your computer... you decided whether to leave him there or use an OS with virtually _no_ attack surface... |
| 13:55.04 | tzafrir_laptop | snapple42, no idea. What do you need it for? |
| 13:55.12 | [TK]D-Fender | Tagor: And there have been security holes that far back... you are VERY behind on things |
| 13:55.54 | Tagor | [TK]D-Fender: you mean with this Asterisk version? Or you mean with the OS? |
| 13:56.01 | fauxalliance | probably both |
| 13:56.59 | fauxalliance | most likely the os.. the bugs in that partucular asterisk version are _well documented_ |
| 13:57.03 | [TK]D-Fender | Tagor: * |
| 13:57.25 | Tagor | [TK]D-Fender: as far as I can see they only have access to a certain extension, not to the system. Would also be a pretty junior hacker that doesn't delete his recorded calls, call logs, etc. |
| 13:57.37 | Tagor | [TK]D-Fender: thanks, will manually install a newer version |
| 13:58.02 | [TK]D-Fender | Tagor: oManually as oppposed tooo? |
| 13:58.56 | Tagor | [TK]D-Fender: using apt-get, instead compile the latest version manually |
| 13:59.04 | fauxalliance | Tagor, you cant assume it was asterisk either... that sounds pretty 'junior' |
| 14:00.00 | tzafrir_laptop | Tagor, what distro is it? |
| 14:00.08 | Tagor | fauxalliance: on this server like anything is blocked out except some sip ports |
| 14:00.59 | *** join/#asterisk jpeeler (~jpeeler@asterisk/digium-software-dev/jpeeler) |
| 14:01.22 | tzafrir_laptop | Tagor, how exactly can you tell that someone got into the Asterisk box? |
| 14:01.40 | fauxalliance | Tagor, I am not disputing that... If it were my hacked boxen... I'd be trying to learn as much as possible from this... |
| 14:01.54 | fauxalliance | funny CDR records i assume.. |
| 14:02.41 | Tagor | tzafrir_laptop: see your private message for the versions, they are making outgoing calls with Asterisk. And I see the person logged in with sip show peers |
| 14:03.09 | Tagor | fauxalliance: that's exactly the reason why I stay with this OS ;) |
| 14:03.26 | tzafrir_laptop | Tagor, hmm... maybe they exploit a bug in your dialplan ? |
| 14:03.36 | fauxalliance | Tagor, OpenBSD "Only two remote holes in the default install, in a heck of a long time!" |
| 14:03.57 | tzafrir_laptop | Look at such a call. Trace exactly how it is routed |
| 14:04.11 | fauxalliance | it's about mitigating risks, not script kiddies... |
| 14:04.23 | snapple42 | <tzafrir_laptop> - Need to simulate a T1 to get TDD/TTY support on SIP.. have it working.. but timing seems off.. very choppy audio |
| 14:04.43 | Tagor | tzafrir_laptop: as far as I can see they just login the normal way on the extension. I can also call the person right now, though he doesn't pickup.. |
| 14:04.47 | tzafrir_laptop | fauxalliance, <Tagor> fauxalliance: on this server like anything is blocked out except some sip ports |
| 14:05.12 | fauxalliance | tzafrir_laptop, i'll take your word for it. |
| 14:05.40 | fauxalliance | 'like anything' is not "everything tcp" |
| 14:05.57 | tzafrir_laptop | snapple42, hmm... dahdi_dynamic_loc creates a span that is a proxy of another local span |
| 14:05.58 | *** join/#asterisk ZeXr0 (~ZeXr0@ip216-239-95-218.vif.net) |
| 14:06.07 | tzafrir_laptop | Not really sure how useful it will be |
| 14:06.27 | Tagor | fauxalliance: 'like anything' means that I have some ports like SSH opened with IP restrictions |
| 14:07.50 | ZeXr0 | Just a quick question : A client what to be able to use different codec, ilbc, g711 and g726. Should I try to convert my soundfile for every codec, or should I use uLaw and let Asterisk convert the audio on the fly ? |
| 14:07.52 | Tagor | tzafrir_laptop: I checked the messages log, but that only says 'Peer X is now reachable' is there a way to get more info on the logins? |
| 14:08.15 | snapple42 | tzafror_laptop: Tried the same thing looping back a TDMOE from one eth to another on same machine.. same result... According to docs there is a config to allow you to simulate a T1->T1. Without the third param in the config it won't monitor... Call goes out on one span, and in on the other.. just choppy audio |
| 14:09.16 | *** join/#asterisk anonymouz666 (~anonymouz@187-28-37-118.poolip.RJO.embratel.net.br) |
| 14:09.25 | fauxalliance | http://ilovetovoip.com/2010/07/what-to-do-when-your-asterisk-is-hacked/ @ Tagor |
| 14:09.26 | [TK]D-Fender | Tagor: Kill off the reg entry in astdb and restart * |
| 14:10.32 | anonymouz666 | [TK]D-Fender: do you know if it's possible to map an offhook (that responds with 486 and there's no active call) to BUSY in asterisk devstate? |
| 14:11.36 | [TK]D-Fender | anonymouz666: Rephrase and expand that a bit please... |
| 14:11.39 | Tagor | [TK]D-Fender: with 'reg entry' you mean the sip user? |
| 14:11.56 | *** join/#asterisk richardf (~savag3@173.116.124.202.static.snap.net.nz) |
| 14:13.31 | anonymouz666 | SIP/100 is an ATA and it is offhook (any calls to in got 486 BUSY response) but it is not in any call. It's just offhook. SIP/100 is also a queue member and it's status is Not In Use. The app_queue keeps sending the call to SIP/100 (due it is not in use) but the ATA is offhook and reply with 486. I am trying to map this 486 to a custom devstate BUSY |
| 14:14.14 | anonymouz666 | then app_queue will forget this member or consider it is a busy to take another action. |
| 14:14.34 | *** join/#asterisk bmg505 (~leon@196-209-7-27.dynamic.isadsl.co.za) |
| 14:14.35 | [TK]D-Fender | Tagor: When they register AstDB is updated including their contact IP.... it could be that if nothing fights to take it back * will consider any qualify to it as validating that it is still good even if you changed the secret and it is not re-reg-ing. |
| 14:14.46 | [TK]D-Fender | Tagor: Which so far you didn't actually confirm |
| 14:15.45 | *** join/#asterisk Qwell (~north@pdpc/sponsor/digium/Qwell) |
| 14:15.45 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 14:15.54 | anonymouz666 | russellb: the func_devstate custom state will be visible to app_queue ? |
| 14:16.11 | Tagor | [TK]D-Fender: what I did is: shutdown *, put the ip in the firewall, changed the password, started * again. And like 10 seconds later another IP successfully logs in |
| 14:16.56 | [TK]D-Fender | Tagor: PASTEBIN everything |
| 14:17.53 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:17.54 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:18.08 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
| 14:18.15 | Tagor | [TK]D-Fender: with everything you mean extensions.conf/sip.conf? Or what other details do you need? |
| 14:18.55 | [TK]D-Fender | Tagor: SIP debug samples, sip.conf. Peer dumps. The works. |
| 14:23.13 | *** join/#asterisk Auz (~Auz@pub00.education.com) |
| 14:25.38 | xheliox | Any anyone think of a clever way where one Asterisk box would call another to verify the PRI is up? E.g. Asterisk 1 calls a specific DID on Asterisk 2 and listens for a tone and if it doesn't hear it, the AGI script would e-mail/page me? |
| 14:27.28 | [TK]D-Fender | xheliox: Call file with basic IVR navigation via SendDtmf, etc |
| 14:28.41 | Tagor | [TK]D-Fender: hold on, just deleted the database entry and changed the pass. Seems they don't get in now. I guess I need to add some bruteforce protection. Thanks for helping out! |
| 14:28.45 | xheliox | Ahh.. have it send DTMF tones to an IVR back on Asterisk 1. I don't know why that was escaping me. |
| 14:28.56 | xheliox | That'd work exactly how I want it to. Thanks. |
| 14:29.37 | [TK]D-Fender | Tagor: You're welcome |
| 14:29.49 | Tagor | Anyone interested in the recorded personal phone calls of this $#%!$ hacker? :P |
| 14:30.04 | fauxalliance | \o |
| 14:30.23 | Tagor | Will upload them later today with the called numbers |
| 14:30.42 | Tagor | Hope this *ss will never hack again :P |
| 14:30.50 | fauxalliance | the numbers could be equally fun,, let's hope he called his mom. |
| 14:31.01 | Tagor | Well there were some pretty long calls |
| 14:31.45 | *** join/#asterisk krion (~seb@unaffiliated/krion) |
| 14:31.46 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 14:31.54 | Tagor | I wondered why my credits ran out so fast and then noticed this |
| 14:32.51 | Tagor | fauxalliance: I'm pretty sure it were friends or family. But the calls are to the Philippines so I don't understand them |
| 14:33.01 | Tagor | I might call some of his family later today |
| 14:33.03 | Tagor | Haha |
| 14:34.14 | fauxalliance | at least you can find humor in the situation Tagor |
| 14:34.34 | *** join/#asterisk dvdevel (~devel@wiggum.digitalcoven.com) |
| 14:34.43 | coppice | tagor doesn't understand tagalog? :-\ |
| 14:35.11 | fauxalliance | Filipino, /me has family in Bacolod City.. and Gen Santos... |
| 14:35.21 | anonymouz666 | plugs an analog line and sends another fax to coppice |
| 14:35.23 | Tagor | fauxalliance: it cost me about 100-150 USD I think, but I will get over it ;) |
| 14:35.55 | Tagor | coppice: do you? if you want I can upload some stuff for you :P |
| 14:40.39 | REdOG | argh, why after I register a sip trunk would * keep sending 107 registers? |
| 14:41.41 | REdOG | fkn broadvoice bans me after too many and I don't know how to tell * to stfu |
| 14:41.58 | anonymouz666 | putnopvut: if a queue member (using ATA) is offhook (but not in use, not in any call) and app_queue try calling and got back the 486 BUSY is it possible from app_queue to set this member as BUSY, or this will not be an easy task? I tried to use func_devstate but the state does not reflects in app_queue |
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| 14:43.33 | putnopvut | anonymouz666: it's something that could be done, but it's not easy. |
| 14:43.50 | anonymouz666 | looks at chan_agent and cries |
| 14:45.26 | anonymouz666 | ok, thanks for the answer. |
| 14:48.13 | dvdevel | greetings, all. |
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| 14:49.37 | dvdevel | i'm seeing a curious problem between two 1.6.2.9 boxes: incoming call on an iax channel, Background() seems to 'double' the dtmf (say in an ivr, type '100' translates it to 110). doesn't happen on a sip channel. any ideas? |
| 14:50.01 | dvdevel | WaitExten() and Read() don't have the same issue... |
| 14:54.30 | fauxalliance | http://readlist.com/lists/lists.digium.com/asterisk-users/10/53319.html @ dvdevel |
| 14:55.02 | fauxalliance | https://issues.asterisk.org/bug_view_advanced_page.php?bug_id=9382 |
| 14:55.25 | fauxalliance | ^^that one |
| 14:58.06 | dvdevel | i suspected something like that might be going on, which is why i was testing against Read() and WaitExten() as well, so why is only Background() affected? And that bug report is against 1.2... |
| 14:58.23 | dvdevel | in doing some searches on my own i saw some similar issues, with varying fixes/workarounds |
| 14:59.38 | dvdevel | one of which was to 'trap' the invalid extensions and redirect - but that doesn't seem too feasibly or manageable in freepbx. that's where this issue was reported to me, so i tested on clean bare asterisk builds and was still able to reproduce it. i wasn't happy. :) |
| 15:02.47 | fauxalliance | they noted they were _not_ going to fix it. |
| 15:03.47 | fauxalliance | as it's minor and as you had aforementioned "well documented" |
| 15:04.19 | dvdevel | i see that. of particular interest to me is that this was precipitated when i updated my core switches from 1.4.21 to 1.6.2.9 |
| 15:04.28 | fauxalliance | ahem dvdevel ; see reelated bug https://issues.asterisk.org/view.php?id=17253 |
| 15:04.51 | *** join/#asterisk bmoraca_work (~bmoraca@66-242-174-254.ceres.bvn.net) |
| 15:05.46 | fauxalliance | patch is |
| 15:05.52 | fauxalliance | 'known to work' |
| 15:06.04 | dvdevel | it can't hurt to try, that's for sure. |
| 15:06.25 | fauxalliance | i think you are on your own with this one dvdevel. |
| 15:06.53 | dvdevel | i think i'll force the customers to SIP |
| 15:07.04 | dvdevel | if i'm on my own, that's my answer |
| 15:08.26 | fauxalliance | patch first... then break out the whip.. |
| 15:08.32 | dvdevel | thanks for the pointers, fauxalliance - i must just not have been striking the right combination of keywords for my search. and i'd looked through dozens of changelogs for reference. |
| 15:08.49 | fauxalliance | is quite intimate with google. |
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| 15:21.08 | jesselang | I have two quad T1 cards in a box. How do I disable span 4? Will commenting out the span= line suffice? It didn't seem to take too kindly to that. |
| 15:22.17 | ChannelZ | just don't allocate the channels |
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| 15:27.06 | *** join/#asterisk Tulga (~chatzilla@203.91.113.10) |
| 15:27.38 | Tulga | I installed card driver and configured chan_dahdi.conf. but I haven't dahdi command on CLI. how to debug problem? |
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| 15:34.31 | tzafrir_laptop | Tulga, module show like chan_dahdi |
| 15:36.41 | Tulga | chan_dahdi.so DAHDI Telephony w/PRI 0 |
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| 15:38.07 | *** part/#asterisk Junior (~Juni@unaffiliated/junior) |
| 15:39.08 | tzafrir_laptop | Tulga, ok. try: module unload chan_dahdi.so module load chan_dahdi.so |
| 15:39.16 | tzafrir_laptop | Do you get any error? |
| 15:39.22 | emmz | hello, does anyone here have experience with the error "Channel SIP/<mysiptrunk> was never answered." in asterisk CLI only when using .call files? All other methods of originating calls are working. But not .call files. |
| 15:39.23 | *** join/#asterisk DelphiWorld (~Delphi@196.20.95.121) |
| 15:39.29 | DelphiWorld | hi hi |
| 15:39.34 | DelphiWorld | doe iaxcomm is fully compatible |
| 15:39.39 | DelphiWorld | with the iax stack of asterisk? |
| 15:39.45 | Tulga | tzafrir_laptop: [Jul 8 23:39:24] WARNING[23287]: chan_dahdi.c:1441 dahdi_open: Unable to open '/dev/dahdi/channel': Permission denied |
| 15:40.08 | tzafrir_laptop | Tulga, I guess that's quite obvious |
| 15:40.43 | tzafrir_laptop | DelphiWorld, AFAIK iaxcomm was never updated to support the "security token" |
| 15:40.55 | DelphiWorld | oh oh oh |
| 15:40.59 | DelphiWorld | so what i need to use |
| 15:41.29 | DelphiWorld | no zoiper |
| 15:41.42 | *** join/#asterisk wcselby (~wcselby@216.110.88.194) |
| 15:41.44 | wcselby | o/ |
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| 15:43.42 | Tulga | tzafrir_laptop: what you mean? |
| 15:45.11 | tzafrir_laptop | Tulga, what user is asterisk running under? |
| 15:45.17 | Tulga | root |
| 15:45.18 | tzafrir_laptop | ls -l /dev/dahdi/channel |
| 15:46.23 | Tulga | crw-rw---- 1 asterisk asterisk 196, 254 Jul 8 23:43 /dev/dahdi/channel |
| 15:46.47 | Tulga | run asterisk under asterisk user? |
| 15:46.49 | emmz | hello, does anyone here have experience with the error "Channel SIP/<mysiptrunk> was never answered." in asterisk CLI only when using .call files? All other methods of originating calls are working. But not .call files. |
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| 15:48.29 | tzafrir_laptop | that's odd. root should have permissins there |
| 15:49.13 | Tulga | yes |
| 15:49.21 | DelphiWorld | tzafrir_laptop: doe iaxclient library is updated ? |
| 15:49.42 | tzafrir_laptop | DelphiWorld, I don't think so |
| 15:50.42 | DelphiWorld | is alone so there is no iaxphone? |
| 15:51.15 | [T]ank | yeah, im back... more questions with h323. :-( |
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| 15:53.51 | wcselby | how can a DNS server provide answers for domains it doesn't know if it's not configured with any forwarders? |
| 15:54.50 | [T]ank | so, here is the configs and output: http://pastebin.com/PsYrutb8 |
| 15:55.19 | [T]ank | logs show that it cannot reach the remote host... which would make sense with the cli output. But, when watching traffic on the network, nothing is being sent at all. |
| 15:55.25 | [T]ank | only arp traffic |
| 15:55.27 | [T]ank | any ideas? |
| 15:55.35 | [T]ank | have a screwed something up in the configs somewhere? |
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| 16:09.00 | [T]ank | anyone? |
| 16:11.04 | *** join/#asterisk stmaher (~stephen@80.68.89.200) |
| 16:11.06 | stmaher | Hi guys.. |
| 16:11.26 | stmaher | Is there a way in asteirsk to send an email to a person when a number hits a certain point in extensions.conf |
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| 16:11.51 | Tulga | tzafrir_laptop: I changed owner of /dev/dahdi/* to root, so now asterisk can load them |
| 16:11.53 | stmaher | we have calls forwarded to mobiles.. but the callerid shows as the PBX outbound number.. and not the originating party |
| 16:11.56 | Tulga | very interesting |
| 16:12.17 | tzafrir_laptop | Tulga, note that those files will be recreated when you load the dahdi drivers |
| 16:12.18 | stmaher | unfortunately our SIP provider wont let us dynamically change the outbound caller ID :-( |
| 16:12.32 | tzafrir_laptop | grep for dahdi under /etc/udev/rules.d |
| 16:13.19 | Tulga | ok |
| 16:13.38 | Gugge | stmaher: you could get asterisk to read the original callerid when you pick up the mobile, before you bridge the call |
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| 16:13.54 | stmaher | Gugge great idea.. could you please point me to the documentation? |
| 16:14.37 | Gugge | not really, i just know it can be done :) |
| 16:14.45 | stmaher | gah :-) |
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| 16:15.07 | stmaher | where is tkdefender when you need him :-) |
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| 16:15.19 | datacompboy | Hi! Anybody knows, how to limit friend peer to have only single connection at time? |
| 16:15.26 | Gugge | Dial with the M() flag, and then a macro using SayDigits() |
| 16:15.32 | Gugge | stmaher: something like that :) |
| 16:15.47 | emmz | hello, does anyone here have experience with the error "Channel SIP/<mysiptrunk> was never answered." in asterisk CLI only when using .call files? All other methods of originating calls are working. But not .call files. |
| 16:16.00 | stmaher | Gugge thanks will give that a go |
| 16:16.13 | datacompboy | emmz: show your .call file |
| 16:17.19 | emmz | datacompboy: Channel: SIP/link2voip/13105929436 |
| 16:17.19 | emmz | Set: ongoing_call=telemed_test.call |
| 16:17.19 | emmz | Callerid: 13055078456 |
| 16:17.20 | emmz | MaxRetries: 2 |
| 16:17.20 | emmz | RetryTime: 60 |
| 16:17.20 | emmz | WaitTime: 30 |
| 16:17.20 | emmz | Context: test_emmz |
| 16:17.21 | emmz | Extension: 5678 |
| 16:17.21 | emmz | Archive: yes |
| 16:17.31 | datacompboy | emmz: use pastebin.ca for that! |
| 16:17.48 | emmz | oh sorry... i will go there now |
| 16:18.29 | datacompboy | try to use Channel: SIP/13105929436@link2voip |
| 16:18.37 | Gugge | datacompboy: You can use GROUP_COUNT in your dialplan |
| 16:19.11 | emmz | datacompboy: ok i will try it! .. let me test... |
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| 16:19.44 | datacompboy | Gugge: i have peer 2000. I have two X-Lite's connected as 2000. when I dial from 2001 to 2000 rigns both 2000'th. Best, is prevent second 2000 X-Lite to register at all, until first one disconnected or timed out |
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| 16:21.31 | Gugge | so you want to limit registrations? |
| 16:22.02 | datacompboy | Gugge: yes. |
| 16:22.19 | Gugge | you can only register one device to one peer/user/friend |
| 16:22.27 | Gugge | when you connect another it replaces the first |
| 16:22.34 | Gugge | nothing you can do about it |
| 16:22.59 | emmz | datacompboy: unfortunately that doesnt work either. It is the strangest thing. All other methods work except call files |
| 16:23.22 | datacompboy | emmz: what "other" method you mean? |
| 16:23.46 | datacompboy | emmz: show your AMI command, used to originate that works |
| 16:23.56 | emmz | meaning i can call external lines with phones connected to PBX or even by using originate command thru CLI |
| 16:24.31 | datacompboy | emmz: then, show your "Dial" command used to originate thru pbx |
| 16:26.33 | emmz | originate SIP/link2voip/xxxx extension 5678@<dialplan> works through CLI otherwise i just call using an extension or softphone. Do you know what i mean?? |
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| 16:28.38 | datacompboy | then, show your dialplan, handled outgoing call from softphone. there should be Dial() command in it |
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| 16:30.13 | datacompboy | emmz: i sometimes use "Channel: SIP/79139257974@mtt \n Application: Dial \n Data: SIP/74956510622@mtt" as .call file, and that correctly connect my cellular with techsupport of provider |
| 16:32.52 | emmz | datacompboy: checking... let me test it out :) |
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| 16:45.44 | emmz | hello, does anyone know why this error occurs "pbx_spool.c: Call failed to go through, reason ( |
| 16:45.44 | emmz | 8) Congestion (circuits busy)"? It only occurs when using .call files |
| 16:46.26 | emmz | that smily face is supposed to be an "8" within parenthesis.... |
| 16:48.55 | fauxalliance | emmz, all circuits are busy... check the outbound routes... |
| 16:49.19 | fauxalliance | do you have to dial a prefix to get a trunk, did you omit it from the call file? |
| 16:49.55 | emmz | fauxalliance: there is no need for a prefix. just 1NXXNXXXXXX |
| 16:50.31 | p3nguin | Check the number you're dialing. If you can dial it from a phone but not a call file, try calling it from the CLI with the originate command. |
| 16:51.03 | emmz | yes CLI works and so does calling from soft phone! error is only from call files |
| 16:51.40 | p3nguin | All signs point to a problem in the call file. |
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| 16:56.17 | emmz | p3nguin: the call file is Channel: SIP/link2voip/xxxx /n Set: ongoing_call=telemed_test.call /n MaxRetries: 2 /n RetryTime: 60 /n WaitTime: 30 /n Context: test_emmz /n Extension: 5678 /n Archive: yes |
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| 16:59.46 | p3nguin | Does the call reach your phone, and that's when you here congestion tones, or you're only seeing it on the CLI? |
| 17:00.23 | emmz | p3nguin: call never reaches phone. i only see it trying and retrying on the CLI |
| 17:01.17 | p3nguin | Do you know if the call actually reaches your outside destination at all? |
| 17:03.04 | emmz | i am testing with my own mobile phone for this. i can reach it when using a soft phone as well as when i use originate command in CLI, but .call files never succeed |
| 17:03.20 | emmz | i have used exact same call files on different system without errors though... |
| 17:04.55 | emmz | p3nguin: to answer your question, no it never reaches it. unless i use other methods besides call files |
| 17:05.28 | *** join/#asterisk pigpen (~mark@fw.seamans.cc) |
| 17:07.01 | fauxalliance | emmz, have you tried manually setting the context as per http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out |
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| 17:17.55 | emmz | fauxalliance: yes we have... Not sure what else it could be. It is the wierdest thing |
| 17:18.11 | emmz | and frustrating... |
| 17:19.40 | p3nguin | Try turning around the connection direction. Connect it internally first, then dial out using dialplan logic. |
| 17:20.14 | *** join/#asterisk sgimeno (~chatzilla@95.122.13.51) |
| 17:21.05 | p3nguin | Right now, you're trying to connect outbound first, and, if that is successful, connect to a phone via dialplan logic. Reverse the operation. |
| 17:21.19 | emmz | p3nguin: not sure what you mean... internally first? to a local extension? |
| 17:21.25 | Naikrovek | yes |
| 17:21.41 | p3nguin | Channel: SIP/5678 |
| 17:21.41 | Naikrovek | i know you weren't asking me, but... |
| 17:21.58 | p3nguin | Context: outbound-calls |
| 17:22.08 | p3nguin | Extension: 3145551212 |
| 17:22.52 | emmz | ohhh... hmmm.. ok .. let me try! (and thank you too Naikrovek) |
| 17:22.57 | p3nguin | I know those aren't your real numbers and context, but hopefully you get the idea. |
| 17:23.44 | p3nguin | If you pick up a phone and dial the external phone number, it works. So that tells me that your outgoing context and dialplan work. Use that to your advantage. |
| 17:28.20 | p3nguin | I know when I get calls from auto-dialers and I have to wait to connect to the callER, it really pisses me off... so if you reverse your operation, it would be more polite as well. |
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| 17:30.42 | emmz | p3nguin: do i need to add "@" with the trunk name in the Extension field? (eg. Extension: 3145551212@<mytrunk>) ? |
| 17:30.59 | p3nguin | no |
| 17:31.03 | p3nguin | See above. |
| 17:31.23 | emmz | i tried but it just dials the extension then hangs up |
| 17:31.27 | p3nguin | Context: <context-name> |
| 17:31.28 | p3nguin | Extension: <ext> |
| 17:32.09 | p3nguin | For Channel, did you speficy your phone, such as SIP/5678? |
| 17:32.48 | emmz | yes, but that didnt work so i tried another extension 5150 and that one is active so it rang but then hung up as soon as i answered |
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| 17:34.27 | p3nguin | Maybe you need to provide some more details on your configuration and possibly even some verbose/debug output. |
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| 17:42.06 | p3nguin | Also, make sure you are moving the file into the asterisk/outgoing directory from a location on the same file system. |
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| 17:48.03 | ZeXr0 | Just a quick question : A client what to be able to use different codec, ilbc, g711 and g726. Should I try to convert my soundfile for every codec, or should I use uLaw and let Asterisk convert the audio on the fly ? |
| 17:48.46 | p3nguin | I made a call file like yours and the receiving phone gets the call and then hears ringing until the local phone picks up. That's not the desired result. |
| 17:49.53 | p3nguin | zexr0: I have several of my sound files in multiple codec formats. It doesn't really provide much performance gain on a small system. |
| 17:52.08 | ZeXr0 | Okay, well it's a small system anyway |
| 17:52.27 | p3nguin | I guess if your processor is overloaded, it would save some clock cycles by not transcoding. |
| 17:52.59 | ZeXr0 | Well it's a big server serving 15 calls at the same time |
| 17:54.12 | p3nguin | On a larger system which could be underpowered, every little bit makes a difference. Do you have much load on the server? |
| 17:54.53 | ZeXr0 | Not really, I think it's a quad core cpu with 16gb of ram, and it's running X server with asterisk and that's about it. |
| 17:55.06 | p3nguin | Why would X be running on a server? |
| 17:55.52 | ZeXr0 | For management reason, seems like someone, somewhere, decided that it would be a good idea to run it in case we need to change settings and that it seems that the tools in linux are easier that config files ... |
| 17:56.27 | p3nguin | Someone needs to take a training course. |
| 17:57.17 | ZeXr0 | p3nguin : Not that I didn't try to change the things in here, but I guess my voice is not strong enought to get by those who used Windows for their entire life, and see linux as something unstable and very hard to learn... |
| 17:57.51 | markfeatherston_ | ZeXr0: I've been there... |
| 17:58.04 | p3nguin | X not only puts unnecessary load on a server computer, but it is considered as a HUGE security vulnerability by most competent admins. |
| 17:58.34 | ZeXr0 | Forget about competent admins in here :P |
| 17:58.40 | leifmadsen | lol |
| 17:59.03 | p3nguin | And the fact that someone can gain physical access to the box to ever USE that X server is even worse! |
| 17:59.50 | markfeatherston_ | ZeXr0: still not a great solution, but you could show them xming + ssh. That way they can run windows, and not have x constantly running on that server. |
| 17:59.56 | pigpen | We call that Dam spooky. |
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| 18:02.07 | *** join/#asterisk rossand (~aross@CPE485b390978ce-CM00159a025ad4.cpe.net.cable.rogers.com) |
| 18:02.46 | *** part/#asterisk rossand (~aross@CPE485b390978ce-CM00159a025ad4.cpe.net.cable.rogers.com) |
| 18:02.51 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 18:03.06 | markfeatherston_ | if I change the variable "TOUCH_MONITOR_OUTPUT" in the middle of a recording, will it rename the files? Or does it have to be changed before Record is called? |
| 18:03.11 | p3nguin | Back to the original question... check your server load at peak operating time. If it is high, consider that transcoding could be adding to that load. Decide if the load is high enough to convert all your sound files to all possible formats. |
| 18:03.12 | *** join/#asterisk kimc (~kimc@76-242-100-1.uvs.livnmi.sbcglobal.net) |
| 18:07.20 | *** join/#asterisk [1]kimc (~kimc@76-242-100-1.uvs.livnmi.sbcglobal.net) |
| 18:11.21 | ZeXr0 | p3nguin : Thanks, I'll look at that |
| 18:13.41 | *** join/#asterisk evil_gordita (~evilgordi@ip70-188-50-186.rn.hr.cox.net) |
| 18:19.33 | *** join/#asterisk jkroon (~jkroon@dsl-244-43-92.telkomadsl.co.za) |
| 18:19.59 | jkroon | is there a rusty here by any chance? |
| 18:25.07 | [T]ank | now that asterisk supports fax, can anyone recommend a good sip provider that supports fax as well that also is reasonably priced? I have been with binfone and have been very happy. but they do not support fax. I have seen some that have nice reseller tools, but i have not found one yet that supports outgoing caller id name. number only. I know the list of providers on the wiki, but looking for a recommendation. In the US. |
| 18:25.55 | Qwell | [T]ank: None support name. |
| 18:26.00 | Qwell | because of how it works.. |
| 18:26.06 | jkroon | [T]ank, i take it then you got T.38 working? |
| 18:26.38 | jkroon | because I've been trying and not been getting it right. |
| 18:27.23 | [T]ank | Qwell: by name, I mean that they subscribe to directory assistance populating CNAM databases. |
| 18:27.55 | [T]ank | binfone does. |
| 18:28.06 | voxter | anyone know, when using external mics with a polycom conference phone (like an ip6000) - shouldnt the internal mics still be usable, but the external mics are an 'addon' or is it one or the other? |
| 18:28.08 | [T]ank | but the cool reseller hosts that I have contacted do not |
| 18:28.26 | Naikrovek | voxter: they all work i believe |
| 18:28.35 | Naikrovek | the addon mics only augment the built-in one |
| 18:28.39 | Naikrovek | probably configurable though |
| 18:28.39 | markfeatherston_ | [T]ank: I haven't tested it myself, but flowroute supports t.38 and is decently priced. You could try them out. |
| 18:28.46 | voxter | Naikrovek: thats what i thought, but testing with a client seems that the internal ones turn off |
| 18:29.02 | Naikrovek | voxter: probably configurable. |
| 18:29.56 | [T]ank | markfeatherston_: thank you |
| 18:31.38 | markfeatherston_ | np |
| 18:38.55 | *** join/#asterisk kfife (~Miranda@home.chicagoventure.com) |
| 18:43.00 | kfife | Anybody know what this means: test_dlinklists.c: http://pastebin.com/UTiHAMFK |
| 18:43.41 | Qwell | it means you enabled tests |
| 18:44.05 | *** join/#asterisk Tobarja (~chatzilla@24.136.151.107) |
| 18:45.57 | kfife | Qwell: Thanks. |
| 18:46.56 | kfife | Qwell: is that enabled by default? |
| 18:47.01 | Qwell | shouldn't be |
| 18:47.36 | kfife | Qwell: Can you tell me which configuration file controls that setting so I can investigate? |
| 18:47.48 | Qwell | menuselect |
| 18:47.55 | kfife | I see. |
| 18:48.07 | *** join/#asterisk githogori (~githogori@SJC-Office-DHCP-135.mail-abuse.org) |
| 18:48.35 | kfife | Again, thanks. |
| 18:52.27 | *** join/#asterisk x1nux (~x1nux@unaffiliated/x1nux) |
| 18:52.30 | x1nux | hi |
| 18:55.37 | *** join/#asterisk [T]ank (~ckwall@206.71.78.158) |
| 18:57.58 | *** join/#asterisk doolittlework (~d@41-134-22-10.dsl.mweb.co.za) |
| 18:58.08 | doolittlework | hi |
| 18:58.17 | *** join/#asterisk ruben23 (~ITadmin@125.212.40.2) |
| 19:00.55 | doolittlework | need some help with misdn, all seems fine but i have 4 bri ports all incomming works on all but one port i can see the call comming in ringing switchboard, but when i answer the call it does not answer the channel, the caller keeps on hearing ringing and i get ingage tone. I checked all the settings all looks good using the same context for all lines, got me baffled? |
| 19:09.49 | *** part/#asterisk x1nux (~x1nux@unaffiliated/x1nux) |
| 19:14.40 | *** join/#asterisk bonux2 (~jkfdjksaf@mail.hegemoncapital.com) |
| 19:15.41 | bonux2 | Everytime I make a call with Xlite the DTMF tones only work for like 30 seconds, then pressing buttons doesnt have any effect in the menu.. this is realy bad when trying to navigated long menus that exceed 30 seconds of usage which most do |
| 19:15.44 | *** join/#asterisk gandhijee (akp@ip67-152-15-148.z15-152-67.customer.algx.net) |
| 19:16.01 | bonux2 | is there any reason why the DTMF tones intermittently work beyond 30 seconds, and sometimes cut off early? |
| 19:16.15 | jplank | depends on what you are using? |
| 19:16.15 | gandhijee | hey, polycom doesn't have licensing for their phones any more, right? |
| 19:16.15 | pabelanger | bonux2: what version of Asterisk? |
| 19:16.18 | jplank | ast version |
| 19:16.22 | jplank | dtmf type |
| 19:16.48 | jplank | gandhijee: I think they brought back the productivity suite if thats what your asking |
| 19:17.17 | gandhijee | does that include the LDAP corp directory stuff? |
| 19:17.23 | jplank | I think so |
| 19:17.37 | gandhijee | i c |
| 19:17.40 | *** join/#asterisk hansin (~hansin@c-76-120-68-209.hsd1.co.comcast.net) |
| 19:17.42 | jplank | it should be in the latest release notes and/or the manual |
| 19:17.47 | jplank | I know its in one of those |
| 19:17.50 | pabelanger | bonux2: https://issues.asterisk.org/view.php?id=17571 |
| 19:18.06 | jplank | there's a grid with all the phones and software |
| 19:18.16 | gandhijee | yeah i saw something in mine manual like a year ago and it made it sound like there weren't any fees for anything any more |
| 19:18.32 | hansin | If I am doing a SIP/VoI |
| 19:18.32 | gandhijee | but i ran across something recently that makes it seem like there is |
| 19:18.48 | p3nguin | bonux2: First of all, x-lite really sucks. Try zoiper. |
| 19:18.58 | hansin | If I am doing a SIP/VoIP only implementation, I really shouldn't need DAHDI and libpri, is that right? |
| 19:19.19 | jkroon | hansin, correct. |
| 19:19.28 | hansin | I have also found Linphone to be a nice client as well... |
| 19:19.30 | pabelanger | hansin: maybe DAHDI for dummy timer |
| 19:19.43 | pabelanger | depends on version of Asterisk however |
| 19:19.55 | markfeatherston_ | hansin: i think meetme needs the dummy timer from dahdi |
| 19:20.19 | bonux2 | How do I get the version |
| 19:20.36 | gandhijee | carefully |
| 19:20.38 | pabelanger | core show version |
| 19:21.22 | hansin | Okay, the reason I ask is that the CentOS Digium add-on repo forces me to install DAHDI and libpri (at least they are dependencies) if all I want is "yum install asterisk". |
| 19:21.40 | jplank | I know they brought back the productivity suite for pay, I don't remember what features it has though |
| 19:21.45 | hansin | I like to keep it simple and not install anything extra if not needed, especially at the kernel level. |
| 19:23.57 | Naikrovek | gandhijee: yes, the ldap stuff is licensed, it's $6 per phone i think |
| 19:24.28 | hansin | ls |
| 19:24.37 | hansin | woops ;) |
| 19:24.38 | gandhijee | Naikrovek: ok thanks, any of you guys know what the cost of a site lic is? |
| 19:25.17 | jplank | lol |
| 19:25.22 | jplank | (3:22:21 PM) Naikrovek: gandhijee: yes, the ldap stuff is licensed, it's $6 per phone i think |
| 19:25.40 | gandhijee | jplank: site lic != individual phone lic |
| 19:25.54 | jplank | oh site license, missed that |
| 19:26.04 | gandhijee | site is blanket coverage, one time pay |
| 19:26.10 | jplank | I didnt' think anyone actually used the productivity suite |
| 19:26.22 | gandhijee | the LDAP feature seems kinda cool |
| 19:26.31 | jplank | LDAP serverside :) |
| 19:26.39 | gandhijee | woukdn't mind playing with it before i try to sell it to customers |
| 19:26.44 | gandhijee | ya |
| 19:27.01 | *** join/#asterisk iscario (~quassel@laureades.davout02.net1.nerim.net) |
| 19:27.30 | jplank | I haven't had any clients request to use LDAP, the closest I've been asked with IMAP storage for voicemail (yea, I know, not even remotely the same thing) |
| 19:28.00 | *** join/#asterisk retentiveboy (~pdugas@69.169.199.82) |
| 19:28.07 | Naikrovek | eh you don't need ldap so much |
| 19:28.21 | Naikrovek | most people write a small PHP app and put a microbrowser button on the phone |
| 19:28.32 | jplank | exactly |
| 19:28.47 | Naikrovek | then you can push "dir" and it'll load the webpage, you can scroll through, click the check, and it'll push the buttons for you i think |
| 19:29.08 | jplank | you could also link the dir button to the microbrowser |
| 19:29.14 | jplank | thats how we do a webdirectory |
| 19:29.29 | jplank | (though you can't seem to do it on a vvx, or at least, we haven't been able to figure it out) |
| 19:29.47 | Naikrovek | i want to write an app that generates the config files on the fly dynamically |
| 19:29.49 | jplank | we only use the phones built in directory for buddy lists |
| 19:30.07 | Naikrovek | so when the phone requests the config, the web server app generates it then |
| 19:30.10 | Naikrovek | pulling from a small db |
| 19:30.18 | jplank | we've been working on the PCM (polycom config manager) thats does that |
| 19:30.24 | jplank | I can't take credit for it, but its awesome |
| 19:30.40 | jplank | no DB though, it parses and creates the cfg files |
| 19:30.48 | Naikrovek | i have a perl script that does that |
| 19:30.51 | Naikrovek | from a text file |
| 19:31.05 | Naikrovek | i'd rather have a web gui so people other than myself can set up new extensiosn |
| 19:31.12 | jplank | yea |
| 19:31.24 | jplank | thats what we have, its a php script |
| 19:31.31 | Naikrovek | perl script is rather slick but a bit manual for my taste |
| 19:31.40 | jplank | even a nice little check box to setup remote phones |
| 19:31.44 | jplank | :) |
| 19:31.59 | Naikrovek | remote phones? |
| 19:32.01 | jplank | (i.e. phones that are going to be remote, can't really do DHCP over WAN) |
| 19:32.03 | Naikrovek | what you talkin' bout willis |
| 19:32.10 | Naikrovek | oh |
| 19:32.26 | Naikrovek | i have phones in india connected to my system here, they have a local dhcp server tho |
| 19:32.33 | jplank | yea |
| 19:32.34 | Naikrovek | so i just point them to my system and they pull their configs |
| 19:32.50 | jplank | but we have it so all you have to do is plug it into the home network once, and its ready to go |
| 19:32.58 | jplank | and you don't even have to plug it into the home network once |
| 19:33.15 | jplank | but if you don't feel like adding IP addresses and ftp username/passwords, its easier |
| 19:33.21 | Naikrovek | yeah mine is like that. i hand the box to the user and the user sets up their phone. when they plug it in it works |
| 19:34.14 | jplank | we were trying to use LLDP to automatically detect when a new phone was plugged into the network, and display a pop up for it (so you don't even need to know the MAC) but we couldn't get it working reliably |
| 19:34.26 | Naikrovek | hmm |
| 19:34.32 | Naikrovek | i don't have lldp here, nor cdp yet. |
| 19:34.53 | jplank | the first time you do an install of over 150 phones, LLDP becomes your friend |
| 19:35.01 | Naikrovek | heh |
| 19:35.08 | Naikrovek | well i don't have any switches capable of it yet |
| 19:35.12 | jplank | ahhh |
| 19:35.22 | jplank | adtran 123x |
| 19:35.27 | jplank | I live by those switches |
| 19:35.36 | Naikrovek | i have some HPs waiting for approval |
| 19:35.47 | korcan | can someone point me to an article on hpw to implement outbound dial code prefix for accounting/ setting CID purposes? |
| 19:36.10 | jplank | Naikrovek: take a look at these http://www.adtran.com/web/page/portal/Adtran/group/4 |
| 19:36.22 | jplank | they are a little pricey, but I assure you, they are worth every cent |
| 19:36.41 | Naikrovek | alright |
| 19:36.49 | jplank | not only does it do every protocol you could through at it (except cdp), but you could also test cable runs with it |
| 19:36.57 | Naikrovek | nice |
| 19:37.10 | *** join/#asterisk Get_The_Fish (~Get_The_F@c-24-8-50-199.hsd1.co.comcast.net) |
| 19:37.11 | jplank | yea, and adtrans support is top notch |
| 19:37.15 | jplank | none of the cisco smartnet BS |
| 19:37.22 | [TK]D-Fender | korcan: "core show function set" "core show function CDR" <- accountcode |
| 19:37.32 | gandhijee | Naikrovek: where in india? |
| 19:37.39 | Naikrovek | hyderabad |
| 19:37.45 | gandhijee | cool |
| 19:37.59 | gandhijee | what kinda of connection you guys have there and who provides it? |
| 19:38.14 | Naikrovek | we have two E1s, don't know the providers off-hand |
| 19:38.24 | Naikrovek | but they're separate and they're not doing BGP, against my advice |
| 19:38.25 | gandhijee | we have an office in Abad |
| 19:38.50 | gandhijee | Naikrovek: how much are those Elinks? |
| 19:39.03 | Naikrovek | no idea how much they run |
| 19:39.09 | *** join/#asterisk SimonPerreault (~SimonPerr@2620:0:230:c000:f53d:debf:13a1:cf2b) |
| 19:39.15 | SimonPerreault | yo |
| 19:39.18 | SimonPerreault | what's up with origsvn? |
| 19:39.53 | *** join/#asterisk eduzimrs (~eduzimrs@201.22.86.124.static.gvt.net.br) |
| 19:39.54 | *** part/#asterisk SimonPerreault (~SimonPerr@2620:0:230:c000:f53d:debf:13a1:cf2b) |
| 19:40.16 | doolittlework | does anyone know what empty_chan_in_stack: cannot empty channel 255 mean---- for misdn lines B410p 4 port isdn |
| 19:43.28 | *** join/#asterisk retentiveboy (~pdugas@69.169.199.82) |
| 19:43.57 | korcan | [TK]D-Fender: thanks :) |
| 19:48.54 | *** part/#asterisk hansin (~hansin@c-76-120-68-209.hsd1.co.comcast.net) |
| 19:51.55 | *** join/#asterisk nosbig (~rgibson@cpe-75-179-4-171.neo.res.rr.com) |
| 19:52.36 | wcselby | the sky is falling, the sky is falling! |
| 19:54.50 | xheliox | looks up |
| 19:54.55 | xheliox | *clunk* |
| 19:54.59 | xheliox | ouch! you're not kidding. |
| 19:57.37 | eduzimrs | i have a FAX in a E1 trunk and i gotta many problems to send/receive is there a codec matter? |
| 19:58.05 | eduzimrs | im using G711A |
| 19:58.30 | path | its more likely network bandwidth |
| 19:59.02 | path | what app are you using to receive fax ? |
| 19:59.05 | nosbig | So, I've been looking at info for installing FreePBX on Debian, and I keep seeing everyone talking about compiling Asterisk and other components from scratch... Why are distro packages so bad to use? |
| 20:00.03 | eduzimrs | path im using a linksys ATA registering a DDR SIP branch |
| 20:02.22 | eduzimrs | path copy? |
| 20:02.56 | [TK]D-Fender | eduzimrs: What kind of E1? |
| 20:05.01 | *** join/#asterisk Benwa (~Benwa@ip-62-235-222-106.dsl.scarlet.be) |
| 20:09.27 | eduzimrs | [TK]D-Fender ISDN |
| 20:09.48 | gloin | Greetings all. I'm trying to do something like System(touch /tmp/`date -R`) in my dialplan, but it doesn't actually create a file that way. What's the trick there? |
| 20:10.46 | [TK]D-Fender | eduzimrs: Ok... you don't seem to understand some crucial differences... a VOICE E1 via a DAHDI card, or a DATA E1 for INTERNET? |
| 20:11.31 | eduzimrs | [TK]D-Fender ohh ok, its a VOICE E1 VIA DAHDI card |
| 20:11.41 | [TK]D-Fender | edu`What ver of *? |
| 20:12.16 | *** join/#asterisk DrCron (rszasz@saxonco.com) |
| 20:13.12 | eduzimrs | [TK]D-Fender sorry what? |
| 20:13.25 | [TK]D-Fender | eduzimrs: What ver of *? |
| 20:14.37 | eduzimrs | [TK]D-Fender i dont mean what youre saying |
| 20:14.51 | gloin | eduzimrs: * means "asterisk" |
| 20:15.01 | gloin | what version of asterisk is what he's asking |
| 20:15.18 | gloin | answered his own question anyways ;) |
| 20:15.27 | eduzimrs | ohh sorry |
| 20:15.54 | tzafrir_laptop | gloin, and this answer is? |
| 20:15.55 | eduzimrs | [TK]D-Fender 1.4.26.3 |
| 20:16.16 | gloin | tzafrir_laptop: I did exten => 2,n,System(echo `date -R` >> /tmp/foobar) |
| 20:16.37 | gloin | and it doesn't create the file for me, but it does dump the output of "date -R" to the end of /tmp/foobar |
| 20:16.38 | [TK]D-Fender | eduzimrs: Expect problems. Even over a local LAN yuo can haev jitter, packet loss etc. You need T.38 which 1.4 does not support except in passthrough mode in 1.4 series |
| 20:17.02 | gloin | is going to write a shell script to do more accurately what he wants |
| 20:18.37 | *** join/#asterisk markfeatherston_ (~mark.feat@71.4.41.98.ptr.us.xo.net) |
| 20:21.09 | eduzimrs | [TK]D-Fender i have set "FAX PASSTHRU = NSE" in my ATA conf. Ther are two others options (Reinvite and NONE) |
| 20:22.04 | [TK]D-Fender | eduzimrs: Ok.... lets try this again. |
| 20:22.10 | doolittlework | sry to trouble where is the isdn_lib.c file |
| 20:22.19 | [TK]D-Fender | eduzimrs: You need * to PROVIDE T.38. 1.4 CANNOT. |
| 20:22.25 | [TK]D-Fender | eduzimrs: This idea id DEAD |
| 20:22.31 | [TK]D-Fender | eduzimrs: UPGRADE |
| 20:27.30 | gloin | doolittlework: nice nick. Try "locate" by any chance? |
| 20:29.05 | xheliox | Using a call file and Application: AGI, is there any way to have it run the AGI script regardless if the remote channel answers? |
| 20:29.55 | eduzimrs | [TK]D-Fender okk, tks i will up |
| 20:30.13 | doolittlework | gloin did the locate but cannot find it |
| 20:30.17 | chuckp | Anyone ever have an Aastra phone that refuses to dial a * ? |
| 20:30.23 | chuckp | the * is coming up as a . |
| 20:30.57 | markfeatherston_ | chuckp: what model? |
| 20:31.03 | chuckp | 57i |
| 20:31.24 | xheliox | xheliox: Call a local channel. xheliox: Hey, great idea. xheliox: Yup. |
| 20:31.25 | chuckp | Ive used the model before and never had a problem |
| 20:31.41 | markfeatherston_ | lol |
| 20:31.42 | doolittlework | how does one implement this on ones server https://issues.asteriks.org/view.php?id=13488 |
| 20:31.44 | chuckp | its almost like its in alpha numeric mode |
| 20:32.16 | markfeatherston_ | i don't have that model to test, sorry. If all else fails, ask on the trixbox aastra endpoints forum. |
| 20:32.25 | markfeatherston_ | make sure you have the latest firmware : / |
| 20:35.44 | [TK]D-Fender | checkout time, BBIAB |
| 20:39.34 | iscario | hi, i have a slight prb with langages (iax accounts) : i configured realtime mysql to read iax accounts prefs. When using the voicemail service, i have a sound in english whereas i switched 'EN' to 'FR' in the database. on the CLI, the good langage was indicated, but the sound was in english anyway... Where is the problem ??? |
| 20:42.27 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 20:44.35 | doolittlework | some please tel me how does one apply this patch https://issues.asterisk.org/view.php?id=13488 |
| 20:45.36 | pabelanger | doolittlework: expain [wget patch] and use the syntax it displays |
| 20:46.18 | pabelanger | s/expain/expand/ |
| 20:46.35 | markfeatherston_ | doolittlework: can you not just update asterisk? that patch was submitted "2008-09-16 08:15". |
| 20:47.37 | doolittlework | do i need to upgrade my asteriks version to another? |
| 20:47.43 | doolittlework | to latest? |
| 20:48.01 | doolittlework | to fix the bug i have on isdn b410p card, |
| 20:48.30 | doolittlework | it locks up an incomming channel,just wont release it |
| 20:49.03 | doolittlework | only restarting asteriks or softhangup channel fixes it |
| 20:50.03 | markfeatherston_ | doolittlework: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.4-current |
| 20:50.12 | markfeatherston_ | there have been security fixes, and a large number of other fixes |
| 20:50.21 | markfeatherston_ | why not just cover all your bases and upgrade? |
| 20:52.50 | *** join/#asterisk retentiveboy (~pdugas@69.169.199.82) |
| 20:54.52 | gloin | argh, so I can't do Set(COUNT--) in a dialplan? |
| 20:55.01 | p3nguin | You could. |
| 20:55.10 | gloin | but it wouldn't decrement, would it? |
| 20:55.52 | gloin | count=count-1 with lotsa extra brackets :( |
| 20:56.23 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 20:56.25 | p3nguin | Set(COUNT=$[${COUNT} - 1]) |
| 20:56.28 | gloin | yep |
| 20:56.39 | jesselang | Can I specify zap channels into a group using a descending order? channels => 23-1 ? |
| 20:56.42 | gloin | kind of hideous visually |
| 20:57.01 | [TK]D-Fender | jesselang: That isn't defining a group |
| 20:57.52 | jesselang | [TK]D-Fender, what I mean is, can I have asterisk use the channels in descending order rather than ascending? |
| 20:58.25 | p3nguin | gloin: example - http://pastebin.com/6YMC1ge6 |
| 20:59.01 | jesselang | channel => 23-1 |
| 20:59.08 | jesselang | In the zapata.conf. |
| 20:59.23 | [TK]D-Fender | jesselang: that does not define a GROUP. |
| 20:59.32 | gloin | p3nguin: thanks, I'm looking at very much the same example in the O'Reilly book. Was hoping there was a cleaner-looking decrement that was newer |
| 20:59.43 | jesselang | [TK]D-Fender, Yes, I understand I must use group = N beforehand. |
| 20:59.47 | [TK]D-Fender | jesselang: that simply takes all of the vaules set about an assigns them to those channels. |
| 21:01.29 | jesselang | [TK]D-Fender, OK. That being the case, how can I have Asterisk try the channels in descending order? |
| 21:01.47 | jesselang | G1, instead of g1 |
| 21:01.48 | jesselang | Right. |
| 21:02.13 | [TK]D-Fender | \o/ |
| 21:02.17 | p3nguin | gloin: It looks pretty straight-forward to me. |
| 21:02.22 | [TK]D-Fender | Clue Acquired |
| 21:02.44 | jesselang | Thanks. |
| 21:02.53 | gloin | p3nguin: sort of. But you've got to admit, "Set(COUNT--)" is a lot cleaner |
| 21:03.06 | gloin | (would be cleaner if it worked heh) |
| 21:03.18 | [TK]D-Fender | gloin: Go make an app for it. |
| 21:03.26 | [TK]D-Fender | gloin: Easy enough |
| 21:03.33 | p3nguin | gloin: What's not clean about the other way? If you can read the syntax, you'll know it doesn't get much easier than setting the COUNT variable to the current value minus one. |
| 21:03.47 | gloin | both of yous :) |
| 21:04.05 | *** join/#asterisk [T]ank (~ckwall@206.71.78.158) |
| 21:04.13 | gloin | It's just a style thing, for whatever reason the syntax used for operators in the dialplan looks gnarly to me |
| 21:04.18 | gloin | keeping in mind I'm new to writing same |
| 21:04.50 | gloin | GotoIf($[${ |
| 21:05.00 | gloin | that kind of thing |
| 21:05.06 | gloin | operators/variables/etc |
| 21:05.13 | *** join/#asterisk wonderworld (~ww@ip-62-143-22-226.unitymediagroup.de) |
| 21:05.18 | p3nguin | It makes sense, but can get difficult when using lots of them |
| 21:05.34 | Kobaz | gloin: use AEL or AGI, and your pain till go away |
| 21:05.38 | Kobaz | s/till/will |
| 21:05.42 | gloin | I'm sure there's a reason for that kind of layout |
| 21:05.45 | p3nguin | OH, so close! |
| 21:06.06 | gloin | Kobaz: AEL has my attention but probably won't end up needing it for this project |
| 21:06.37 | Kobaz | i've completely moved away from extensions.conf because of the horrible syntax (other when it's absolutly required, like generating dynamic dialplan) |
| 21:07.07 | Kobaz | s/when/then |
| 21:07.08 | Kobaz | dang |
| 21:07.23 | Kobaz | s/when/then when/ |
| 21:08.09 | gloin | yeah, we'll see |
| 21:08.52 | gloin | once I get a couple more really basic things nailed down, I have to figure out how to create a group of external numbers to be dialed out round-robin until a trigger event (somebody wakes up and presses two) |
| 21:09.10 | gloin | using AMI to set up the call |
| 21:09.18 | gloin | may use AMI to handle that part as well |
| 21:10.11 | gloin | sigh, or learn AEL and do it that way :( |
| 21:16.01 | *** join/#asterisk bandu (kvirc@pool-71-164-233-190.dllstx.fios.verizon.net) |
| 21:16.09 | [TK]D-Fender | AEL has nothing to do with AMI |
| 21:16.26 | wcselby | Not true [TK]D-Fender......They both start with the letter A |
| 21:16.49 | [TK]D-Fender | reaches for his ClueBat (tm) and starts swatting |
| 21:16.53 | wcselby | they both also contain one vowel |
| 21:17.14 | wcselby | haha, i guess that would be better if I said they contained one consonant, or two vowels |
| 21:17.20 | wcselby | or one vowel other than A |
| 21:18.24 | wcselby | sorry, it's been a rather......day here and I've got to left it out somehow |
| 21:18.30 | *** join/#asterisk Mish- (~mish@125-236-219-18.adsl.xtra.co.nz) |
| 21:19.33 | *** join/#asterisk drclue (~drclue@ip-65-49-163-54.wireless.dyn.beamspeed.net) |
| 21:19.38 | Mish- | I'm about to build a very basic home Asterisk server with a single PSTN line using a "SPA3000", is there any reason I wouldn't use "Asterisk 1.6.2"? |
| 21:19.56 | pabelanger | Mish-: no |
| 21:21.42 | Mish- | Excellent, thanks. |
| 21:22.32 | Mish- | Is there a good web configuration gui, or am I best to stick with the raw conf files? |
| 21:23.18 | p3nguin | Web configuration? You just lost the interest of half the people here. |
| 21:23.25 | drclue | Howdy All, I'm looking for guinea pigs for my HTML5/WebSockets Javascript Asterisk AMI interface. Execute any AMI command , receive all AMI events, connected browsers can "chat" any XML structures between them across multiple Asterisk servers. Especially interested in iPhone people as I want to start testing AsterClick on cellphones really soon, but everyone is invited |
| 21:23.52 | wcselby | Mish- - if you want web conf, try looking at the AsteriskNOW ISO image (it builds the server and everything for you). |
| 21:24.26 | drclue | If you want to edit config files , AMI has support for that , which means that AsterClick can help you too |
| 21:24.42 | [TK]D-Fender | Mish-: Best to DIY like the rest of us |
| 21:25.05 | [TK]D-Fender | drclue: pimp[-1] |
| 21:25.27 | wcselby | Mish- - but I agree with p3nguin and [TK]D-Fender, roll your own. Check out the asterisk book for more info |
| 21:25.29 | wcselby | ~book |
| 21:25.30 | 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 |
| 21:25.37 | wcselby | Mish- ^^^ |
| 21:25.57 | *** join/#asterisk bmg505 (~leon@196-209-7-27.dynamic.isadsl.co.za) |
| 21:26.17 | drclue | @fender , brother , you live in the dial plan , and I respect that. I don't live in the dial plan , so please brother , respect that |
| 21:28.03 | [TK]D-Fender | drclue: And AMI has nothing to do with controlling all the actions a call is going to take. |
| 21:29.01 | [TK]D-Fender | drclue: So yes... you are as "in" the dialplan as the rest of us. |
| 21:29.01 | [TK]D-Fender | drclue: Externally yanking stuff around isn't so common a need |
| 21:29.01 | [TK]D-Fender | drcuUseful sure, but not even vaguely comparable |
| 21:29.02 | drclue | @fender , whatever your response had to do with what I said, fine. If one wants to speak from the dial plan , then you can use my FastAGI tools |
| 21:31.20 | drclue | @fender It's interesting that with all the web based Asterisk exploitations that you would say there is no need for web based tools. I guess I would simply have to agree that YOU don't need web based tools , because your a dial plan genius |
| 21:32.31 | [TK]D-Fender | drclue: You don't seem to be clueing in.... |
| 21:32.46 | [TK]D-Fender | drclue: AMI has NOTHING to do with DIALPLAN. |
| 21:32.55 | drclue | There are many folks who are more at home with Javascript, and PHP than there folks comfortable with the dialplan |
| 21:32.56 | [TK]D-Fender | drmessano: I made a single directed point. |
| 21:33.04 | [TK]D-Fender | drclue: ^^ |
| 21:33.38 | *** join/#asterisk Mhaddog (~Mhaddog@z65-50-118-232.ips.direcpath.com) |
| 21:34.00 | drclue | @fender Hey your entitled to your views , but I don't understand why you argue with folks just to argue |
| 21:35.12 | [TK]D-Fender | drclue: Dialplan processes a call. AMI does not. It isn't an "opinion". It is a "fact". |
| 21:35.45 | drclue | This is the #asterisk channel not the #dialplan channel , so every way of skinning asterisk is valid conversation |
| 21:37.15 | drclue | AMI receives events at each stage of a call , and is free to issue any AMI command it likes to any active channel. For dialplan based interactions I use my FastAGI system, but for everything else , that is why I created AsterClick |
| 21:38.28 | gloin | hmm |
| 21:38.48 | gloin | turns out I'm now the only person in #dialplan |
| 21:38.49 | [TK]D-Fender | drSo what does AsterClick have to do with your "FastAGI system" ? |
| 21:39.53 | Kobaz | i prefer SlowAGI |
| 21:40.17 | drclue | @fendor You view the entire world through the lenses of the dialplan , and that is fine. I have dialplans that export everything to FastAGI and have hardly any dialplan at all and do all the things the dialplan could. The Asterclick FastAMI is just a flip side of that coin. Your still relevant , have no fear, I'm not burning your dialplan throne, but the dial plan has little use in what I do. |
| 21:40.51 | [TK]D-Fender | drclue: You keep dodging the fac that your AsterClick AMI tools still has nothing to do with call processing. |
| 21:41.05 | [TK]D-Fender | drclue: This is astill all after the fact manipulations. |
| 21:41.48 | drclue | @fendor , that's odd , as I do process calls via AMI. Of course I also process calls via AGI and often have them work together |
| 21:42.40 | [TK]D-Fender | drclue: call comes in. His the dialplan. Not AMI. AMI may receive a notification that an exten is being executed, but it is not asked what to do. |
| 21:42.49 | *** join/#asterisk quintana (~sylvain@aghnar.doowan.net) |
| 21:43.04 | [TK]D-Fender | drclue: drclue Nor does AMI control the complete flow of a call. it is not "executed" against the call. |
| 21:43.18 | p3nguin | wonders if it is harder to type tk and press the tab-key to correctly address [tk]d-fender or to manually mistype @fendor every time. |
| 21:43.30 | [TK]D-Fender | drclue: call processing is linear, AMI just throws events around "whenever" |
| 21:43.31 | Mish- | Does anyone know of a device that lets a standard phone plug in and then connects it to SIP? |
| 21:43.34 | creativx | @fendrar: eoh hai |
| 21:43.44 | Kobaz | drclue: think of AMI as a 'view into the system'... you can listen for what the system is doing... or you can send the system some instructions |
| 21:43.47 | p3nguin | mish-: yep |
| 21:43.55 | drclue | @fendor You like to do things in the dialplan , and that is fine , differnet strokes for different folks, but some of us find the dial plan annoying. Thats just us. Just as you seem to find ansers that travel beyond the dial plan annoying |
| 21:44.01 | [TK]D-Fender | Mish-: SPA-3102 has a port for a line as well as a phone |
| 21:44.02 | p3nguin | mish-: SPA-3102, SPA-3000, maybe SPA-2000 |
| 21:44.07 | [TK]D-Fender | Mish-: (2) total |
| 21:44.09 | Kobaz | drclue: and the dialplan is the flow of the call... which can be redirected with the AMI... but overall the dialplan is the flow master |
| 21:44.29 | [TK]D-Fender | Mish-: most of their other products are just FXS devices for you to plug in phones. |
| 21:44.44 | p3nguin | mish-: Any normal ATA provides what you've asked for, actually. |
| 21:44.47 | [TK]D-Fender | Kobaz: Call FLOW does not get directed to AMI <- |
| 21:44.54 | Kobaz | [TK]D-Fender: i didn't say that |
| 21:44.57 | [TK]D-Fender | Kobaz: Kobaz Try again |
| 21:45.03 | Kobaz | what i said is correct |
| 21:45.31 | Kobaz | you can move the flow to different parts of the dialplan, using the ami (channel redirect) |
| 21:45.37 | [TK]D-Fender | Kobaz: If you meant in terms of the "point in the dialplan", then I'll accept that :) |
| 21:45.52 | Kobaz | yeah |
| 21:46.13 | Kobaz | drclue: if you are 'programming your dialplan' in AMI... you're doing it wrong |
| 21:46.13 | [TK]D-Fender | Kobaz: It doesn't however start issuing apps against and and completely replace the dialplan for multiple steps |
| 21:46.23 | drclue | @Kobaz A pissing post war over the dialplan versus outside the dialplan is amusing at best. I've seen these aprroach pissing wars for 30 years in various technologies but, even if I say your way is fine and my way is another , yet you want to piss the post, it's a freee world |
| 21:46.33 | [TK]D-Fender | Kobaz: No, right now he is double-selling his tools. |
| 21:47.03 | Kobaz | drclue: you are comparing apples and oranges |
| 21:47.08 | drclue | AMI and AGI together with but cursory dialplan entries make an awesome combination |
| 21:47.23 | [TK]D-Fender | drclue: If you compared learning * dialplan (the more usual set of apps) to your FastAGI solution then there wouldn't be an argument. You are however comparing it to **AMI** This is a complete apples & oranges scenario |
| 21:47.30 | [TK]D-Fender | drclue: STOP DOUBLE-SELLING |
| 21:47.38 | gloin | cripes, you guys, take it to PM |
| 21:47.38 | Kobaz | drclue: there is no 'outside dialplan' |
| 21:47.54 | *** join/#asterisk quintana (~sylvain@aghnar.doowan.net) |
| 21:47.58 | [TK]D-Fender | AGI is an alternative to complete dialplan logic. AMI is NOT |
| 21:48.21 | Kobaz | gloin: the lurkers might find it informative |
| 21:48.37 | gloin | Kobaz: I might, but the emotional context tends to override the factual context. |
| 21:48.44 | Kobaz | some people here have never touched the ami |
| 21:48.55 | Kobaz | gloin: i know... tk gets emotional :P |
| 21:49.03 | [TK]D-Fender | gloin: Probelm is the facts were distorted on entry. |
| 21:49.18 | Kobaz | [TK]D-Fender: are you going to be at astricon |
| 21:49.22 | d|i|g|i|t|a|l | :P |
| 21:49.24 | drclue | @fender , I don't particularly understand your thought of double selling. My real sale here is externalizing life from the dialplan via the other interfaces Asterisk thought wise to provide AGI/AMI. Obviously , you know more than the folks who wrote the Asterisk server |
| 21:49.53 | Mish- | [TK]D-Fender: Excuse my n00bness, but I've just ordered a "SPA3000", would it be possible to connect it to the phone line, a standard phone and have asterisk take care of the "routing" etc for the phone plugged in to the SPA3000? |
| 21:50.00 | Kobaz | drclue: many of us here are also asterisk developers |
| 21:50.21 | Kobaz | Mish-: yes... read the asterisk book too |
| 21:50.31 | [TK]D-Fender | drclue: AMI has nothing to do with taking away the dialplan. it was never a tool for that. |
| 21:50.52 | [TK]D-Fender | Mish-: SPA-3000 is the old discontinued version of the SPA-3102 |
| 21:51.12 | [TK]D-Fender | Mish-: I hope you got it dirt cheap compared to its current version. |
| 21:51.38 | drclue | @kobaz Great , lad to see Asterisk developers in the conversation. FastAGI and AMI should then by those be understood to be a powerful combination that can truly minimize the need for the traditional dialplan if a particular implementor decided to take that approach |
| 21:52.36 | p3nguin | mish-: If the SPA-3000 has voip-to-pstn and pstn-to-voip gateway, then that's what it was intended to do. |
| 21:52.46 | drclue | @fender Sorry if I did not make it clear enough that overall I like using both AGI and AMI in concert |
| 21:52.55 | Kobaz | drclue: if you really want to... you can use the ami to reduce the need on 'dialplan'... but it's not what it's designed to do... nor is it the recommended approach |
| 21:53.01 | Mish- | [TK]D-Fender: About 2/3 of the cost, is it much worse than the newer model? |
| 21:53.24 | Kobaz | i use agi/ami all the time |
| 21:53.38 | gloin | so I've got a dialplan puzzle: my context is built around the outbound dialer recipe here: http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out+deliver+message (outboundmsg1) but rather than have the T extension just hangup, I want to do 3 cycles, thus my whole thing on the count-- earlier |
| 21:53.42 | Kobaz | but they work together... one is not the replacement for the other |
| 21:53.45 | *** join/#asterisk idespinner (~idespinne@cpe-76-93-115-243.socal.res.rr.com) |
| 21:53.56 | gloin | thing is, when I do that, I end up back at the S extension and the count gets reset |
| 21:54.01 | [TK]D-Fender | Mish-: Not as good at handling echo problems should they arise, not under warranty, not getting firmware updates, etc. If it works it'll be fine though. |
| 21:54.03 | WIMPy | The nice thing about using AMI is that you don't have to fork processes, so it's very resource efficient. |
| 21:54.11 | gloin | I suppose I could count up instead of down though |
| 21:54.13 | *** join/#asterisk quintana (~sylvain@aghnar.doowan.net) |
| 21:54.19 | Kobaz | WIMPy: depends on your design |
| 21:54.24 | gloin | can you increment an uninitialized variable in a dialplan? |
| 21:54.43 | Kobaz | WIMPy: you can write a single process fastagi server and accomplish the same resource efficienty |
| 21:54.43 | idespinner | possible to switch all graphs to show 12hour format instead of 24 hour? |
| 21:54.47 | Kobaz | efficiency |
| 21:54.53 | [TK]D-Fender | gloin: If you didn't initialize it... you just shot yourself in the foot :) |
| 21:54.58 | Kobaz | gloin: you need to set it first |
| 21:55.09 | gloin | [TK]D-Fender: that's the thing, if I set the variable, it happens in the S extension |
| 21:55.12 | WIMPy | And what you want to do in the first place, but compared to AGI it can increase efficiency dramatically. |
| 21:55.21 | gloin | and the T extension takes me back to S and reinitializes the variable |
| 21:55.30 | Kobaz | WIMPy: completely different animals |
| 21:55.49 | [TK]D-Fender | gloin: No, YOUR one does. Maybe you should skip to a priority AFTER the one that INITIALES it <- |
| 21:55.54 | Kobaz | WIMPy: i think we already went through this with drclue |
| 21:55.55 | WIMPy | Kobaz: But you;d still need to connect once per call, wouldn't you? |
| 21:56.10 | Kobaz | WIMPy: you would |
| 21:56.19 | gloin | [TK]D-Fender: tried doing that, can't seem to get it to skip down to the priority I want though |
| 21:56.26 | [TK]D-Fender | gloin: Go pay attention to where you are jumping to |
| 21:56.27 | gloin | or can you do priority 1, 2, n, n, n? |
| 21:56.29 | Kobaz | WIMPy: but simulating dialplan with the ami is just a nightmare waiting to happen |
| 21:56.39 | WIMPy | Kobaz: So on a lot of calls you'd save quite some overhead. |
| 21:56.49 | [TK]D-Fender | gloin: You can just to hard numbered (you can calculate the n's if you want), or use LABELS |
| 21:56.52 | Kobaz | and as always it depends on what you're doing |
| 21:56.57 | drclue | @kobaz I basically decided long ago that everyone looking at asterisk including myself are sorta like the blind men and the elephant proverb. What I decided to do is not the only way to Rome with each method being valid to that person using it as long as it works. I decided to minimize the dialplan, while others decided to live in the dialplan. Nothing wrong with either approach. The only thing that might be wrong is crying b |
| 21:57.12 | WIMPy | Kobaz: I like it. |
| 21:57.55 | Kobaz | drclue: if it works, that's fine... but all we're trying to do is help people not misuderstand what these components do |
| 21:57.56 | [TK]D-Fender | Kobaz: Problem is nothing keeps going back to AMI to let it keep a call alive and feed it more. |
| 21:58.02 | Mish- | Next question, what's the most preferred free windows soft phone application? |
| 21:58.32 | [TK]D-Fender | Kobaz: It was never meant for this. dialplan (and AGI) is linear. AMI is not |
| 21:58.47 | [TK]D-Fender | Mish-: X-Lite and Zoiper |
| 21:58.56 | gloin | Well, here's the context that gets called, fails |
| 21:58.59 | Kobaz | [TK]D-Fender: i could imagine you had one dialplan context that was a looping noop... and every so often you fed the ami a redirect to go do something, and then a redirect to go back to the noop loop |
| 21:59.05 | gloin | http://pastebin.ca/1896919 |
| 21:59.21 | gloin | says [Jul 8 14:57:49] NOTICE[23099]: pbx.c:3744 pbx_extension_helper: No such label 'startalert' in extension 't' in context 'alertcall' |
| 21:59.24 | gloin | [Jul 8 14:57:49] WARNING[23099]: pbx.c:9625 pbx_parseable_goto: Priority 'startalert' must be a number > 0, or valid label |
| 21:59.28 | *** join/#asterisk retentiveboy (~pdugas@69.169.199.82) |
| 21:59.28 | gloin | in cCLI |
| 21:59.29 | Kobaz | WIMPy: drclue [TK]D-Fender... but then you're re-inventing dialplan... doing what dialplan does for you to begin with |
| 21:59.30 | gloin | -c |
| 21:59.36 | drclue | @kobaz so cool , we are in agreement that any particular path to success is valid. Now if everyone could simply agree to that tenant without post pissing than we would be fine |
| 21:59.36 | WIMPy | found a simple Wait fine. |
| 22:00.04 | gloin | am I wrong in waiting for the T extension to get called here? |
| 22:00.20 | Kobaz | drclue, WIMPy: it's more like a best practices sort of issue |
| 22:00.39 | [TK]D-Fender | Kobaz: is there a place where AMI can execute a specific command AND have the dialplanc ontinue where it is? |
| 22:00.58 | Kobaz | if you go into a car mechanic channel, and tell them you are replacing the engine using a butter knife and spoon... they will 1) laugh at you... and 2) instruct you as to the recommended approach |
| 22:01.07 | [TK]D-Fender | gloin: No. You are wrong in where you JUMP to afterwards./ |
| 22:01.19 | gloin | right |
| 22:01.31 | [TK]D-Fender | gloin: : No such label 'startalert' <-- like it says... you didn't use the label right |
| 22:01.41 | gloin | ok |
| 22:01.49 | Kobaz | drclue, WIMPy: so... the ami/agi question... is kind of like that for us |
| 22:01.49 | gloin | exten => s,n(startalert),Playback(system-crashed) |
| 22:01.57 | gloin | that's where I'm setting the label, right? |
| 22:02.07 | Kobaz | [TK]D-Fender: don't know of a nice way to do that other than several redirects |
| 22:02.26 | [TK]D-Fender | gloin: Remember that the label represents a priority in a given exten. It is no just a singular name you can use within the scope of a context to span multiple extens. |
| 22:02.42 | [TK]D-Fender | gloin: Goto(s,startalert) <----- |
| 22:02.58 | gloin | gah! |
| 22:03.00 | gloin | thanks |
| 22:03.01 | gloin | :) |
| 22:03.02 | drclue | The dialplan is a favorite pissing post for those who have invested a lot of time in it. All respect to folks who approach the situation that way , as for many situations the dial plan is fine, but typically I'm involved in real time applications where the dialplan is in that instance a boat anchor. Either reality is fine IMHO, but the work I do is with realities far more dynamic than the diaplan itself is reasonable for |
| 22:03.12 | Kobaz | [TK]D-Fender: Or... you have a while (1) { TryExec(${command}) }... and then set channel variables using ami, to tell dialplan what to do |
| 22:03.49 | gloin | [TK]D-Fender: I was thinking of the label as a discrete element, didn't think of it as a priority |
| 22:04.14 | [TK]D-Fender | Kobaz: First try to dream that AMI could even catch the end of an exten executing (don't think that it notifies at the END, just the start), then the race to stop it from going CIRCULAR |
| 22:04.29 | Kobaz | drclue: so... give us an overview of what you've done to move your dependency on dialplan over to ami |
| 22:04.42 | [TK]D-Fender | Kobaz: Regardless of even being vaguely trigerrable, we still fall into the realm of the insane |
| 22:04.47 | Kobaz | hehe |
| 22:05.01 | drclue | @Kobaz It's a tag team match between AGI and AMI |
| 22:05.30 | [TK]D-Fender | Want to process your entire call in AGI (aside from the singular line that calls it), SURE. Whatever. |
| 22:05.40 | Kobaz | [TK]D-Fender: you would need a tryexec command, set command == '' afterwards.. send a userevent once the command is done... and then in your loop, wait(1) and try again if command=='' |
| 22:05.45 | [TK]D-Fender | claiming to use AMI to completely run your calls = BS. |
| 22:05.49 | Kobaz | drclue: that doesn't tell me much |
| 22:05.49 | drclue | Right now , we are replacing HUDlight and other type applications with AsterClick |
| 22:06.02 | Kobaz | drclue: i use ami and agi together all the time... but i'm still using dialplan |
| 22:06.13 | [TK]D-Fender | Kobaz: And your hack requires all sorts of vague, messy, and DOOMED dialplan to mis-implement the best you can ;) |
| 22:06.28 | [TK]D-Fender | HUDlight isn't a call processing substitute either |
| 22:06.31 | Kobaz | [TK]D-Fender: it'll work... i didn't say it was good... |
| 22:06.43 | [TK]D-Fender | More FOP-grade attendant monitoring. |
| 22:07.12 | [TK]D-Fender | Kobaz: And firing off a nuke under your ass may get you into space... at least little BITS of you |
| 22:07.15 | drclue | @Kobaz You imply your an Asterisk developer , so just think outside the dialplan and think what is possible with coordinated FastAGI/AMI and you'll have pretty much the answer to your question |
| 22:07.39 | Kobaz | i'm not a core developer, but i do contribute |
| 22:07.41 | *** join/#asterisk TimeRider (~steve@109.224.131.68) |
| 22:08.16 | Kobaz | drclue: i know what's possible with coordinated dialplan/ami... I do it all the tiome |
| 22:08.19 | Kobaz | time |
| 22:08.31 | Kobaz | drclue: you haven't explained how you've removed your dependency on dialplan though |
| 22:08.45 | Kobaz | you keep saying how dialplan was your boat anchor... i'm interested to see what you did, and why |
| 22:09.33 | Kobaz | and i know enough about the internals of asterisk to tell you with 100% certainty that ami is not a replacement for dialplan |
| 22:09.38 | *** join/#asterisk Lantizia (~Lantizia@erebus.seaquake.net) |
| 22:10.10 | Kobaz | because the channel *must* be executing dialplan for it to be of any use at all |
| 22:10.14 | Kobaz | it's just how asterisk works |
| 22:10.55 | WIMPy | Which in it self is some sort of limitation. |
| 22:11.08 | Kobaz | i don't see it as a limitation at all |
| 22:11.27 | Kobaz | if you replaced it with something different... call it "system x"... it would still amount to be the same thing |
| 22:11.37 | [TK]D-Fender | Kobaz: Would be nifty to see an immediate hook from channels so that AGI, etc could be used directly without any shell of a dialplan.. but that isn't the case. |
| 22:11.42 | drclue | 1@kobaz while do you keep thinking that AGI/AMI have to be separate? |
| 22:11.46 | Kobaz | the channel needs to sit in an executor that feeds it things to do... and that's what dialplan does |
| 22:12.11 | Kobaz | [TK]D-Fender: that would be cool |
| 22:12.12 | WIMPy | The trouble comes with supplementary features that are not call related. |
| 22:12.35 | [TK]D-Fender | they COMPLEMENT each other as fuctionality goes. there are not the SAME. |
| 22:12.41 | Kobaz | [TK]D-Fender: it would be nice to be able to Play(foo) to channel via the ami, whenever you wanted to... |
| 22:12.53 | [TK]D-Fender | totally different PIECES |
| 22:13.00 | drclue | @kobaz FastAGI is able to make a call to a socket , AMI works with sockets , the middleground can do anything it likes |
| 22:13.18 | Kobaz | drclue: they have to be seperate because they are seperate... if you want them to be the same thing... we have open development contribution... feel free to rewrite the call flow handling and submit your patches |
| 22:13.26 | [TK]D-Fender | drclue: And my file server works with sockets too, so I guess AMI is a FILE SERVER. |
| 22:13.41 | [TK]D-Fender | drclue: Your straw-man is about ready to be set ablaze |
| 22:14.27 | drclue | AMI is a command and response interface , FastAGI is a command and response interface. Am I not free to coordinate the two? |
| 22:14.41 | Kobaz | coordinate them by all means |
| 22:14.54 | Kobaz | it's an excellent use of the system... best tool for the job |
| 22:15.38 | Kobaz | drclue: you still haven't answered my question... i'm interested to see what you've done to remove your dependency on dialplan |
| 22:16.52 | [TK]D-Fender | Hard to coordinate the linear and the non-linear |
| 22:16.56 | Kobaz | it is |
| 22:16.59 | drclue | @kobaz Well , simply think what I could get away with coordinating the two and you have probably a pretty good basic idea of what I'm up to. For the most part I could in a dialplan hand everything off to FastAGI if I wanted to. And in AMI I am free to process each and every event and issue any AMI command I like. Between the two I have very little need for the dialplan |
| 22:17.08 | Kobaz | but it's doable |
| 22:17.29 | Kobaz | uhh |
| 22:17.33 | Kobaz | haha |
| 22:17.43 | Kobaz | FastAGI *is* dialplan |
| 22:18.00 | Kobaz | i think we're argueing over semantics |
| 22:19.17 | drclue | @kobaz Yes FastAGI is something I have to place in the dialplan , but how much of the dialplan do you think I could handle in FastAGI? How many lines of dialplan would I need for a fully exploited FastAGI? |
| 22:19.33 | Kobaz | but fastagi is dialplan |
| 22:19.49 | Kobaz | you can do anything in fastagi that you can do in extensions.conf |
| 22:20.03 | russellb | and more :-) |
| 22:20.07 | Kobaz | fastagi is a way to have an outside script control the dialplan |
| 22:20.10 | Kobaz | russellb: that too |
| 22:20.26 | Kobaz | even though it's an outside script... it's still dialplan |
| 22:20.50 | drclue | Emm , perhaps we are debating terminology as to me in my perception , dialplan is that within the dialplan , not that handled outside the dialplan |
| 22:20.56 | Kobaz | 18:17 <Kobaz> i think we're argueing over semantics |
| 22:20.58 | Kobaz | i already said that |
| 22:20.59 | Kobaz | heh |
| 22:21.06 | *** join/#asterisk [sr] (~Unknowned@pal-213-228-140-150.netvisao.pt) |
| 22:21.08 | [sr] | hi |
| 22:21.09 | Kobaz | <PROTECTED> |
| 22:21.09 | Kobaz | <PROTECTED> |
| 22:21.17 | drclue | Cross posting at it's finest :) |
| 22:21.23 | Kobaz | 2: the meaning of a word, phrase, sentence, or text; "a petty argument about semantics" |
| 22:22.20 | p3nguin | drclue: its finest, actually. |
| 22:22.30 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 22:22.48 | drclue | @kobaz To me , the dialplan is that text file that folks like fender do so well. That wich is externally called by the the dialplan that is not a standard application is where I live |
| 22:23.21 | Kobaz | if you talk to any asterisk peeps... dialplan is extensions.conf/extensions.ael/agi/fastagi |
| 22:23.38 | Kobaz | it's just a terminology issue since the beginning of this whole thing |
| 22:24.18 | p3nguin | The way I see it: If it contains rules for how to handle called or calling numbers, it's dialplan. |
| 22:24.25 | Kobaz | exactly |
| 22:24.42 | Kobaz | p3nguin: with one addition... that it's tied to the channel |
| 22:24.49 | p3nguin | Of course it doesn't have to be limited to only those two factors. |
| 22:24.59 | Kobaz | ie: sending redirect commands to ami... i don't consider to be dialplan |
| 22:25.16 | drclue | @kobaz I certainly agree , various folks and the elephant see the rope,the wall , the tree trunk etc. To me the dialplan is that text file with the windows ini syntax. Once I have extricated myself from that context I'm something else |
| 22:26.15 | Kobaz | if you talk to someone other than me, or p3nguin [TK]D-Fender or russellb.... you'll still confuse the hell out of them when you say you 'arent using dialplan' |
| 22:26.34 | drclue | @kobaz Why are we stuck on this idea that AMI and FastAGI need to be other than friends? |
| 22:26.57 | russellb | I HEARD MY NAME |
| 22:26.59 | Kobaz | not sure what you mean |
| 22:27.02 | Kobaz | licks russellb |
| 22:27.34 | Kobaz | [TK]D-Fender: oh... you didn't say before.... are you going to astricon |
| 22:28.05 | [TK]D-Fender | russellb: I still don't get what all this talk about narcissism has to do with ME... |
| 22:28.31 | Kobaz | drclue: agi/dialplan and ami can surely be friends... they make good friends |
| 22:28.43 | Kobaz | [TK]D-Fender: astricon? |
| 22:28.56 | [TK]D-Fender | Kobaz: Nope... never been. Low likelyhood of ever happening, but if you're ever up this way I'll but the first round :0 |
| 22:29.04 | Kobaz | aww |
| 22:29.13 | Kobaz | which where is this way |
| 22:29.24 | [TK]D-Fender | Kobaz: Montreal |
| 22:29.27 | Kobaz | astricon is near DC this year... it's like a 4 hour drive |
| 22:29.28 | Kobaz | ah |
| 22:29.28 | drclue | When I started working with computers , there was no Apple , there was no Microsoft , so perhaps I have a weird way of looking at things. When I started working with Asterisk as a programmer , I looked at the dialplan and puked. As I said , there is nothing at all wrong with folks who view Asterisk from the perspective of the dialplan, but for me I wanted out of that and have put in the effort to do so |
| 22:29.35 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 22:29.55 | Kobaz | drclue: you don't like the extensions language |
| 22:30.05 | Kobaz | drclue: i understand... i don't like it either |
| 22:30.16 | russellb | extensions language (either .conf or .ael) == dialplan |
| 22:30.37 | Kobaz | and with agi, your sending control of the dialplan to an outside app |
| 22:30.40 | Kobaz | that's all |
| 22:30.42 | Kobaz | it's not magic |
| 22:30.46 | drclue | @kobaz As the type of engineer I am I wanted the nearest escape hatch that would provide the broadest level of support |
| 22:30.50 | [TK]D-Fender | drclue: You are never completely free, and while you can jump to AGI within the first step or two, do not ever mix the terms & purpose of AGI & AMI together. |
| 22:31.03 | russellb | lots of people combine AGI and AMI |
| 22:31.04 | Kobaz | [TK]D-Fender: he was just confused |
| 22:31.09 | [TK]D-Fender | drclue: Completely separate purposes |
| 22:31.10 | russellb | you have to for a lot of more fancy things |
| 22:31.16 | russellb | disagree! |
| 22:31.40 | [TK]D-Fender | Kobaz: Yeah, imagine his as a driver being confused about a brake pedal and an accelerator pedal. |
| 22:31.46 | Kobaz | [TK]D-Fender: yes they are for seperate purposes... but i go and send out ami commands in my agi scripts all the time |
| 22:31.51 | [TK]D-Fender | Kobaz: Just let me know what roads to avoid first |
| 22:31.52 | drclue | @fender Perhaps your MANY efforts at FastAGI/AMI have not ser ved you well, but they have been serving me in a grand fashion |
| 22:32.20 | Kobaz | drclue: i think it's just a language/semantics issue |
| 22:32.59 | [TK]D-Fender | Kobaz: Yes, and you can pretty much issue them from non AGI Dialplan as well. But again... you can issue mail-server queue commands from there. So is AMI now a mail-server processing engine? |
| 22:33.11 | [TK]D-Fender | Kobaz: its the open ended BS that is just out of hand |
| 22:33.22 | Kobaz | i'm actually going to move more of my dialplan logic to be async... have my agi script wait for ami events in order to do the next thing |
| 22:33.40 | russellb | Kobaz: have you seen the async AGI stuff for that? |
| 22:33.40 | Kobaz | [TK]D-Fender: heh |
| 22:33.43 | [TK]D-Fender | Kobaz: AMI doesn't process calls. AMI can smack them around after, but this is after-the-fact. |
| 22:33.50 | drclue | @kobaz You are probably correct that semantics are playing a large role in the discussion. I'm having a ton of success in what I'm doing so I must conclude that the ney sayers and I are simply having mis-understandings in verbage |
| 22:33.53 | Kobaz | [TK]D-Fender: yeah we know |
| 22:34.05 | Kobaz | russellb: no i haven't... what is it? |
| 22:34.22 | russellb | it's something newer ... but it does exactly what you just said |
| 22:34.31 | Kobaz | trunk? |
| 22:34.36 | russellb | a call goes into AGI and processes AGI commands sent asyncronously via AMI |
| 22:34.47 | russellb | maybe even 1.6.something ... |
| 22:34.49 | russellb | i'll look |
| 22:35.16 | Kobaz | russellb: that's kind of like my while (1) {tryExec..} thing i just came up with minutes ago |
| 22:35.16 | russellb | it's in 1.6.2 at least |
| 22:35.47 | Kobaz | i have 1.6.2 on a box |
| 22:35.52 | Kobaz | i would be interested in monkeying with it |
| 22:35.57 | drclue | FastAGI talks to a socket , AMI talks to a socket. The magic that happens behind those sockets has very few limts |
| 22:36.08 | Kobaz | any docs, or app commands to look at? |
| 22:36.36 | Kobaz | drclue: yeah, but fenders point is that just because it's a socket doesn't make them interchangeable or the same |
| 22:37.29 | *** join/#asterisk boodu (~antoine@host-115-126-167-240.adsl.nautile.nc) |
| 22:37.34 | boodu | hello |
| 22:37.50 | russellb | Kobaz: yeah, docs would be nice, i'm having to read the code to figure it out right now ... |
| 22:37.57 | Kobaz | russellb: even if it's a built in app, i might wind up using my own method anyway to accomodate things like timeouts and whatnot |
| 22:38.03 | russellb | you start it using AGI(agi:async) |
| 22:38.07 | Kobaz | ah |
| 22:38.19 | russellb | and then you add commands to the queue using either a CLI command, or a manager action |
| 22:38.35 | Kobaz | interesting |
| 22:38.57 | russellb | and then there is an AGI command "async agi break" or something that will have it exit the AGI if you want |
| 22:39.10 | Kobaz | sounds like what i want |
| 22:39.47 | Kobaz | right now i have lots of busywait loops... and it really puts a big load hit on asterisk once i have dozens of channels hitting these event loops |
| 22:40.27 | drclue | @kobaz On a fairly regular basis I do make updates to things on-line , and while they do not always reflect each and every thing I'm up to , they are themselves useful. You can look at the publicly available code at asterclick.drclue.net for the AMI HTML5 Asterisk interface. I'm not totally ready to release the other code. One can find older versions of my FastAGI code via google |
| 22:40.36 | Kobaz | imagine like 100 channels all doing getvars, wait(0.rand), tons of more stuff, and then waiting again |
| 22:40.55 | russellb | awesome, no docs other than the built in brief docs for the manager action and the asyncagi break AGI command ... |
| 22:41.03 | russellb | nowhere that even tells you how to start an async AGI session |
| 22:41.06 | russellb | our documentation sucks |
| 22:41.31 | Kobaz | haha |
| 22:41.35 | Kobaz | russellb: i could have told you that :) |
| 22:41.41 | russellb | yep, it's not a secret |
| 22:41.48 | Kobaz | although i did help kpflemming with some docs on local channels |
| 22:42.01 | russellb | yeah, much appreciated |
| 22:42.14 | *** part/#asterisk Get_The_Fish (~Get_The_F@c-24-8-50-199.hsd1.co.comcast.net) |
| 22:42.45 | Kobaz | i need to finish my group vars patches so they make it into 1.8 |
| 22:43.16 | drclue | The only reason I do not release the AGI/AMI code is that I'm not totally happy with it, and while it does work I don't want to marry it just yet |
| 22:43.52 | drclue | That is why for now , the FastAGI and AMI code units are released separately |
| 22:44.53 | russellb | Kobaz: the cut off date is approaching very fast |
| 22:45.00 | Kobaz | oo |
| 22:45.01 | russellb | couple of weeks or so ... |
| 22:45.11 | Kobaz | i made all the reviewboard changes except for a few |
| 22:45.12 | drclue | The internal versions currently have three lines of dialplan to do everything |
| 22:53.24 | drclue | @kobaz Given the likes of yourself , I would greatly appreciate you and others of your ilk trying out the AsterClick interface and making suggestions of how I might make it better |
| 22:54.35 | WIMPy | Oh, that async agi/ami thing sounds extremely interesting. |
| 22:54.54 | drclue | One of the things I;m trying to find right now is a couple of Android and iPhone developers to see if I can get AsterClick applications running on a few cellphones |
| 22:55.22 | Kobaz | drclue: it looks interesting |
| 22:55.57 | Kobaz | now if only my unit tests are all successful |
| 22:56.50 | Kobaz | drclue: if you buy me an android, i'll run astriclick on it for you |
| 22:57.06 | drclue | @kobaz Thanks for for the interest, now if I can only find a few folks to beat on the code as relates to their own aspirations so that I can make the code more rounded. |
| 22:57.46 | gloin | weird issue here: http://pastebin.ca/1896956 |
| 22:57.54 | *** join/#asterisk retentiveboy (~pdugas@69.169.199.82) |
| 22:58.03 | gloin | When it switches from one extension to another, there's a 4-5 second pause that I want to eliminate |
| 22:58.11 | gloin | is that just a default somewhere? |
| 22:58.22 | Kobaz | there are no built in pauses |
| 22:58.30 | Kobaz | paste your dialplan output too |
| 22:58.34 | drclue | @kobaz I have a couple of industry sponsors for the code who have been underwriting the coding effort , buying me equipment and such , so it might be possible to get you some equipment |
| 22:58.51 | Kobaz | drclue: cool |
| 22:59.15 | gloin | Kobaz: http://pastebin.ca/1896957 |
| 22:59.24 | Kobaz | exten => s,n,WaitExten(5,m) |
| 22:59.33 | gloin | right |
| 22:59.34 | Kobaz | you sure it's not pausing because of one of your wait()'s |
| 22:59.50 | drclue | @kobaz Simply hit the site and put your request in on the forums and I will see if one of the companies will make a deal |
| 22:59.54 | Kobaz | gloin: no no... not that |
| 22:59.55 | gloin | no, even if I press 1, it should go to s,1 immediately, right? |
| 23:00.00 | Kobaz | gloin: console output i mean |
| 23:00.03 | gloin | ah |
| 23:00.09 | Kobaz | gloin: and in your console output... write in where it pauses |
| 23:00.54 | Kobaz | drclue: tomorrow... i'm about to bust out in a little bit |
| 23:01.16 | drclue | Currently companies like MBIT, cloudvox.com and others are underwriting the costs of development , so if one can make a reasonable argument , they will write a check or send equipment |
| 23:02.00 | gloin | Kobaz: http://pastebin.ca/1896959 |
| 23:02.05 | gloin | Look for "LONG PAUSE" |
| 23:02.22 | Corydon76-dig | gloin: do you have any extensions in "outgoing-pbx" that start with "1"? |
| 23:02.25 | gloin | I pressed the "1" key which should just do Goto s,1 |
| 23:02.28 | *** join/#asterisk Ziaeon (~Alchemist@adsl-074-166-171-132.sip.mia.bellsouth.net) |
| 23:02.38 | Kobaz | and how long is the pause |
| 23:02.39 | *** join/#asterisk p3nguin_ (gpz5GvdFkf@cpe-0210-4bff-fe2b-9074.dhcp.a2infotech.com) |
| 23:02.40 | Kobaz | is it 5 seconds? |
| 23:02.47 | Kobaz | what if you change waitexten to '2' |
| 23:02.52 | Kobaz | is it then 2 seconds? |
| 23:03.05 | gloin | Corydon76-dig: no, outgoing pbx is just to talk to shoretel trunk, Kobaz I'll try your hint |
| 23:03.36 | Corydon76-dig | gloin: and what about "incoming"? |
| 23:03.44 | Kobaz | if it's your waitexten... then you have digit timeout due to dialplan matching overlap |
| 23:04.15 | gloin | no, set waitexten to 2 and pause was at least five seconds (and I did reload) |
| 23:04.34 | gloin | I'm wondering about the "CDR updated" part of the call |
| 23:04.44 | Kobaz | turn off cdr... see if that does anything |
| 23:05.04 | Corydon76-dig | That's not related |
| 23:05.16 | Kobaz | it sounds like dialplan timeout |
| 23:05.28 | Kobaz | Corydon76-dig: i've had weird things happen with i/o |
| 23:05.43 | Corydon76-dig | That's just the cdr structure getting updated from the channel structure. Nothing to do with IO |
| 23:05.48 | gloin | turn off CDR in modules? |
| 23:05.51 | Kobaz | oh okay |
| 23:06.01 | Kobaz | Corydon76-dig: it wouldn't write it out to the db at that point? |
| 23:06.11 | drclue | MBIT has contributed thousands of dollars to the coding effort , while cloudvox.com has donated hundreds of dollars in equipment and servers. Others of made contributions as well. We are hoping for enough donations to bring on-board additional staff |
| 23:06.14 | Mish- | Anyone had expereince with lines like: "chan_sip.c: Failed to authenticate device "0275588204" <sip:0275588204@202.180.76.163>;tag=as683f3e90" appearing in the logs when someone attempts to call in through a SIP provider, the call never connects. |
| 23:06.15 | Corydon76-dig | Nope, not until the call is finished |
| 23:06.50 | Kobaz | yeah... it's been a while since i did regular cdr |
| 23:07.28 | Corydon76-dig | gloin: pastebin your ENTIRE dialplan, not just the one context |
| 23:09.08 | gloin | http://pastebin.ca/1896968 |
| 23:09.17 | Kobaz | Corydon76-dig: i cant stand the standard cdr system, i wrote my own that's more realtime and much easier to trace |
| 23:09.58 | Corydon76-dig | gloin: As I said, it's youre outbound-pbx that is conflicting |
| 23:10.12 | gloin | matching? |
| 23:10.25 | Corydon76-dig | _XXXX CAN match any extension that starts with "1" |
| 23:10.37 | gloin | hm |
| 23:10.44 | *** join/#asterisk retentiveboy (~pdugas@69.169.199.82) |
| 23:10.45 | Kobaz | yeah, that's the spot |
| 23:10.48 | gloin | we have valid extensions within the shoretel system that do start with one |
| 23:10.48 | Kobaz | it's a dialplan digit timeout |
| 23:10.54 | Corydon76-dig | and so it has to wait TIMEOUT(digits) seconds before the extension is considered complete |
| 23:11.21 | Kobaz | Corydon76-dig: you might want to consider using Read()... and explicitly match on the things you want |
| 23:11.24 | Kobaz | er |
| 23:11.24 | Kobaz | gloin: |
| 23:11.27 | Kobaz | not you |
| 23:12.03 | Kobaz | gloin: or make a new context and don't include outbound-pbx and incoming... and just have 1 and 2 |
| 23:12.33 | Kobaz | do you need to have access to outbound-pbx and incoming? |
| 23:12.45 | Kobaz | if not you can just remove them from that context and it should fix it |
| 23:13.52 | Corydon76-dig | If you do need access to other extensions, I'd suggest using "3" to jump out to another context that includes those extensions |
| 23:14.25 | gloin | I won't, apparently. |
| 23:14.56 | gloin | I was operating under the notion that I needed that outbound context to reach an external number |
| 23:15.11 | Kobaz | not necessarily |
| 23:15.30 | Kobaz | you can do _3XXXX => Goto(outbound-context-...etc |
| 23:15.52 | gloin | well, that was kind of weird |
| 23:15.59 | gloin | I commented out the outbound context |
| 23:16.09 | gloin | pauses went away immediately |
| 23:16.16 | gloin | works fine calling my shoretel extension |
| 23:16.18 | Kobaz | it's a digit matching overlap timeout |
| 23:16.25 | Kobaz | in your asterisk dialplan |
| 23:16.27 | gloin | I see, and thanks for that. |
| 23:16.34 | gloin | Now I tried having it call my cellphone |
| 23:16.49 | gloin | and it started playing the alert messages before my cellphone even rang (which it eventually did) |
| 23:16.53 | Kobaz | if it doesn't match something immediatly, it will wait for more digits |
| 23:17.47 | gloin | makes sense |
| 23:17.58 | gloin | but why would it detect a ring on the cellphone as a pickup? |
| 23:18.07 | Kobaz | it's your cell carrier |
| 23:18.18 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 23:18.19 | gloin | lovely |
| 23:18.23 | Kobaz | many cell providers will answer() the call immediatly, and then ring your cell phone |
| 23:18.26 | Kobaz | it's a pain in the ass |
| 23:18.58 | gloin | nods |
| 23:19.06 | gloin | Verizon being a pain in the ass? Perish the thought! |
| 23:19.10 | Kobaz | heh |
| 23:19.20 | gloin | has anyone come up with a reasonable workaround? |
| 23:20.21 | Kobaz | not me |
| 23:20.40 | gloin | Wait(15) at the beginning of the call... |
| 23:20.44 | gloin | or 30 |
| 23:20.47 | gloin | crap :( |
| 23:21.07 | Kobaz | off the top of my head, you can do what google voice does, and wait for someone to pick up and dial 1 to signal 'hey i'm here' |
| 23:21.21 | Kobaz | you can do it in asterisk, it would be some tricky dialplan |
| 23:21.30 | Kobaz | and it's almost time to bust out |
| 23:21.36 | gloin | yeah, it's about that time |
| 23:21.39 | gloin | thanks for your help |
| 23:21.47 | gloin | Kobaz and [TK]D-Fender and Corydon76-dig |
| 23:21.50 | gloin | etc |
| 23:22.12 | gloin | (FWIW, T-Mobile does the same answer() thing as verizon |
| 23:22.13 | gloin | ) |
| 23:25.22 | Kobaz | gym time |
| 23:25.35 | drclue | @fender If one were allowed to ask for anything they wanted in an AMI/AGI tool , what would YOU ask for? |
| 23:26.44 | Kobaz | drclue: it would need to make me coffee, give me much luvin, and not fight back |
| 23:27.20 | [TK]D-Fender | Kobaz: My normal dialplan does that already... |
| 23:27.24 | [TK]D-Fender | Kobaz: LITERALLY |
| 23:27.49 | Kobaz | even the hot and steamy luvin? |
| 23:28.20 | drclue | @kobaz I might be able to to address the "not fight back" portion of your request, but the make coffee and give luvin aspect would be application specific :) |
| 23:28.23 | *** join/#asterisk obnauticus (~obnauticu@about/windows/regular/obnauticus) |
| 23:28.28 | Kobaz | heh |
| 23:28.29 | Kobaz | anyways |
| 23:28.33 | Kobaz | i'm out |
| 23:29.01 | [TK]D-Fender | drclue: AGI isn't a "tool" so much. Tools do specific tasks when needed. AGI is always personal. AMI as a tools has many specific profitable goals. Which is why FOP, HUD, exist |
| 23:29.05 | *** join/#asterisk ariel_ (~chatzilla@c-24-127-196-248.hsd1.fl.comcast.net) |
| 23:29.41 | [TK]D-Fender | drclue: They let you monitor stuf. trigger recordings, redirect calls, monitor Meetmes, etc. Track hints, phone status. use for controlling a concurrent outbound auto-dialer, etc |
| 23:29.57 | [TK]D-Fender | drclue: EVERYBODY will have different needs & wants. |
| 23:30.05 | [TK]D-Fender | drclue: I specifically need NOTHING, |
| 23:30.07 | *** join/#asterisk valajbeg (~hamo@b207c4.pptp-gw52.cable-internet.GlobalNET.ba) |
| 23:30.30 | [TK]D-Fender | drclue: All the extra AMI & AGI anyone could code means little to my personal usage. *I* do not need it |
| 23:30.51 | [TK]D-Fender | drclue: My calls come in. They follow my extremely functional and minimalist dialplan and I'm happy. |
| 23:31.12 | [TK]D-Fender | drclue: Others may want a comprehensive voice conferencing management tool though |
| 23:31.21 | [TK]D-Fender | drclue: but that would not be "me" |
| 23:32.03 | [TK]D-Fender | drclue: Maybe AMI monitoring to do better management of call parking. |
| 23:32.23 | [TK]D-Fender | drclue: I use AMI to provide live call Queue stats on my phones. |
| 23:32.35 | [TK]D-Fender | drclue: For those not using queues... it'd be useless. |
| 23:33.04 | *** join/#asterisk retentiveboy (~pdugas@69.169.199.82) |
| 23:33.36 | [TK]D-Fender | drclue: Advice I'd give you : be very clear in the goals each tool you implement is meant to acheive and how it fits into a user's complete picture. If the fit isn't right it might devalidate the tool altogether. |
| 23:34.21 | drclue | @fender The particular application "HUD" may be having some issues relating to the latest version of Asterisk. Now while they may address those things , AsterClick places the power to address things in everyones hands as opposed to waiting for a gift. I'm happy that your personal knowledge and expertise in addressing everything with miles of dialplan is such as it is and in a certain way that is laudabel , but I'm aiming to add |
| 23:34.22 | [TK]D-Fender | drclue: the purpose and usability will determine if anyone cares. Get too specific and if it isn't something that really everyone could use a better version of and doesn't ahve to tailor to them in any SPECIAL way then hey, why not |
| 23:35.34 | [TK]D-Fender | drclue: Please note that you seem to have taken our discussion as being against the benifits that your new AsteriskClick project offers. This is NOT the case. You mentioned it as part of a solution to probalme that did NOT have any relationship to the things it offers. |
| 23:35.56 | [TK]D-Fender | drclue: I'd sure your tool has all sorts of appeal to untold numbers of users. just not HIM <-------- |
| 23:36.53 | [TK]D-Fender | drclue: Please understand this. I was talking about that ONE PERSON. and one ASPECT he was dealing with. AsterClick != replacement for dialplan. Whatever your mystery AGI stuff does... maybe, but I have no functional details on what it offers as you claim it replaces all other processing logic. |
| 23:37.23 | [TK]D-Fender | drclue: I'd like to THINK that it covers at least the cheap-ass stuff most GUI's do like IVR's, cookie-cutter call flow, etc |
| 23:37.29 | drclue | @fender You have not made as much as a single positive comment towards AsterClick, and while you are certainly not required to do so, I've been of the hopes that folks like yourself could offer positive feedback and help improve the offering |
| 23:38.11 | [TK]D-Fender | drclue: I didn not say a single NEGATIVE thing about it. I just said that it was not a tool that had any relationship to that one person's personal goal <- |
| 23:38.35 | [TK]D-Fender | drclue: You seem to be taking that as "my tool is useless". I did NOT say that. |
| 23:39.04 | [TK]D-Fender | drclue: You were offering a hammer to someone who needed a screwdriver. I'm sure you made a very nice hammer. It simply wasn't what HE needed |
| 23:39.12 | [TK]D-Fender | drclue: Am I clear yet? |
| 23:40.04 | drclue | @fender Lets not argue about your previous postings than , but rather , lets see what we can do towards making this effort a positive contribution to the community that addresses both the community at large and those things that you yourself might find to be a positive contribution |
| 23:40.09 | [TK]D-Fender | "Hi, I need a screwdriver!", "Here, this hammer will do the job for you. It is very nice!" "But I have to tighten this screw, you hammer isn't meant to do that kind of job!" |
| 23:41.07 | [TK]D-Fender | drclue: that WAS the only "argument" I had. the relationship between the goal in mind (dialplan / linear call processing), and AMI (event driven system monitoring and minor call manipulation) |
| 23:41.29 | [TK]D-Fender | drclue: If you can follow and accept this we can consider that conversation concluded |
| 23:42.28 | [TK]D-Fender | drclue: If you want to get peoples ideas on things you might want to add to your project it'd help to know what it does already offer and what your intention was in making it the way you did |
| 23:42.30 | drclue | @fender there is no need to limit the AsterClick interaction to AMI. I have AGI tools tool as well and aim to make both AMI and AGI work together , so lets look at it from that perspective if you like |
| 23:43.05 | [TK]D-Fender | drclue: Well I know nothing about the specific functionality it offers right now. |
| 23:43.31 | [TK]D-Fender | drclue: And unfortunately myy time is up for tonight. Feel free to bring it up as a topic of its own whenever you see me next. |
| 23:44.23 | [TK]D-Fender | drclue: Any suggestions I make will be blinded by the fact I don't even own an Apple device to run that from if you are still Aple-centric in your coding. |
| 23:44.24 | drclue | Projects like AJAM and other Javascript tools in no way can do what AsterClick does , as those tools rely on bandwidth hogging polling while AsterClick is event driven. You do know the difference right? |
| 23:44.41 | [TK]D-Fender | drclue: If it is normal browser friendly then I could do more. |
| 23:45.30 | [TK]D-Fender | drclue: I'm not in a postiion to comment on your particular approach not know just how much is client vs server. |
| 23:45.36 | [TK]D-Fender | drclue: But we can look at thqat later. |
| 23:45.48 | [TK]D-Fender | drclue: So consider this bridge un-burned. |
| 23:45.57 | drclue | @fender Ithe AsterClick tool does not care of you have an apple device , a desktop application , or whatever, although it does aim to provide development avenues for many popular situations |
| 23:46.01 | [TK]D-Fender | heads off for the night... |
| 23:46.29 | [TK]D-Fender | drclue: Ok, you can show me in more appropriate detail later and you are welcome to my insights. |
| 23:47.10 | [TK]D-Fender | is actually going now... |
| 23:47.28 | drclue | The basic interface for AsterClick is based upon XML and HTML5 WebSockets , currently supported by Chrome Dev, Firefox Beta , Safari main and soon even IE. |
| 23:48.16 | gloin | is it possible to set a group of off-site numbers within *? |
| 23:48.37 | gloin | basically I want to set a round-robin group of cellphones to call with all the above fun |
| 23:48.38 | Gugge | what is the correct way to specify a sound language from the dialplan in 1.6.1 ? |
| 23:48.50 | gloin | GROUP() function seems to be for channels if I read correctly |
| 23:49.43 | drclue | @fender The dev versions of the code both server side and client side are publicly available at asterclick.drclue.net and will be refined by the comments of those trying it out prior to Alpha , beta and production releases |
| 23:49.57 | bmoraca_work | gloin: you can use static agents with local extensions to accomodate a round-robin of agents |
| 23:50.07 | bmoraca_work | in the queues, that is |
| 23:51.03 | gloin | bmoraca_work: and then just forward the agent extensions to the externals? |
| 23:51.33 | bmoraca_work | gloin: no...the static queue members would be the Local channel extensions for those outside numbers |
| 23:51.34 | carrar | Y*A*W*M |
| 23:51.38 | carrar | N |
| 23:51.44 | gloin | :P |
| 23:51.44 | bmoraca_work | i shouldn't have used the term "agent" |
| 23:52.52 | bmoraca_work | gloin: member => Local/5551234@outbound-context |
| 23:55.19 | gloin | so with my weird setup it'd be Local/SIP/ShoreTel/5551212@outbound-pbx then? |
| 23:55.54 | bmoraca_work | no |
| 23:56.19 | gloin | eh |
| 23:56.40 | gloin | dialing out goes * > ShoreTel SIP Trunk > PTSN |
| 23:56.44 | bmoraca_work | your outbound-pbx context, is that the context that identifies which dialed numbers go where? |
| 23:57.12 | bmoraca_work | if so, you would do Local/5551212@outbound-pbx |
| 23:57.29 | gloin | ah |
| 23:57.37 | bmoraca_work | local extensions do nothing more than redirect a channel to a specific extension within the targeted context |
| 23:57.45 | gloin | and then how do I assign those agent entries to a single extension? |
| 23:57.50 | bmoraca_work | or, rather, i should say "the current channel" |
| 23:58.04 | bmoraca_work | those local channels are members of a queue |
| 23:59.00 | bmoraca_work | so when someone dials that, you say exten => 1234,1,Queue(externagents)...yada yada yada |
| 23:59.26 | bmoraca_work | within queues.conf, you specify your queue to have your round-robin deal |
| 23:59.36 | bmoraca_work | http://www.voip-info.org/wiki/view/Asterisk+call+queues |
| 23:59.44 | gloin | heh |
| 23:59.49 | gloin | I'm at that page just now :) |