| 00:00.14 | delmar | file, right. I use busy detect actually.. and it works ok. |
| 00:00.19 | delmar | file, but its dodgy I hear |
| 00:00.29 | delmar | file, anyway.. since u understand what im on about... |
| 00:01.15 | delmar | file, the problem i have is that... the call hits the * box, which has tdm400p in it... dials out via FXO-4 .. which has a gsm gateway on it.. just providing 2wire dialtone to the FXO... |
| 00:01.29 | file | I already don't like where this is going |
| 00:01.57 | delmar | file, the problem is.. as soon as the call is placed out the FXO.. and the caller is connected... there is a huge amount of silence while the gateway gets the call established on the gsm network |
| 00:02.14 | joburg | you get gsm gateways that has a tone while it's processing the call |
| 00:02.22 | delmar | file, i can play ring tones and all that.. right up to when the FXO goes "Answered" ... then we are waiting for ages... for the call to get going |
| 00:02.28 | joburg | maybe thats your solution |
| 00:02.33 | delmar | joburg, right. |
| 00:02.39 | delmar | joburg, that sounds damn perfect. |
| 00:03.15 | delmar | file, so what i was wondering is.. if there is a way to inject some noise.. either to the Zap channel.. or on the SIP side.. or.. something |
| 00:03.41 | delmar | file, for about 10seconds.. long enuf for the gsm call progress tones to kick in. |
| 00:03.51 | file | ...no |
| 00:04.02 | delmar | nah i didnt think so either. |
| 00:04.55 | delmar | so.. back to my remote-access DISA thingie problem... |
| 00:05.08 | delmar | joburg, I hear what u are saying regards dial patterns... |
| 00:06.34 | delmar | joburg, however.. for example... cell phone numbers here.. come in different lengths... 02XXXXXXX and 02XXXXXXXXX sorta thing. so I guess I can do 02XXXXXXX. |
| 00:07.11 | delmar | surely there must be a way to lengthen the wait time to at least 3secs. right now it seems shorter than that. |
| 00:07.18 | joburg | no rather have them all seperate |
| 00:07.25 | compu73rg33k | are voip plans cheap? |
| 00:08.06 | delmar | joburg, how so? |
| 00:08.30 | joburg | iow : have _02XXXXXXX and _02XXXXXXXXX |
| 00:09.33 | delmar | joburg, that said.... |
| 00:09.51 | delmar | joburg, im not sure the matching has anything to do with it... |
| 00:10.15 | delmar | joburg, the call hasnt been placed via the dialplan at this point.... |
| 00:10.59 | *** part/#asterisk Z_God (n=Z_God@jabber.xs4all.nl) |
| 00:10.59 | delmar | joburg, but ill give this a go |
| 00:10.59 | delmar | joburg, the longer match before the shorter I would assume, otherwise it will match the shorter one always... |
| 00:11.39 | joburg | if you dont have them both the shorter one will not be exepted |
| 00:11.55 | *** join/#asterisk ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 00:12.53 | joburg | instead of DISA you can use a ivr and force the user the end with a # |
| 00:14.22 | joburg | I allow DISA only based on the callers Callder ID for example , adding to the security then taking them to a menu where their calls gets processed |
| 00:15.17 | delmar | joburg, I wouldnt consider CallerID to be a form of security. |
| 00:15.30 | delmar | joburg, a security hole in most respects |
| 00:16.03 | joburg | you can also add a pin coupled to a callerid |
| 00:16.14 | delmar | joburg, so .. the pattern match for cell phones .. not such a big issue.... and national dialing.. are fixed length also.. but waht about international? |
| 00:16.28 | joburg | it also depends in what country you are i guess |
| 00:16.47 | delmar | I can think of a fairly short number already... 00XXXXXXXX |
| 00:16.57 | delmar | but there are plenty that will be longer than this |
| 00:17.29 | joburg | the same goes 4 international yes - have all the possibilities seperate |
| 00:17.34 | delmar | anyway.. im still thinking that its a wait time issue ahead of the matching which is done in the dialplan when the call is actually being placed.. so ill go test that a sec |
| 00:17.43 | delmar | joburg, u reckon? |
| 00:18.44 | delmar | joburg, I think it would be easier if the damn thing would wait a sec or two longer for the user to finish entering :P |
| 00:18.52 | joburg | gottago - its past 2am already! |
| 00:19.19 | joburg | cheera |
| 00:19.39 | *** part/#asterisk joburg (n=voipmagi@vc-196-207-36-176.3g.vodacom.co.za) |
| 00:32.13 | *** join/#asterisk vaq (n=vaggie@0x57306388.rdnxx5.adsl-dhcp.tele.dk) |
| 00:32.15 | vaq | Hello |
| 00:32.20 | vaq | is anybody online? |
| 00:32.39 | vaq | My asterisk is up and running with a VoIP provider, now how can i connect skype with my asterisk server |
| 00:33.42 | vaq | ? |
| 00:34.15 | Nugget | you can't |
| 00:35.01 | vaq | what? |
| 00:35.06 | vaq | Not with skypeOut? |
| 00:35.08 | crochat | Nugget: That's not true... he can with a strange manipulation ;-) |
| 00:35.20 | vaq | how |
| 00:36.01 | vaq | ? |
| 00:36.22 | Nugget | well you can buy one of those cheap skype fxs thingeys and then plug it into a sip or iax fxo device. but I didn't expect that was what you were really asking. |
| 00:36.23 | crochat | In my house, I can call Skype contact through my Asterisk server, and I can also receive Skype calls on my VoIP phones (SIP, IAX, etc...) |
| 00:36.30 | hads | telnet |
| 00:36.30 | Nugget | telnet is eeeeeeevil! |
| 00:36.55 | vaq | crochat how? |
| 00:37.00 | vaq | please tell me how |
| 00:37.12 | crochat | vaq: I wrote a howto, but it's in french... sorry ;-) |
| 00:37.17 | crochat | vaq: http://www.allo.ch/phpbb2/viewtopic.php?t=15502 |
| 00:37.29 | Nugget | those nutty swiss. |
| 00:37.47 | vaq | crochat: i cant read that, can we go in private? |
| 00:38.04 | file | Nugget: !!! |
| 00:38.07 | Nugget | moo |
| 00:38.33 | crochat | vaq: No, sorry, it's 02:38 AM in Switzerland... I'll go in bed ! |
| 00:38.45 | Nugget | crochat: where in switzerland are you? |
| 00:39.27 | crochat | Nugget: La Chaux-de-Fonds... the highest city (city=more than 10000 persons) in Europe |
| 00:39.31 | file | it's a trap! don't answer |
| 00:39.36 | file | Nugget is from the CIA |
| 00:39.39 | delmar | bah. I was right. its nothing to do with pattern matching. |
| 00:39.49 | crochat | file: lol |
| 00:40.15 | Un1x | anyone know a site where i can purchase a DID? |
| 00:40.27 | delmar | Un1x, what state? |
| 00:40.44 | crochat | vaq: You can ask for a translation on the allo.ch forum, or write me an email... but I haven't much time now ! I have an exam in one week :-( |
| 00:40.52 | delmar | Un1x, you can get a free Iowa number from trxtel |
| 00:41.09 | *** join/#asterisk Mportnoy (n=test@201.199.76.194) |
| 00:41.19 | Un1x | delmar i dont need usa i need canadian numbers :D |
| 00:41.34 | crochat | Nugget: Basel is one hour (in car) far from me ;-) |
| 00:41.36 | delmar | Un1x, no clue. |
| 00:41.45 | Nugget | basel is nice. :) |
| 00:42.00 | crochat | Nugget: Geneva is better ;-) |
| 00:42.26 | crochat | Nugget: For me, Geneva is the most beautiful city in Switzerland |
| 00:42.36 | Nugget | yes, but basel is the weirdest. :) |
| 00:42.38 | crochat | Nugget: You should go in Geneva |
| 00:42.43 | Nugget | I've been |
| 00:42.53 | crochat | Oh, good |
| 00:42.56 | Nugget | I have been to basel and zurich and geneva |
| 00:43.44 | crochat | Ok guys ! Sorry, vaq ! Probably somebody here can translate my howto for you, or you can ask Google ;-)) |
| 00:44.04 | *** join/#asterisk webman (n=chatzill@200.179.233.220.exetel.com.au) |
| 00:44.10 | Nugget | @+ |
| 00:44.13 | crochat | Gooooooooooogle powa ;-) |
| 00:44.28 | crochat | Bye ! |
| 00:44.47 | crochat | Nugget: Tu parles français ? |
| 00:44.59 | Nugget | non |
| 00:45.02 | crochat | Nugget: Oups, je vais me faire kicker... héhé |
| 00:45.34 | vaq | My asterisk is up and running with a VoIP provider, now how can i connect skype with my asterisk server |
| 00:45.39 | vaq | does anybody have any good howtos |
| 00:46.09 | webman | vaq: you need an FXO interface, and a skype FXS interface, or vice versa |
| 00:46.22 | vaq | no |
| 00:46.27 | vaq | i can do it with asterisk and skypeOut |
| 00:46.34 | crochat | vaq: Good luck ! I must say that you must have an Asterisk server, and a f*** Windows computer running Skype... and skipe2sip software |
| 00:46.50 | vaq | crochat: I got skype2Sip! |
| 00:46.56 | crochat | s/skipe2sip/skype2sip/ |
| 00:46.59 | vaq | crochat: but im getting: Aug 20 02:44:35 NOTICE[19322]: chan_sip.c:7708 handle_request: Registration from '<sip:vaq@10.0.0.1>' failed for '10.0.0.104' |
| 00:47.38 | vaq | ? |
| 00:48.10 | crochat | vaq: If your Skype contact name is "vaq", you can't call "vaq", even for testing ! |
| 00:48.27 | vaq | crochat my skype name is vaggie1 |
| 00:48.34 | vaq | and my asterisk account name is vaq |
| 00:48.41 | crochat | s/vaq/vaggiel/ |
| 00:48.51 | vaq | + |
| 00:48.55 | vaq | ?? |
| 00:49.00 | crochat | forget it |
| 00:49.13 | vaq | thanks for your great help |
| 00:49.51 | crochat | vaq: Try to translate my howto with Google |
| 00:50.13 | vaq | i dont know how to do that, but i followed your guide for setting up the sip.conf |
| 00:50.15 | vaq | didnt work out |
| 00:50.26 | crochat | vaq: Or register on the forum, and ask for a translation ! Perhaps I could make that today ! ok ? |
| 00:50.48 | vaq | ok |
| 00:51.24 | crochat | Good night, guys ! |
| 00:51.31 | crochat | ++ |
| 00:51.38 | Mportnoy | ANyone from Costa Rica? |
| 00:51.38 | *** join/#asterisk RoyK (n=roy@ti211210a080-2688.bb.online.no) |
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| 01:54.05 | *** join/#asterisk ManxPower (n=eric@69-2-85-41.wan.networktel.net) |
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| 02:15.13 | *** join/#asterisk talljon84 (n=jonathan@24-159-230-113.dhcp.mdsn.wi.charter.com) |
| 02:16.34 | talljon84 | Is anyone using the new 8.x SIP firmware for the Cisco phones? I have a question about the behavior of the MWI light with it: It flashed when there was a new message in the 7.5 firmware but doesn't under 8.0. Is there a new paramater that I need to configure or is anyone aware of a bug? |
| 02:18.01 | qai | Question for those dialplan guru's - my VOIP service provider allows adapter connected users and asterisk server connected users. He has a feature for the adapter connected users that allows them to initiate 3-way calling by dialing the first number, pressing hook flash, then appending *23 to the beginning of the second #, then via another hook flash, all three parties are connected. Anyone have an idea what the *23 dialplan would look like? |
| 02:24.59 | qai | talljon84 - http://www.voip-info.org/wiki/index.php?page=Asterisk+phone+cisco+79xx Seems as though there are a few issues being reported. |
| 02:29.52 | talljon84 | qai: thanks |
| 02:30.44 | *** join/#asterisk Gabriel25 (n=gabe@user-12lcg7s.cable.mindspring.com) |
| 02:31.33 | Gabriel25 | hi guys I find http://nerdvittles.com/index.php?p=141 Mailcall for asterisk |
| 02:31.50 | Gabriel25 | I don1t have trixbox ... but I want to use this ... |
| 02:32.39 | Gabriel25 | where can I add the basic code ? In extension.conf ? |
| 02:32.42 | Gabriel25 | but where ? |
| 02:47.57 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 02:57.23 | *** join/#asterisk LoneShadow (n=duh@59.92.140.214) |
| 02:58.30 | LoneShadow | anyone know for what reasons spa3k's line1 would keep going off hook ? |
| 03:03.17 | *** join/#asterisk slackuser_ (n=FullT@201009027088.user.veloxzone.com.br) |
| 03:03.27 | *** join/#asterisk switch (n=switch@61.206.115.5.user.ad.il24.net) [NETSPLIT VICTIM] |
| 03:03.58 | *** join/#asterisk dos000 (n=dos000@i216-58-65-76.cybersurf.com) |
| 03:04.06 | *** join/#asterisk coppice (n=chatzill@229.166.17.210.dyn.pacific.net.hk) |
| 03:08.33 | *** join/#asterisk nailbags|work (n=nailbags@149.171.94.134) [NETSPLIT VICTIM] |
| 03:09.34 | *** join/#asterisk signuts (n=signuts@sig.triton.net) |
| 03:11.25 | *** join/#asterisk yxa (i=lonari@cm121.gamma228.maxonline.com.sg) |
| 03:13.08 | *** join/#asterisk sgorilla3 (n=sgorilla@user-0cetq2d.cable.mindspring.com) |
| 03:45.27 | *** join/#asterisk L|NUX (n=linux@202.5.145.56) |
| 03:52.10 | *** join/#asterisk sid (i=emanresu@tor/regular/sid) |
| 04:04.18 | sid | Windows binary anywhere? |
| 04:04.23 | *** join/#asterisk ReD-MaN (i=redman@dhcp-0-2-b3-9a-4a-5b.cpe.quickclic.net) |
| 04:05.29 | sid | I wanted to setup a system..so I could use my computer(in New York) and connect to my brother in law's computer in Romania..then from his computer jump onto his phone line...and make phone calls with his phone line. So I don't have to pay rates per minute from America(New York) to Europe(Romania). I can just pay rates from his city in Romania to his town in Romania. heh |
| 04:05.54 | ManxPower | sid, the best thing to do is use Skype |
| 04:06.25 | ManxPower | or FWD or any number of similar services |
| 04:06.45 | sid | ManxPower: right, but I want to call my aunt and uncles, and they don't have computers. |
| 04:07.30 | ManxPower | sid, Asterisk does not work on Windows. Asterisk requires Linux. In order for you to set up a system like you want you will have to learn telecom, networking, Linux, and VoIP. |
| 04:07.43 | ManxPower | Do you really want to do that to save a few dollars? |
| 04:07.44 | sid | damn |
| 04:07.56 | ManxPower | sid, Asterisk is phone system. |
| 04:07.56 | sid | It's like $200 a month for phone bill |
| 04:08.05 | ManxPower | sid, How much do you pay per min? |
| 04:08.08 | sid | Verizon and Vonage both offer $0.33 cents per minute |
| 04:09.02 | sid | I use Debian GNU / Linux sid on my laptop right now. |
| 04:09.09 | sid | heh, I can apt-get install asterisk and I'm done in 5 seconds. |
| 04:09.15 | sid | But my brother in law uses Windows XP |
| 04:09.25 | ManxPower | Are you calling cell phones? |
| 04:09.39 | ManxPower | Teliax has calls to Romania for 8 - 10 cents/min |
| 04:12.08 | ManxPower | sid, in theory you can get a SIPura box that has FXO (phone line) and FXS (phone) ports on it and use that to get your call out to the Romanian PSTN, no asterisk needed |
| 04:21.20 | Gabriel25 | who is from romania ? |
| 04:21.22 | Gabriel25 | :) |
| 04:22.10 | Gabriel25 | sid ... |
| 04:24.01 | Gabriel25 | sid maybe you want to check http://www.telefonip.ro/ |
| 04:24.45 | sid | ManxPower: I don't know what calling cell phones are |
| 04:25.44 | ManxPower | sid, mobile - cell phone |
| 04:26.48 | sid | She uses motorolla cell phone |
| 04:28.21 | ManxPower | calling mobile phones is expensive in most of the world. |
| 04:28.56 | file | JunK-Y: one of your patches is silly |
| 04:29.03 | Gabriel25 | especially in Romania |
| 04:29.29 | Gabriel25 | ivn if you are in romania From cel to cel it`s 15 c an min |
| 04:32.28 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 04:32.33 | shmaltz | hi every1 |
| 04:54.50 | pcm | it's late |
| 05:19.40 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-25-149.cybersurf.com) |
| 05:21.25 | *** join/#asterisk ^Hitch-Dubai (n=none@ner-as11379.alshamil.net.ae) |
| 05:22.14 | ^Hitch-Dubai | ho |
| 05:22.16 | ^Hitch-Dubai | hi |
| 05:23.57 | *** part/#asterisk sgorilla3 (n=sgorilla@user-0cetq2d.cable.mindspring.com) |
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| 05:41.36 | novafirst | hi guys |
| 05:42.20 | novafirst | this is my first visit to this channel, but probably not the last one |
| 05:44.20 | novafirst | I have one problem. Asterisk is trying to run under asterisk:asterisk username/group but I have another username/group set so how do I direct asterisk not to use default one |
| 05:47.00 | intralanman | are you using an rc script to start it? |
| 05:47.05 | intralanman | and what OS? |
| 05:50.08 | *** part/#asterisk sid (i=emanresu@tor/regular/sid) |
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| 06:03.10 | *** part/#asterisk pcm (n=pcm@68.159.139.234) |
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| 07:15.56 | adelas | hey, what is easier or better for a noobie with alot of customization need, trixbox or freepbx (which one has more support ;))? |
| 07:18.01 | *** join/#asterisk Assid (i=assid@203.115.83.215) |
| 07:18.58 | hads|home | adelas: Both are supported in #freepbx I believe. |
| 07:19.33 | adelas | is there any major difference? |
| 07:20.01 | hads|home | I don't think you'll find many people here using either (me included). |
| 07:20.23 | hads|home | freepbx is a web GUI that is used by trixbox. |
| 07:20.31 | adelas | ah okay |
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| 07:22.48 | Gabriel25 | how I make asterisk to ...read a text file ? |
| 07:23.28 | Gabriel25 | I want to create an extension when people call that extension to start reading a text file |
| 07:23.50 | hads|home | What do you mean by reading? Text to speech? |
| 07:24.00 | Gabriel25 | yes |
| 07:24.27 | Gabriel25 | i have to do it in extension_custom.conf |
| 07:24.38 | Gabriel25 | but I don`t know how ... |
| 07:24.46 | hads|home | festival and cepstral are the most common from what I know. |
| 07:25.04 | Gabriel25 | festival I do have it installed |
| 07:25.10 | hads|home | If you are using A@H/Trixbox/FreePBX then you will get more help in #freepbx |
| 07:25.22 | Gabriel25 | I don`t |
| 07:25.33 | Gabriel25 | i have asterisk installed on fedora 5 |
| 07:26.12 | hads|home | I don't use TTS so I can't really help. |
| 07:26.32 | Gabriel25 | Ohh ok |
| 07:33.21 | delmar | has anyone setup something like the portable-extensions idea on here http://www.voip-info.org/wiki/view/Asterisk+Tips+follow+me ?? |
| 07:36.26 | delmar | my brain is on a go-slow tonight. I can see how that is all working in terms of the activation/deactivation and all that... what I don't get is the first part of the portable-extensions context. |
| 07:37.36 | benjk | you can always get a good night's rest and look at it again tomorrow ;) |
| 07:37.57 | delmar | 7:30pm here |
| 07:38.04 | delmar | so im a little ways off from sleep. |
| 07:38.44 | benjk | well, you said "my brain is on a go-slow tonight", so I figured it may be a good idea to go to sleep early and get up early |
| 07:39.07 | delmar | but yeah.... im a little tired. got 3hrs sleep, up from 4:30, then got a couple hours nap this avo, but im still a bit buggered. |
| 07:39.13 | benjk | I do that sometimes when I get stuck |
| 07:39.50 | delmar | for sure. i was lookin at something last night and decided I would look at it with a fresh brain in the morning.. not that i was too fresh after 3hrs sleep this morning but.. i spotted the problem instantly after that. |
| 07:40.06 | benjk | heh |
| 07:41.16 | benjk | I don't know what portable extensions means, but as for follow me, I use astdb |
| 07:41.21 | delmar | someone wanna take a look at http://www.voip-info.org/wiki/view/Asterisk+Tips+follow+me .. at the portable-extensions context and run through a few things with me? I have some questions about how this is implimented really. |
| 07:41.56 | delmar | ok. well thats all I want really.. some way to run follow-me ... |
| 07:41.57 | *** join/#asterisk af_ (n=af@ip-173-144.sn1.eutelia.it) |
| 07:42.13 | delmar | benjk, so maybe your way is another option I can look at... |
| 07:42.28 | benjk | it involves a bit of macro programming though |
| 07:42.50 | delmar | benjk, the example there on that link... is that not using the asteriskDB ? |
| 07:42.51 | benjk | but basically what I do is, I only have a single extension for everybody |
| 07:43.30 | benjk | I define my local extension range in a global variable LER = 30XX for example |
| 07:43.39 | benjk | then I have a local context |
| 07:43.58 | benjk | exten => _${LER},1,.... |
| 07:44.14 | benjk | which calls a custom macro |
| 07:44.27 | benjk | each extension has a dictionary in astdb |
| 07:44.46 | benjk | one of which is an unconditional call forward |
| 07:45.04 | benjk | and another of which is a follow me forward |
| 07:45.28 | delmar | ok |
| 07:46.50 | benjk | CLI> database show ext2001 |
| 07:47.15 | benjk | CLI> database show ext2001 |
| 07:47.15 | benjk | <PROTECTED> |
| 07:47.15 | benjk | <PROTECTED> |
| 07:47.15 | benjk | <PROTECTED> |
| 07:47.26 | benjk | and a bunch of others |
| 07:47.29 | delmar | ok |
| 07:47.35 | benjk | everything is controlled by the entries in astdb |
| 07:47.50 | delmar | right. |
| 07:47.54 | benjk | for each active extension there is such a dictionary |
| 07:48.04 | benjk | ext2001, ext2002, ext2003 etc etc etc |
| 07:48.14 | delmar | yeah ok that part I get |
| 07:48.21 | delmar | it's really the dialplan side I need to get going. |
| 07:48.24 | benjk | then the custom marco looks at those and decides what to do |
| 07:49.04 | benjk | if it sees there is a number in /extXXXX/followme, then it will dial that after a dial timeout with dialstatus NOANSWER |
| 07:49.54 | benjk | basically I call Dial() always with the g flag for "continue in the dialplan" |
| 07:49.58 | delmar | ah thats something i had not really thought about.. followme is really.. something that happens after the other phone rang for a bit... |
| 07:50.21 | benjk | then I look at the value of DIALSTATUS and depending on that I do whatever the next action should be |
| 07:50.26 | delmar | what I wanna do is have it so an extension can be remotely set for call-forward |
| 07:50.37 | benjk | same thing |
| 07:50.55 | benjk | the /extXXXX/callforward entry in the database controls that |
| 07:50.58 | delmar | not if follow me .. does it's thing after ringing the extension for X seconds before forwarding |
| 07:51.10 | benjk | if the macro sees a number there, it will dial that straight away |
| 07:51.12 | Assid | err,, if im using manager api.. and i want to make a call using a macro .. how would i do that? |
| 07:51.26 | Assid | its what should i use in the channel |
| 07:51.41 | benjk | the thing is you have to work out the order in which to process things |
| 07:51.56 | *** join/#asterisk BrainSurg (n=paul@d141-204-36.home.cgocable.net) |
| 07:52.00 | benjk | a DND setting would be the first thing to look at |
| 07:52.09 | BrainSurg | greetings |
| 07:52.10 | benjk | if DND is set, you go straight to voicemail |
| 07:52.13 | BrainSurg | Salvete omnes |
| 07:52.13 | delmar | benjk, sounds interesting. do you have your macro and stuff posted anywhere? I might like to see if I can make that work. |
| 07:52.20 | benjk | then next is unconditional call forward |
| 07:52.32 | benjk | if that is set you dial the forwarding number straight away |
| 07:52.49 | benjk | the next is the actual peer where the extension can be reached |
| 07:53.04 | benjk | then if this times out, you examine the follow me entry |
| 07:53.22 | benjk | its 3500 lines of code :D |
| 07:53.54 | delmar | benjk, all I wanna really do right now is thus... if i go upstairs and forget to set fowarding on the Polycom downstairs... I wanna pickup the phone upstairs and tell asterisk to send calls for the other extension .. to the phone upstairs.. without having to run down and do it :P |
| 07:54.01 | delmar | benjk, as an example .. :P |
| 07:54.05 | benjk | its a universal dialplan that takes a whole bunch of things into account, different countries, loads of user control |
| 07:54.50 | delmar | benjk, still sounds pretty cool. you should rip out any sensitive bits and post it up sometime !! |
| 07:54.52 | benjk | an example is not as easy as you make it sound because all my macros call each other |
| 07:55.04 | BrainSurg | Quick question for all: Can one suggest a good book for learning to configure asterisk? |
| 07:55.07 | hads|home | delmar: Simple way; you could just setup an extension which sets a value in the db which sets up which phone to call. |
| 07:55.13 | hads|home | ~thebook |
| 07:55.20 | jbot | rumour has it, thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 07:55.20 | sevard | the book! |
| 07:55.25 | benjk | for what you want to do its really not that complicated |
| 07:55.26 | BrainSurg | oooh |
| 07:55.40 | benjk | set up a structure in the database for each extension you have |
| 07:55.51 | delmar | BrainSurg, also... www.voip-info.org is a really good place to get info |
| 07:56.03 | benjk | like database put extXXXX callforward nnnnnnnnn |
| 07:56.21 | hads|home | Although take everything on the wiki with a grain of salt - some of it is out of date or just wrong. |
| 07:56.21 | benjk | database put extXXXX followme nnnnnnnnn |
| 07:56.21 | BrainSurg | Thanks, very helpful. |
| 07:56.24 | benjk | etc |
| 07:56.27 | delmar | hads|home I think this is what this thing on http://www.voip-info.org/wiki/view/Asterisk+Tips+follow+me is doing? |
| 07:56.56 | benjk | then write a little macro [macro-dial-extension] |
| 07:57.12 | benjk | that macro should first check if there is a value in callforward |
| 07:57.19 | benjk | if there is, you dial that number instead |
| 07:57.24 | benjk | if not you continue |
| 07:57.41 | benjk | then it should check where the extension lives |
| 07:57.41 | delmar | I think .. what is on here.. at http://www.voip-info.org/wiki/view/Asterisk+Tips+follow+me ... will do it... and I understand whats in that example quite well... accept the part at the top... im like.. WTF is it going to do here... |
| 07:58.00 | hads|home | delmar: Not really what I'm talking about there. |
| 07:58.01 | benjk | <PROTECTED> |
| 07:58.13 | benjk | then you dial that route |
| 07:58.34 | benjk | then after the call times out you check followme and if there is a value you dial that |
| 07:58.40 | benjk | that's the entire flow |
| 07:58.53 | benjk | you can change the database from anywhere |
| 07:59.11 | benjk | you can set up a short dial extension for letting users change their settings by phone |
| 07:59.23 | benjk | you can log in via ssh and change settings on the CLI |
| 07:59.29 | hads|home | For your simple case you could just add a db key which has the extension you want to call (you could use a global variable too) and then on your incoming extension just to a Dial(${VALUE_FROM_DB}) |
| 07:59.30 | benjk | and you can change via AMI |
| 08:00.22 | benjk | so no matter what situation you're in and what location you are, you can change the forwarding and follow me numbers for any given extension |
| 08:01.24 | delmar | ok |
| 08:01.29 | delmar | cool. I think i have a few ideas now. |
| 08:01.48 | benjk | this scheme is most flexible and you can extend it over time to add more parameters |
| 08:01.52 | benjk | like DND |
| 08:02.31 | benjk | dial timeout when internal, dial timeout when forwarding, mailbox number etc etc |
| 08:02.45 | benjk | I even have a VIP caller list for each extension |
| 08:03.01 | delmar | sounds pretty cool |
| 08:03.10 | benjk | so users can enter their important customers/friends/etc and treat incoming calls from them differently |
| 08:03.17 | hads|home | benjk and I are talking about pretty much the same thing, his way is just more expandable. |
| 08:03.51 | benjk | this is just an outcome of adding more stuff over time |
| 08:04.12 | benjk | once you draw parameters from the database, you may as well draw everything from there |
| 08:04.31 | *** join/#asterisk remiss (i=bofh@191.80-203-38.nextgentel.com) |
| 08:04.32 | delmar | yeah. sounds pretty powerful |
| 08:04.44 | benjk | and you don't have to do this all in one go |
| 08:04.51 | benjk | you just start with forwarding |
| 08:04.58 | benjk | later on, you add DND |
| 08:05.06 | benjk | then follow me, etc etc |
| 08:05.24 | benjk | essential is that each extension has its own dictionary |
| 08:05.31 | benjk | or family as asterisk docs call it |
| 08:07.30 | delmar | ok |
| 08:07.31 | benjk | its also helpful to be aware of the different types of forwards |
| 08:07.48 | benjk | unconditional forward, forward on busy, forward on no answer etc |
| 08:08.00 | benjk | follow me is basically a fwd-on-no-answer |
| 08:08.07 | delmar | yeah |
| 08:08.40 | delmar | what i wanna do is setup unconditional forward, which is able to be activated for a given extension.. from any other extension. |
| 08:08.45 | delmar | for now at least |
| 08:09.32 | benjk | you should distinguish between the storing and retrieval of the foward parameter itself |
| 08:09.39 | benjk | and the ways to change it |
| 08:09.49 | benjk | those should be implemented independently |
| 08:11.10 | benjk | the macro that dials extensions should only be interested in that database field, it should not have to worry about how the number got into the database |
| 08:12.12 | benjk | likewise, your IVR or whatever instrument you use to change the forwarding number should only have to worry about putting the value into the database, it need not know about the macro that will read from it |
| 08:15.55 | *** join/#asterisk darkgamer20 (n=chatzill@adsl-71-146-141-161.dsl.pltn13.sbcglobal.net) |
| 08:18.16 | *** join/#asterisk Assid (i=assid@203.115.83.215) |
| 08:18.19 | Assid | heya |
| 08:18.36 | Assid | err.. how do i get the caller id of an incoming call using manager |
| 08:18.42 | delmar | benjk, ok thanks for the tips dude. |
| 08:18.47 | benjk | welcome |
| 08:19.14 | Assid | hey benjk: whats your experience with manager api |
| 08:19.19 | darkgamer20 | is there a way to direct a call from an external number to another external number if an external number calls my asterisk server? |
| 08:19.43 | *** join/#asterisk Arno[Slack] (i=100@master.infinityperl.org) |
| 08:21.27 | hads|home | darkgamer20: Asterisk doesn't really care if the number is internal or external. |
| 08:22.58 | darkgamer20 | hads|home: hmm oh ok, so lets say I use Dial(ZAP/r1/REDIRECTION NUMBER) to direct the call to that number would i be charged for making the call? |
| 08:23.19 | hads|home | Yes |
| 08:23.59 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 08:24.20 | darkgamer20 | is there a way to not get charged and redirect the call? |
| 08:25.35 | delmar | unless the number u are directing to .. costs you nothing when you dial it normally... No |
| 08:25.52 | benjk | there is if you have SS7 |
| 08:25.57 | benjk | and a class 5 switch |
| 08:26.08 | hads|home | Well yeah |
| 08:26.14 | benjk | but you'd not be asking if you had that ;) |
| 08:26.18 | delmar | i thought of that also.. but wasnt gonna bother going into that :P |
| 08:26.21 | *** join/#asterisk svenna (n=svenna@p548D124D.dip0.t-ipconnect.de) |
| 08:26.26 | darkgamer20 | haha yea |
| 08:26.45 | adelas | do you guys know of any sip voip provider thats cheap and good? |
| 08:26.53 | benjk | in some countries BRI also has caller pays call deflection |
| 08:26.53 | adelas | like viatalk or junctionnetworks? |
| 08:27.12 | darkgamer20 | what about in the US? |
| 08:27.45 | benjk | the US doesn't really like BRI so even in those places where you get it, they probably don't bother to implement fancy features |
| 08:28.40 | darkgamer20 | oh |
| 08:30.31 | benjk | Assid, I don't really make much use of the manager interface, I think it ought to be replaced by something decent |
| 08:44.02 | remiss | anyone else used the execif-app here? i can't seem to get it to work... |
| 08:44.03 | remiss | <PROTECTED> |
| 08:44.03 | remiss | Aug 20 10:42:30 WARNING[6146]: app_while.c:110 execif_exec: Count not find application! ( SayDigits) |
| 08:44.36 | remiss | e.g. ExecIf( some test, SayDigits, 123) |
| 08:46.58 | *** join/#asterisk pcm (n=pcm@68.159.139.234) |
| 08:47.43 | pcm | anyone has cheap server hosting for asterisk ? |
| 08:51.36 | *** join/#asterisk moon06 (n=michael@cim06-1-82-228-240-97.fbx.proxad.net) |
| 08:57.14 | hads|home | remiss: From your WARNING message it looks like ExecIf might not like the spaces in your arguments, try taking them out. |
| 08:57.43 | benjk | that's because those kids don't know how to parse arguments |
| 08:57.59 | benjk | all they do is strchr, strrchr and strsep |
| 08:58.24 | benjk | I have replaced loads of those |
| 08:58.54 | remiss | hads|home: thanks, mate :) |
| 08:59.04 | hads|home | np |
| 08:59.13 | remiss | *put it in the wiki* |
| 09:02.47 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 09:08.40 | *** join/#asterisk muppetmaster (n=jasongoe@169.red-81-184-73.user.auna.net) |
| 09:08.50 | muppetmaster | Hello |
| 09:09.03 | muppetmaster | With the new AJAM interface, it is easy to do a request/response, get status, etc. |
| 09:09.31 | muppetmaster | But is there a way to do a callback that would allow one to monitor all events and build state models for various devices. Or does one still need to go to the traditional Manager API port for this? |
| 09:10.19 | *** join/#asterisk Jenocin (i=jenocin@99.3.118.70.cfl.res.rr.com) |
| 09:10.30 | Jenocin | hey people, anyone using voicestick ? |
| 09:11.29 | muppetmaster | also, according to the readme, show http is meant to give you a list of all functions, but that command does not exist |
| 09:11.33 | muppetmaster | Only http show status seems to |
| 09:12.50 | *** join/#asterisk abatista (n=Ariel@dsl-20-177.cofs.net) |
| 09:15.39 | remiss | is there a better wiki than the voip-info stuff? |
| 09:15.57 | hads|home | Not really |
| 09:21.37 | tzafrir | remiss, better technically or more "informed"? |
| 09:22.28 | benjk | you should be glad you've got that wiki |
| 09:22.47 | benjk | when we started out, there was no documentation, none whatsoever |
| 09:22.59 | tzafrir | remiss, pastebin the relevant parts of your dialplan |
| 09:23.01 | benjk | other than the source code |
| 09:23.32 | hads|home | benjk: You're so hard done by ;) |
| 09:23.39 | benjk | heh |
| 09:23.54 | tzafrir | Yeah. And we had to write the binaries by hand |
| 09:23.56 | benjk | I am just saying, people don't know how good they have it |
| 09:24.24 | hads|home | I know, just giving you a hard time :) |
| 09:25.13 | benjk | no, we had to custom order binaries from a distant voodoo master by sending in rat tails and monkey skulls |
| 09:25.23 | remiss | tzafrir: i've figured things out now.. it's just that voip-info is slow and not that good.. |
| 09:25.39 | remiss | but it's better than nothing :) |
| 09:26.22 | hads|home | The fact that Asterisk has changed and the wiki hasn't kept up can make it a little difficult though. |
| 09:27.19 | benjk | well, you can always take the headers literally "Asterisk, a telephony toolkit" |
| 09:27.37 | benjk | then assemble what you need from the available parts |
| 09:27.44 | benjk | and leave those parts you don't need |
| 09:28.18 | benjk | not everybody buys the latest model of the car they drive just because there is a new model number |
| 09:28.36 | benjk | usually you drive it for a number of years |
| 09:28.59 | benjk | most people would be better off if they took a lesson from that for their use of software |
| 09:33.27 | muppetmaster | No one is informed on AJAM? |
| 09:33.50 | delmar | benjk, i agree |
| 09:34.42 | delmar | benjk, 340days up time on my colocation box. i used to be always wanting the latest kernel.. or latest software... but the thing just runs non-stop. i don't wanna touch it apart from a few minor software updates.. patches.. security issues.. the usual |
| 09:35.01 | benjk | yep |
| 09:35.06 | delmar | benjk, I forgot to add forced module unloading tho.. so its gonna get a kernel compile soon.. but yeah |
| 09:35.07 | benjk | way to go |
| 09:35.39 | gordonjcp | benjk: and some of us deliberately choose to drive 15-year-old cars because they are technically superior to anything new on the market ;-) |
| 09:35.59 | gordonjcp | (which is why I like Citroens and VMS) |
| 09:36.34 | benjk | heh |
| 09:36.57 | benjk | You gotta love the DS |
| 09:37.07 | gordonjcp | I would indeed love a DS, but I can't afford one |
| 09:37.13 | benjk | heh |
| 09:37.26 | gordonjcp | not really what you'd want to use as a daily driver anyway |
| 09:37.44 | benjk | a friend of mine in Europe used to be a driving salesman, he had a phobia of flying |
| 09:37.49 | gordonjcp | I have a CX and an XM as my daily drivers |
| 09:37.54 | benjk | so he drove the entire continent |
| 09:38.02 | gordonjcp | I can't be arsed flying, driving is much more fun |
| 09:38.23 | benjk | according to him, you can drive from Denmark to Munich in one go in a Merc |
| 09:38.37 | gordonjcp | I can believe it |
| 09:38.37 | benjk | but to Athens you can only do it in a Citroen |
| 09:38.40 | gordonjcp | lol |
| 09:38.48 | gordonjcp | they are very very comfortable |
| 09:39.11 | gordonjcp | benjk: my XM is as comfortable as a Bentley but can outhandle a Subaru Impreza |
| 09:39.12 | benjk | well, you probably know the difference between German and French car making |
| 09:39.23 | gordonjcp | benjk: yes |
| 09:39.31 | delmar | gotta love the speed limits in parts of EU.. or lack of. |
| 09:39.39 | benjk | the Germans would build the most advanced car with the latest tech and when they are done they realise ... |
| 09:39.41 | delmar | arent there parts of canada that are limitless too? |
| 09:39.42 | benjk | Oh shit!!! |
| 09:39.48 | gordonjcp | German cars, they need to fasten two parts so they make one extra thick and drill a blind hole and tap it |
| 09:39.49 | benjk | four people ought to go in there |
| 09:39.51 | benjk | dang |
| 09:39.56 | benjk | what are we going to do now? |
| 09:40.04 | benjk | so they put four chairs in |
| 09:40.21 | gordonjcp | French cars, they have a sort of spring washer, a stud with a hole up the middle, a kind of locking screw with a bristol spline head, and a thing a bit like a rawlplug |
| 09:40.35 | benjk | the French on the other hand will take 4 of the most comfortable living room arm chairs and build the car around it |
| 09:41.01 | gordonjcp | it's documented over four pages in the manual, and when you get to it, it's actually rusted into a blob because they didn't bother to cadmium-plate the steel bits, and made part of it out of aluminium |
| 09:41.15 | gordonjcp | benjk: haha |
| 09:41.35 | gordonjcp | benjk: I've actually fallen asleep in the CX, just parked up outside the house |
| 09:41.42 | benjk | heh |
| 09:41.48 | gordonjcp | sat for a moment to listen to the end of the news or something, and nodded off |
| 09:41.56 | benjk | when I lived in France I had a GSA |
| 09:41.59 | gordonjcp | my gf can fall asleep in the car driving across town |
| 09:42.04 | gordonjcp | GSAs rock! |
| 09:42.08 | benjk | the predecessor of the BX |
| 09:42.44 | gordonjcp | good for embarrassing ricers when the lights change... |
| 09:42.44 | benjk | that was 10 years old second or third hand, but it was marvellous |
| 09:43.08 | benjk | one of the best cars I had was a Renault 16TX |
| 09:43.31 | benjk | and I also had just about every BMW (3, 5, 7) and Merc |
| 09:44.04 | gordonjcp | http://www.gjcp.net/citroen_gsa.jpg |
| 09:44.08 | gordonjcp | ^ my old GSA |
| 09:44.18 | gordonjcp | I had a Merc 230TE, that was good |
| 09:44.31 | benjk | a French car may have a problem with things like wipers not working when the radio is on and it rains |
| 09:44.39 | benjk | or you may have a door lock rust off |
| 09:45.10 | gordonjcp | benjk: yes, my XM does occasionally complain about the ABS being out of use (in French) when it's low on petrol and raining |
| 09:45.19 | gordonjcp | the ABS is perfectly ok, it's just grumbling |
| 09:45.35 | benjk | but the stuff that you absolutely need to keep moving (and which is more expensive to repair or replace) will be rock solid |
| 09:45.41 | gordonjcp | yup |
| 09:45.49 | gordonjcp | the plumbing scares people, but it's easy to do |
| 09:45.57 | benjk | a German car will never have a door handle rust off |
| 09:46.13 | gordonjcp | I've got to replace a couple of bits of the hydraulic pipe in the CX |
| 09:46.17 | benjk | and it will never show any weird electrical behaviour with radio, wipers and stuff |
| 09:46.30 | gordonjcp | benjk: yeah but just you try and buy a set of brake discs ;-) |
| 09:46.51 | benjk | but if you have a tiny little problem with the cooling and you don't immediately go to the next garage, you will look at a hefty bill for a new engine |
| 09:47.02 | benjk | because it;ll die in an instant |
| 09:47.20 | benjk | a French car will go 20.000 kms with no oil |
| 09:47.20 | gordonjcp | benjk: BMW E30 cracked rockers, perchance? |
| 09:47.57 | benjk | I had a BMW 525 which lost a tiny little bit of cooling water |
| 09:48.08 | benjk | I noticed it barely |
| 09:48.25 | benjk | I thought I could drive the 30 kms back home and bring it to the garage there |
| 09:48.35 | benjk | engine exitus after 5kms |
| 09:49.08 | benjk | I also had a Renault 4 which I drove for 20.000 without any oil |
| 09:49.14 | benjk | I didn't know it then |
| 09:49.39 | benjk | but the gear shift was so heavy that I got back pain from the exercise |
| 09:49.53 | gordonjcp | heh |
| 09:49.53 | benjk | and after the winter season, I was to change the oil |
| 09:49.58 | benjk | what oil? |
| 09:50.03 | benjk | there was no oil in there |
| 09:50.08 | gordonjcp | benjk: my XM dropped its coolant one dark and wet morning |
| 09:50.14 | benjk | ok, so lets put some oil in |
| 09:50.26 | benjk | and afterwards the gear shift was soft like butter |
| 09:50.30 | benjk | oh well |
| 09:50.41 | *** join/#asterisk apardo (n=apardo@87.217.146.232) |
| 09:50.42 | gordonjcp | I thought the cloudy stuff I could see in my mirror was just spray, until after about two miles the heater wasn't working |
| 09:50.46 | benjk | a German car would have never ever forgiven that |
| 09:50.54 | gordonjcp | and then after another mile, a big red <STOP> light came on |
| 09:51.03 | gordonjcp | split the top radiator hose |
| 09:51.14 | gordonjcp | seems to have survived, and that's the rather fragile V6-24 engine |
| 09:51.53 | benjk | yeah, the French have to build their cars that way because there are regions in France where there is no garage for 50 kms |
| 09:52.11 | benjk | in Germany you have a garage every 5 kms or so, no matter how far in the country side you are |
| 09:52.46 | *** join/#asterisk DarKnesS_WolF (n=wolf@212.103.170.135) |
| 09:52.59 | benjk | also, the French tend to design their stuff so that a blacksmith in some remote village can fix it |
| 09:54.53 | benjk | if your door handle rusts off, what's the problem? have it wielded back on by your local blacksmith, don't make a fuzz |
| 09:55.01 | gordonjcp | tie it on with string |
| 09:55.09 | benjk | or duct tape |
| 09:55.19 | benjk | but if your engine dies, that's it |
| 09:55.41 | benjk | turns the entire car into a pile of junk |
| 09:55.49 | coppice | string. that's luxury. when I was lad we would have had to hold all the pieces of the car together with our bare hands |
| 09:55.58 | benjk | :D |
| 09:56.24 | benjk | that's British cars for you, not even compatible with string or duct tape |
| 09:57.11 | coppice | people are strange about the concept of "serious problem" with cars. to many people a lot of total breakdowns don't count as serious, because the cause is something trivial breaking |
| 09:57.42 | benjk | indeed |
| 09:57.43 | Un1x | anyone do AGI scripting for money :D? |
| 09:58.29 | benjk | sure, there are plenty of folks who do that |
| 09:58.43 | benjk | I think there's even a channel for that |
| 09:58.54 | benjk | asterisk-biz or so, not entirely sure though |
| 09:59.26 | Un1x | no i think it's the mailiing list |
| 09:59.34 | Un1x | not sure if there is a channle would like to know if there is tho :D |
| 09:59.38 | remiss | anyone know how to get festival to go into the background instead of hanging around? |
| 10:00.12 | benjk | I usually write my stuff as a macro first and if it gets out of hand, I turn it into an app |
| 10:00.20 | benjk | so I bypass the whole AGI thing |
| 10:01.44 | Un1x | hmm |
| 10:01.50 | Un1x | well i need this app to call people from a list |
| 10:01.57 | Un1x | so im not entirely sure what to use |
| 10:02.07 | Un1x | weather to make it .c or agi or phpi dont know :S |
| 10:03.42 | benjk | not sure what you mean "call people from a list" |
| 10:03.51 | benjk | do you mean a predictive dialer? |
| 10:04.12 | benjk | like telemarketers use? |
| 10:04.28 | benjk | if so, there is vicidial for asterisk |
| 10:05.45 | benjk | otherwise, you may want to explain in more detail |
| 10:08.02 | gordonjcp | benjk: you need to get Lucas string, the ends are a slightly different size |
| 10:08.05 | Un1x | benjk i just need to call people from a list play a certain recording and wait for input via DAILPAD and have that input stored into a txt file |
| 10:10.14 | benjk | I see |
| 10:10.59 | benjk | could be done either way |
| 10:11.23 | benjk | of course it'll be least clutter if it is an app |
| 10:12.11 | benjk | how fast do you need this? |
| 10:13.15 | SwK | gordoncjp have that already contact us at the office on monday |
| 10:13.20 | SwK | asteriasgi.com |
| 10:13.23 | SwK | or email sales@ |
| 10:13.38 | SwK | errrr unix |
| 10:13.41 | benjk | for a piece of Lucas string? |
| 10:13.43 | benjk | :D |
| 10:16.50 | benjk | gordonjcp, are you running * on VMS? |
| 10:19.31 | gordonjcp | sadly not |
| 10:19.42 | benjk | why not then? |
| 10:19.44 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 10:19.53 | gordonjcp | heh |
| 10:20.03 | gordonjcp | I wonder if it could be ported... |
| 10:20.27 | gordonjcp | I'm actually running it on NetBSD-i386 |
| 10:20.27 | gordonjcp | teh boring |
| 10:20.27 | benjk | I would just give it a try |
| 10:21.06 | benjk | I sold my last AXP and also my last VAX a few years back, couldn't really try |
| 10:21.21 | gordonjcp | I've got a microvax II and a microvax 3300 |
| 10:22.00 | benjk | if you have the POSIX environment ("Unix services for VMS" IIRC) it should build out of the box |
| 10:22.15 | benjk | without Zaptel of course |
| 10:22.56 | benjk | would be interesting to do IAX over LAT |
| 10:23.16 | benjk | and then do a cluster failover in the middle of a call |
| 10:25.04 | benjk | the 3xxx have DSSI cluster capability, you'd need to get one more though ;) |
| 10:31.33 | gordonjcp | benjk: I can't get one more |
| 10:31.43 | gordonjcp | I need to get rid of some stuff first... |
| 10:31.53 | benjk | heh, sounds familar |
| 10:32.13 | benjk | like I said, I sold my last DEC stuff a few years back |
| 10:32.16 | gordonjcp | sell at least one car and find a new home for the MVII, I think is the current deal |
| 10:32.24 | benjk | including a complete set of printed documentation |
| 10:32.54 | gordonjcp | ! |
| 10:33.09 | benjk | to the guy who bought it I said he'll need to bring a van for the documentation |
| 10:33.14 | gordonjcp | heh |
| 10:33.17 | benjk | he thought I was joking |
| 10:33.30 | benjk | should have seen his face when I showed him the boxes |
| 10:33.36 | gordonjcp | I fitted a fair chunk of greywall, the MVII, the MV3300 and some other bits into the back of my CX |
| 10:33.49 | gordonjcp | the MVII is in a rack about 40" tall |
| 10:33.57 | benjk | yeah, I know |
| 10:33.57 | gordonjcp | *just* fitted across the back seat |
| 10:34.24 | benjk | the documentation fills a 2m by 3m booksheld |
| 10:34.30 | benjk | bookshelf |
| 10:34.51 | benjk | in those nice old grey DEC folders |
| 10:35.03 | gordonjcp | yup |
| 10:35.17 | gordonjcp | some of mine have the horizontal fold that acts as a stand... |
| 10:35.48 | gordonjcp | mine has two Fujitsu Eagle drives, the MVII in a BA23, and a big tape drive |
| 10:36.18 | benjk | yes those are the folders I was talking about |
| 10:38.35 | benjk | everything was solid back in those days |
| 10:38.41 | benjk | even the folders |
| 10:43.38 | Jenocin | hey people, anyone using voicestick ? |
| 11:01.30 | *** join/#asterisk tsurk0 (n=tsurko@85.187.160.157) |
| 11:16.02 | *** join/#asterisk RoyK (n=roy@ti211210a080-2688.bb.online.no) |
| 11:32.06 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 11:47.30 | *** join/#asterisk RoyK (n=roy@ti211210a080-2688.bb.online.no) |
| 11:52.13 | remiss | anyone here gotton sphinx to work? |
| 11:54.34 | *** join/#asterisk rogier (n=rogier@83-67-dsl.ipact.nl) |
| 12:08.40 | *** join/#asterisk DarKnesS_WolF (n=wolf@212.103.170.135) |
| 12:16.35 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 12:16.38 | *** join/#asterisk ToTo (n=ToTo@host212-207.pool872.interbusiness.it) |
| 12:33.59 | RoyK | <PROTECTED> |
| 12:35.34 | xheliox | <PROTECTED> |
| 12:47.05 | *** join/#asterisk silentz (n=silentz@202.69.106.2) |
| 12:49.03 | *** join/#asterisk oej (n=oej@63.116.149.163) |
| 13:04.43 | *** join/#asterisk Shoko (i=btshoko@btshoko.de) |
| 13:06.24 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 13:08.21 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 13:15.17 | RoyK | hi. if trapping all calls from PSTN with an exten => _X.,1,....., is there a way to later reject calls from certain numbers, having the switch trying them on the next PRI? |
| 13:26.11 | xheliox | sd25-rm |
| 13:26.18 | xheliox | erm. |
| 13:26.22 | xheliox | wrong window. |
| 13:26.59 | *** join/#asterisk png6 (n=png@host49-71.etanet.se) |
| 13:27.27 | png6 | if I want to reach a sertain extension when I dial in, and only have one number - is the only way a menu system that the user meets when he first call? |
| 13:28.04 | png6 | or can I dial 012345 and then press the extensionnumber (where 012345 is my phone number) |
| 13:29.23 | png6 | are you with me? |
| 13:46.10 | *** join/#asterisk queuetue (n=scott@toronto-HSE-ppp4122670.sympatico.ca) |
| 13:46.51 | queuetue | Hi. How do I indicate "asterisk" in an asterisk dialplan? It's a hard combo to google for. :) |
| 13:47.38 | *** join/#asterisk somegeek_ (i=levin@tor/regular/somegeek) |
| 13:53.22 | *** join/#asterisk mrec_ (n=revenger@p85.212.42.171.tisdip.tiscali.de) |
| 13:54.32 | rogier | Hello there. What is currently considered the best and most up to date documentation for asterisk ? I checked the asterisk handbook on digium's site, but it says "Last edited 3/3/03" ! |
| 13:56.11 | RoyK | hi. if trapping all calls from PSTN with an exten => _X.,1,....., is there a way to later reject calls from certain numbers, having the switch trying them on the next PRI? will setting a certain hangup cause help? |
| 13:57.55 | remiss | rogier: voip-info.org perhaps.. |
| 13:58.19 | rogier | remiss, ok, will check there |
| 14:01.06 | *** join/#asterisk dongs (n=HPUX@h193107.ppp.asahi-net.or.jp) |
| 14:01.14 | dongs | hey whats the word on this -> http://www.intel.com/products/desktop/adapters/600sm/index.htm |
| 14:01.36 | *** join/#asterisk oej (n=oej@63.116.149.163) |
| 14:04.07 | *** join/#asterisk |dennis| (n=dennis@200.32.215.82) |
| 14:07.28 | dongs | so? |
| 14:08.30 | ManxPower | NO! |
| 14:08.35 | dongs | no! what |
| 14:08.44 | ManxPower | voip-info.org is FULL of wrong information |
| 14:08.45 | xheliox | Just no, alright? |
| 14:09.04 | ManxPower | check /path/to/src/asterisk/docs also "show applications" in the asterisk CLI, also The Book |
| 14:09.06 | ManxPower | ~docs |
| 14:09.07 | jbot | i guess docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 14:09.10 | ManxPower | ~book |
| 14:09.11 | jbot | i heard book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 14:09.23 | ManxPower | use voip-info.org is your last source |
| 14:09.57 | xheliox | voip-info.org is a good source of examples, but the person presenting the examples may or may not know what the f* they're doing. |
| 14:10.41 | ManxPower | Many of the examples are wrong. |
| 14:10.49 | xheliox | For example, I have entries over there, and if people are following my suggestions, there's no telling how screwed they'll end up. ;) |
| 14:10.59 | ManxPower | much of the docs apply to 1.0.x, but not to 1.2.x, but there is no indication of that |
| 14:11.45 | ManxPower | much of the info on the asterisk apps is out of date, "show applications" and "show application X" is better (in the CLI |
| 14:12.04 | xheliox | show application X is excellent. |
| 14:16.14 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:19.19 | mrec_ | -- Executing Playback("ALSA/plug:au600capture", "invalid") in new stack |
| 14:19.24 | mrec_ | isn't this a bit misleading? |
| 14:19.44 | mrec_ | output is set to au600playback in the alsa.conf file |
| 14:19.56 | mrec_ | but it tries to access the input device which makes no sense |
| 14:20.09 | mrec_ | not in that "Executing Playback" context actually |
| 14:20.39 | rogier | ManxPower, ok thanks for that notice ! |
| 14:22.04 | rogier | How do you rate the O'Reilly pdf hosted at asteriskdocs.org ? It's from august 2005 and says it covers asterisk 1.2 , which I currently have installed. |
| 14:22.06 | dongs | huh |
| 14:22.21 | dongs | hey whats the word on this -> http://www.intel.com/products/desktop/adapters/600sm/index.htm |
| 14:23.24 | mrec_ | dongs: are you looking for something like that: http://www.packetizer.com/products/au600/ ? |
| 14:23.28 | rogier | dongs, I'm just starting out on this asterisk adventure. I have no idea how good that is ..... |
| 14:24.09 | dongs | no, i found that particular card on intel's site and im wondering whats the big deal |
| 14:24.12 | *** join/#asterisk tmccrary (n=tmccrary@d14-69-160-83.try.wideopenwest.com) |
| 14:24.30 | tmccrary | Is the TE410P pretty reliable for use as a general data T1 interface card? |
| 14:24.44 | tmccrary | w/hdlc |
| 14:24.45 | dongs | yes. |
| 14:34.18 | *** join/#asterisk ehtesham (n=md_ehtes@dialpool-210-214-246-176.maa.sify.net) |
| 14:35.22 | *** join/#asterisk joe (n=nnjsauer@66.107.33.195) |
| 14:36.36 | RoyK | hi. if trapping all calls from PSTN with an exten => _X.,1,....., is there a way to later reject calls from certain numbers, having the switch trying them on the next PRI? will setting a certain hangup cause help? |
| 14:36.49 | mrec_ | bah asterisk's alsa implementation sucks |
| 14:37.34 | RoyK | mrec_: really? I thought the whole of asterisk was perfect?? |
| 14:38.28 | mrec_ | hehe |
| 14:39.06 | mrec_ | alsa itself is already a big bad documented beast .. |
| 14:39.15 | qai | Question for those dialplan guru's - my VOIP service provider allows adapter connected users and asterisk server connected users. He has a feature for the adapter connected users that allows them to initiate 3-way calling by dialing the first number, pressing hook flash, then appending *23 to the beginning of the second #, then via another hook flash, all three parties are connected. Anyone have an idea what the *23 dialplan would look like? |
| 14:46.45 | *** join/#asterisk I-MOD (n=opticron@gateway.digium.com) |
| 14:47.53 | *** part/#asterisk ehtesham (n=md_ehtes@dialpool-210-214-246-176.maa.sify.net) |
| 14:58.05 | *** join/#asterisk Rez (i=lorez@freenode/staff/lorez) |
| 14:59.24 | *** part/#asterisk dongs (n=HPUX@h193107.ppp.asahi-net.or.jp) |
| 15:00.27 | *** join/#asterisk daniel_bergamini (n=daniel_b@70-41-166-149.cust.wildblue.net) |
| 15:18.21 | *** join/#asterisk Dibbler_ (n=Dibbler@dsl-217-155-254-174.zen.co.uk) |
| 15:33.02 | mrec_ | is matthew fredrickson still around? |
| 15:33.26 | mrec_ | wonder what he thinks an output/input device should be |
| 15:35.14 | yxa | trying to do variable multiplication: syntax error: syntax error, unexpected $end, expecting '-' or '!' or '(' or '<token>'; Input: |
| 15:41.33 | Corydon76-home | yxa: you passed it an empty string |
| 15:45.25 | mrec_ | is there anything special about the alsa implementation? I really really wonder why I don't get a single tone out of it using alsa |
| 15:45.28 | *** join/#asterisk SplasPood (n=jwb@gate.lga2.us.voxel.net) |
| 15:45.33 | mrec_ | the device itself works fine with aplay and arecord |
| 15:45.48 | mrec_ | there's just one thing aplay needs to run in order to get arecord work with it |
| 15:46.20 | yxa | Corydon-w no, its Set(CALLDURSEC=$[${CALLDURSECS} * 60]); |
| 15:46.58 | *** join/#asterisk coppice (n=chatzill@229.166.17.210.dyn.pacific.net.hk) |
| 15:51.45 | yxa | unless I found a bug |
| 15:56.48 | *** join/#asterisk mmealling (n=michael@c-24-98-171-50.hsd1.ga.comcast.net) |
| 15:57.23 | mrec_ | grr I hate it |
| 16:01.24 | *** join/#asterisk SplasPood (n=jwb@gate.lga2.us.voxel.net) |
| 16:01.35 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) [NETSPLIT VICTIM] |
| 16:06.02 | Corydon76-home | yxa: is CALLDURSECS perhaps ""? |
| 16:06.32 | Corydon76-home | because that isn't a variable that I know of |
| 16:08.45 | *** join/#asterisk jwsh (n=jwsh@ip67-93-24-114.z24-93-67.customer.algx.net) |
| 16:09.42 | mrec_ | could it be that this alsa module is a bit weird and broken? |
| 16:10.07 | mrec_ | there's a nosound flag in the source which is set to 0 .. |
| 16:10.12 | mrec_ | actually it should be set to 1 ... |
| 16:11.53 | jwsh | Does SIP not support callerID, or am I doing something wrong? |
| 16:12.36 | jwsh | no matter what I set CALLERID(num) / CALLERID(name) to, I just get "asterisk" on my SIP phone as the callerID |
| 16:12.38 | *** join/#asterisk Assid (i=assid@203.115.83.215) |
| 16:13.27 | jwsh | if I call from another sip phone it shows that phone's callerID, but when I call in from outside it doesn't display it |
| 16:16.17 | Assid | are you getting the caller id from your incoming provider? |
| 16:17.30 | crochat | !seen vaq |
| 16:17.44 | jwsh | actually I'm not, which is another issue alltogether (they didn't set it up apparently). I'm hard coding it in the dialplan |
| 16:18.40 | *** join/#asterisk jbsolutios (n=jbenson@87-194-2-120.bethere.co.uk) |
| 16:18.55 | jwsh | eg: exten => s,6,Set(CALLERID(name) = "Unavailable") |
| 16:20.05 | crochat | jwsh: What's your Asterisk version ? |
| 16:22.23 | *** join/#asterisk BugKham (i=CKGLOB@61.47.106.94) |
| 16:22.49 | jwsh | crochat: asterisk 1.2.7.1 - apparently if I hard code it in zapata.conf it works? |
| 16:22.55 | BugKham | which variable contains the agi exit status? |
| 16:23.01 | jwsh | but if I set it from the dialplan it doesn't? |
| 16:25.26 | crochat | jwsh: Did you try with the variable CALLERIDNUM ? |
| 16:25.34 | crochat | jwsh: As if it was an older Asterisk version... |
| 16:25.37 | jwsh | yup |
| 16:25.42 | jbsolutios | Hi all - could anyone confirm that RTCP support is going to be included in 1.4 please (from bug 2863)? Thanks |
| 16:26.32 | jbsolutios | to stop the problem we are having whereby a Cisco gateway terminates voicemail calls over around 38 seconds because no RTP audio is being received |
| 16:31.33 | jbsolutios | anyone |
| 16:31.36 | jbsolutios | <PROTECTED> |
| 16:32.42 | jwsh | crochat: very weird - it appears to ignore what I set from the dialplan |
| 16:34.11 | jwsh | crochat: but it'll pass along whatever I setup in zapata.conf |
| 16:36.02 | jwsh | oh well, good enough for now |
| 16:37.14 | muppetmaster | Hello all, anyone here familiar with AJAM? |
| 16:37.34 | *** join/#asterisk TheCops (i=nobody@got.securebinary.com) |
| 16:38.06 | TheCops | Someone using IBM Xseries 345 with Digium or sangoma board ? |
| 16:39.57 | *** join/#asterisk joburg (n=voipmagi@vc-196-207-37-206.3g.vodacom.co.za) |
| 16:40.13 | joburg | hi from south africa |
| 16:40.38 | crochat | joburg: Hi from Switzerland ;-) |
| 16:41.08 | muppetmaster | Hi from Barcelona |
| 16:41.27 | crochat | Hi from the Earth |
| 16:41.41 | TheCops | lo |
| 16:41.51 | coppice | hi from !south afria && !switzerland && !barcelona |
| 16:44.08 | *** join/#asterisk adorah (n=Administ@87.68.169.132.cable.012.net.il) |
| 16:46.57 | BugKham | hi, anyone knows which variable contains the agi exit status? |
| 16:47.02 | BugKham | in the dialplan |
| 16:47.07 | blitzrage | check README.variables |
| 16:47.25 | blitzrage | I don't think you can get the exit status codes though of an app |
| 16:51.45 | *** join/#asterisk RoyK (n=roy@gprs-ggsn5-nat.mobil.telenor.no) |
| 16:57.29 | joburg | if the variable exist you should be able to |
| 16:58.59 | TheCops | What kind of PCI Rhino PCI T1 card is needed? |
| 17:00.28 | JunK-Y | AGISTATUS? |
| 17:01.45 | *** join/#asterisk KDan (i=nobody@sleek.sleektech.nl) |
| 17:03.23 | BugKham | blitzrage, joburg : actually, I just wanna pass a variable from agi to the dialplan |
| 17:04.05 | joburg | be back soon |
| 17:04.53 | *** join/#asterisk techie (n=gus@adsl-068-209-242-072.sip.mia.bellsouth.net) |
| 17:08.23 | mrec_ | why has the driver to take care about DTMF detection?! |
| 17:08.30 | *** join/#asterisk Flauto (n=zhao@adsl-75-3-139-218.dsl.chcgil.sbcglobal.net) |
| 17:08.31 | blitzrage | ? |
| 17:08.39 | blitzrage | that wasn't really a question... |
| 17:09.04 | JunK-Y | BugKham: use the chanvar called AGISTATUS |
| 17:09.10 | mrec_ | hmm? |
| 17:09.25 | Flauto | i am installing asterisk on centos, anything i need to prepare the installation |
| 17:09.27 | mrec_ | well I read I have to implement DTMF tone detection by myself into a driver if I want to use it.. |
| 17:09.41 | BugKham | JunK-Y: yeap, I had it done now |
| 17:10.42 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust944.leed.cable.ntl.com) |
| 17:10.45 | techie | anyone know how to meet dependencies related to codec_gsm.so using menuselect? i installed libgsm but it wont compile |
| 17:12.20 | JunK-Y | techie: which lib exactly? |
| 17:13.15 | KDan | which is the recommended php-agi lib to use? I'm trying to use http://eder.us/projects/phpagi/ but the record_file function isn't working for some reason... was wondering if there's another php agi lib that's a bit more documented? |
| 17:13.54 | techie | on debian, libgsm1 |
| 17:14.34 | JunK-Y | techie: on that ubuntu box, this is the correct one, u need to ./configure again, then make menuselect |
| 17:14.38 | JunK-Y | (again) |
| 17:19.07 | *** join/#asterisk Q6 (n=Q9@85.100.193.143) |
| 17:25.08 | techie | JunK: it compiles but it crashes on startup (Broken Pipe) |
| 17:25.12 | techie | any ideas? |
| 17:25.46 | JunK-Y | whats the backtrace saying? |
| 17:25.57 | *** join/#asterisk carrar (i=tim@osburn.com) |
| 17:26.47 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 17:26.55 | *** part/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 17:29.18 | Flauto | any thing i need to know for installing asterisk on centos 4.3 |
| 17:29.37 | carrar | yeah let me know how it goes |
| 17:29.42 | carrar | I run it on 4.2 |
| 17:29.47 | carrar | have issues with 4.3 |
| 17:30.00 | carrar | could have just been me |
| 17:30.49 | *** join/#asterisk oelewapperke (n=oelewapp@85.158.215.1) |
| 17:30.53 | Flauto | what problem did you have |
| 17:31.02 | oelewapperke | anyone seen this before ? : Aug 20 19:29:05 NOTICE[5413] channel.c: Dropping incompatible voice frame on IAX2/kotjeleuven-4 of format g729 since our native format has changed to gsm |
| 17:31.06 | carrar | or maybe it was the unistim drive I was using had issues |
| 17:31.10 | carrar | driver |
| 17:31.25 | carrar | (nortel ip phone driver) |
| 17:31.29 | techie | no core file, just crashes, interesting |
| 17:31.46 | Flauto | oh |
| 17:31.49 | *** join/#asterisk marv (n=ilovekim@c-71-228-189-127.hsd1.al.comcast.net) |
| 17:31.58 | *** join/#asterisk anonymouz666 (n=anonymou@200.218.193.6) |
| 17:32.12 | JunK-Y | techie: start it with -g |
| 17:32.46 | techie | I did |
| 17:32.53 | techie | and with -dddddddd |
| 17:33.06 | techie | [codec_gsm.so]output: fwrite: Broken pipe |
| 17:33.47 | JunK-Y | techie: on debian? |
| 17:34.37 | techie | yes Sir |
| 17:34.41 | joburg | techie : what is the broken pipe complaining about? what mod ? |
| 17:34.52 | techie | codec_gsm.so |
| 17:35.38 | *** join/#asterisk topping (n=topping@207.47.6.201.static.nextweb.net) |
| 17:36.36 | *** part/#asterisk Q6 (n=Q9@85.100.193.143) |
| 17:40.43 | KDan | which is the recommended php-agi lib to use? I'm trying to use http://eder.us/projects/phpagi/ but the record_file function isn't working for some reason... was wondering if there's another php agi lib that's a bit more documented? |
| 17:51.07 | oelewapperke | how do you check what codecs your install supports ? |
| 17:52.10 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 17:54.18 | Assid | show codecs |
| 18:00.01 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 18:00.05 | shmaltz | hi every1 |
| 18:00.25 | *** join/#asterisk clyrrad (n=ddd@CPE001195f553c7-CM0011aea484a4.cpe.net.cable.rogers.com) |
| 18:00.49 | clyrrad | Good Afternoon Folks |
| 18:01.46 | *** join/#asterisk luchshiy (n=anonymou@d212-53-104-193.cust.tele2.ch) |
| 18:01.55 | KDan | why would this agi record() call fail? ::: $agi->record_file("/asterisk/test", "gsm", "0#", 10000, null, true, 1); |
| 18:02.23 | KDan | it just doesn't record anything... test.gsm stays size 0, and the php script never exits |
| 18:04.08 | clyrrad | my guess is your record_file call is not correct |
| 18:04.21 | clyrrad | if the agi hangs and never exists then its getting stuck in there |
| 18:04.45 | KDan | the asterisk console also never says anything about recording stuff - although i do hear the beep |
| 18:04.48 | clyrrad | also ps aux and check if you have multiple copies of the agi running - if so kill them |
| 18:04.59 | KDan | yeah i've killed them |
| 18:05.15 | clyrrad | check the syntax of record_file |
| 18:05.22 | clyrrad | do you need "/asterisk/test" |
| 18:05.28 | clyrrad | or just "asterisk/test" |
| 18:05.29 | KDan | it is correct according to the docs |
| 18:05.34 | clyrrad | it can be something really simple like that |
| 18:05.44 | KDan | (which admittedly are very sparse docs) |
| 18:05.57 | KDan | http://eder.us/projects/phpagi/phpagi/api-docs/phpAGI/AGI.html#record_file |
| 18:05.57 | clyrrad | yea im betting its a mistake in the parameters |
| 18:06.12 | KDan | <PROTECTED> |
| 18:06.22 | clyrrad | not sure if this wil make a difference but remove the spaces between parameters |
| 18:06.28 | muppetmaster | I am trying to use Dial(Jingle/realadd@gmail.com) to an address that does have the Gtalk client installed on Windows, but I keep getting the message 'no jingle capable client to talk' back from the CLI and the call does not go through. |
| 18:06.32 | KDan | nah, in php it makes no difference |
| 18:06.37 | muppetmaster | Any ideas? |
| 18:06.51 | clyrrad | also have you instantiated your agi object? |
| 18:06.56 | KDan | yep |
| 18:07.02 | KDan | i've tested it with say_digits |
| 18:07.03 | clyrrad | used new |
| 18:07.06 | clyrrad | ok |
| 18:07.29 | *** join/#asterisk alpinus (n=alpinus@81.219.54.250) |
| 18:07.38 | *** join/#asterisk razu_ (n=razu@87-119-182-130.tll.elisa.ee) |
| 18:07.47 | clyrrad | take out "gsm" and put "wav" |
| 18:08.28 | KDan | ok |
| 18:09.36 | clyrrad | also you should be in the format like this $AGI->record_file($wavfile, 'wav', '0123456789', 1080000, 1); |
| 18:09.58 | KDan | k now it creates a 44-bytes long wav file but still hangs |
| 18:10.12 | clyrrad | ah so we getting somewhere |
| 18:10.19 | clyrrad | its your parameters that are wrong then |
| 18:10.33 | KDan | well not really, since the wav still is effectively useless |
| 18:10.36 | KDan | and the script still hangs |
| 18:11.01 | clyrrad | like I said your arguments must be wrong becase it never exits the record_file call |
| 18:11.07 | clyrrad | have you changed your format like this $AGI->record_file($wavfile, 'wav', '0123456789', 1080000, 1); |
| 18:12.21 | clyrrad | all exampls that i find are using single quote instead of double - i know PHP dont care - but try and see if thats whats snagging it |
| 18:13.04 | KDan | $agi->record_file('/asterisk/test', 'wav', '0123456789', 1080000, 1); |
| 18:13.07 | KDan | trying with this now |
| 18:13.12 | clyrrad | ok |
| 18:13.22 | clyrrad | here is a 3 line working example i found |
| 18:13.22 | clyrrad | <PROTECTED> |
| 18:13.23 | clyrrad | <PROTECTED> |
| 18:13.23 | clyrrad | <PROTECTED> |
| 18:14.06 | clyrrad | looks like filename should be in double quotes - and the parameters in single quotes |
| 18:14.13 | *** join/#asterisk toerkeium (i=oo@201.216.206.221) |
| 18:14.23 | KDan | hmm, crap, looks like the last one screwed the asterisk ports or something... gotta restart asterisk.. |
| 18:14.24 | clyrrad | unless they are integers then obviously you dont need quotes |
| 18:14.49 | clyrrad | yea - put the file name in double quotes as in the example above |
| 18:15.33 | KDan | created the 44-byte wav again... php stil lhanging |
| 18:15.42 | toerkeium | mornings |
| 18:15.42 | clyrrad | paste your line |
| 18:15.51 | KDan | single/double quotes in php are only for escaping purposes |
| 18:16.07 | KDan | ie they use single quotes fo rthe second param there so that they can have '""' instead of "\"\"" |
| 18:16.09 | clyrrad | I know - but AGI strange sometimes |
| 18:16.14 | KDan | ok |
| 18:16.30 | KDan | will try the same as above line for line then (Except for the file name |
| 18:17.42 | KDan | $agi->stream_file("beep", '""'); |
| 18:17.42 | KDan | $agi->record_file("/asterisk/test", 'gsm', '0', 3000); |
| 18:17.43 | KDan | $agi->stream_file("beep", '""'); |
| 18:17.46 | KDan | trying now |
| 18:18.46 | KDan | same result |
| 18:18.53 | clyrrad | actually |
| 18:18.56 | KDan | (except 0-byte gsm instead of 44-byte wav) |
| 18:18.57 | clyrrad | here is the function prototype |
| 18:19.00 | clyrrad | array, record_file (string $file, string $format, [string $escape_digits = ''], [integer $timeout = -1], [integer $offset = NULL], [boolean $beep = false], [integer $silence = NULL]) |
| 18:19.40 | KDan | that's what i pasted earlier :-) |
| 18:19.54 | clyrrad | so $agi->record_file("/asterisk/test", 'GSM', '0', 3000,NULL,1,0); |
| 18:20.11 | clyrrad | I found it here: http://phpagi.sourceforge.net/phpagi2/docs/phpAGI/AGI.html#methodrecord_file |
| 18:20.48 | KDan | yeah that's the same description as on the eder.us site |
| 18:21.01 | clyrrad | what happens when you use the line i put above |
| 18:21.22 | KDan | trying |
| 18:22.16 | KDan | Aug 20 19:22:45 WARNING[25136]: file.c:988 ast_writefile: No such format 'GSM' |
| 18:22.18 | KDan | :-) |
| 18:22.21 | KDan | trying lowercase |
| 18:22.52 | clyrrad | its case sensitive? lol |
| 18:23.41 | KDan | same as before... hangs and produces a 0-byte .gsm |
| 18:23.45 | KDan | :-( |
| 18:23.54 | clyrrad | can you paste bin your PHP |
| 18:23.54 | KDan | would have thought recording would be easy!!! :-P |
| 18:23.57 | KDan | sure |
| 18:24.32 | KDan | http://textpaste.net/spni35 |
| 18:25.05 | clyrrad | may be a dumb question... but.... |
| 18:25.28 | clyrrad | your script is ../ below phpagi/phpagi.php right? |
| 18:25.28 | KDan | yep |
| 18:25.30 | clyrrad | ok |
| 18:25.31 | KDan | :-) |
| 18:25.46 | clyrrad | so saydigits is working? |
| 18:25.49 | clyrrad | you hear it |
| 18:25.50 | KDan | yep |
| 18:25.52 | clyrrad | ok |
| 18:26.00 | *** join/#asterisk Flauto (n=zhao@adsl-75-3-139-218.dsl.chcgil.sbcglobal.net) |
| 18:26.01 | KDan | and i hear the beep too |
| 18:26.04 | clyrrad | streamfile beep work? |
| 18:26.04 | clyrrad | ok |
| 18:26.11 | KDan | in the case of this latest sample, i actually hear two beeps |
| 18:26.23 | Flauto | make[1]: *** No rule to make target `../makeopts'. Stop. |
| 18:26.23 | Flauto | make[1]: Leaving directory `/usr/src/asterisk/menuselect' |
| 18:26.23 | Flauto | make: *** [clean] Error 2 |
| 18:26.28 | clyrrad | ok |
| 18:26.37 | clyrrad | take out the NULL so that its just ,, |
| 18:26.38 | Flauto | i got this when i installing asterisk on centos |
| 18:26.46 | KDan | that won't work in php |
| 18:26.55 | Qwell | Flauto: Do you have the absolute latest revision of trunk? |
| 18:27.07 | Flauto | no |
| 18:27.13 | Qwell | What do you have? |
| 18:27.13 | Flauto | so, i should get 1.2? |
| 18:27.22 | clyrrad | and put a $agi->say_digits("12345"); after record just to see if it really is not getting there |
| 18:27.37 | Qwell | Flauto: Just do a `svn update`, then try again |
| 18:27.51 | Flauto | okay |
| 18:27.54 | clyrrad | howdy Qwell |
| 18:28.34 | clyrrad | KDan - does it do the last say_digits ? |
| 18:28.44 | Flauto | qwell, i got this same thing after the update |
| 18:28.47 | KDan | let me try that |
| 18:29.38 | techie | menuselect hell |
| 18:29.48 | KDan | nope, can't hear the digits |
| 18:29.51 | KDan | (At the end) |
| 18:30.13 | Flauto | i thought i would run into problems on zaptel installation with centos, but i did not at all |
| 18:30.25 | clyrrad | damn... its like that function is hanging waiting on something.... how are you trying to end the recording? |
| 18:31.07 | KDan | pressing 0. i'm going to try doing the recording directly from the dialplan, see if that works (thanks techie for the suggestion) |
| 18:31.11 | clyrrad | and does the function ever reach its timeout parameter and stop? |
| 18:31.25 | KDan | no, it doesn't seem to reach them. let me just try doing a standard record |
| 18:31.30 | KDan | see whether that works.. |
| 18:32.40 | Flauto | qwell, i got the same thing |
| 18:32.48 | Flauto | after updated |
| 18:32.58 | Flauto | anything else i can do |
| 18:33.10 | Qwell | Don't use trunk, if you don't know how to figure out what broke... |
| 18:34.58 | KDan | exten => _X.,4,Record(dan-message:gsm) |
| 18:35.02 | KDan | also hangs :-( |
| 18:35.31 | KDan | with a 0-byte .gsm |
| 18:36.04 | clyrrad | i wonder if its not getting the DTMF to tell it to stop recording |
| 18:36.21 | Qwell | # terminates Record(), not 0 |
| 18:36.36 | KDan | i tried # as well. let me try again to make sure. |
| 18:37.37 | KDan | same result |
| 18:37.55 | KDan | it just doesn't hear the DTMFs, and never puts anything in the file... could it be that x-lite doesn't send the DTMF's properly? |
| 18:37.56 | clyrrad | it looks like its not getting the DTMF from your phone to tell it to stop |
| 18:37.56 | Qwell | set a maxduration of like 5 |
| 18:38.19 | KDan | k |
| 18:38.44 | clyrrad | what version of Asterisk are you running KDan? |
| 18:39.15 | KDan | exten => _X.,4,Record(dan-message:gsm|5|5) |
| 18:39.17 | KDan | 1.2.10 |
| 18:40.04 | KDan | ok it did time out and hang up |
| 18:40.15 | KDan | but the .gsm file is still empty |
| 18:40.33 | clyrrad | so looks like i may be correct its not getting the DTMF when you press # |
| 18:40.42 | clyrrad | are you using an OLD phone? |
| 18:40.46 | clyrrad | or is it a newer one |
| 18:40.52 | KDan | i'm using the latest version of x-lite |
| 18:40.58 | clyrrad | ahhhhhhhhh |
| 18:41.02 | KDan | about as new as it gets :-) |
| 18:41.05 | clyrrad | do you have a hardware phone |
| 18:41.15 | KDan | yes, but my asterisk is not connected to the pstn atm |
| 18:41.22 | clyrrad | does not have to be |
| 18:41.26 | Qwell | hardware phone isn't going to solve anything |
| 18:41.41 | clyrrad | well i want to rule out if its his config in the software phone |
| 18:41.42 | Qwell | Just make sure you have the same dtmfmode in both sides |
| 18:42.01 | clyrrad | if the hardware phone works and the software one dont - then we know the issue is in the phone and not asterisk |
| 18:42.08 | clyrrad | anyway im betting its the DTMF |
| 18:42.12 | clyrrad | what do you have it set to? |
| 18:42.40 | KDan | haven't even touched the x-lite dtmf setting |
| 18:42.48 | KDan | trying to find it |
| 18:42.54 | clyrrad | im betting its wrong |
| 18:42.55 | yatesy | just by chance, does anyone know of a device that'll do FXO and FXS in one box that then connects to a machine running asterisk using ethernet? i can't get a PCI card caus the asterisk machine runs OpenBSD |
| 18:42.58 | clyrrad | let me know what its set to |
| 18:43.08 | Qwell | yatesy: sipura spa3000 |
| 18:43.09 | clyrrad | also its connected with SIP right? |
| 18:43.17 | KDan | yes |
| 18:43.30 | clyrrad | ok find the DTMF setting |
| 18:43.49 | KDan | can't see a DTMF setting in x-lite |
| 18:44.12 | clyrrad | its there you just gotta find it |
| 18:44.57 | yatesy | Qwell: thanks, looking into it now |
| 18:45.46 | joburg | <PROTECTED> |
| 18:48.40 | KDan | even if the dtmf's are not being sent, surely the recording should work? |
| 18:48.52 | KDan | ie cut off after 5 seconds and dump what's already been recorded? |
| 18:49.05 | clyrrad | one step at a time my friend |
| 18:49.09 | clyrrad | first you need to find that setting |
| 18:49.32 | KDan | yeah, it's not in the snazzy x-lite interface. it appears they don't care about such mundane settings :-) |
| 18:49.51 | clyrrad | that is a pretty important setting |
| 18:50.34 | KDan | perhaps... got a better softphone to recommend? |
| 18:50.51 | clyrrad | I only ever used XLite |
| 18:50.57 | clyrrad | but if you have a hardware phone try it |
| 18:51.05 | clyrrad | that will rule out so many variables |
| 18:51.09 | KDan | can't - no pstn or hardware connection |
| 18:51.15 | clyrrad | im betting your config is fine and hte softphone is screwed |
| 18:51.17 | KDan | just plain old IP |
| 18:51.29 | clyrrad | you have an IP phone? |
| 18:51.36 | clyrrad | ohhh you have just a regular phone...? |
| 18:51.38 | KDan | xlite is an IP phone |
| 18:51.45 | clyrrad | no a hardware IP phone |
| 18:51.50 | KDan | no, i don't |
| 18:52.14 | clyrrad | no ata? |
| 18:52.26 | KDan | ata? |
| 18:52.44 | clyrrad | k that answers my question hehe you dont have it |
| 18:52.48 | joburg | dial from your console |
| 18:52.50 | KDan | :-) |
| 18:52.57 | KDan | joburg: how do i do that? |
| 18:53.01 | clyrrad | errrr the softphone is your prolblem it would bet money on it if I was a gambling man |
| 18:53.20 | joburg | use the dial command from your console |
| 18:53.34 | joburg | then use it again to dial the digits |
| 18:53.45 | KDan | but then it definitely won't record anything? |
| 18:53.57 | joburg | to end simply use the hangup command |
| 18:54.26 | joburg | it will record if you have a speaker/microphone |
| 18:54.39 | joburg | the console is the best softphone for asterisk |
| 18:55.10 | KDan | well, the console is running on a dell server in my friend's bathroom, 2 hours away by train :-) |
| 18:55.17 | KDan | so i'd need a pretty serious microphone |
| 18:55.34 | yatesy | bathroom?! haha what a legend! |
| 18:55.37 | remiss | KDan: you can make it.. don't have to go to bed yet, right? |
| 18:55.54 | KDan | remiss: nah, only got a massively busy day at work tomorrow... don't need sleep :-) |
| 18:56.03 | KDan | yatesy: coolest room in the house :-P |
| 18:56.29 | joburg | well if you send it digits it will porbably record the digits..... |
| 18:56.39 | KDan | (note: he does not take his showers in that bathroom... he has another one) |
| 18:57.01 | remiss | james blunt is ok MOH, right? |
| 18:57.10 | yatesy | KDan: ah ok heh |
| 18:57.26 | adelas | can asterisk support video ? |
| 18:57.30 | Qwell | adelas: yes |
| 18:57.38 | adelas | your bsing me/ |
| 18:57.41 | adelas | ?? |
| 18:58.17 | adelas | is there even a softphone that has video? |
| 18:58.24 | *** join/#asterisk test34 (n=test34@unaffiliated/test34) |
| 18:58.41 | KDan | gonna try a different softphone first |
| 18:58.59 | adelas | Qwell, are you serious? |
| 18:59.08 | Qwell | yes, it supports video |
| 18:59.16 | clyrrad | KDan - i bet its the softphone - it will fix your problems im sure |
| 19:00.08 | KDan | clyrrad: let's hope so! |
| 19:00.36 | adelas | Qwell, is the video feature on by default? just need video clients? |
| 19:00.37 | joburg | url of this video softphone ? |
| 19:01.01 | Qwell | adelas: sip.conf, videosupport=yes |
| 19:02.15 | adelas | Qwell, are there any free video clients out there?(softphone)? |
| 19:02.41 | Qwell | whatever gnomemeeting renamed itself to, I believe |
| 19:02.44 | Qwell | ekiga or something |
| 19:03.01 | KDan | hmm, great, 3CX Phone gives a "protocol error, layer 2". NEXT! |
| 19:03.40 | KDan | let's try an iax phone... *sigh* |
| 19:03.54 | adelas | awesome, thanks |
| 19:09.29 | KDan | YAY |
| 19:09.35 | KDan | it works with IAX + ePhone |
| 19:11.01 | *** join/#asterisk SwK_ (n=Silik0nJ@12-218-74-89.client.mchsi.com) |
| 19:13.41 | adelas | hey does anyone know if its possible to have a cisco conference phone to work with asterisk? |
| 19:13.46 | adelas | b/c it dosn't support sip.. |
| 19:13.58 | *** join/#asterisk alpinus (n=alpinus@81.219.54.125) |
| 19:13.59 | adelas | or so it says, but its a polycom based phoned |
| 19:14.57 | adelas | cisco 7935 |
| 19:14.58 | joburg | is the cisco a skinny phone? |
| 19:15.05 | KDan | fyi, the php works fine now with the iax ip-phone |
| 19:15.12 | adelas | yea |
| 19:15.23 | KDan | so the error was with the soft-phone - thanks everyone :-) |
| 19:15.50 | KDan | >> goes to grab some food |
| 19:16.08 | adelas | any ideas if its possible? |
| 19:16.40 | joburg | not if it's skinny |
| 19:17.01 | adelas | well its cisco thingy, so |
| 19:17.53 | joburg | you can convert cisco skinny to sip but it's a mission! |
| 19:18.09 | adelas | nope. theres no sip converting for this crap |
| 19:18.26 | adelas | its not a regulart cisco 7960/7940, but a conference phone |
| 19:19.04 | adelas | i tried to do some bsing update with the phone, and tftp server says file not found ect |
| 19:19.10 | adelas | and file not matched |
| 19:21.46 | *** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin) |
| 19:22.01 | *** part/#asterisk joburg (n=voipmagi@vc-196-207-37-206.3g.vodacom.co.za) |
| 19:22.51 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 19:36.24 | rogier | I'm having some issues with the sip module. Firstly it won't load automatically. Is that default behaviour ? Then, if I load it from the terminal, it loads, but won't bind to port 5060. If I try to load it through modules.con, the load fails. |
| 19:37.06 | rogier | Loading through modules.conf failes with: WARNING[12590] loader.c: /usr/lib/asterisk/modules/chan_sip.so: undefined symbol: ast_park_call |
| 19:37.17 | rogier | How can I effectively debug this ? |
| 19:37.53 | rogier | I use 1.2.10 |
| 19:38.02 | rogier | asterisk version that is..... |
| 19:41.04 | *** join/#asterisk VoicePulse (n=contact@unaffiliated/voicepulse) |
| 19:46.46 | *** join/#asterisk marv (n=ilovekim@c-71-228-189-127.hsd1.al.comcast.net) |
| 19:49.34 | *** part/#asterisk techie (n=gus@adsl-068-209-242-072.sip.mia.bellsouth.net) |
| 19:49.35 | *** join/#asterisk smurf (n=smurf@debian/developer/smurf) |
| 19:50.43 | *** join/#asterisk TrixVox (n=Spy007@c-69-248-121-104.hsd1.nj.comcast.net) |
| 19:57.15 | Un1x | isn't there a channel |
| 19:57.19 | Un1x | for Asterisk Biz? |
| 19:57.30 | Un1x | there is a mailing list but no channel weird :/ |
| 20:00.59 | ruskie | hmmm if I want to use multiple sip providers through asterisk to dial out I guess I need to assign them extensions right? |
| 20:01.00 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 20:02.40 | Un1x | ruskie: id assume so or you could build a AGI script to select |
| 20:03.11 | *** join/#asterisk shodan (n=shodan@ip015.96-113-216.pppoe1.joliette.intermonde.net) |
| 20:03.28 | ruskie | hmm anyone have a sample config for ekiga? |
| 20:04.30 | shodan | anyone got the manufacturer's site for netweb 301 phones ? a friend just lended me his but forgot the cds and manuals |
| 20:05.16 | ruskie | hmm how about for incoming calls? I have this in my default: exten => s,1,dial(SIP/205) will this work for all or do I need to do anything else? |
| 20:06.03 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 20:06.23 | *** join/#asterisk dusan2 (i=dusan@209-223-47-160-static.oplink.net) |
| 20:07.34 | *** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin) |
| 20:10.58 | oelewapperke | how can this be : |
| 20:10.58 | *** join/#asterisk |dennis| (n=dennis@200.32.215.82) |
| 20:10.58 | oelewapperke | Aug 20 19:28:36 WARNING[25605] channel.c: Unable to find a codec translation path from gsm to g729 |
| 20:11.05 | oelewapperke | when show codecs shows both gsm and g729 in my asterisk installation |
| 20:14.26 | file | oelewapperke: show codecs does not show you what codecs are installed |
| 20:15.00 | *** join/#asterisk ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 20:28.06 | *** join/#asterisk stinkpad (n=cunted@cpc2-broo4-0-0-cust398.renf.cable.ntl.com) |
| 20:28.51 | stinkpad | is there any way to detect remote hangup on an x100p in the uk? |
| 20:29.43 | *** join/#asterisk _deg_ (n=deg@201-40-223-25.ctame704.dsl.brasiltelecom.net.br) |
| 20:34.20 | *** join/#asterisk |dennis| (n=dennis@200.32.215.82) |
| 20:37.10 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 20:38.02 | KDan | not strictly an asterisk question, but since .mp3 is read-only in asterisk, what's the recommended software to use to convert recorded wavs to mp3s as soon as they're finished recording? |
| 20:38.11 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 20:42.36 | *** join/#asterisk Dibbler_ (n=Dibbler@dsl-217-155-254-174.zen.co.uk) |
| 20:44.31 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 20:45.55 | *** part/#asterisk sevard (n=sev@c-67-188-173-23.hsd1.ca.comcast.net) |
| 20:47.15 | pcm | Un1x: what do you want done ? |
| 20:47.25 | pcm | ups; |
| 20:47.39 | pcm | anyone needs a bounty coded/made ? |
| 20:50.33 | rogier | if I set autoload=yes in modules.conf , all modules should be loaded automatically right ? Not working here.... |
| 20:51.00 | rogier | Any tips what I should be looking for to make that work ? |
| 20:51.03 | pcm | and what doesn't load ? |
| 20:51.27 | rogier | Well, specifcally the chan_sip.so module, but also no codec is loaded |
| 20:51.50 | pcm | the reason is that chan_sip propably blocks |
| 20:51.59 | pcm | since one of the ports it tries to request is used by something else |
| 20:52.12 | pcm | stop asterisk |
| 20:52.13 | pcm | netstat -l |
| 20:52.24 | pcm | check if 5060 or other port is used ... |
| 20:52.29 | pcm | kill the guilty process |
| 20:52.31 | pcm | or reboot the box |
| 20:52.34 | pcm | try again |
| 20:52.38 | rogier | If I hardcode chan_sip.so in modules.conf to be loaded, it does indeed block, but if I load it from the asterisk shell, everything works... |
| 20:52.58 | pcm | ok, then try this |
| 20:53.03 | pcm | put unload => chan_sip.so |
| 20:53.08 | pcm | and from CLI load chan_sip.so |
| 20:53.25 | pcm | it propably will block the CLI ... |
| 20:53.36 | pcm | and then you know something there goes bad |
| 20:54.02 | *** join/#asterisk Nebukadneza (n=daddel9@i3ED6F6B0.versanet.de) |
| 20:54.32 | rogier | or everything, rather I mean, the module loads without errors, I can register with a softphoen an it registers to voip providers. I cannot yet call the pbx with my softphone to run the echo test for example, without any codecs loaded. |
| 20:54.40 | rogier | Ok, will try that ocm, thanks. |
| 20:54.47 | rogier | err, *pcm* |
| 20:54.57 | pcm | forgiven |
| 20:58.02 | *** join/#asterisk Stormy (n=maryann@p3m/member/Stormy) |
| 20:58.38 | Stormy | Could someone tell me of a utility that I can see my microphone on some kind of a sound meter in linux. I'm trying to debug this thing and i'm not sure if my microphone is even working |
| 21:00.03 | rogier | ah, pcm, you put me on the right trach. It was actually chan_phone.so that was blocking the autoloading of all modules. Thanks |
| 21:04.29 | *** join/#asterisk rg1_ (n=rg1@www.airlinksystems.com) |
| 21:04.50 | KDan | Stormy: is there a skype for linux? |
| 21:05.17 | KDan | yep there is~ |
| 21:05.29 | rg1_ | When I setup for "Record" function, and I'm looking for "Silence" to indicate the person has stopped talking - is there a way to adjust for ambient noise (i.e. in a car on a cell phone) |
| 21:06.03 | rg1_ | like maybe be able to specify what "noise" is recording range or something? |
| 21:07.55 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.41.91) |
| 21:09.27 | shodan | is there some reasonably (under 200$usd) priced voip videophone that works with * ? |
| 21:09.43 | rg1_ | darkness - you know anything about adjusting the frequency range that asterisk recognizes as "voice"? |
| 21:09.49 | *** join/#asterisk callee (n=unknown@2002:5387:41e5:0:0:0:0:1) |
| 21:10.14 | callee | hi, i have a short question: can a normal isdn modem using spandsp send fax? |
| 21:10.30 | callee | or even better recieve faxes? |
| 21:10.57 | callee | i am asking because i would like to get rid of an analog modem |
| 21:13.41 | pcm | I think it would |
| 21:13.56 | pcm | ISDN BRI .... card under vmisdn ? |
| 21:15.18 | *** part/#asterisk pcm (n=pcm@68.159.139.234) |
| 21:15.30 | *** join/#asterisk pcm (n=pcm@68.159.139.234) |
| 21:15.31 | *** join/#asterisk Jenocin (i=jenocin@99.3.118.70.cfl.res.rr.com) |
| 21:15.48 | *** part/#asterisk pcm (n=pcm@68.159.139.234) |
| 21:15.58 | *** join/#asterisk pcm (n=pcm@68.159.139.234) |
| 21:16.39 | *** part/#asterisk brand-new-nick (n=pcm@68.159.139.234) |
| 21:17.27 | *** join/#asterisk AJaymn (n=boiwonde@70.59.126.206) |
| 21:19.43 | *** join/#asterisk daniel_bergamini (n=daniel_b@70-41-166-149.cust.wildblue.net) |
| 21:20.00 | *** join/#asterisk asterisk_bounty (n=pcm@68.159.139.234) |
| 21:20.04 | KDan | i have a system (based on asterisk) that occasionally records .wav files to a directory... I want some sort of script to check this directory ever half-second, and if the file is there, to quickly encode it to mp3 and move it to another directory... what would be the best encoder/utility/magic button to use? |
| 21:20.29 | *** part/#asterisk asterisk_bounty (n=pcm@68.159.139.234) |
| 21:20.52 | callee | it is a hfc-based card running with misdn |
| 21:21.05 | callee | dunno yet what bri is |
| 21:22.36 | daniel_bergamini | KDan I think mplayer will do conversions like that, cron is likely your best bet |
| 21:23.05 | KDan | thanks |
| 21:23.07 | daniel_bergamini | KDan http://gimpel.gi.funpic.de/wiki/index.php?title=Howto:convert_aac/mp4_to_wav/mp3/ogg_on_Linux |
| 21:23.16 | hads | Indeed, also if you have a lot of them it may be better to do the transcoding on another box |
| 21:23.33 | KDan | yeah as it scales up we no doubt will |
| 21:23.59 | hads | Oh and lame may be a better choice if you want mp3 |
| 21:24.26 | KDan | sweet, great link |
| 21:24.27 | daniel_bergamini | yeah that's true |
| 21:24.31 | KDan | thanks both of you :-) |
| 21:24.38 | daniel_bergamini | sorry that example did use lame |
| 21:24.58 | hads | :) |
| 21:25.53 | Jenocin | what do you guys consider to be the best trunk provider? voicepulse? |
| 21:25.57 | hads | BTW cron only goes down to 1 minute, not seconds. |
| 21:26.18 | daniel_bergamini | you could just use a looped script |
| 21:26.24 | *** join/#asterisk asterisk_bounty (n=pcm@68.159.139.234) |
| 21:26.37 | hads | Yeah, or decide if you really need it every second :) |
| 21:26.39 | daniel_bergamini | ok I give up, what username and password do I use to install freePBX? |
| 21:27.26 | hads | daniel_bergamini: No idea sorry, but the people over at #freepbx may be of more assistance. I don't think many people here use it. |
| 21:27.31 | daniel_bergamini | whether I try my account or root I get "Connecting to database..FAILED" |
| 21:27.35 | daniel_bergamini | oh yeah? |
| 21:27.56 | daniel_bergamini | ok then I don't necessarily need to use it I'm just having trouble with the basic configuration and figured it might help |
| 21:28.37 | hads | FreePBX will take you from basic config to having mucho complex config files in a matter of seconds. |
| 21:28.58 | hads | What exactly are you having trouble with? |
| 21:29.05 | KDan | hads: do need it every second. every half-second preferrably! |
| 21:29.14 | KDan | i guess i'll use a looping php script :-) |
| 21:29.44 | hads | PHP wasn't really designed for daemon processes, but OK. |
| 21:30.55 | daniel_bergamini | I'm trying to get a basic setup going |
| 21:31.04 | daniel_bergamini | right now I'd be happy with X-lite being able to login |
| 21:31.51 | hads | daniel_bergamini: Have you read 'the book'? |
| 21:32.03 | hads | ~thebook |
| 21:32.05 | jbot | it has been said that thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 21:32.10 | *** join/#asterisk uk-wombat (n=root@82.163.6.212) |
| 21:32.28 | daniel_bergamini | yeah I got a copy of that book, but it droned on and on. will that step me through everything |
| 21:32.51 | KDan | i found it was really very readable |
| 21:33.03 | hads | If it was too wordy for you then there is a chapter in there with sample configs that you might like to copy from. |
| 21:34.05 | *** join/#asterisk evisu (n=hIRC@bzq-88-154-45-231.red.bezeqint.net) |
| 21:38.51 | uk-wombat | Hi, has anyone managed to compile libss7? |
| 21:41.59 | *** join/#asterisk Amilcar_ (n=Email@201.11.187.247) |
| 21:43.56 | hmmhesays | daniel_bergamini: you want to skip the research you better have the cash to compensate |
| 21:46.05 | daniel_bergamini | is this bad? WARNING[6892]: res_odbc.c:565 odbc_obj_connect: res_odbc: Error SQLConnect=-1 errno=0 [unixODBC][Driver Manager]Data source name not found, and no default driver specified |
| 21:46.22 | daniel_bergamini | hmmhesays I have time to compensate for both |
| 21:46.31 | daniel_bergamini | this is just a home project I'm playing with |
| 21:46.42 | shodan | I have a problem, my dad used to own a business in a nearby town but now he moved and his phone number can't move , so he asked another business to host his phone number so he can use call transfert to the new number , but now I just installed asterisk at the new location (residential aera) he has 2 phone lines and apparently I can't stack phone lines in a residential aera , so is there a way I can have people who call t |
| 21:46.42 | shodan | he old number not hit a busy signal if there is already someone on the line ? maybe I can redirect to a voip number and have that number redirect to another number alternating between the 2 lines ? |
| 21:48.00 | KDan | shodan: you could set it up with a DID provider so that it goes straight into asterisk through SIP or AIX |
| 21:48.11 | KDan | not sure if they allow you to transfer an existing number though |
| 21:48.50 | KDan | (some of them must, i guess) |
| 21:49.29 | shodan | the problem is that the internet connection at the new location is not reliable enough so I can't use it for the phone |
| 21:49.49 | KDan | ah... well that would be a problem :-) |
| 21:51.03 | shodan | it's probably good enough for extra outbound lines , but I can't risk the connection being down to cause a phone blackout (it's a WISP :\ ) |
| 21:51.19 | KDan | WISP? |
| 21:51.28 | daniel_bergamini | wireless isp? |
| 21:51.32 | shodan | yes |
| 21:51.47 | shodan | it's the only thing available here |
| 21:52.54 | hmmhesays | callforward on busy should be something your telco providers |
| 21:53.06 | hmmhesays | or unconditional forward |
| 21:53.08 | *** join/#asterisk Spacy (n=spacy@p508C79F4.dip.t-dialin.net) |
| 21:53.11 | daniel_bergamini | yeah if I recall it's something like *68 |
| 21:53.17 | hmmhesays | you coul duse that also shodan |
| 21:53.56 | Spacy | Good evening... or whatever time it is at your location ;) |
| 21:54.02 | shmaltz | Spacy, hi |
| 21:54.14 | shodan | oh forward on busy would be great I could set that on the first line and essentially my line would be stacked |
| 21:54.38 | shodan | I'm using Bell Canada btw, I'll call them to see if it's available |
| 21:55.28 | *** join/#asterisk DrRighteous (n=DrRighte@ool-457843d1.dyn.optonline.net) |
| 21:56.56 | shmaltz | shodan, in most cases the telco will block anything more than one channel from being forwarded on regular POTS |
| 21:57.37 | shmaltz | and by "on regular POTS" I mean that the phone number does NOT belong to a cell phone |
| 21:57.54 | rg1_ | ANYONE know if you can make some adjustment on asterisk to account for ambient noise - so that for it to detect "silence" it will do that if someone is not talking - but yet there is background noise? |
| 21:58.50 | shodan | shmaltz, ok , so since my line is already forwarded it won't get forwarded a second time ? |
| 21:58.52 | shmaltz | rg1, not yet |
| 21:59.01 | hmmhesays | if you call forward onconditional you can just forward all calls to a voip number |
| 21:59.08 | hmmhesays | wow this vnc is farked up |
| 21:59.09 | shmaltz | shodan, exactly, only the first caller will get forwarded |
| 21:59.10 | rg1_ | shmaltz - you know of any plans for them to do that? |
| 21:59.24 | Spacy | I have a problem recording voice mail (* trunk) and would like to know if anyone experienced the same problem. The recording is very choppy. It's coming in on chan_capi (Fritz card PCI). I tested connection with SIP softphone and had no probs so far. So I think its a voicemail (recording) problem. Playback of menu-messages is just fine. |
| 21:59.31 | shmaltz | rg1, no clue |
| 22:00.14 | shmaltz | hmmhesays, you could forward to voip number, but only the first caller will get to the forwarded number, and the 2nd will get a busy signal |
| 22:00.23 | *** join/#asterisk Skyelar (n=planet@222-153-145-60.jetstream.xtra.co.nz) |
| 22:01.00 | shmaltz | shodan, the best -and most expensive - workaround to my knowledge, is to convert the phone number to a remote forwarding number with the telco, and purchase mutiple callpaths |
| 22:01.52 | shmaltz | shodan the seceond best workaround and a bit cheaper is to PORT the number (not forward) to a VoIP provider (like vongage) and use their call fowarding services, which usualy will allow you for more than one call to be forwarded. |
| 22:02.50 | shodan | bell said "that's not possible" (or something) :\ are they BSing me ? |
| 22:03.04 | shodan | but I wasn't very specific on doing that |
| 22:04.00 | shodan | it already costs 53$cad/month just to keep the number and forward so that would be great |
| 22:04.02 | shmaltz | shodan, that whats not possible? |
| 22:05.17 | shodan | shmaltz, well it was my dad speaking with the phone rep and I wasn't there, she told him the current hosting in another business and forwarding was the only way he could keep his number |
| 22:05.37 | shodan | she probably didn't consider switching to the competition a possibility ;) |
| 22:05.39 | oelewapperke | file: then how do I check what codecs are installed ? |
| 22:06.21 | shmaltz | shodan, here: |
| 22:06.23 | shmaltz | http://enterprise.bell.ca/en/default.asp?sid=39&did=239 |
| 22:06.46 | Spacy | nobody experiencing problems with current trunk voicemail? or is just nobody using it? *g* |
| 22:06.47 | *** join/#asterisk draco_710 (n=tlambeth@12-214-163-60.client.mchsi.com) |
| 22:07.28 | shmaltz | Spacy, I would assume the former, which of course implies that you should read the docs |
| 22:07.38 | daniel_bergamini | join #freepbx |
| 22:07.42 | daniel_bergamini | heh oopsies |
| 22:09.08 | shmaltz | shodan, you followed the link? |
| 22:09.11 | shodan | shmaltz, that Remote Call Forwarding, it costs long distance to the receiver for each call right ? |
| 22:09.16 | shodan | yes |
| 22:09.43 | shmaltz | shodan, of course, so does it now, it's actualy cheaper than what you pay now ($53) |
| 22:09.58 | Skyelar | With the Asterisk 1.2 Manager API, is there any way to match up a synchronous Originate with the associated Newchannel? |
| 22:10.11 | shmaltz | also each talkpath will cost some more money (like $8 per month in the US) |
| 22:10.29 | shmaltz | Skyelar, you mean brdige 2 known channels? |
| 22:11.00 | shmaltz | Skyelar, the best you can hope for is dump them both in a meetme room |
| 22:11.08 | *** join/#asterisk Darthclue (n=Darthclu@adsl-69-153-22-31.dsl.snantx.swbell.net) |
| 22:11.13 | Skyelar | shmaltz: I'm doing an Originate, and I want to be able to find the unique ID of the created channel so I can follow the calls progress |
| 22:11.36 | shmaltz | Skyelar, you don't need unitque ID for that |
| 22:12.52 | Skyelar | shmaltz: this is in the manager API, not the dialplan - the success response to the Originate doesn't contain any information that I can see for matching up later Newchannel / Newexten / etc. events to the initial Originate |
| 22:13.11 | shodan | shmaltz, I haven't received the first bill yet but are you saying I will have to pay long distance charge on top of the 53$/month ? the old number is in 450-754 and the new number is in 450-755 I can call this number toll-free but it costs long distance to forward ? |
| 22:13.13 | shmaltz | SKyekar, yes it does |
| 22:13.24 | shmaltz | shodan, yes |
| 22:13.29 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 22:13.40 | shmaltz | if you forwarded it to a number that incures LD charges |
| 22:14.30 | KDan | the poor little broke phone companies have to get their money *somewherE* |
| 22:14.50 | shmaltz | KDan, exactly |
| 22:15.48 | Skyelar | shmaltz: all I get back from Originate is "Response: Success\nActionID: XXX\nMessage: Originate successfully queued" - shortly afterwards, there's a Newchannel event, but I can find nothing to match the two together (_reliably_ - without looking at just Channel, which leaves me open to a race condition) |
| 22:16.59 | shmaltz | Skyelar, the channel info will always hold: |
| 22:17.01 | shmaltz | 1. The channel you originated the call to + numbers if any. |
| 22:17.02 | shmaltz | 2. The other end of the channel (dial command etc.) which in most practical cases is unique enough |
| 22:17.58 | *** part/#asterisk tmccrary (n=tmccrary@d14-69-160-83.try.wideopenwest.com) |
| 22:20.04 | Skyelar | shmaltz: you understand I'm talking about the manager interface, right? I can't see how to tell if a Newchannel event is the result of my Originate, or of someone else calling them at the same time. |
| 22:20.33 | shmaltz | Skyelar, I do understand you are talking about the manager API. |
| 22:21.05 | Skyelar | shmaltz: is this race just something I'll have to live with for the moment? |
| 22:21.06 | shmaltz | Skyelar, I personaly have never worked with the manager API, but I'm sure that the manager API has got as much info as asterisk CLI command: show channels |
| 22:21.31 | shmaltz | if it doesn't then you can use shell: asterisk -rx "show channels" |
| 22:21.39 | shmaltz | then a grep for you channel |
| 22:21.53 | shmaltz | then: asterisk -rx "show channel justgrepted" |
| 22:22.28 | *** join/#asterisk Budairc (n=budairc@mercurio.mhnet.com.br) |
| 22:23.01 | shmaltz | Skyelar, there are plenty of ways to do what you want, I just gave you one, and for some reason I'm sure that the manager API gives you all the info |
| 22:23.03 | shmaltz | gtg |
| 22:23.06 | shmaltz | good day guys |
| 22:23.20 | Skyelar | shmaltz: unfortunately it doesn't, so I'll have to keep working on it... or patching it :) |
| 22:23.30 | Skyelar | shmaltz: thanks for your help |
| 22:23.47 | shmaltz | Skylear, patching is not the solution as the info is available as I showed you thru the shell |
| 22:25.13 | Skyelar | (belately) unfortunately not in any way I can see |
| 22:25.19 | rg1_ | for Record() - is there an easy/any way to know if the recording was stopped because of "Silence" or if the user pressed the "*" key? |
| 22:27.17 | Skyelar | rg1_: it doesn't appear to return any differently either way, at least in 1.2.10 |
| 22:30.57 | *** join/#asterisk Kumba0 (n=kumba@210-208.124-70.tampabay.res.rr.com) |
| 22:34.07 | Spacy | Hm. can someone confirm that app_voicemail in latest trunk (last change was 45hours ago) is recording messages fine? or could anyone imagine a cause why my recorded messages are chopped? |
| 22:39.19 | Kumba0 | In asterisk the help command doesn't list any zap commands... any ideas why? |
| 22:42.06 | adelas | hey for the cisco SCCP (cisco confereence phone 7935), is it possible to get it to work? |
| 22:47.23 | shodan | damn, I just found out my dad took a 1 year contract with bell :( too bad.. |
| 22:47.59 | shodan | anyone knows what is the *number for forward on busy with bell canada ? |
| 22:51.04 | *** join/#asterisk litage (n=nick@203.220.55.70) |
| 23:00.48 | Kumba0 | zap isn't even listed on the channel type |
| 23:00.54 | *** part/#asterisk Jenocin (i=jenocin@99.3.118.70.cfl.res.rr.com) |
| 23:07.23 | *** join/#asterisk wwalker (n=wwalker@pdpc/supporter/sustaining/wwalker) |
| 23:09.29 | wwalker | anyone have a pointer to a very simple extensions.conf? the sample is 350 lines of every feature and macro around. I'd like a starting point I can explain to people. just a couple of extensions, outgoing call context, incoming call context maybe. I'm fairly new to asterisk (well used it a long time, but as an AGI developer, not as one who configures it |
| 23:15.14 | *** join/#asterisk techie (n=gus@adsl-068-209-242-072.sip.mia.bellsouth.net) |
| 23:19.07 | *** join/#asterisk Kumba_ (n=kumba@210-208.124-70.tampabay.res.rr.com) |
| 23:19.26 | *** join/#asterisk linlin (i=linlin@c-67-173-38-87.hsd1.il.comcast.net) |
| 23:20.02 | Kumba_ | I've installed two X100p's, set them up in Zaptel, according to 'ztcfg -vvv' it's installed both cards as channel 1/2... set the cards up in zapata.conf... but I dont see any zap commands/channels in asterisk... |
| 23:20.05 | Kumba_ | any ideas? |
| 23:20.53 | hads | Does show modules show chan_zap.so |
| 23:21.33 | Kumba_ | lemme see... |
| 23:21.59 | *** join/#asterisk doolph (n=doolph@200.46.148.58) |
| 23:22.01 | doolph | hi |
| 23:26.37 | hads | Spacy: Voicemail works fine for me. |
| 23:26.50 | Kumba_ | is there a way to get show modules to dump to a file? it scrolls out of buffer... |
| 23:27.10 | Kumba_ | when I do a show modules like chan_zap.so |
| 23:27.13 | Kumba_ | it returns nothing |
| 23:27.20 | Kumba_ | says no modules found |
| 23:27.32 | hads | asterisk -rx "show modules" > file |
| 23:28.14 | hads | It sounds liike you don't have zap support. |
| 23:28.26 | Kumba_ | is it from compiling asterisk before zap? |
| 23:29.18 | hads | Yes, if you did that then it will be. |
| 23:29.32 | Kumba_ | ok... so recompile asterisk... |
| 23:29.33 | hads | You need to compile and install zaptel and then compile Asterisk |
| 23:30.02 | Kumba_ | Then I can have all the fun and excitement of fiddling through a dialplan :D |
| 23:31.15 | *** join/#asterisk quid246 (n=quid24@CPE00131078ba5d-CM000f9f7eff1e.cpe.net.cable.rogers.com) |
| 23:33.08 | *** join/#asterisk uk-wombat (n=root@82.163.6.212) |
| 23:34.18 | hmmhesays | ok i'm sick of writing dialplan today |
| 23:34.22 | *** join/#asterisk hunmonk (n=hunmonk@pool-71-97-41-106.dfw.dsl-w.verizon.net) |
| 23:35.05 | *** join/#asterisk nailbags|work (n=nailbags@149.171.94.134) |
| 23:36.41 | hunmonk | hi all. i'm an open source developer trying to configure asterisk for the first time, and having a little trouble. i was hoping somebody here could help me figure out a few things, and i'm willing to pay for it. anybody interested? |
| 23:37.11 | Kumba_ | hmmhesays: You can write my dialplan if you like... it's simple :) |
| 23:37.15 | Kumba_ | i'll even buy you a beer? |
| 23:38.05 | hunmonk | Kumba_: i'll pay better than a beer :) |
| 23:38.36 | Kumba_ | I only know how to compile asterisk wrong... |
| 23:38.43 | Kumba_ | sorry |
| 23:39.00 | hmmhesays | sure for cash |
| 23:39.18 | hunmonk | Kumba_: i've got it compiled and installed just fine. the problem is the dialplan, i think |
| 23:40.14 | hunmonk | hmmhesays: i can pay via paypal. i have 3 or 4 basic things i'd like to get working, and i'd like somebody to help me figure out where i'm screwing up :) |
| 23:40.32 | Kumba_ | Yeah... the dialplan is fun... |
| 23:40.37 | hmmhesays | what are you looking to do? |
| 23:40.42 | Spacy | hads: thx for the info. any idea what i could try to get rid of the choppy messages? |
| 23:40.59 | Kumba_ | Anyone got a dialplan command reference link somewhere? (gives me the command, and options for it) |
| 23:41.08 | hmmhesays | use ztdummy |
| 23:41.17 | hmmhesays | hunmonk wha are you looking to do? |
| 23:41.20 | hunmonk | hmmhesays: 1) place outbound IAX calls to anywhere, including another IAX user |
| 23:41.34 | hmmhesays | ok.. |
| 23:41.39 | hunmonk | 2) configure so an IAX user can call the demo |
| 23:41.57 | hunmonk | 3) configure so an IAX user can call a voicemail box |
| 23:42.14 | hads | Kumba_: 'show application $foo' from the CLI |
| 23:42.26 | hunmonk | 4) possibly get a cheap FXO card set up :) |
| 23:42.33 | hads | Spacy: Sorry, I don't know if I'll be much help there. |
| 23:42.35 | hunmonk | hmmhesays: that's it |
| 23:42.41 | hunmonk | hmmhesays: think you can help? |
| 23:42.41 | hmmhesays | i wouldn't both er with cheap fxo |
| 23:42.51 | hmmhesays | hehe, yeah |
| 23:43.15 | hunmonk | hmmhesays: well, that's not critical. i think when it goes production it'll be all VOIP |
| 23:43.21 | hmmhesays | what iax2 client you got? |
| 23:43.43 | hunmonk | hmmhesays: couple of different ones. iaxcomm for windows. loudhush for mac |
| 23:43.47 | hmmhesays | i'm building a dialplan right now for a 120 user box |
| 23:43.55 | hmmhesays | so where is your issue? |
| 23:43.59 | wwalker | If I call exten 307 and this line is hit, will the argument passed to the macro be 307? exten => NXX,3,Macro(stdexten,${EXTEN}) |
| 23:44.23 | Spacy | hads: what coded do you use to record and playback the messages? gsm, wav? |
| 23:44.28 | hunmonk | hmmhesays: at the moment the peers i have set up can't seem to place any outbound calls |
| 23:44.46 | hunmonk | hmmhesays: lemme pull an error message for you |
| 23:44.50 | hmmhesays | set verbose 5 and make a call |
| 23:45.11 | hunmonk | hmmhesays: trying |
| 23:45.19 | hads | Spacy: In voicemail.conf? format=wav49|gsm|wav |
| 23:45.24 | *** join/#asterisk marv (n=ilovekim@c-71-228-189-127.hsd1.al.comcast.net) |
| 23:46.16 | *** join/#asterisk TrixVox (n=Spy007@c-69-248-121-104.hsd1.nj.comcast.net) |
| 23:46.27 | Spacy | hads: okay thx. i tested all of these on their own too. So the codec doesn't seem to be the problem either. -.- |
| 23:46.27 | hmmhesays | bah i'll be back |
| 23:47.51 | *** join/#asterisk kavit (n=kavit@ppp244-74.static.internode.on.net) |
| 23:48.13 | kavit | how do I know if a particular patch was included in a release or not? |
| 23:49.18 | Kumba_ | Can asterisk send SMS messages? |
| 23:49.18 | *** join/#asterisk hoytbowUE (i=webspinn@203.11.107.242) |
| 23:50.30 | hunmonk | uh oh. my tech support left. anyone else willing to help me set up a few dialplan things for some cold hard cash?? :) |
| 23:53.04 | *** join/#asterisk jeebusmobile (n=jeebusmo@12.180.154.130) |
| 23:53.55 | *** join/#asterisk techie (n=gus@adsl-068-209-242-072.sip.mia.bellsouth.net) |
| 23:54.06 | daniel_bergamini | on a simple asterist setup, I just want to add an extension I can use with xlite. what is the quickest route to that end? |
| 23:54.10 | *** join/#asterisk hmmhesays (n=hmmhesay@24-117-135-28.cpe.cableone.net) |
| 23:54.13 | hmmhesays | bah |
| 23:54.13 | hmmhesays | back |
| 23:54.36 | hunmonk | hmmhesays: http://pastebin.ca/141436 |
| 23:55.02 | hunmonk | hmmhesays: i didnt find that error message terribly informative :) |
| 23:55.28 | hmmhesays | iax2 debug |
| 23:55.36 | hmmhesays | will tell you more |
| 23:56.41 | Kumba_ | looks like you didn't supply a login/pass for IAX? |
| 23:56.57 | Kumba_ | that's my guess anyways |
| 23:57.32 | hunmonk | Kumba_: maybe. as of yet i haven't found much info on how to call another iax phone from an iax phone :) |
| 23:57.59 | kavit | does anyone know if 1.2.10 fixes this http://bugs.digium.com/view.php?id=7403&nbn=5 ? |
| 23:58.03 | hmmhesays | so what are you looking to do for a final install? |
| 23:58.06 | hunmonk | hmmhesays: ok, gimme a second to get that set up, and i'll also pastebin the users i have set up, and the stuff in the dialplan.. |
| 23:59.47 | *** part/#asterisk hads (n=hads@203.109.245.87) |
